Changelog update

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
imc: use correct position for flcoeffs2 calculation
2015-07-19 10:56:11 +02:00 · 2015-07-19 10:14:31 +02:00 · 2015-07-19 10:01:30 +02:00 · 2015-07-19 10:01:18 +02:00 · 2015-07-19 10:00:07 +02:00 · 2015-07-19 09:43:20 +02:00
1424 changed files with 17546 additions and 108455 deletions
--- a/.gitignore
+++ b/.gitignore
@@ -28,7 +28,6 @@
 /ffserver
 /config.*
 /coverage.info
-/avversion.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -37,7 +36,7 @@
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
-/doc/examples/avio_dir_cmd
+/doc/examples/avio_list_dir
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
@@ -63,7 +62,6 @@
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
-/tests/checkasm/checkasm
 /tests/data/
 /tests/pixfmts.mak
 /tests/rotozoom
--- a/210
+++ b/210
@@ -1,134 +1,94 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.

-version 2.8.2
- various fixes in the aac_fixed decoder
- various fixes in softfloat
- swresample/resample: increase precision for compensation
- lavf/mov: add support for sidx fragment indexes
- avformat/mxfenc: Only store user comment related tags when needed
- tests/fate/avformat: Fix fate-lavf
- doc/ffmpeg: Clarify that the sdp_file option requires an rtp output.
- ffmpeg: Don't try and write sdp info if none of the outputs had an rtp format.
- apng: use correct size for output buffer
- jvdec: avoid unsigned overflow in comparison
- avcodec/jpeg2000dec: Clip all tile coordinates
- avcodec/microdvddec: Check for string end in 'P' case
- avcodec/dirac_parser: Fix undefined memcpy() use
- avformat/xmv: Discard remainder of packet on error
- avformat/xmv: factor return check out of if/else
- avcodec/mpeg12dec: Do not call show_bits() with invalid bits
- avcodec/faxcompr: Add missing runs check in decode_uncompressed()
- libavutil/channel_layout: Check strtol*() for failure
- avformat/mpegts: Only start probing data streams within probe_packets
- avcodec/hevc_ps: Check chroma_format_idc
- avcodec/ffv1dec: Check for 0 quant tables
- avcodec/mjpegdec: Reinitialize IDCT on BPP changes
- avcodec/mjpegdec: Check index in ljpeg_decode_yuv_scan() before using it
- avutil/file_open: avoid file handle inheritance on Windows
- avcodec/h264_slice: Disable slice threads if there are multiple access units in a packet
- avformat/hls: update cookies on setcookie response
- opusdec: Don't run vector_fmul_scalar on zero length arrays
- avcodec/opusdec: Fix extra samples read index
- avcodec/ffv1: Initialize vlc_state on allocation
- avcodec/ffv1dec: update progress in case of broken pointer chains
- avcodec/ffv1dec: Clear slice coordinates if they are invalid or slice header decoding fails for other reasons
- rtsp: Allow $ as interleaved packet indicator before a complete response header
- videodsp: don't overread edges in vfix3 emu_edge.
- avformat/mp3dec: improve junk skipping heuristic
- concatdec: fix file_start_time calculation regression
- avcodec: loongson optimize h264dsp idct and loop filter with mmi
- avcodec/jpeg2000dec: Clear properties in jpeg2000_dec_cleanup() too
- avformat/hls: add support for EXT-X-MAP
- avformat/hls: fix segment selection regression on track changes of live streams
- configure: Require libkvazaar < 0.7.
- avcodec/vp8: Do not use num_coeff_partitions in thread/buffer setup
+version <next>:

+version 2.7.2:
+- imc: use correct position for flcoeffs2 calculation
+- hevc: check slice address length
+- snow: remove an obsolete av_assert2
+- webp: fix infinite loop in webp_decode_frame
+- wavpack: limit extra_bits to 32 and use get_bits_long
+- ffmpeg: only count got_output/errors in decode_error_stat
+- ffmpeg: exit_on_error if decoding a packet failed
+- pthread_frame: forward error codes when flushing
+- huffyuvdec: validate image size
+- wavpack: use get_bits_long to read up to 32 bits
+- nutdec: check maxpos in read_sm_data before returning success
+- s302m: fix arithmetic exception
+- vc1dec: use get_bits_long and limit the read bits to 32
+- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
+- avcodec/vp8: Check buffer size in vp8_decode_frame_header()
+- avcodec/vp8: Fix null pointer dereference in ff_vp8_decode_free()
+- avcodec/diracdec: Check for hpel_base allocation failure
+- avcodec/rv34: Clear pointers in ff_rv34_decode_init_thread_copy()
+- avfilter/af_aresample: Check ff_all_* for allocation failures
+- avcodec/pthread_frame: clear priv_data, avoid stale pointer in error case
+- swscale/utils: Clear pix buffers
+- avutil/fifo: Fix the case where func() returns less bytes than requested in av_fifo_generic_write()
+- ffmpeg: Fix cleanup after failed allocation of output_files
+- avformat/mov: Fix deallocation when MOVStreamContext failed to allocate
+- ffmpeg: Fix crash with ost->last_frame allocation failure
+- ffmpeg: Fix cleanup with ost = NULL
+- avcodec/pthread_frame: check avctx on deallocation
+- avcodec/sanm: Reset sizes in destroy_buffers()
+- avcodec/alac: Clear pointers in allocate_buffers()
+- bytestream2: set the reader to the end when reading more than available
+- avcodec/utils: use a minimum 32pixel width in  avcodec_align_dimensions2() for H.264
+- avcodec/mpegvideo: Clear pointers in ff_mpv_common_init()
+- oggparsedirac: check return value of init_get_bits
+- wmalosslessdec: reset frame->nb_samples on packet loss
+- wmalosslessdec: avoid reading 0 bits with get_bits
+- Put a space between string literals and macros.
+- avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2()
+- avcodec/aacsbr: check that the element type matches before applying SBR
+- avcodec/h264_slice: Use w/h from the AVFrame instead of mb_w/h
+- vp9/update_prob: prevent out of bounds table read
+- avfilter/vf_transpose: Fix rounding error
+- avcodec/h264_refs: discard mismatching references
+- avcodec/mjpegdec: Fix small picture upscale
+- avcodec/pngdec: Check values before updating context in decode_fctl_chunk()
+- avcodec/pngdec: Copy IHDR & plte state from last thread
+- avcodec/pngdec: Require a IHDR chunk before fctl
+- avcodec/pngdec: Only allow one IHDR chunk
+- wmavoice: limit wmavoice_decode_packet return value to packet size
+- swscale/swscale_unscaled: Fix rounding difference with RGBA output between little and big endian
+- ffmpeg: Do not use the data/size of a bitstream filter after failure
+- swscale/x86/rgb2rgb_template: fix signedness of v in shuffle_bytes_2103_{mmx,mmxext}
+- vda: unlock the pixel buffer base address.
+- swscale/rgb2rgb_template: Fix signedness of v in shuffle_bytes_2103_c()
+- swscale/rgb2rgb_template: Implement shuffle_bytes_0321_c and fix shuffle_bytes_2103_c on BE
+- swscale/rgb2rgb_template: Disable shuffle_bytes_2103_c on big endian
+- swr: Remember previously set int_sample_format from user
+- swresample: soxr implementation for swr_get_out_samples()
+- avformat/swfdec: Do not error out on pixel format changes
+- ffmpeg_opt: Fix forcing fourccs
+- configure: Check for x265_api_get
+- swscale/x86/rgb2rgb_template: don't call emms on sse2/avx functions
+- swscale/x86/rgb2rgb_template: add missing xmm clobbers
+- library.mak: Workaround SDL redefining main and breaking fate tests on mingw
+- vaapi_h264: fix RefPicList[] field flags.

-version 2.8.1:
- swscale: fix ticket #4881
+version 2.7.1:
+- postproc: fix unaligned access
+- avformat: clarify what package needs to be compiled with SSL support
+- avcodec/libx264: Avoid reconfig on equivalent aspect ratios
+- avcodec/flacenc: Fix Invalid Rice order
+- tls_gnutls: fix hang on disconnection
+- avcodec/hevc_ps: Only discard overread VPS if a previous is available
+- ffmpeg: Free last_frame instead of just unref
+- avcodec/ffv1enc: fix bps for >8bit yuv when not explicitly set
+- avio: fix potential crashes when combining ffio_ensure_seekback + crc
+- examples/demuxing_decoding: use properties from frame instead of video_dec_ctx
+- h264: er: Copy from the previous reference only if compatible
 - doc: fix spelling errors
- hls: only seek if there is an offset
- asfdec: add more checks for size left in asf packet buffer
- asfdec: alloc enough space for storing name in asf_read_metadata_obj
- avcodec/pngdec: Check blend_op.
- h264_mp4toannexb: fix pps offfset fault when there are more than one sps in avcc
- avcodec/h264_mp4toannexb_bsf: Use av_freep() to free spspps_buf
- avformat/avidec: Workaround broken initial frame
- avformat/hls: fix some cases of HLS streams which require cookies
- avcodec/pngdec: reset has_trns after every decode_frame_png()
- lavf/img2dec: Fix memory leak
- avcodec/mp3: fix skipping zeros
- avformat/srtdec: make sure we probe a number
- configure: check for ID3D11VideoContext
- avformat/vobsub: compare correct packet stream IDs
- avformat/srtdec: more lenient first line probing
- avformat/srtdec: fix number check for the first character
- avcodec/mips: build fix for MSA 64bit
- avcodec/mips: build fix for MSA
- avformat/httpauth: Add space after commas in HTTP/RTSP auth header
- libavformat/hlsenc: Use of uninitialized memory unlinking old files
- avcodec/x86/sbrdsp: Fix using uninitialized upper 32bit of noise
- avcodec/ffv1dec: Fix off by 1 error in quant_table_count check
- avcodec/ffv1dec: Explicitly check read_quant_table() return value
- dnxhddata: correct weight tables
- dnxhddec: decode and use interlace mb flag
- swscale: fix ticket #4877
- avcodec/rangecoder: Check e
- avcodec/ffv1: separate slice_count from max_slice_count
- swscale: fix ticket 4850
- cmdutils: Filter dst/srcw/h
- avutil/log: fix zero length gnu_printf format string warning
- lavf/webvttenc: Require webvtt file to contain exactly one WebVTT stream.
- swscale/swscale: Fix "unused variable" warning
- avcodec/mjpegdec: Fix decoding RGBA RCT LJPEG
- MAINTAINERS: add 2.8, drop 2.2
- doc: mention libavcodec can decode Opus natively
- hevc: properly handle no_rasl_output_flag when removing pictures from the DPB
- avfilter/af_ladspa: process all channels for nb_handles > 1
- configure: add libsoxr to swresample's pkgconfig
- lavc: Fix compilation with --disable-everything --enable-parser=mpeg4video.
-
-version 2.8:
- colorkey video filter
- BFSTM/BCSTM demuxer
- little-endian ADPCM_THP decoder
- Hap decoder and encoder
- DirectDraw Surface image/texture decoder
- ssim filter
- optional new ASF demuxer
- showvolume filter
- Many improvements to the JPEG 2000 decoder
- Go2Meeting decoding support
- adrawgraph audio and drawgraph video filter
- removegrain video filter
- Intel QSV-accelerated MPEG-2 video and HEVC encoding
- Intel QSV-accelerated MPEG-2 video and HEVC decoding
- Intel QSV-accelerated VC-1 video decoding
- libkvazaar HEVC encoder
- erosion, dilation, deflate and inflate video filters
- Dynamic Audio Normalizer as dynaudnorm filter
- Reverse video and areverse audio filter
- Random filter
- deband filter
- AAC fixed-point decoding
- sidechaincompress audio filter
- bitstream filter for converting HEVC from MP4 to Annex B
- acrossfade audio filter
- allyuv and allrgb video sources
- atadenoise video filter
- OS X VideoToolbox support
- aphasemeter filter
- showfreqs filter
- vectorscope filter
- waveform filter
- hstack and vstack filter
- Support DNx100 (1440x1080@8)
- VAAPI hevc hwaccel
- VDPAU hevc hwaccel
- framerate filter
- Switched default encoders for webm to VP9 and Opus
- Removed experimental flag from the JPEG 2000 encoder
+- configure: only disable VSX for !ppc64el
+- ffmpeg_opt: Check for localtime() failure
+- avformat/singlejpeg: fix standalone compilation
+- configure: Disable VSX on unspecified / generic CPUs
+- avformat: Fix bug in parse_rps for HEVC.
+- takdec: ensure chan2 is a valid channel index
+- avcodec/h264_slice: Use AVFrame dimensions for grayscale handling


 version 2.7:
--- a/LICENSE.md
+++ b/LICENSE.md
@@ -16,7 +16,6 @@ Specifically, the GPL parts of FFmpeg are:
 - optional x86 optimizations in the files
  - `libavcodec/x86/flac_dsp_gpl.asm`
  - `libavcodec/x86/idct_mmx.c`
-  - `libavfilter/x86/vf_removegrain.asm`
 - libutvideo encoding/decoding wrappers in
  `libavcodec/libutvideo*.cpp`
 - the X11 grabber in `libavdevice/x11grab.c`
--- a/32
+++ b/32
@@ -14,6 +14,7 @@ patches and related discussions.
 Project Leader
 ==============

+Michael Niedermayer
  final design decisions


@@ -42,7 +43,7 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================

-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
 build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
 project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
 presets                                 Robert Swain
@@ -137,7 +138,6 @@ Codecs:
  4xm.c                                 Michael Niedermayer
  8bps.c                                Roberto Togni
  8svx.c                                Jaikrishnan Menon
-  aacenc*, aaccoder.c                   Rostislav Pehlivanov
  aasc.c                                Kostya Shishkov
  ac3*                                  Justin Ruggles
  alacenc.c                             Jaikrishnan Menon
@@ -171,7 +171,6 @@ Codecs:
  dvbsubdec.c                           Anshul Maheshwari
  dxa.c                                 Kostya Shishkov
  eacmv*, eaidct*, eat*                 Peter Ross
-  evrc*                                 Paul B Mahol
  exif.c, exif.h                        Thilo Borgmann
  ffv1*                                 Michael Niedermayer
  ffwavesynth.c                         Nicolas George
@@ -201,7 +200,6 @@ Codecs:
  libcelt_dec.c                         Nicolas George
  libdirac*                             David Conrad
  libgsm.c                              Michel Bardiaux
-  libkvazaar.c                          Arttu Ylä-Outinen
  libopenjpeg.c                         Jaikrishnan Menon
  libopenjpegenc.c                      Michael Bradshaw
  libschroedinger*                      David Conrad
@@ -239,7 +237,6 @@ Codecs:
  qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
  qdrw.c                                Kostya Shishkov
  qpeg.c                                Kostya Shishkov
-  qsv*                                  Ivan Uskov
  qtrle.c                               Mike Melanson
  ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
  resample2.c                           Michael Niedermayer
@@ -301,12 +298,11 @@ Codecs:

 Hardware acceleration:
  crystalhd.c                           Philip Langdale
-  dxva2*                                Hendrik Leppkes, Laurent Aimar
+  dxva2*                                Laurent Aimar
  libstagefright.cpp                    Mohamed Naufal
  vaapi*                                Gwenole Beauchesne
  vda*                                  Sebastien Zwickert
-  vdpau*                                Philip Langdale, Carl Eugen Hoyos
-  videotoolbox*                         Sebastien Zwickert
+  vdpau*                                Carl Eugen Hoyos


 libavdevice
@@ -338,7 +334,6 @@ Generic parts:
  graphdump.c                           Nicolas George

 Filters:
-  f_drawgraph.c                         Paul B Mahol
  af_adelay.c                           Paul B Mahol
  af_aecho.c                            Paul B Mahol
  af_afade.c                            Paul B Mahol
@@ -349,21 +344,14 @@ Filters:
  af_astreamsync.c                      Nicolas George
  af_atempo.c                           Pavel Koshevoy
  af_biquads.c                          Paul B Mahol
-  af_chorus.c                           Paul B Mahol
  af_compand.c                          Paul B Mahol
  af_ladspa.c                           Paul B Mahol
  af_pan.c                              Nicolas George
-  af_sidechaincompress.c                Paul B Mahol
  af_silenceremove.c                    Paul B Mahol
-  avf_aphasemeter.c                     Paul B Mahol
  avf_avectorscope.c                    Paul B Mahol
  avf_showcqt.c                         Muhammad Faiz
  vf_blend.c                            Paul B Mahol
-  vf_colorchannelmixer.c                Paul B Mahol
  vf_colorbalance.c                     Paul B Mahol
-  vf_colorkey.c                         Timo Rothenpieler
-  vf_colorlevels.c                      Paul B Mahol
-  vf_deband.c                           Paul B Mahol
  vf_dejudder.c                         Nicholas Robbins
  vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
  vf_drawbox.c/drawgrid                 Andrey Utkin
@@ -374,16 +362,12 @@ Filters:
  vf_il.c                               Paul B Mahol
  vf_lenscorrection.c                   Daniel Oberhoff
  vf_mergeplanes.c                      Paul B Mahol
-  vf_neighbor.c                         Paul B Mahol
  vf_psnr.c                             Paul B Mahol
-  vf_random.c                           Paul B Mahol
  vf_scale.c                            Michael Niedermayer
  vf_separatefields.c                   Paul B Mahol
-  vf_ssim.c                             Paul B Mahol
  vf_stereo3d.c                         Paul B Mahol
  vf_telecine.c                         Paul B Mahol
  vf_yadif.c                            Michael Niedermayer
-  vf_zoompan.c                          Paul B Mahol

 Sources:
  vsrc_mandelbrot.c                     Michael Niedermayer
@@ -400,7 +384,6 @@ Generic parts:

 Muxers/Demuxers:
  4xm.c                                 Mike Melanson
-  aadec.c                               Vesselin Bontchev (vesselin.bontchev at yandex dot com)
  adtsenc.c                             Robert Swain
  afc.c                                 Paul B Mahol
  aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
@@ -432,7 +415,6 @@ Muxers/Demuxers:
  gxf.c                                 Reimar Doeffinger
  gxfenc.c                              Baptiste Coudurier
  hls.c                                 Anssi Hannula
-  hls encryption (hlsenc.c)             Christian Suloway
  idcin.c                               Mike Melanson
  idroqdec.c                            Mike Melanson
  iff.c                                 Jaikrishnan Menon
@@ -518,7 +500,6 @@ Muxers/Demuxers:
  wvenc.c                               Paul B Mahol

 Protocols:
-  async.c                               Zhang Rui
  bluray.c                              Petri Hintukainen
  ftp.c                                 Lukasz Marek
  http.c                                Ronald S. Bultje
@@ -554,7 +535,7 @@ Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver, Hendrik Leppkes
+Windows MSVC                            Matthew Oliver
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
@@ -564,11 +545,11 @@ x86                                     Michael Niedermayer
 Releases
 ========

-2.8                                     Michael Niedermayer
 2.7                                     Michael Niedermayer
 2.6                                     Michael Niedermayer
 2.5                                     Michael Niedermayer
 2.4                                     Michael Niedermayer
+2.2                                     Michael Niedermayer

 If you want to maintain an older release, please contact us

@@ -599,7 +580,6 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
-Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
 Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
--- a/10
+++ b/10
@@ -31,10 +31,7 @@ $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_o
 OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
 OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-ifndef CONFIG_VIDEOTOOLBOX
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
-endif
-OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
+OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
 OBJS-ffserver                 += ffserver_config.o

 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
@@ -63,7 +60,6 @@ include $(SRC_PATH)/common.mak

 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
-FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)

 all: $(AVPROGS)

@@ -85,7 +81,7 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS       \
-               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               LOONGSON3-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS

 define RESET
 $(1) :=
@@ -179,7 +175,7 @@ clean::

 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h

 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
--- a/2
+++ b/2
@@ -1 +1 @@
-2.8.2
+2.7.2
--- a/10
+++ b/10
@@ -1,10 +1,10 @@

-              ┌────────────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 2.8 "Feynman" │
-              └────────────────────────────────────────┘
+              ┌─────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 2.7 "Nash" │
+              └─────────────────────────────────────┘

-   The FFmpeg Project proudly presents FFmpeg 2.8 "Feynman", about 3
-   months after the release of FFmpeg 2.7.
+   The FFmpeg Project proudly presents FFmpeg 2.7 "Nash", about 3
+   months after the release of FFmpeg 2.6.

   A complete Changelog is available at the root of the project, and the
   complete Git history on http://source.ffmpeg.org.
--- a/arch.mak
+++ b/arch.mak
@@ -8,7 +8,7 @@ OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
-OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
+OBJS-$(HAVE_LOONGSON3) += $(LOONGSON3-OBJS)  $(LOONGSON3-OBJS-yes)

 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
--- a/cmdutils.c
+++ b/cmdutils.c
@@ -63,7 +63,7 @@

 static int init_report(const char *env);

-AVDictionary *sws_dict;
+struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;

@@ -73,13 +73,20 @@ int hide_banner = 0;

 void init_opts(void)
 {
-    av_dict_set(&sws_dict, "flags", "bicubic", 0);
+
+    if(CONFIG_SWSCALE)
+        sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
+                              NULL, NULL, NULL);
 }

 void uninit_opts(void)
 {
+#if CONFIG_SWSCALE
+    sws_freeContext(sws_opts);
+    sws_opts = NULL;
+#endif
+
    av_dict_free(&swr_opts);
-    av_dict_free(&sws_dict);
    av_dict_free(&format_opts);
    av_dict_free(&codec_opts);
    av_dict_free(&resample_opts);
@@ -522,7 +529,7 @@ static const AVOption *opt_find(void *obj, const char *name, const char *unit,
    return o;
 }

-#define FLAGS (o->type == AV_OPT_TYPE_FLAGS && (arg[0]=='-' || arg[0]=='+')) ? AV_DICT_APPEND : 0
+#define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
 int opt_default(void *optctx, const char *opt, const char *arg)
 {
    const AVOption *o;
@@ -558,24 +565,14 @@ int opt_default(void *optctx, const char *opt, const char *arg)
    }
 #if CONFIG_SWSCALE
    sc = sws_get_class();
-    if (!consumed && (o = opt_find(&sc, opt, NULL, 0,
-                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
-        struct SwsContext *sws = sws_alloc_context();
-        int ret = av_opt_set(sws, opt, arg, 0);
-        sws_freeContext(sws);
-        if (!strcmp(opt, "srcw") || !strcmp(opt, "srch") ||
-            !strcmp(opt, "dstw") || !strcmp(opt, "dsth") ||
-            !strcmp(opt, "src_format") || !strcmp(opt, "dst_format")) {
-            av_log(NULL, AV_LOG_ERROR, "Directly using swscale dimensions/format options is not supported, please use the -s or -pix_fmt options\n");
-            return AVERROR(EINVAL);
-        }
+    if (!consumed && opt_find(&sc, opt, NULL, 0,
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
+        // XXX we only support sws_flags, not arbitrary sws options
+        int ret = av_opt_set(sws_opts, opt, arg, 0);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
            return ret;
        }
-
-        av_dict_set(&sws_dict, opt, arg, FLAGS);
-
        consumed = 1;
    }
 #else
@@ -649,7 +646,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
    *g             = octx->cur_group;
    g->arg         = arg;
    g->group_def   = l->group_def;
-    g->sws_dict    = sws_dict;
+#if CONFIG_SWSCALE
+    g->sws_opts    = sws_opts;
+#endif
    g->swr_opts    = swr_opts;
    g->codec_opts  = codec_opts;
    g->format_opts = format_opts;
@@ -658,7 +657,9 @@ static void finish_group(OptionParseContext *octx, int group_idx,
    codec_opts  = NULL;
    format_opts = NULL;
    resample_opts = NULL;
-    sws_dict    = NULL;
+#if CONFIG_SWSCALE
+    sws_opts    = NULL;
+#endif
    swr_opts    = NULL;
    init_opts();

@@ -714,8 +715,9 @@ void uninit_parse_context(OptionParseContext *octx)
            av_dict_free(&l->groups[j].codec_opts);
            av_dict_free(&l->groups[j].format_opts);
            av_dict_free(&l->groups[j].resample_opts);
-
-            av_dict_free(&l->groups[j].sws_dict);
+#if CONFIG_SWSCALE
+            sws_freeContext(l->groups[j].sws_opts);
+#endif
            av_dict_free(&l->groups[j].swr_opts);
        }
        av_freep(&l->groups);
@@ -1322,12 +1324,12 @@ static void print_codec(const AVCodec *c)
    if (c->type == AVMEDIA_TYPE_VIDEO ||
        c->type == AVMEDIA_TYPE_AUDIO) {
        printf("    Threading capabilities: ");
-        switch (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
-                                   AV_CODEC_CAP_SLICE_THREADS)) {
-        case AV_CODEC_CAP_FRAME_THREADS |
-             AV_CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
-        case AV_CODEC_CAP_FRAME_THREADS: printf("frame");           break;
-        case AV_CODEC_CAP_SLICE_THREADS: printf("slice");           break;
+        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
+                                   CODEC_CAP_SLICE_THREADS)) {
+        case CODEC_CAP_FRAME_THREADS |
+             CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
+        case CODEC_CAP_FRAME_THREADS: printf("frame");           break;
+        case CODEC_CAP_SLICE_THREADS: printf("slice");           break;
        default:                      printf("no");              break;
        }
        printf("\n");
@@ -1501,11 +1503,11 @@ static void print_codecs(int encoder)

        while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
            printf(" %c", get_media_type_char(desc->type));
-            printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
-            printf((codec->capabilities & AV_CODEC_CAP_DR1)           ? "D" : ".");
+            printf((codec->capabilities & CODEC_CAP_FRAME_THREADS) ? "F" : ".");
+            printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : ".");
+            printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
+            printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
+            printf((codec->capabilities & CODEC_CAP_DR1)           ? "D" : ".");

            printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : "");
            if (strcmp(codec->name, desc->name))
@@ -1579,10 +1581,10 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                *(descr_cur++) = '>';
            }
            pad = i ? filter->outputs : filter->inputs;
-            for (j = 0; pad && avfilter_pad_get_name(pad, j); j++) {
+            for (j = 0; pad && pad[j].name; j++) {
                if (descr_cur >= descr + sizeof(descr) - 4)
                    break;
-                *(descr_cur++) = get_media_type_char(avfilter_pad_get_type(pad, j));
+                *(descr_cur++) = get_media_type_char(pad[j].type);
            }
            if (!j)
                *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
@@ -1873,6 +1875,64 @@ int read_yesno(void)
    return yesno;
 }

+int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
+{
+    int64_t ret;
+    FILE *f = av_fopen_utf8(filename, "rb");
+
+    if (!f) {
+        ret = AVERROR(errno);
+        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
+               strerror(errno));
+        return ret;
+    }
+
+    ret = fseek(f, 0, SEEK_END);
+    if (ret == -1) {
+        ret = AVERROR(errno);
+        goto out;
+    }
+
+    ret = ftell(f);
+    if (ret < 0) {
+        ret = AVERROR(errno);
+        goto out;
+    }
+    *size = ret;
+
+    ret = fseek(f, 0, SEEK_SET);
+    if (ret == -1) {
+        ret = AVERROR(errno);
+        goto out;
+    }
+
+    *bufptr = av_malloc(*size + 1);
+    if (!*bufptr) {
+        av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
+        ret = AVERROR(ENOMEM);
+        goto out;
+    }
+    ret = fread(*bufptr, 1, *size, f);
+    if (ret < *size) {
+        av_free(*bufptr);
+        if (ferror(f)) {
+            ret = AVERROR(errno);
+            av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
+                   filename, strerror(errno));
+        } else
+            ret = AVERROR_EOF;
+    } else {
+        ret = 0;
+        (*bufptr)[(*size)++] = '\0';
+    }
+
+out:
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", av_err2str(ret));
+    fclose(f);
+    return ret;
+}
+
 FILE *get_preset_file(char *filename, size_t filename_size,
                      const char *preset_name, int is_path,
                      const char *codec_name)
@@ -2046,10 +2106,7 @@ double get_rotation(AVStream *st)
    theta -= 360*floor(theta/360 + 0.9/360);

    if (fabs(theta - 90*round(theta/90)) > 2)
-        av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
-               "If you want to help, upload a sample "
-               "of this file to ftp://upload.ffmpeg.org/incoming/ "
-               "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
+        av_log_ask_for_sample(NULL, "Odd rotation angle\n");

    return theta;
 }
--- a/cmdutils.h
+++ b/cmdutils.h
@@ -46,7 +46,7 @@ extern const int program_birth_year;

 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
-extern AVDictionary *sws_dict;
+extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;
@@ -277,7 +277,7 @@ typedef struct OptionGroup {
    AVDictionary *codec_opts;
    AVDictionary *format_opts;
    AVDictionary *resample_opts;
-    AVDictionary *sws_dict;
+    struct SwsContext *sws_opts;
    AVDictionary *swr_opts;
 } OptionGroup;

@@ -529,6 +529,18 @@ int show_colors(void *optctx, const char *opt, const char *arg);
 */
 int read_yesno(void);

+/**
+ * Read the file with name filename, and put its content in a newly
+ * allocated 0-terminated buffer.
+ *
+ * @param filename file to read from
+ * @param bufptr location where pointer to buffer is returned
+ * @param size   location where size of buffer is returned
+ * @return >= 0 in case of success, a negative value corresponding to an
+ * AVERROR error code in case of failure.
+ */
+int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
+
 /**
 * Get a file corresponding to a preset file.
 *
--- a/common.mak
+++ b/common.mak
@@ -118,9 +118,8 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)

-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
-STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))

 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
--- a/compat/windows/mslink
+++ b/compat/windows/mslink
@@ -1,9 +0,0 @@
-#!/bin/sh
-
-LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
-if [ -x "$LINK_EXE_PATH" ]; then
-    "$LINK_EXE_PATH" $@
-else
-    link $@
-fi
-exit $?
--- a/346
+++ b/346
@@ -155,7 +155,6 @@ Hardware accelerators:
  --disable-vaapi          disable VAAPI code [autodetect]
  --disable-vda            disable VDA code [autodetect]
  --disable-vdpau          disable VDPAU code [autodetect]
-  --enable-videotoolbox    enable VideoToolbox code [autodetect]

 Individual component options:
  --disable-everything     disable all components listed below
@@ -223,7 +222,6 @@ External library support:
  --enable-libgsm          enable GSM de/encoding via libgsm [no]
  --enable-libiec61883     enable iec61883 via libiec61883 [no]
  --enable-libilbc         enable iLBC de/encoding via libilbc [no]
-  --enable-libkvazaar      enable HEVC encoding via libkvazaar [no]
  --enable-libmfx          enable HW acceleration through libmfx
  --enable-libmodplug      enable ModPlug via libmodplug [no]
  --enable-libmp3lame      enable MP3 encoding via libmp3lame [no]
@@ -241,7 +239,6 @@ External library support:
  --enable-libschroedinger enable Dirac de/encoding via libschroedinger [no]
  --enable-libshine        enable fixed-point MP3 encoding via libshine [no]
  --enable-libsmbclient    enable Samba protocol via libsmbclient [no]
-  --enable-libsnappy       enable Snappy compression, needed for hap encoding [no]
  --enable-libsoxr         enable Include libsoxr resampling [no]
  --enable-libspeex        enable Speex de/encoding via libspeex [no]
  --enable-libssh          enable SFTP protocol via libssh [no]
@@ -324,7 +321,6 @@ Toolchain options:
  --extra-cxxflags=ECFLAGS add ECFLAGS to CXXFLAGS [$CXXFLAGS]
  --extra-ldflags=ELDFLAGS add ELDFLAGS to LDFLAGS [$LDFLAGS]
  --extra-ldexeflags=ELDFLAGS add ELDFLAGS to LDEXEFLAGS [$LDEXEFLAGS]
-  --extra-ldlibflags=ELDFLAGS add ELDFLAGS to LDLIBFLAGS [$LDLIBFLAGS]
  --extra-libs=ELIBS       add ELIBS [$ELIBS]
  --extra-version=STRING   version string suffix []
  --optflags=OPTFLAGS      override optimization-related compiler flags
@@ -377,7 +373,7 @@ Optimization options (experts only):
  --disable-mipsdspr2      disable MIPS DSP ASE R2 optimizations
  --disable-msa            disable MSA optimizations
  --disable-mipsfpu        disable floating point MIPS optimizations
-  --disable-mmi            disable Loongson SIMD optimizations
+  --disable-loongson3      disable Loongson-3 SIMD optimizations
  --disable-fast-unaligned consider unaligned accesses slow

 Developer options (useful when working on FFmpeg itself):
@@ -479,7 +475,7 @@ sh_quote(){
 }

 cleanws(){
-    echo "$@" | sed 's/^ *//;s/[[:space:]][[:space:]]*/ /g;s/ *$//'
+    echo "$@" | sed 's/^ *//;s/  */ /g;s/ *$//;s/\\r//g'
 }

 filter(){
@@ -795,10 +791,6 @@ add_ldexeflags(){
    append LDEXEFLAGS $($ldflags_filter "$@")
 }

-add_ldlibflags(){
-    append LDLIBFLAGS $($ldflags_filter "$@")
-}
-
 add_stripflags(){
    append ASMSTRIPFLAGS "$@"
 }
@@ -1312,6 +1304,12 @@ check_host_cpp_condition(){
 EOF
 }

+apply(){
+    file=$1
+    shift
+    "$@" < "$file" > "$file.tmp" && mv "$file.tmp" "$file" || rm "$file.tmp"
+}
+
 cp_if_changed(){
    cmp -s "$1" "$2" && echo "$2 is unchanged" && return
    mkdir -p "$(dirname $2)"
@@ -1337,7 +1335,7 @@ COMPONENT_LIST="

 EXAMPLE_LIST="
    avio_reading_example
-    avio_dir_cmd_example
+    avio_list_dir_example
    decoding_encoding_example
    demuxing_decoding_example
    extract_mvs_example
@@ -1382,7 +1380,6 @@ EXTERNAL_LIBRARY_LIST="
    libgsm
    libiec61883
    libilbc
-    libkvazaar
    libmfx
    libmodplug
    libmp3lame
@@ -1399,7 +1396,6 @@ EXTERNAL_LIBRARY_LIST="
    libschroedinger
    libshine
    libsmbclient
-    libsnappy
    libsoxr
    libspeex
    libssh
@@ -1465,7 +1461,6 @@ HWACCEL_LIST="
    vaapi
    vda
    vdpau
-    videotoolbox
    xvmc
 "

@@ -1594,9 +1589,7 @@ ARCH_EXT_LIST_MIPS="
 "

 ARCH_EXT_LIST_LOONGSON="
-    loongson2
    loongson3
-    mmi
 "

 ARCH_EXT_LIST_X86_SIMD="
@@ -1686,7 +1679,6 @@ HEADERS_LIST="
    dev_video_bktr_ioctl_bt848_h
    dev_video_meteor_ioctl_meteor_h
    direct_h
-    dirent_h
    dlfcn_h
    d3d11_h
    dxva_h
@@ -1727,9 +1719,7 @@ MATH_FUNCS="
    atan2f
    cbrt
    cbrtf
-    copysign
    cosf
-    erf
    exp2
    exp2f
    expf
@@ -1782,7 +1772,6 @@ SYSTEM_FUNCS="
    jack_port_get_latency_range
    kbhit
    localtime_r
-    lstat
    lzo1x_999_compress
    mach_absolute_time
    MapViewOfFile
@@ -1796,7 +1785,6 @@ SYSTEM_FUNCS="
    pthread_cancel
    sched_getaffinity
    SetConsoleTextAttribute
-    SetConsoleCtrlHandler
    setmode
    setrlimit
    Sleep
@@ -1836,7 +1824,6 @@ TYPES_LIST="
    CONDITION_VARIABLE_Ptr
    socklen_t
    struct_addrinfo
-    struct_dcadec_exss_info_matrix_encoding
    struct_group_source_req
    struct_ip_mreq_source
    struct_ipv6_mreq
@@ -1900,10 +1887,8 @@ CONFIG_EXTRA="
    faandct
    faanidct
    fdctdsp
-    flacdsp
    fmtconvert
    frame_thread_encoder
-    g722dsp
    gcrypt
    gmp
    golomb
@@ -1921,9 +1906,7 @@ CONFIG_EXTRA="
    iirfilter
    imdct15
    intrax8
-    ividsp
    jpegtables
-    libx262
    lgplv3
    llauddsp
    llviddsp
@@ -1934,7 +1917,6 @@ CONFIG_EXTRA="
    mpegaudiodsp
    mpegvideo
    mpegvideoenc
-    mss34dsp
    pixblockdsp
    qpeldsp
    qsv
@@ -1945,19 +1927,12 @@ CONFIG_EXTRA="
    riffenc
    rtpdec
    rtpenc_chain
-    rv34dsp
    sinewin
-    snappy
    startcode
-    texturedsp
-    texturedspenc
    tpeldsp
    videodsp
    vp3dsp
-    vp56dsp
-    vp8dsp
    wma_freqs
-    wmv2dsp
 "

 CMDLINE_SELECT="
@@ -2065,7 +2040,7 @@ mips32r2_deps="mips"
 mips32r5_deps="mips"
 mips64r6_deps="mips"
 msa_deps="mips"
-mmi_deps="mips"
+loongson3_deps="mips"

 altivec_deps="ppc"
 dcbzl_deps="ppc"
@@ -2137,22 +2112,19 @@ me_cmp_select="fdctdsp idctdsp pixblockdsp"
 mpeg_er_select="error_resilience"
 mpegaudio_select="mpegaudiodsp"
 mpegaudiodsp_select="dct"
-mpegvideo_select="blockdsp h264chroma hpeldsp idctdsp me_cmp mpeg_er videodsp"
+mpegvideo_select="blockdsp h264chroma hpeldsp idctdsp me_cmp videodsp"
 mpegvideoenc_select="me_cmp mpegvideo pixblockdsp qpeldsp"
 qsvdec_select="qsv"
 qsvenc_select="qsv"

 # decoders / encoders
 aac_decoder_select="imdct15 mdct sinewin"
-aac_fixed_decoder_select="mdct sinewin"
-aac_encoder_select="audio_frame_queue iirfilter lpc mdct sinewin"
+aac_encoder_select="audio_frame_queue iirfilter mdct sinewin"
 aac_latm_decoder_select="aac_decoder aac_latm_parser"
 ac3_decoder_select="ac3_parser ac3dsp bswapdsp fmtconvert mdct"
 ac3_fixed_decoder_select="ac3_parser ac3dsp bswapdsp mdct"
 ac3_encoder_select="ac3dsp audiodsp mdct me_cmp"
 ac3_fixed_encoder_select="ac3dsp audiodsp mdct me_cmp"
-adpcm_g722_decoder_select="g722dsp"
-adpcm_g722_encoder_select="g722dsp"
 aic_decoder_select="golomb idctdsp"
 alac_encoder_select="lpc"
 als_decoder_select="bswapdsp"
@@ -2181,7 +2153,6 @@ cook_decoder_select="audiodsp mdct sinewin"
 cscd_decoder_select="lzo"
 cscd_decoder_suggest="zlib"
 dca_decoder_select="fmtconvert mdct"
-dds_decoder_select="texturedsp"
 dirac_decoder_select="dwt golomb videodsp mpegvideoenc"
 dnxhd_decoder_select="blockdsp idctdsp"
 dnxhd_encoder_select="aandcttables blockdsp fdctdsp idctdsp mpegvideoenc pixblockdsp"
@@ -2199,8 +2170,8 @@ ffv1_encoder_select="rangecoder"
 ffvhuff_decoder_select="huffyuv_decoder"
 ffvhuff_encoder_select="huffyuv_encoder"
 fic_decoder_select="golomb"
-flac_decoder_select="flacdsp golomb"
-flac_encoder_select="bswapdsp flacdsp golomb lpc"
+flac_decoder_select="golomb"
+flac_encoder_select="bswapdsp golomb lpc"
 flashsv_decoder_select="zlib"
 flashsv_encoder_select="zlib"
 flashsv2_encoder_select="zlib"
@@ -2211,9 +2182,9 @@ fourxm_decoder_select="blockdsp bswapdsp"
 fraps_decoder_select="bswapdsp huffman"
 g2m_decoder_select="blockdsp idctdsp jpegtables zlib"
 g729_decoder_select="audiodsp"
-h261_decoder_select="mpegvideo"
+h261_decoder_select="mpeg_er mpegvideo"
 h261_encoder_select="aandcttables mpegvideoenc"
-h263_decoder_select="h263_parser h263dsp mpegvideo qpeldsp"
+h263_decoder_select="error_resilience h263_parser h263dsp mpeg_er mpegvideo qpeldsp"
 h263_encoder_select="aandcttables h263dsp mpegvideoenc"
 h263i_decoder_select="h263_decoder"
 h263p_decoder_select="h263_decoder"
@@ -2224,21 +2195,12 @@ h264_qsv_decoder_deps="libmfx"
 h264_qsv_decoder_select="h264_mp4toannexb_bsf h264_parser qsvdec h264_qsv_hwaccel"
 h264_qsv_encoder_deps="libmfx"
 h264_qsv_encoder_select="qsvenc"
-hap_decoder_select="snappy texturedsp"
-hap_encoder_deps="libsnappy"
-hap_encoder_select="texturedspenc"
 hevc_decoder_select="bswapdsp cabac golomb videodsp"
-hevc_qsv_encoder_deps="libmfx"
-hevc_qsv_decoder_deps="libmfx"
-hevc_qsv_decoder_select="hevc_mp4toannexb_bsf hevc_parser qsvdec hevc_qsv_hwaccel"
-hevc_qsv_encoder_select="qsvenc"
 huffyuv_decoder_select="bswapdsp huffyuvdsp llviddsp"
 huffyuv_encoder_select="bswapdsp huffman huffyuvencdsp llviddsp"
 iac_decoder_select="imc_decoder"
 imc_decoder_select="bswapdsp fft mdct sinewin"
 indeo3_decoder_select="hpeldsp"
-indeo4_decoder_select="ividsp"
-indeo5_decoder_select="ividsp"
 interplay_video_decoder_select="hpeldsp"
 jpegls_decoder_select="golomb mjpeg_decoder"
 jpegls_encoder_select="golomb"
@@ -2268,25 +2230,19 @@ mpc7_decoder_select="bswapdsp mpegaudiodsp"
 mpc8_decoder_select="mpegaudiodsp"
 mpeg_xvmc_decoder_deps="X11_extensions_XvMClib_h"
 mpeg_xvmc_decoder_select="mpeg2video_decoder"
-mpegvideo_decoder_select="mpegvideo"
-mpeg1video_decoder_select="mpegvideo"
+mpegvideo_decoder_select="error_resilience mpeg_er mpegvideo"
+mpeg1video_decoder_select="error_resilience mpeg_er mpegvideo"
 mpeg1video_encoder_select="aandcttables mpegvideoenc h263dsp"
-mpeg2video_decoder_select="mpegvideo"
+mpeg2video_decoder_select="error_resilience mpeg_er mpegvideo"
 mpeg2video_encoder_select="aandcttables mpegvideoenc h263dsp"
-mpeg2_qsv_decoder_deps="libmfx"
-mpeg2_qsv_decoder_select="qsvdec mpeg2_qsv_hwaccel"
-mpeg2_qsv_encoder_deps="libmfx"
-mpeg2_qsv_encoder_select="qsvenc"
 mpeg4_decoder_select="h263_decoder mpeg4video_parser"
 mpeg4_encoder_select="h263_encoder"
-msa1_decoder_select="mss34dsp"
 msmpeg4v1_decoder_select="h263_decoder"
 msmpeg4v2_decoder_select="h263_decoder"
 msmpeg4v2_encoder_select="h263_encoder"
 msmpeg4v3_decoder_select="h263_decoder"
 msmpeg4v3_encoder_select="h263_encoder"
-mss2_decoder_select="vc1_decoder"
-mts2_decoder_select="mss34dsp"
+mss2_decoder_select="error_resilience mpeg_er qpeldsp vc1_decoder"
 mxpeg_decoder_select="mjpeg_decoder"
 nellymoser_decoder_select="mdct sinewin"
 nellymoser_encoder_select="audio_frame_queue mdct sinewin"
@@ -2305,12 +2261,12 @@ ra_144_decoder_select="audiodsp"
 ralf_decoder_select="golomb"
 rawvideo_decoder_select="bswapdsp"
 rtjpeg_decoder_select="me_cmp"
-rv10_decoder_select="h263_decoder"
+rv10_decoder_select="error_resilience h263_decoder h263dsp mpeg_er"
 rv10_encoder_select="h263_encoder"
-rv20_decoder_select="h263_decoder"
+rv20_decoder_select="error_resilience h263_decoder h263dsp mpeg_er"
 rv20_encoder_select="h263_encoder"
-rv30_decoder_select="golomb h264pred h264qpel mpegvideo rv34dsp"
-rv40_decoder_select="golomb h264pred h264qpel mpegvideo rv34dsp"
+rv30_decoder_select="error_resilience golomb h264chroma h264pred h264qpel mpeg_er mpegvideo videodsp"
+rv40_decoder_select="error_resilience golomb h264chroma h264pred h264qpel mpeg_er mpegvideo videodsp"
 shorten_decoder_select="golomb"
 sipr_decoder_select="lsp"
 snow_decoder_select="dwt h264qpel hpeldsp me_cmp rangecoder videodsp"
@@ -2334,23 +2290,20 @@ truemotion2_decoder_select="bswapdsp"
 truespeech_decoder_select="bswapdsp"
 tscc_decoder_select="zlib"
 twinvq_decoder_select="mdct lsp sinewin"
-txd_decoder_select="texturedsp"
 utvideo_decoder_select="bswapdsp"
 utvideo_encoder_select="bswapdsp huffman huffyuvencdsp"
 vble_decoder_select="huffyuvdsp"
-vc1_decoder_select="blockdsp h263_decoder h264qpel intrax8 qpeldsp startcode"
+vc1_decoder_select="blockdsp error_resilience h263_decoder h264chroma h264qpel intrax8 mpeg_er qpeldsp startcode"
 vc1image_decoder_select="vc1_decoder"
-vc1_qsv_decoder_deps="libmfx"
-vc1_qsv_decoder_select="qsvdec vc1_qsv_hwaccel"
 vorbis_decoder_select="mdct"
 vorbis_encoder_select="mdct"
 vp3_decoder_select="hpeldsp vp3dsp videodsp"
-vp5_decoder_select="h264chroma hpeldsp videodsp vp3dsp vp56dsp"
-vp6_decoder_select="h264chroma hpeldsp huffman videodsp vp3dsp vp56dsp"
+vp5_decoder_select="h264chroma hpeldsp videodsp vp3dsp"
+vp6_decoder_select="h264chroma hpeldsp huffman videodsp vp3dsp"
 vp6a_decoder_select="vp6_decoder"
 vp6f_decoder_select="vp6_decoder"
-vp7_decoder_select="h264pred videodsp vp8dsp"
-vp8_decoder_select="h264pred videodsp vp8dsp"
+vp7_decoder_select="h264pred videodsp"
+vp8_decoder_select="h264pred videodsp"
 vp9_decoder_select="videodsp vp9_parser"
 webp_decoder_select="vp8_decoder"
 wmalossless_decoder_select="llauddsp"
@@ -2362,8 +2315,8 @@ wmav2_encoder_select="mdct sinewin wma_freqs"
 wmavoice_decoder_select="lsp rdft dct mdct sinewin"
 wmv1_decoder_select="h263_decoder"
 wmv1_encoder_select="h263_encoder"
-wmv2_decoder_select="blockdsp h263_decoder idctdsp intrax8 videodsp wmv2dsp"
-wmv2_encoder_select="h263_encoder wmv2dsp"
+wmv2_decoder_select="blockdsp h263_decoder idctdsp intrax8 videodsp"
+wmv2_encoder_select="h263_encoder"
 wmv3_decoder_select="vc1_decoder"
 wmv3image_decoder_select="wmv3_decoder"
 zerocodec_decoder_select="zlib"
@@ -2374,22 +2327,18 @@ zmbv_encoder_select="zlib"

 # hardware accelerators
 crystalhd_deps="libcrystalhd_libcrystalhd_if_h"
-d3d11va_deps="d3d11_h dxva_h ID3D11VideoDecoder ID3D11VideoContext"
+d3d11va_deps="d3d11_h dxva_h ID3D11VideoDecoder"
 dxva2_deps="dxva2api_h DXVA2_ConfigPictureDecode"
 vaapi_deps="va_va_h"
-vda_deps="VideoDecodeAcceleration_VDADecoder_h pthreads CoreServices_CoreServices_h"
-vda_extralibs="-framework CoreFoundation -framework VideoDecodeAcceleration -framework QuartzCore -framework CoreServices"
+vda_deps="VideoDecodeAcceleration_VDADecoder_h pthreads"
+vda_extralibs="-framework CoreFoundation -framework VideoDecodeAcceleration -framework QuartzCore"
 vdpau_deps="vdpau_vdpau_h vdpau_vdpau_x11_h"
-videotoolbox_deps="VideoToolbox_VideoToolbox_h pthreads CoreServices_CoreServices_h"
-videotoolbox_extralibs="-framework CoreFoundation -framework VideoToolbox -framework CoreMedia -framework QuartzCore -framework CoreServices"
 xvmc_deps="X11_extensions_XvMClib_h"

 h263_vaapi_hwaccel_deps="vaapi"
 h263_vaapi_hwaccel_select="h263_decoder"
 h263_vdpau_hwaccel_deps="vdpau"
 h263_vdpau_hwaccel_select="h263_decoder"
-h263_videotoolbox_hwaccel_deps="videotoolbox"
-h263_videotoolbox_hwaccel_select="h263_decoder"
 h264_crystalhd_decoder_select="crystalhd h264_mp4toannexb_bsf h264_parser"
 h264_d3d11va_hwaccel_deps="d3d11va"
 h264_d3d11va_hwaccel_select="h264_decoder"
@@ -2412,17 +2361,10 @@ h264_vdpau_decoder_deps="vdpau"
 h264_vdpau_decoder_select="h264_decoder"
 h264_vdpau_hwaccel_deps="vdpau"
 h264_vdpau_hwaccel_select="h264_decoder"
-h264_videotoolbox_hwaccel_deps="videotoolbox"
-h264_videotoolbox_hwaccel_select="h264_decoder"
 hevc_d3d11va_hwaccel_deps="d3d11va DXVA_PicParams_HEVC"
 hevc_d3d11va_hwaccel_select="hevc_decoder"
 hevc_dxva2_hwaccel_deps="dxva2 DXVA_PicParams_HEVC"
 hevc_dxva2_hwaccel_select="hevc_decoder"
-hevc_qsv_hwaccel_deps="libmfx"
-hevc_vaapi_hwaccel_deps="vaapi VAPictureParameterBufferHEVC"
-hevc_vaapi_hwaccel_select="hevc_decoder"
-hevc_vdpau_hwaccel_deps="vdpau VdpPictureInfoHEVC"
-hevc_vdpau_hwaccel_select="hevc_decoder"
 mpeg_vdpau_decoder_deps="vdpau"
 mpeg_vdpau_decoder_select="mpeg2video_decoder"
 mpeg_xvmc_hwaccel_deps="xvmc"
@@ -2431,8 +2373,6 @@ mpeg1_vdpau_decoder_deps="vdpau"
 mpeg1_vdpau_decoder_select="mpeg1video_decoder"
 mpeg1_vdpau_hwaccel_deps="vdpau"
 mpeg1_vdpau_hwaccel_select="mpeg1video_decoder"
-mpeg1_videotoolbox_hwaccel_deps="videotoolbox"
-mpeg1_videotoolbox_hwaccel_select="mpeg1video_decoder"
 mpeg1_xvmc_hwaccel_deps="xvmc"
 mpeg1_xvmc_hwaccel_select="mpeg1video_decoder"
 mpeg2_crystalhd_decoder_select="crystalhd"
@@ -2440,14 +2380,10 @@ mpeg2_d3d11va_hwaccel_deps="d3d11va"
 mpeg2_d3d11va_hwaccel_select="mpeg2video_decoder"
 mpeg2_dxva2_hwaccel_deps="dxva2"
 mpeg2_dxva2_hwaccel_select="mpeg2video_decoder"
-mpeg2_qsv_hwaccel_deps="libmfx"
-mpeg2_qsv_hwaccel_select="qsvdec_mpeg2"
 mpeg2_vaapi_hwaccel_deps="vaapi"
 mpeg2_vaapi_hwaccel_select="mpeg2video_decoder"
 mpeg2_vdpau_hwaccel_deps="vdpau"
 mpeg2_vdpau_hwaccel_select="mpeg2video_decoder"
-mpeg2_videotoolbox_hwaccel_deps="videotoolbox"
-mpeg2_videotoolbox_hwaccel_select="mpeg2video_decoder"
 mpeg2_xvmc_hwaccel_deps="xvmc"
 mpeg2_xvmc_hwaccel_select="mpeg2video_decoder"
 mpeg4_crystalhd_decoder_select="crystalhd"
@@ -2457,8 +2393,6 @@ mpeg4_vdpau_decoder_deps="vdpau"
 mpeg4_vdpau_decoder_select="mpeg4_decoder"
 mpeg4_vdpau_hwaccel_deps="vdpau"
 mpeg4_vdpau_hwaccel_select="mpeg4_decoder"
-mpeg4_videotoolbox_hwaccel_deps="videotoolbox"
-mpeg4_videotoolbox_hwaccel_select="mpeg4_decoder"
 msmpeg4_crystalhd_decoder_select="crystalhd"
 vc1_crystalhd_decoder_select="crystalhd"
 vc1_d3d11va_hwaccel_deps="d3d11va"
@@ -2471,8 +2405,6 @@ vc1_vdpau_decoder_deps="vdpau"
 vc1_vdpau_decoder_select="vc1_decoder"
 vc1_vdpau_hwaccel_deps="vdpau"
 vc1_vdpau_hwaccel_select="vc1_decoder"
-vc1_qsv_hwaccel_deps="libmfx"
-vc1_qsv_hwaccel_select="qsvdec_vc1"
 wmv3_crystalhd_decoder_select="crystalhd"
 wmv3_d3d11va_hwaccel_select="vc1_d3d11va_hwaccel"
 wmv3_dxva2_hwaccel_select="vc1_dxva2_hwaccel"
@@ -2482,9 +2414,9 @@ wmv3_vdpau_hwaccel_select="vc1_vdpau_hwaccel"

 # parsers
 h264_parser_select="h264_decoder"
-hevc_parser_select="golomb"
+hevc_parser_select="hevc_decoder"
 mpegvideo_parser_select="mpegvideo"
-mpeg4video_parser_select="h263dsp mpegvideo qpeldsp"
+mpeg4video_parser_select="error_resilience h263dsp mpeg_er mpegvideo qpeldsp"
 vc1_parser_select="mpegvideo startcode vc1_decoder"

 # bitstream_filters
@@ -2506,7 +2438,6 @@ libgsm_ms_decoder_deps="libgsm"
 libgsm_ms_encoder_deps="libgsm"
 libilbc_decoder_deps="libilbc"
 libilbc_encoder_deps="libilbc"
-libkvazaar_encoder_deps="libkvazaar"
 libmodplug_demuxer_deps="libmodplug"
 libmp3lame_encoder_deps="libmp3lame"
 libmp3lame_encoder_select="audio_frame_queue"
@@ -2544,7 +2475,6 @@ libvpx_vp9_encoder_deps="libvpx"
 libwavpack_encoder_deps="libwavpack"
 libwebp_encoder_deps="libwebp"
 libwebp_anim_encoder_deps="libwebp"
-libx262_encoder_deps="libx262"
 libx264_encoder_deps="libx264"
 libx264rgb_encoder_deps="libx264"
 libx264rgb_encoder_select="libx264_encoder"
@@ -2561,7 +2491,6 @@ nvenc_hevc_encoder_deps="nvenc"
 # demuxers / muxers
 ac3_demuxer_select="ac3_parser"
 asf_demuxer_select="riffdec"
-asf_o_demuxer_select="riffdec"
 asf_muxer_select="riffenc"
 asf_stream_muxer_select="asf_muxer"
 avi_demuxer_select="riffdec exif"
@@ -2679,7 +2608,6 @@ x11grab_indev_deps="x11grab"
 x11grab_xcb_indev_deps="libxcb"

 # protocols
-async_protocol_deps="pthreads"
 bluray_protocol_deps="libbluray"
 ffrtmpcrypt_protocol_deps="!librtmp_protocol"
 ffrtmpcrypt_protocol_deps_any="gcrypt gmp openssl"
@@ -2743,7 +2671,6 @@ cropdetect_filter_deps="gpl"
 delogo_filter_deps="gpl"
 deshake_filter_select="pixelutils"
 drawtext_filter_deps="libfreetype"
-dynaudnorm_filter_deps="copysign erf"
 ebur128_filter_deps="gpl"
 eq_filter_deps="gpl"
 fftfilt_filter_deps="avcodec"
@@ -2778,13 +2705,10 @@ repeatfields_filter_deps="gpl"
 resample_filter_deps="avresample"
 sab_filter_deps="gpl swscale"
 scale_filter_deps="swscale"
-scale2ref_filter_deps="swscale"
 select_filter_select="pixelutils"
 smartblur_filter_deps="gpl swscale"
 showcqt_filter_deps="avcodec"
 showcqt_filter_select="fft"
-showfreqs_filter_deps="avcodec"
-showfreqs_filter_select="fft"
 showspectrum_filter_deps="avcodec"
 showspectrum_filter_select="rdft"
 spp_filter_deps="gpl avcodec"
@@ -2804,7 +2728,7 @@ zoompan_filter_deps="swscale"

 # examples
 avio_reading="avformat avcodec avutil"
-avio_dir_cmd="avformat avutil"
+avio_list_dir="avformat avutil"
 avcodec_example_deps="avcodec avutil"
 decoding_encoding_example_deps="avcodec avformat avutil"
 demuxing_decoding_example_deps="avcodec avformat avutil"
@@ -2876,7 +2800,11 @@ ln_s="ln -s -f"
 nm_default="nm -g"
 objformat="elf"
 pkg_config_default=pkg-config
-ranlib_default="ranlib"
+if ranlib 2>&1 | grep -q "\-D "; then
+    ranlib_default="ranlib -D"
+else
+    ranlib_default="ranlib"
+fi
 strip_default="strip"
 yasmexe_default="yasm"
 windres_default="windres"
@@ -2917,7 +2845,7 @@ sws_max_filter_size_default=256
 set_default sws_max_filter_size

 # Enable hwaccels by default.
-enable d3d11va dxva2 vaapi vda vdpau videotoolbox xvmc
+enable d3d11va dxva2 vaapi vda vdpau xvmc
 enable xlib

 # build settings
@@ -2970,9 +2898,8 @@ if test -f configure; then
    source_path=.
 else
    source_path=$(cd $(dirname "$0"); pwd)
-    case "$source_path" in
-        *[[:blank:]]*) die "Out of tree builds are impossible with whitespace in source path." ;;
-    esac
+    echo "$source_path" | grep -q '[[:blank:]]' &&
+        die "Out of tree builds are impossible with whitespace in source path."
    test -e "$source_path/config.h" &&
        die "Out of tree builds are impossible with config.h in source dir."
 fi
@@ -3072,9 +2999,6 @@ for opt do
        --extra-ldexeflags=*)
            add_ldexeflags $optval
        ;;
-        --extra-ldlibflags=*)
-            add_ldlibflags $optval
-        ;;
        --extra-libs=*)
            add_extralibs $optval
        ;;
@@ -3210,14 +3134,9 @@ case "$toolchain" in
        else
            cc_default="c99wrap cl"
        fi
-        ld_default="$source_path/compat/windows/mslink"
+        ld_default="link"
        nm_default="dumpbin -symbols"
        ar_default="lib"
-        case "$arch" in
-        arm*)
-            as_default="armasm"
-            ;;
-        esac
        target_os_default="win32"
        # Use a relative path for TMPDIR. This makes sure all the
        # ffconf temp files are written with a relative path, avoiding
@@ -3259,11 +3178,7 @@ cc_default="${cross_prefix}${cc_default}"
 cxx_default="${cross_prefix}${cxx_default}"
 nm_default="${cross_prefix}${nm_default}"
 pkg_config_default="${cross_prefix}${pkg_config_default}"
-if ${cross_prefix}${ranlib_default} 2>&1 | grep -q "\-D "; then
-    ranlib_default="${cross_prefix}${ranlib_default} -D"
-else
-    ranlib_default="${cross_prefix}${ranlib_default}"
-fi
+ranlib_default="${cross_prefix}${ranlib_default}"
 strip_default="${cross_prefix}${strip_default}"
 windres_default="${cross_prefix}${windres_default}"

@@ -3414,7 +3329,6 @@ msvc_common_flags(){
            -lavifil32)           echo vfw32.lib ;;
            -lavicap32)           echo vfw32.lib user32.lib ;;
            -l*)                  echo ${flag#-l}.lib ;;
-            -LARGEADDRESSAWARE)   echo $flag ;;
            -L*)                  echo -libpath:${flag#-L} ;;
            *)                    echo $flag ;;
        esac
@@ -3543,7 +3457,6 @@ tms470_flags(){
 probe_cc(){
    pfx=$1
    _cc=$2
-    first=$3

    unset _type _ident _cc_c _cc_e _cc_o _flags _cflags
    unset _ld_o _ldflags _ld_lib _ld_path
@@ -3554,8 +3467,8 @@ probe_cc(){
        true # no-op to avoid reading stdin in following checks
    elif $_cc -v 2>&1 | grep -q '^gcc.*LLVM'; then
        _type=llvm_gcc
-        gcc_extra_ver=$(expr "$($_cc --version 2>/dev/null | head -n1)" : '.*\((.*)\)')
-        _ident="llvm-gcc $($_cc -dumpversion 2>/dev/null) $gcc_extra_ver"
+        gcc_extra_ver=$(expr "$($_cc --version | head -n1)" : '.*\((.*)\)')
+        _ident="llvm-gcc $($_cc -dumpversion) $gcc_extra_ver"
        _depflags='-MMD -MF $(@:.o=.d) -MT $@'
        _cflags_speed='-O3'
        _cflags_size='-Os'
@@ -3566,14 +3479,8 @@ probe_cc(){
        gcc_pkg_ver=$(expr "$gcc_version" : '[^ ]* \(([^)]*)\)')
        gcc_ext_ver=$(expr "$gcc_version" : ".*$gcc_pkg_ver $gcc_basever \\(.*\\)")
        _ident=$(cleanws "gcc $gcc_basever $gcc_pkg_ver $gcc_ext_ver")
-        case $gcc_basever in
-            2*) _depflags='-MMD -MF $(@:.o=.d) -MT $@' ;;
-        esac
-        if [ "$first" = true ]; then
-            case $gcc_basever in
-                4.2*)
-                warn "gcc 4.2 is outdated and may miscompile FFmpeg. Please use a newer compiler." ;;
-            esac
+        if ! $_cc -dumpversion | grep -q '^2\.'; then
+            _depflags='-MMD -MF $(@:.o=.d) -MT $@'
        fi
        _cflags_speed='-O3'
        _cflags_size='-Os'
@@ -3625,7 +3532,7 @@ probe_cc(){
        _flags_filter=tms470_flags
    elif $_cc -v 2>&1 | grep -q clang; then
        _type=clang
-        _ident=$($_cc --version 2>/dev/null | head -n1)
+        _ident=$($_cc --version | head -n1)
        _depflags='-MMD -MF $(@:.o=.d) -MT $@'
        _cflags_speed='-O3'
        _cflags_size='-Os'
@@ -3688,16 +3595,16 @@ probe_cc(){
        _flags='-nologo -Qdiag-error:4044,10157'
        # -Qvec- -Qsimd- to prevent miscompilation, -GS, fp:precise for consistency
        # with MSVC which enables it by default.
-        _cflags='-D_USE_MATH_DEFINES -Qms0 -Qvec- -Qsimd- -GS -fp:precise'
+        _cflags='-D_USE_MATH_DEFINES -FIstdlib.h -Dstrtoll=_strtoi64 -Qms0 -Qvec- -Qsimd- -GS -fp:precise'
        disable stripping
-    elif $_cc -nologo- 2>&1 | grep -q Microsoft; then
+    elif $_cc 2>&1 | grep -q Microsoft; then
        _type=msvc
        _ident=$($_cc 2>&1 | head -n1)
        _DEPCMD='$(DEP$(1)) $(DEP$(1)FLAGS) $($(1)DEP_FLAGS) $< 2>&1 | awk '\''/including/ { sub(/^.*file: */, ""); gsub(/\\/, "/"); if (!match($$0, / /)) print "$@:", $$0 }'\'' > $(@:.o=.d)'
        _DEPFLAGS='$(CPPFLAGS) $(CFLAGS) -showIncludes -Zs'
        _cflags_speed="-O2"
        _cflags_size="-O1"
-        if $_cc -nologo- 2>&1 | grep -q Linker; then
+        if $_cc 2>&1 | grep -q Linker; then
            _ld_o='-out:$@'
        else
            _ld_o='-Fe$@'
@@ -3708,7 +3615,7 @@ probe_cc(){
        _ld_lib='lib%.a'
        _ld_path='-libpath:'
        _flags='-nologo'
-        _cflags='-D_USE_MATH_DEFINES -D_CRT_SECURE_NO_WARNINGS'
+        _cflags='-D_USE_MATH_DEFINES -D_CRT_SECURE_NO_WARNINGS -Dinline=__inline -FIstdlib.h -Dstrtoll=_strtoi64'
        disable stripping
    elif $_cc --version 2>/dev/null | grep -q ^cparser; then
        _type=cparser
@@ -3737,7 +3644,7 @@ set_ccvars(){
    fi
 }

-probe_cc cc "$cc" "true"
+probe_cc cc "$cc"
 cflags_filter=$_flags_filter
 cflags_speed=$_cflags_speed
 cflags_size=$_cflags_size
@@ -4043,7 +3950,7 @@ elif enabled mips; then
            check_cflags "-mtune=i6400 -mabi=64"
            check_ldflags "-mabi=64"
        ;;
-        loongson*)
+        loongson3*)
            disable mipsfpu
            disable mips32r2
            disable mips32r5
@@ -4051,24 +3958,14 @@ elif enabled mips; then
            disable mipsdspr1
            disable mipsdspr2
            disable msa
-            enable local_aligned_8 local_aligned_16 local_aligned_32
+            enable local_aligned_8 local_aligned_16
            enable simd_align_16
            enable fast_64bit
            enable fast_clz
            enable fast_cmov
            enable fast_unaligned
            disable aligned_stack
-            case $cpu in
-                loongson3*)
-                    cpuflags="-march=loongson3a -mhard-float -fno-expensive-optimizations"
-                ;;
-                loongson2e)
-                    cpuflags="-march=loongson2e -mhard-float -fno-expensive-optimizations"
-                ;;
-                loongson2f)
-                    cpuflags="-march=loongson2f -mhard-float -fno-expensive-optimizations"
-                ;;
-            esac
+            cpuflags="-march=loongson3a -mhard-float"
        ;;
        generic)
            disable mips32r5
@@ -4341,8 +4238,6 @@ case $target_os in
        else
            target_os=mingw32
        fi
-        decklink_outdev_extralibs="$decklink_outdev_extralibs -lole32 -loleaut32"
-        decklink_indev_extralibs="$decklink_indev_extralibs -lole32 -loleaut32"
        LIBTARGET=i386
        if enabled x86_64; then
            LIBTARGET="i386:x86-64"
@@ -4551,19 +4446,8 @@ probe_libc(){
        # in such new versions and producing binaries requiring windows 7.0.
        # Therefore explicitly set the default to XP unless the user has
        # set something else on the command line.
-        # Don't do this if WINAPI_FAMILY is set and is set to a non-desktop
-        # family. For these cases, configure is free to use any functions
-        # found in the SDK headers by default. (Alternatively, we could force
-        # _WIN32_WINNT to 0x0602 in that case.)
        check_${pfx}cpp_condition stdlib.h "defined(_WIN32_WINNT)" ||
-            { check_${pfx}cpp <<EOF && add_${pfx}cppflags -D_WIN32_WINNT=0x0502; }
-#ifdef WINAPI_FAMILY
-#include <winapifamily.h>
-#if !WINAPI_FAMILY_PARTITION(WINAPI_PARTITION_DESKTOP)
-#error not desktop
-#endif
-#endif
-EOF
+            add_${pfx}cppflags -D_WIN32_WINNT=0x0502
    elif check_${pfx}cpp_condition stddef.h "defined __KLIBC__"; then
        eval ${pfx}libc_type=klibc
    elif check_${pfx}cpp_condition sys/cdefs.h "defined __BIONIC__"; then
@@ -4792,8 +4676,6 @@ elif enabled alpha; then

 elif enabled arm; then

-    enabled msvc && check_cpp_condition stddef.h "defined _M_ARMT" && enable thumb
-
    check_cpp_condition stddef.h "defined __thumb__" && check_cc <<EOF && enable_weak thumb
 float func(float a, float b){ return a+b; }
 EOF
@@ -4802,9 +4684,7 @@ EOF

    if     check_cpp_condition stddef.h "defined __ARM_PCS_VFP"; then
        enable vfp_args
-    elif check_cpp_condition stddef.h "defined _M_ARM_FP && _M_ARM_FP >= 30"; then
-        enable vfp_args
-    elif ! check_cpp_condition stddef.h "defined __ARM_PCS || defined __SOFTFP__" && [ $target_os != darwin ]; then
+    elif ! check_cpp_condition stddef.h "defined __ARM_PCS || defined __SOFTFP__"; then
        case "${cross_prefix:-$cc}" in
            *hardfloat*)         enable vfp_args;   fpabi=vfp ;;
            *) check_ld "cc" <<EOF && enable vfp_args && fpabi=vfp || fpabi=soft ;;
@@ -4843,15 +4723,11 @@ EOF

 elif enabled mips; then

-    enabled loongson2 && check_inline_asm loongson2 '"dmult.g $8, $9, $10"'
-    enabled loongson3 && check_inline_asm loongson3 '"gsldxc1 $f0, 0($2, $3)"'
-    enabled mmi && check_inline_asm mmi '"punpcklhw $f0, $f0, $f0"'
-
    # Enable minimum ISA based on selected options
    if enabled mips64 && (enabled mipsdspr1 || enabled mipsdspr2); then
        add_cflags "-mips64r2"
        add_asflags "-mips64r2"
-    elif enabled mips64 && enabled mipsfpu && disabled loongson2 && disabled loongson3; then
+    elif enabled mips64 && enabled mipsfpu && disabled loongson3; then
        add_cflags "-mips64"
        add_asflags "-mips64"
    elif enabled mipsdspr1 || enabled mipsdspr2; then
@@ -4876,6 +4752,7 @@ elif enabled mips; then
     check_inline_asm mipsfpu   '"madd.d $f0, $f2, $f4, $f6"'
    enabled msa       && check_cflags "-mmsa" && check_ldflags "-mmsa" &&
     check_inline_asm msa       '"addvi.b $w0, $w1, 1"'
+    enabled loongson3 && check_inline_asm loongson3 '"gsldxc1 $f0, 0($2, $3)"'

    enabled mips32r5 && add_asflags "-mips32r5 -mfp64"
    enabled mips64r6 && add_asflags "-mips64r6 -mfp64"
@@ -5092,7 +4969,6 @@ check_func_headers conio.h kbhit
 check_func_headers io.h setmode
 check_func_headers lzo/lzo1x.h lzo1x_999_compress
 check_func_headers stdlib.h getenv
-check_func_headers sys/stat.h lstat

 check_func_headers windows.h CoTaskMemFree -lole32
 check_func_headers windows.h GetProcessAffinityMask
@@ -5101,7 +4977,6 @@ check_func_headers windows.h GetSystemTimeAsFileTime
 check_func_headers windows.h MapViewOfFile
 check_func_headers windows.h PeekNamedPipe
 check_func_headers windows.h SetConsoleTextAttribute
-check_func_headers windows.h SetConsoleCtrlHandler
 check_func_headers windows.h Sleep
 check_func_headers windows.h VirtualAlloc
 check_struct windows.h "CONDITION_VARIABLE" Ptr
@@ -5109,9 +4984,7 @@ check_func_headers glob.h glob
 enabled xlib &&
    check_func_headers "X11/Xlib.h X11/extensions/Xvlib.h" XvGetPortAttribute -lXv -lX11 -lXext

-check_header CoreServices/CoreServices.h
 check_header direct.h
-check_header dirent.h
 check_header dlfcn.h
 check_header d3d11.h
 check_header dxva.h
@@ -5134,7 +5007,6 @@ check_header valgrind/valgrind.h
 check_header vdpau/vdpau.h
 check_header vdpau/vdpau_x11.h
 check_header VideoDecodeAcceleration/VDADecoder.h
-check_header VideoToolbox/VideoToolbox.h
 check_header windows.h
 check_header X11/extensions/XvMClib.h
 check_header asm/types.h
@@ -5145,14 +5017,9 @@ check_lib2 "windows.h psapi.h" GetProcessMemoryInfo -lpsapi

 check_struct "sys/time.h sys/resource.h" "struct rusage" ru_maxrss

-check_type "windows.h dxva.h" "DXVA_PicParams_HEVC" -DWINAPI_FAMILY=WINAPI_FAMILY_DESKTOP_APP -D_CRT_BUILD_DESKTOP_APP=0
+check_type "windows.h dxva.h" "DXVA_PicParams_HEVC"
 check_type "windows.h d3d11.h" "ID3D11VideoDecoder"
-check_type "windows.h d3d11.h" "ID3D11VideoContext"
-check_type "d3d9.h dxva2api.h" DXVA2_ConfigPictureDecode -D_WIN32_WINNT=0x0602
-
-check_type "va/va.h" "VAPictureParameterBufferHEVC"
-
-check_type "vdpau/vdpau.h" "VdpPictureInfoHEVC"
+check_type "d3d9.h dxva2api.h" DXVA2_ConfigPictureDecode -D_WIN32_WINNT=0x0600

 if ! disabled w32threads && ! enabled pthreads; then
    check_func_headers "windows.h process.h" _beginthreadex &&
@@ -5195,7 +5062,6 @@ check_lib math.h sin -lm && LIBM="-lm"
 disabled crystalhd || check_lib libcrystalhd/libcrystalhd_if.h DtsCrystalHDVersion -lcrystalhd || disable crystalhd

 atan2f_args=2
-copysign_args=2
 ldexpf_args=2
 powf_args=2

@@ -5223,8 +5089,7 @@ enabled libcelt           && require libcelt celt/celt.h celt_decode -lcelt0 &&
                             { check_lib celt/celt.h celt_decoder_create_custom -lcelt0 ||
                               die "ERROR: libcelt must be installed and version must be >= 0.11.0."; }
 enabled libcaca           && require_pkg_config caca caca.h caca_create_canvas
-enabled libdcadec         && require_pkg_config dcadec libdcadec/dca_context.h dcadec_context_create &&
-                             check_struct libdcadec/dca_context.h "struct dcadec_exss_info" matrix_encoding
+enabled libdcadec         && require_pkg_config dcadec libdcadec/dca_context.h dcadec_context_create
 enabled libfaac           && require2 libfaac "stdint.h faac.h" faacEncGetVersion -lfaac
 enabled libfdk_aac        && { use_pkg_config fdk-aac "fdk-aac/aacenc_lib.h" aacEncOpen ||
                               { require libfdk_aac fdk-aac/aacenc_lib.h aacEncOpen -lfdk-aac &&
@@ -5240,7 +5105,6 @@ enabled libgsm            && { for gsm_hdr in "gsm.h" "gsm/gsm.h"; do
                                   check_lib "${gsm_hdr}" gsm_create -lgsm && break;
                               done || die "ERROR: libgsm not found"; }
 enabled libilbc           && require libilbc ilbc.h WebRtcIlbcfix_InitDecode -lilbc
-enabled libkvazaar        && require_pkg_config "kvazaar < 0.7.0" kvazaar.h kvz_api_get
 enabled libmfx            && require_pkg_config libmfx "mfx/mfxvideo.h" MFXInit
 enabled libmodplug        && require_pkg_config libmodplug libmodplug/modplug.h ModPlug_Load
 enabled libmp3lame        && require "libmp3lame >= 3.98.3" lame/lame.h lame_set_VBR_quality -lmp3lame
@@ -5261,8 +5125,7 @@ enabled libschroedinger   && require_pkg_config schroedinger-1.0 schroedinger/sc
 enabled libshine          && require_pkg_config shine shine/layer3.h shine_encode_buffer
 enabled libsmbclient      && { use_pkg_config smbclient libsmbclient.h smbc_init ||
                               require smbclient libsmbclient.h smbc_init -lsmbclient; }
-enabled libsnappy         && require snappy snappy-c.h snappy_compress -lsnappy
-enabled libsoxr           && require libsoxr soxr.h soxr_create -lsoxr && LIBSOXR="-lsoxr"
+enabled libsoxr           && require libsoxr soxr.h soxr_create -lsoxr
 enabled libssh            && require_pkg_config libssh libssh/sftp.h sftp_init
 enabled libspeex          && require_pkg_config speex speex/speex.h speex_decoder_init -lspeex
 enabled libstagefright_h264 && require_cpp libstagefright_h264 "binder/ProcessState.h media/stagefright/MetaData.h
@@ -5278,33 +5141,13 @@ enabled libvidstab        && require_pkg_config "vidstab >= 0.98" vid.stab/libvi
 enabled libvo_aacenc      && require libvo_aacenc vo-aacenc/voAAC.h voGetAACEncAPI -lvo-aacenc
 enabled libvo_amrwbenc    && require libvo_amrwbenc vo-amrwbenc/enc_if.h E_IF_init -lvo-amrwbenc
 enabled libvorbis         && require libvorbis vorbis/vorbisenc.h vorbis_info_init -lvorbisenc -lvorbis -logg
-
 enabled libvpx            && {
-    enabled libvpx_vp8_decoder && {
-        use_pkg_config "vpx >= 0.9.1" "vpx/vpx_decoder.h vpx/vp8dx.h" vpx_codec_vp8_dx ||
-            check_lib2 "vpx/vpx_decoder.h vpx/vp8dx.h" vpx_codec_dec_init_ver -lvpx ||
-                die "ERROR: libvpx decoder version must be >=0.9.1";
-    }
-    enabled libvpx_vp8_encoder && {
-        use_pkg_config "vpx >= 0.9.7" "vpx/vpx_encoder.h vpx/vp8cx.h" vpx_codec_vp8_cx ||
-            check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_enc_init_ver VP8E_SET_MAX_INTRA_BITRATE_PCT" -lvpx ||
-                die "ERROR: libvpx encoder version must be >=0.9.7";
-    }
-    enabled libvpx_vp9_decoder && {
-        use_pkg_config "vpx >= 1.3.0" "vpx/vpx_decoder.h vpx/vp8dx.h" vpx_codec_vp9_dx ||
-            check_lib2 "vpx/vpx_decoder.h vpx/vp8dx.h" "vpx_codec_vp9_dx" -lvpx ||
-                disable libvpx_vp9_decoder;
-    }
-    enabled libvpx_vp9_encoder && {
-        use_pkg_config "vpx >= 1.3.0" "vpx/vpx_encoder.h vpx/vp8cx.h" vpx_codec_vp9_cx ||
-            check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_vp9_cx VP9E_SET_AQ_MODE" -lvpx ||
-                disable libvpx_vp9_encoder;
-    }
-    if disabled_all libvpx_vp8_decoder libvpx_vp9_decoder libvpx_vp8_encoder libvpx_vp9_encoder; then
-        die "libvpx enabled but no supported decoders found"
-    fi
-}
-
+    enabled libvpx_vp8_decoder && { check_lib2 "vpx/vpx_decoder.h vpx/vp8dx.h" vpx_codec_dec_init_ver -lvpx ||
+                                    die "ERROR: libvpx decoder version must be >=0.9.1"; }
+    enabled libvpx_vp8_encoder && { check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_enc_init_ver VP8E_SET_MAX_INTRA_BITRATE_PCT" -lvpx ||
+                                    die "ERROR: libvpx encoder version must be >=0.9.7"; }
+    enabled libvpx_vp9_decoder && { check_lib2 "vpx/vpx_decoder.h vpx/vp8dx.h" "vpx_codec_vp9_dx" -lvpx || disable libvpx_vp9_decoder; }
+    enabled libvpx_vp9_encoder && { check_lib2 "vpx/vpx_encoder.h vpx/vp8cx.h" "vpx_codec_vp9_cx VP9E_SET_AQ_MODE" -lvpx || disable libvpx_vp9_encoder; } }
 enabled libwavpack        && require libwavpack wavpack/wavpack.h WavpackOpenFileOutput  -lwavpack
 enabled libwebp           && {
    enabled libwebp_encoder      && require_pkg_config "libwebp >= 0.2.0" webp/encode.h WebPGetEncoderVersion
@@ -5313,9 +5156,7 @@ enabled libx264           && { use_pkg_config x264 "stdint.h x264.h" x264_encode
                               { require libx264 x264.h x264_encoder_encode -lx264 &&
                                 warn "using libx264 without pkg-config"; } } &&
                             { check_cpp_condition x264.h "X264_BUILD >= 118" ||
-                               die "ERROR: libx264 must be installed and version must be >= 0.118."; } &&
-                             { check_cpp_condition x264.h "X264_MPEG2" &&
-                               enable libx262; }
+                               die "ERROR: libx264 must be installed and version must be >= 0.118."; }
 enabled libx265           && require_pkg_config x265 x265.h x265_api_get &&
                             { check_cpp_condition x265.h "X265_BUILD >= 57" ||
                               die "ERROR: libx265 version must be >= 57."; }
@@ -5390,9 +5231,6 @@ if ! disabled sdl; then
            disable sdl
        fi
    fi
-    if test $target_os = "mingw32"; then
-        sdl_libs="$sdl_libs -mconsole"
-    fi
 fi
 enabled sdl && add_cflags $sdl_cflags && add_extralibs $sdl_libs

@@ -5556,7 +5394,6 @@ check_disable_warning -Wno-pointer-sign
 check_ldflags -Wl,--warn-common
 check_ldflags -Wl,-rpath-link=libpostproc:libswresample:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil:libavresample
 enabled rpath && add_ldexeflags -Wl,-rpath,$libdir
-enabled rpath && add_ldlibflags -Wl,-rpath,$libdir
 test_ldflags -Wl,-Bsymbolic && append SHFLAGS -Wl,-Bsymbolic

 # add some strip flags
@@ -5723,30 +5560,8 @@ elif enabled_any msvc icl; then
    fi
    # msvcrt10 x64 incorrectly enables log2, only msvcrt12 (MSVC 2013) onwards actually has log2.
    check_cpp_condition crtversion.h "_VC_CRT_MAJOR_VERSION >= 12" || disable log2
-    # The CRT headers contain __declspec(restrict) in a few places, but if redefining
-    # restrict, this might break. MSVC 2010 and 2012 fail with __declspec(__restrict)
-    # (as it ends up if the restrict redefine is done before including stdlib.h), while
-    # MSVC 2013 and newer can handle it fine.
-    # If this declspec fails, force including stdlib.h before the restrict redefinition
-    # happens in config.h.
-    if [ $_restrict != restrict ]; then
-        check_cc <<EOF || add_cflags -FIstdlib.h
-__declspec($_restrict) void* foo(int);
-EOF
-    fi
-    check_func strtoll || add_cflags -Dstrtoll=_strtoi64
 fi

-for pfx in "" host_; do
-    varname=${pfx%_}cc_type
-    eval "type=\$$varname"
-    if [ $type = "msvc" ]; then
-        check_${pfx}cc <<EOF || add_${pfx}cflags -Dinline=__inline
-static inline int foo(int a) { return a; }
-EOF
-    fi
-done
-
 case $as_type in
    clang)
        add_asflags -Qunused-arguments
@@ -5849,7 +5664,6 @@ enabled removelogo_filter   && prepend avfilter_deps "avformat avcodec swscale"
 enabled resample_filter && prepend avfilter_deps "avresample"
 enabled sab_filter          && prepend avfilter_deps "swscale"
 enabled scale_filter    && prepend avfilter_deps "swscale"
-enabled scale2ref_filter    && prepend avfilter_deps "swscale"
 enabled showspectrum_filter && prepend avfilter_deps "avcodec"
 enabled smartblur_filter    && prepend avfilter_deps "swscale"
 enabled subtitles_filter    && prepend avfilter_deps "avformat avcodec"
@@ -5927,7 +5741,7 @@ if enabled mips; then
    echo "MIPS DSP R1 enabled       ${mipsdspr1-no}"
    echo "MIPS DSP R2 enabled       ${mipsdspr2-no}"
    echo "MIPS MSA enabled          ${msa-no}"
-    echo "LOONGSON MMI enabled      ${mmi-no}"
+    echo "LOONGSON3 enabled         ${loongson3-no}"
 fi
 if enabled ppc; then
    echo "AltiVec enabled           ${altivec-no}"
@@ -5958,10 +5772,6 @@ test -n "$random_seed" &&
    echo "random seed               ${random_seed}"
 echo

-echo "Enabled programs:"
-print_enabled '' $PROGRAM_LIST | print_3_columns
-echo
-
 echo "External libraries:"
 print_enabled '' $EXTERNAL_LIBRARY_LIST | print_3_columns
 echo
@@ -6052,7 +5862,6 @@ DEPWINDRES=$dep_cc
 DOXYGEN=$doxygen
 LDFLAGS=$LDFLAGS
 LDEXEFLAGS=$LDEXEFLAGS
-LDLIBFLAGS=$LDLIBFLAGS
 SHFLAGS=$(echo $($ldflags_filter $SHFLAGS))
 ASMSTRIPFLAGS=$ASMSTRIPFLAGS
 YASMFLAGS=$YASMFLAGS
@@ -6170,7 +5979,6 @@ enabled getenv || echo "#define getenv(x) NULL" >> $TMPH

 mkdir -p doc
 mkdir -p tests
-mkdir -p tests/api
 echo "@c auto-generated by configure" > doc/config.texi

 print_config ARCH_   "$config_files" $ARCH_LIST
@@ -6265,4 +6073,4 @@ pkgconfig_generate libavfilter   "FFmpeg audio/video filtering library" "$LIBAVF
 pkgconfig_generate libpostproc   "FFmpeg postprocessing library"        "$LIBPOSTPROC_VERSION"   ""
 pkgconfig_generate libavresample "Libav audio resampling library"       "$LIBAVRESAMPLE_VERSION" "$LIBM"
 pkgconfig_generate libswscale    "FFmpeg image rescaling library"       "$LIBSWSCALE_VERSION"    "$LIBM"
-pkgconfig_generate libswresample "FFmpeg audio resampling library"      "$LIBSWRESAMPLE_VERSION" "$LIBM $LIBSOXR"
+pkgconfig_generate libswresample "FFmpeg audio resampling library"      "$LIBSWRESAMPLE_VERSION" "$LIBM"
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,42 +15,6 @@ libavutil:     2014-08-09

 API changes, most recent first:

-------- 8< --------- FFmpeg 2.8 was cut here -------- 8< ---------
-
-2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h
-  Deprecate old VA-API context (vaapi_context) fields that were only
-  set and used by libavcodec. They are all managed internally now.
-
-2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h
-  Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that
-  indicates the nature of the underlying storage: a VA surface. This
-  yields the same value as AV_PIX_FMT_VAAPI_VLD.
-  Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO,
-  AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD.
-
-2015-08-02 - lavu 54.30.100 / 54.17.0
-  9ed59f1 / 7a7df34c -  Add av_blowfish_alloc().
-  a130ec9 / ae365453 -  Add av_rc4_alloc().
-  9ca1997 / 5d8bea3b -  Add av_xtea_alloc().
-  3cf08e9 / d9e8b47e -  Add av_des_alloc().
-
-2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h
-  94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
-  444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
-  29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
-              to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE.
-
-2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h
-  Add AV_PKT_DATA_QUALITY_STATS to export the quality value, PSNR, and pict_type
-  of an AVPacket.
-
-2015-07-16 - 8dad213 - lavc 56.49.100
-  Add av_codec_get_codec_properties(), FF_CODEC_PROPERTY_LOSSLESS
-  and FF_CODEC_PROPERTY_CLOSED_CAPTIONS
-
-2015-07-03 - d563e13 / 83212943 - lavu 54.28.100 / 56.15.0
-  Add av_version_info().
-
 -------- 8< --------- FFmpeg 2.7 was cut here -------- 8< ---------

 2015-06-04 - cc17b43 - lswr  1.2.100
@@ -728,9 +692,6 @@ API changes, most recent first:
   av_ripemd_update()
   av_ripemd_final()

-2013-06-10 - 82ef670 - lavu 52.35.101 - hmac.h
-  Add AV_HMAC_SHA224, AV_HMAC_SHA256, AV_HMAC_SHA384, AV_HMAC_SHA512
-
 2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
  Add av_realloc_array and av_reallocp_array

--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.

-PROJECT_NUMBER         = 2.8.2
+PROJECT_NUMBER         = 2.7.2

 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
--- a/doc/Makefile
+++ b/doc/Makefile
@@ -36,7 +36,7 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)

-DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
+DOC_EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE)     += avio_list_dir
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
 DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
--- a/doc/codecs.texi
+++ b/doc/codecs.texi
@@ -475,9 +475,6 @@ per-block quantization parameter (QP)
 motion vector
@item dct_coeff

-@item green_metadata
-display complexity metadata for the upcoming frame, GoP or for a given duration.
-
@item skip

@item startcode
@@ -1045,11 +1042,7 @@ Possible values:
@item color_primaries @var{integer} (@emph{decoding/encoding,video})
@item color_trc @var{integer} (@emph{decoding/encoding,video})
@item colorspace @var{integer} (@emph{decoding/encoding,video})
-
@item color_range @var{integer} (@emph{decoding/encoding,video})
-If used as input parameter, it serves as a hint to the decoder, which
-color_range the input has.
-
@item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})

@item log_level_offset @var{integer}
--- a/doc/decoders.texi
+++ b/doc/decoders.texi
@@ -25,13 +25,6 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.

-@section hevc
-
-HEVC / H.265 decoder.
-
-Note: the @option{skip_loop_filter} option has effect only at level
-@code{all}.
-
@section rawvideo

 Raw video decoder.
@@ -195,25 +188,6 @@ without this library.
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS

-@section dvbsub
-
-@subsection Options
-
-@table @option
-@item compute_clut
-@table @option
-@item -1
-Compute clut if no matching CLUT is in the stream.
-@item 0
-Never compute CLUT
-@item 1
-Always compute CLUT and override the one provided in the stream.
-@end table
-@item dvb_substream
-Selects the dvb substream, or all substreams if -1 which is default.
-
-@end table
-
@section dvdsub

 This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
--- a/doc/demuxers.texi
+++ b/doc/demuxers.texi
@@ -18,12 +18,6 @@ enabled demuxers.

 The description of some of the currently available demuxers follows.

-@section aa
-
-Audible Format 2, 3, and 4 demuxer.
-
-This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
-
@section applehttp

 Apple HTTP Live Streaming demuxer.
@@ -118,47 +112,6 @@ file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.

-@item @code{inpoint @var{timestamp}}
-In point of the file. When the demuxer opens the file it instantly seeks to the
-specified timestamp. Seeking is done so that all streams can be presented
-successfully at In point.
-
-This directive works best with intra frame codecs, because for non-intra frame
-ones you will usually get extra packets before the actual In point and the
-decoded content will most likely contain frames before In point too.
-
-For each file, packets before the file In point will have timestamps less than
-the calculated start timestamp of the file (negative in case of the first
-file), and the duration of the files (if not specified by the @code{duration}
-directive) will be reduced based on their specified In point.
-
-Because of potential packets before the specified In point, packet timestamps
-may overlap between two concatenated files.
-
-@item @code{outpoint @var{timestamp}}
-Out point of the file. When the demuxer reaches the specified decoding
-timestamp in any of the streams, it handles it as an end of file condition and
-skips the current and all the remaining packets from all streams.
-
-Out point is exclusive, which means that the demuxer will not output packets
-with a decoding timestamp greater or equal to Out point.
-
-This directive works best with intra frame codecs and formats where all streams
-are tightly interleaved. For non-intra frame codecs you will usually get
-additional packets with presentation timestamp after Out point therefore the
-decoded content will most likely contain frames after Out point too. If your
-streams are not tightly interleaved you may not get all the packets from all
-streams before Out point and you may only will be able to decode the earliest
-stream until Out point.
-
-The duration of the files (if not specified by the @code{duration}
-directive) will be reduced based on their specified Out point.
-
-@item @code{file_packet_metadata @var{key=value}}
-Metadata of the packets of the file. The specified metadata will be set for
-each file packet. You can specify this directive multiple times to add multiple
-metadata entries.
-
@item @code{stream}
 Introduce a stream in the virtual file.
 All subsequent stream-related directives apply to the last introduced
@@ -420,26 +373,13 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv

 MPEG-2 transport stream demuxer.

-This demuxer accepts the following options:
@table @option
-@item resync_size
-Set size limit for looking up a new synchronization. Default value is
-65536.

@item fix_teletext_pts
-Override teletext packet PTS and DTS values with the timestamps calculated
+Overrides teletext packet PTS and DTS values with the timestamps calculated
 from the PCR of the first program which the teletext stream is part of and is
 not discarded. Default value is 1, set this option to 0 if you want your
 teletext packet PTS and DTS values untouched.
-
-@item ts_packetsize
-Output option carrying the raw packet size in bytes.
-Show the detected raw packet size, cannot be set by the user.
-
-@item scan_all_pmts
-Scan and combine all PMTs. The value is an integer with value from -1
-to 1 (-1 means automatic setting, 1 means enabled, 0 means
-disabled). Default value is -1.
@end table

@section rawvideo
--- a/doc/developer.texi
+++ b/doc/developer.texi
@@ -543,10 +543,6 @@ tools/trasher, the noise bitstream filter, and
 should not crash, end in a (near) infinite loop, or allocate ridiculous
 amounts of memory when fed damaged data.

-@item
-Did you test your decoder or demuxer against sample files?
-Samples may be obtained at @url{http://samples.ffmpeg.org}.
-
@item
 Does the patch not mix functional and cosmetic changes?

@@ -637,10 +633,6 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.

-Everyone is welcome to review patches. Also if you are waiting for your patch
-to be reviewed, please consider helping to review other patches, that is a great
-way to get everyone's patches reviewed sooner.
-
@anchor{Regression tests}
@section Regression tests

--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -1342,30 +1342,6 @@ disabled
 A description of some of the currently available video encoders
 follows.

-@section jpeg2000
-
-The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
-option can be used to set the encoding quality. Lossless encoding
-can be selected with @code{-pred 1}.
-
-@subsection Options
-
-@table @option
-@item format
-Can be set to either @code{j2k} or @code{jp2} (the default) that
-makes it possible to store non-rgb pix_fmts.
-
-@end table
-
-@section snow
-
-@subsection Options
-
-@table @option
-@item iterative_dia_size
-dia size for the iterative motion estimation
-@end table
-
@section libtheora

 libtheora Theora encoder wrapper.
@@ -1440,153 +1416,113 @@ You need to explicitly configure the build with @code{--enable-libvpx}.

@subsection Options

-The following options are supported by the libvpx wrapper. The
-@command{vpxenc}-equivalent options or values are listed in parentheses
-for easy migration.
-
-To reduce the duplication of documentation, only the private options
-and some others requiring special attention are documented here. For
-the documentation of the undocumented generic options, see
-@ref{codec-options,,the Codec Options chapter}.
-
-To get more documentation of the libvpx options, invoke the command
-@command{ffmpeg -h encoder=libvpx}, @command{ffmpeg -h encoder=libvpx-vp9} or
-@command{vpxenc --help}. Further information is available in the libvpx API
-documentation.
+Mapping from FFmpeg to libvpx options with conversion notes in parentheses.

@table @option

-@item b (@emph{target-bitrate})
-Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
-expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in
-kilobits/s.
+@item threads
+g_threads

-@item g (@emph{kf-max-dist})
+@item profile
+g_profile

-@item keyint_min (@emph{kf-min-dist})
+@item vb
+rc_target_bitrate

-@item qmin (@emph{min-q})
+@item g
+kf_max_dist

-@item qmax (@emph{max-q})
+@item keyint_min
+kf_min_dist

-@item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz})
-Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
-specified in milliseconds, the libvpx wrapper converts this value as follows:
-@code{buf-sz = bufsize * 1000 / bitrate},
-@code{buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6}.
+@item qmin
+rc_min_quantizer

-@item rc_init_occupancy (@emph{buf-initial-sz})
-Set number of bits which should be loaded into the rc buffer before decoding
-starts. Note @command{vpxenc}'s option is specified in milliseconds, the libvpx
-wrapper converts this value as follows:
-@code{rc_init_occupancy * 1000 / bitrate}.
+@item qmax
+rc_max_quantizer

-@item undershoot-pct
-Set datarate undershoot (min) percentage of the target bitrate.
+@item bufsize, vb
+rc_buf_sz
+@code{(bufsize * 1000 / vb)}

-@item overshoot-pct
-Set datarate overshoot (max) percentage of the target bitrate.
+rc_buf_optimal_sz
+@code{(bufsize * 1000 / vb * 5 / 6)}

-@item skip_threshold (@emph{drop-frame})
+@item rc_init_occupancy, vb
+rc_buf_initial_sz
+@code{(rc_init_occupancy * 1000 / vb)}

-@item qcomp (@emph{bias-pct})
+@item rc_buffer_aggressivity
+rc_undershoot_pct

-@item maxrate (@emph{maxsection-pct})
-Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
-percentage of the target bitrate, the libvpx wrapper converts this value as
-follows: @code{(maxrate * 100 / bitrate)}.
+@item skip_threshold
+rc_dropframe_thresh

-@item minrate (@emph{minsection-pct})
-Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
-percentage of the target bitrate, the libvpx wrapper converts this value as
-follows: @code{(minrate * 100 / bitrate)}.
+@item qcomp
+rc_2pass_vbr_bias_pct

-@item minrate, maxrate, b @emph{end-usage=cbr}
-@code{(minrate == maxrate == bitrate)}.
+@item maxrate, vb
+rc_2pass_vbr_maxsection_pct
+@code{(maxrate * 100 / vb)}

-@item crf (@emph{end-usage=cq}, @emph{cq-level})
+@item minrate, vb
+rc_2pass_vbr_minsection_pct
+@code{(minrate * 100 / vb)}

-@item quality, deadline (@emph{deadline})
-@table @samp
-@item best
-Use best quality deadline. Poorly named and quite slow, this option should be
-avoided as it may give worse quality output than good.
-@item good
-Use good quality deadline. This is a good trade-off between speed and quality
-when used with the @option{cpu-used} option.
-@item realtime
-Use realtime quality deadline.
+@item minrate, maxrate, vb
+@code{VPX_CBR}
+@code{(minrate == maxrate == vb)}
+
+@item crf
+@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
+
+@item quality
+@table @option
+@item @var{best}
+@code{VPX_DL_BEST_QUALITY}
+@item @var{good}
+@code{VPX_DL_GOOD_QUALITY}
+@item @var{realtime}
+@code{VPX_DL_REALTIME}
@end table

-@item speed, cpu-used (@emph{cpu-used})
-Set quality/speed ratio modifier. Higher values speed up the encode at the cost
-of quality.
+@item speed
+@code{VP8E_SET_CPUUSED}

-@item nr (@emph{noise-sensitivity})
+@item nr
+@code{VP8E_SET_NOISE_SENSITIVITY}

-@item static-thresh
-Set a change threshold on blocks below which they will be skipped by the
-encoder.
+@item mb_threshold
+@code{VP8E_SET_STATIC_THRESHOLD}

-@item slices (@emph{token-parts})
-Note that FFmpeg's @option{slices} option gives the total number of partitions,
-while @command{vpxenc}'s @option{token-parts} is given as
-@code{log2(partitions)}.
+@item slices
+@code{VP8E_SET_TOKEN_PARTITIONS}

@item max-intra-rate
-Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0
-means unlimited.
+@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}

@item force_key_frames
@code{VPX_EFLAG_FORCE_KF}

@item Alternate reference frame related
@table @option
-@item auto-alt-ref
-Enable use of alternate reference frames (2-pass only).
-@item arnr-max-frames
-Set altref noise reduction max frame count.
-@item arnr-type
-Set altref noise reduction filter type: backward, forward, centered.
-@item arnr-strength
-Set altref noise reduction filter strength.
-@item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
-Set number of frames to look ahead for frametype and ratecontrol.
+@item vp8flags altref
+@code{VP8E_SET_ENABLEAUTOALTREF}
+@item @var{arnr_max_frames}
+@code{VP8E_SET_ARNR_MAXFRAMES}
+@item @var{arnr_type}
+@code{VP8E_SET_ARNR_TYPE}
+@item @var{arnr_strength}
+@code{VP8E_SET_ARNR_STRENGTH}
+@item @var{rc_lookahead}
+g_lag_in_frames
@end table

-@item error-resilient
-Enable error resiliency features.
+@item vp8flags error_resilient
+g_error_resilient

-@item VP9-specific options
-@table @option
-@item lossless
-Enable lossless mode.
-@item tile-columns
-Set number of tile columns to use. Note this is given as
-@code{log2(tile_columns)}. For example, 8 tile columns would be requested by
-setting the @option{tile-columns} option to 3.
-@item tile-rows
-Set number of tile rows to use. Note this is given as @code{log2(tile_rows)}.
-For example, 4 tile rows would be requested by setting the @option{tile-rows}
-option to 2.
-@item frame-parallel
-Enable frame parallel decodability features.
-@item aq-mode
-Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
-cyclic refresh).
-@item colorspace @emph{color-space}
-Set input color space. The VP9 bitstream supports signaling the following
-colorspaces:
-@table @option
-@item @samp{rgb} @emph{sRGB}
-@item @samp{bt709} @emph{bt709}
-@item @samp{unspecified} @emph{unknown}
-@item @samp{bt470bg} @emph{bt601}
-@item @samp{smpte170m} @emph{smpte170}
-@item @samp{smpte240m} @emph{smpte240}
-@item @samp{bt2020_ncl} @emph{bt2020}
-@end table
-@end table
+@item aq_mode
+@code{VP9E_SET_AQ_MODE}

@end table

@@ -2324,30 +2260,6 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.

-@section libkvazaar
-
-Kvazaar H.265/HEVC encoder.
-
-Requires the presence of the libkvazaar headers and library during
-configuration. You need to explicitly configure the build with
-@option{--enable-libkvazaar}.
-
-@subsection Options
-
-@table @option
-
-@item b
-Set target video bitrate in bit/s and enable rate control.
-
-@item threads
-Set number of encoding threads.
-
-@item kvazaar-params
-Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
-by commas (,). See kvazaar documentation for a list of options.
-
-@end table
-
@c man end VIDEO ENCODERS

@chapter Subtitles Encoders
--- a/doc/examples/Makefile
+++ b/doc/examples/Makefile
@@ -11,14 +11,13 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)

-EXAMPLES=       avio_dir_cmd                       \
+EXAMPLES=       avio_list_dir                      \
                avio_reading                       \
                decoding_encoding                  \
                demuxing_decoding                  \
                extract_mvs                        \
                filtering_video                    \
                filtering_audio                    \
-                http_multiclient                   \
                metadata                           \
                muxing                             \
                remuxing                           \
--- a/doc/examples/avio_list_dir.c
+++ b/doc/examples/avio_list_dir.c
@@ -54,13 +54,28 @@ static const char *type_string(int type)
    return "<UNKNOWN>";
 }

-static int list_op(const char *input_dir)
+int main(int argc, char *argv[])
 {
+    const char *input_dir = NULL;
    AVIODirEntry *entry = NULL;
    AVIODirContext *ctx = NULL;
    int cnt, ret;
    char filemode[4], uid_and_gid[20];

+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc != 2) {
+        fprintf(stderr, "usage: %s input_dir\n"
+                "API example program to show how to list files in directory "
+                "accessed through AVIOContext.\n", argv[0]);
+        return 1;
+    }
+    input_dir = argv[1];
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+    avformat_network_init();
+
    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
        goto fail;
@@ -99,81 +114,6 @@ static int list_op(const char *input_dir)

  fail:
    avio_close_dir(&ctx);
-    return ret;
-}
-
-static int del_op(const char *url)
-{
-    int ret = avpriv_io_delete(url);
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
-    return ret;
-}
-
-static int move_op(const char *src, const char *dst)
-{
-    int ret = avpriv_io_move(src, dst);
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
-    return ret;
-}
-
-
-static void usage(const char *program_name)
-{
-    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
-            "API example program to show how to manipulate resources "
-            "accessed through AVIOContext.\n"
-            "OPERATIONS:\n"
-            "list      list content of the directory\n"
-            "move      rename content in directory\n"
-            "del       delete content in directory\n",
-            program_name);
-}
-
-int main(int argc, char *argv[])
-{
-    const char *op = NULL;
-    int ret;
-
-    av_log_set_level(AV_LOG_DEBUG);
-
-    if (argc < 2) {
-        usage(argv[0]);
-        return 1;
-    }
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-    avformat_network_init();
-
-    op = argv[1];
-    if (strcmp(op, "list") == 0) {
-        if (argc < 3) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = list_op(argv[2]);
-        }
-    } else if (strcmp(op, "del") == 0) {
-        if (argc < 3) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = del_op(argv[2]);
-        }
-    } else if (strcmp(op, "move") == 0) {
-        if (argc < 4) {
-            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
-            ret = AVERROR(EINVAL);
-        } else {
-            ret = move_op(argv[2], argv[3]);
-        }
-    } else {
-        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
-        ret = AVERROR(EINVAL);
-    }
-
    avformat_network_deinit();

    return ret < 0 ? 1 : 0;
--- a/doc/examples/decoding_encoding.c
+++ b/doc/examples/decoding_encoding.c
@@ -245,7 +245,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;

@@ -521,7 +521,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
        /* the picture is allocated by the decoder, no need to free it */
        snprintf(buf, sizeof(buf), outfilename, *frame_count);
        pgm_save(frame->data[0], frame->linesize[0],
-                 frame->width, frame->height, buf);
+                 avctx->width, avctx->height, buf);
        (*frame_count)++;
    }
    if (pkt->data) {
@@ -538,13 +538,13 @@ static void video_decode_example(const char *outfilename, const char *filename)
    int frame_count;
    FILE *f;
    AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;

    av_init_packet(&avpkt);

    /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
-    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);

    printf("Decode video file %s to %s\n", filename, outfilename);

@@ -561,8 +561,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
        exit(1);
    }

-    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
-        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
+    if(codec->capabilities&CODEC_CAP_TRUNCATED)
+        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */

    /* For some codecs, such as msmpeg4 and mpeg4, width and height
       MUST be initialized there because this information is not
--- a/doc/examples/filtering_video.c
+++ b/doc/examples/filtering_video.c
@@ -38,10 +38,7 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>

-const char *filter_descr = "scale=78:24,transpose=cclock";
-/* other way:
-   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
- */
+const char *filter_descr = "scale=78:24";

 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
--- a/doc/examples/http_multiclient.c
+++ b/doc/examples/http_multiclient.c
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2015 Stephan Holljes
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat multi-client network API usage example.
- *
- * @example http_multiclient.c
- * This example will serve a file without decoding or demuxing it over http.
- * Multiple clients can connect and will receive the same file.
- */
-
-#include <libavformat/avformat.h>
-#include <libavutil/opt.h>
-#include <unistd.h>
-
-void process_client(AVIOContext *client, const char *in_uri)
-{
-    AVIOContext *input = NULL;
-    uint8_t buf[1024];
-    int ret, n, reply_code;
-    char *resource = NULL;
-    while ((ret = avio_handshake(client)) > 0) {
-        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
-        // check for strlen(resource) is necessary, because av_opt_get()
-        // may return empty string.
-        if (resource && strlen(resource))
-            break;
-    }
-    if (ret < 0)
-        goto end;
-    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
-    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
-        reply_code = 200;
-    } else {
-        reply_code = AVERROR_HTTP_NOT_FOUND;
-    }
-    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
-        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
-        goto end;
-    }
-    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
-
-    while ((ret = avio_handshake(client)) > 0);
-
-    if (ret < 0)
-        goto end;
-
-    fprintf(stderr, "Handshake performed.\n");
-    if (reply_code != 200)
-        goto end;
-    fprintf(stderr, "Opening input file.\n");
-    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
-        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
-               av_err2str(ret));
-        goto end;
-    }
-    for(;;) {
-        n = avio_read(input, buf, sizeof(buf));
-        if (n < 0) {
-            if (n == AVERROR_EOF)
-                break;
-            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
-                   av_err2str(n));
-            break;
-        }
-        avio_write(client, buf, n);
-        avio_flush(client);
-    }
-end:
-    fprintf(stderr, "Flushing client\n");
-    avio_flush(client);
-    fprintf(stderr, "Closing client\n");
-    avio_close(client);
-    fprintf(stderr, "Closing input\n");
-    avio_close(input);
-}
-
-int main(int argc, char **argv)
-{
-    av_log_set_level(AV_LOG_TRACE);
-    AVDictionary *options = NULL;
-    AVIOContext *client = NULL, *server = NULL;
-    const char *in_uri, *out_uri;
-    int ret, pid;
-    if (argc < 3) {
-        printf("usage: %s input http://hostname[:port]\n"
-               "API example program to serve http to multiple clients.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    in_uri = argv[1];
-    out_uri = argv[2];
-
-    av_register_all();
-    avformat_network_init();
-
-    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
-        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
-        return ret;
-    }
-    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
-        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
-        return ret;
-    }
-    fprintf(stderr, "Entering main loop.\n");
-    for(;;) {
-        if ((ret = avio_accept(server, &client)) < 0)
-            goto end;
-        fprintf(stderr, "Accepted client, forking process.\n");
-        // XXX: Since we don't reap our children and don't ignore signals
-        //      this produces zombie processes.
-        pid = fork();
-        if (pid < 0) {
-            perror("Fork failed");
-            ret = AVERROR(errno);
-            goto end;
-        }
-        if (pid == 0) {
-            fprintf(stderr, "In child.\n");
-            process_client(client, in_uri);
-            avio_close(server);
-            exit(0);
-        }
-        if (pid > 0)
-            avio_close(client);
-    }
-end:
-    avio_close(server);
-    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-    return 0;
-}
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -172,7 +172,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,

    /* Some formats want stream headers to be separate. */
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
 }

 /**************************************************************/
@@ -230,7 +230,7 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
    /* increment frequency by 110 Hz per second */
    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

-    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
+    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
        nb_samples = 10000;
    else
        nb_samples = c->frame_size;
--- a/doc/examples/qsvdec.c
+++ b/doc/examples/qsvdec.c
@@ -405,7 +405,7 @@ int main(int argc, char **argv)
    decoder_ctx->codec_id = AV_CODEC_ID_H264;
    if (video_st->codec->extradata_size) {
        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
-                                            AV_INPUT_BUFFER_PADDING_SIZE);
+                                            FF_INPUT_BUFFER_PADDING_SIZE);
        if (!decoder_ctx->extradata) {
            ret = AVERROR(ENOMEM);
            goto finish;
--- a/doc/examples/remuxing.c
+++ b/doc/examples/remuxing.c
@@ -101,7 +101,7 @@ int main(int argc, char **argv)
        }
        out_stream->codec->codec_tag = 0;
        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
    }
    av_dump_format(ofmt_ctx, 0, out_filename, 1);

--- a/doc/examples/transcode_aac.c
+++ b/doc/examples/transcode_aac.c
@@ -192,7 +192,7 @@ static int open_output_file(const char *filename,
     * Mark the encoder so that it behaves accordingly.
     */
    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;

    /** Open the encoder for the audio stream to use it later. */
    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
--- a/doc/examples/transcoding.c
+++ b/doc/examples/transcoding.c
@@ -161,7 +161,7 @@ static int open_output_file(const char *filename)
        }

        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;

    }
    av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -449,7 +449,7 @@ static int flush_encoder(unsigned int stream_index)
    int got_frame;

    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                AV_CODEC_CAP_DELAY))
+                CODEC_CAP_DELAY))
        return 0;

    while (1) {
--- a/doc/ffmpeg.texi
+++ b/doc/ffmpeg.texi
@@ -280,15 +280,13 @@ data read from the input file.
 When used as an output option (before an output filename), stop writing the
 output after its duration reaches @var{duration}.

-@var{duration} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.

 -to and -t are mutually exclusive and -t has priority.

@item -to @var{position} (@emph{output})
 Stop writing the output at @var{position}.
-@var{position} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.

 -to and -t are mutually exclusive and -t has priority.

@@ -297,8 +295,8 @@ Set the file size limit, expressed in bytes.

@item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. Note that in most formats it is not possible to seek exactly,
-so @command{ffmpeg} will seek to the closest seek point before @var{position}.
+@var{position}. Note the in most formats it is not possible to seek exactly, so
+@command{ffmpeg} will seek to the closest seek point before @var{position}.
 When transcoding and @option{-accurate_seek} is enabled (the default), this
 extra segment between the seek point and @var{position} will be decoded and
 discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
@@ -307,13 +305,7 @@ will be preserved.
 When used as an output option (before an output filename), decodes but discards
 input until the timestamps reach @var{position}.

-@var{position} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-@item -sseof @var{position} (@emph{input/output})
-
-Like the @code{-ss} option but relative to the "end of file". That is negative
-values are earlier in the file, 0 is at EOF.
+@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.

@item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
@@ -328,7 +320,7 @@ the time duration specified in @var{offset}.
@item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.

-@var{date} must be a date specification,
+@var{date} must be a time duration specification,
 see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.

@item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
@@ -698,10 +690,6 @@ is not specified, the value of the @var{DISPLAY} environment variable is used
 For DXVA2, this option should contain the number of the display adapter to use.
 If this option is not specified, the default adapter is used.
@end table
-
-@item -hwaccels
-List all hardware acceleration methods supported in this build of ffmpeg.
-
@end table

@section Audio Options
@@ -1205,9 +1193,9 @@ The option is intended for cases where features are needed that cannot be
 specified to @command{ffserver} but can be to @command{ffmpeg}.

@item -sdp_file @var{file} (@emph{global})
-Print sdp information for an output stream to @var{file}.
+Print sdp information to @var{file}.
 This allows dumping sdp information when at least one output isn't an
-rtp stream. (Requires at least one of the output formats to be rtp).
+rtp stream.

@item -discard (@emph{input})
 Allows discarding specific streams or frames of streams at the demuxer.
@@ -1311,6 +1299,47 @@ If no such file is found, then ffmpeg will search for a file named

@c man end OPTIONS

+@chapter Tips
+@c man begin TIPS
+
+@itemize
+@item
+For streaming at very low bitrates, use a low frame rate
+and a small GOP size. This is especially true for RealVideo where
+the Linux player does not seem to be very fast, so it can miss
+frames. An example is:
+
+@example
+ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
+@end example
+
+@item
+The parameter 'q' which is displayed while encoding is the current
+quantizer. The value 1 indicates that a very good quality could
+be achieved. The value 31 indicates the worst quality. If q=31 appears
+too often, it means that the encoder cannot compress enough to meet
+your bitrate. You must either increase the bitrate, decrease the
+frame rate or decrease the frame size.
+
+@item
+If your computer is not fast enough, you can speed up the
+compression at the expense of the compression ratio. You can use
+'-me zero' to speed up motion estimation, and '-g 0' to disable
+motion estimation completely (you have only I-frames, which means it
+is about as good as JPEG compression).
+
+@item
+To have very low audio bitrates, reduce the sampling frequency
+(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
+
+@item
+To have a constant quality (but a variable bitrate), use the option
+'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
+quality).
+
+@end itemize
+@c man end TIPS
+
@chapter Examples
@c man begin EXAMPLES

--- a/doc/ffplay.texi
+++ b/doc/ffplay.texi
@@ -47,17 +47,9 @@ Disable video.
@item -sn
 Disable subtitles.
@item -ss @var{pos}
-Seek to @var{pos}. Note that in most formats it is not possible to seek
-exactly, so @command{ffplay} will seek to the nearest seek point to
-@var{pos}.
-
-@var{pos} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+Seek to a given position in seconds.
@item -t @var{duration}
-Play @var{duration} seconds of audio/video.
-
-@var{duration} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+play <duration> seconds of audio/video
@item -bytes
 Seek by bytes.
@item -nodisp
--- a/doc/fftools-common-opts.texi
+++ b/doc/fftools-common-opts.texi
@@ -36,10 +36,8 @@ Possible forms of stream specifiers are:
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
 thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
-for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
-streams, 'V' only matches video streams which are not attached pictures, video
-thumbnails or cover arts.  If @var{stream_index} is given, then it matches
+@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
+'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
 stream number @var{stream_index} of this type. Otherwise, it matches all
 streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
--- a/doc/filter_design.txt
+++ b/doc/filter_design.txt
@@ -98,7 +98,7 @@ Buffer references ownership and permissions
    The AVFilterLink structure has a few AVFilterBufferRef fields. The
    cur_buf and out_buf were used with the deprecated
    start_frame/draw_slice/end_frame API and should no longer be used.
-    src_buf and partial_buf are used by libavfilter internally
+    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
    and must not be accessed by filters.

  Reference permissions
--- a/doc/filters.texi
+++ b/doc/filters.texi
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -145,14 +145,6 @@ x265 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
@end float

-@section kvazaar
-
-FFmpeg can make use of the kvazaar library for HEVC encoding.
-
-Go to @url{https://github.com/ultravideo/kvazaar} and follow the
-instructions for installing the library. Then pass
-@code{--enable-libkvazaar} to configure to enable it.
-
@section libilbc

 iLBC is a narrowband speech codec that has been made freely available
@@ -200,17 +192,6 @@ end user having AviSynth or AvxSynth installed - they'll only need to be
 installed to use AviSynth scripts (obviously).
@end float

-@section Intel QuickSync Video
-
-FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
-of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
-dispatcher, which loads the actual decoding libraries.
-
-The dispatcher is open source and can be downloaded from
-@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
-with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
-locate the dispatcher's @code{.pc} files.
-

@chapter Supported File Formats, Codecs or Features

@@ -226,10 +207,6 @@ library:
@item 4xm                       @tab   @tab X
    @tab 4X Technologies format, used in some games.
@item 8088flex TMV              @tab   @tab X
-@item AAX                       @tab   @tab X
-    @tab Audible Enhanced Audio format, used in audiobooks.
-@item AA                        @tab   @tab X
-    @tab Audible Format 2, 3, and 4, used in audiobooks.
@item ACT Voice                 @tab   @tab X
    @tab contains G.729 audio
@item Adobe Filmstrip           @tab X @tab X
@@ -266,8 +243,6 @@ library:
    @tab Used in Z and Z95 games.
@item Brute Force & Ignorance   @tab   @tab X
    @tab Used in the game Flash Traffic: City of Angels.
-@item BFSTM                     @tab   @tab X
-    @tab Audio format used on the Nintendo WiiU (based on BRSTM).
@item BRSTM                     @tab   @tab X
    @tab Audio format used on the Nintendo Wii.
@item BWF                       @tab X @tab X
@@ -298,7 +273,6 @@ library:
@item Deluxe Paint Animation    @tab   @tab X
@item DFA                       @tab   @tab X
    @tab This format is used in Chronomaster game
-@item DirectDraw Surface        @tab   @tab X
@item DSD Stream File (DSF)     @tab   @tab X
@item DV video                  @tab X @tab X
@item DXA                       @tab   @tab X
@@ -503,7 +477,6 @@ library:
    @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
@item True Audio                @tab   @tab X
@item VC-1 test bitstream       @tab X @tab X
-@item Vidvox Hap                @tab X @tab X
@item Vivo                      @tab   @tab X
@item WAV                       @tab X @tab X
@item WavPack                   @tab X @tab X
@@ -690,8 +663,6 @@ following image formats are supported:
    @tab Sorenson H.263 used in Flash
@item Forward Uncompressed   @tab     @tab  X
@item Fraps                  @tab     @tab  X
-@item Go2Meeting             @tab     @tab  X
-    @tab fourcc: G2M2, G2M3
@item Go2Webinar             @tab     @tab  X
    @tab fourcc: G2M4
@item H.261                  @tab  X  @tab  X
@@ -700,7 +671,7 @@ following image formats are supported:
@item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
    @tab encoding supported through external library libx264 and OpenH264
@item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through external library libx265 and libkvazaar
+    @tab encoding supported through the external library libx265
@item HNM version 4          @tab     @tab  X
@item HuffYUV                @tab  X  @tab  X
@item HuffYUV FFmpeg variant @tab  X  @tab  X
@@ -865,7 +836,7 @@ following image formats are supported:
@item Name @tab Encoding @tab Decoding @tab Comments
@item 8SVX exponential       @tab     @tab  X
@item 8SVX fibonacci         @tab     @tab  X
-@item AAC+                   @tab  E  @tab  IX
+@item AAC+                   @tab  E  @tab  X
    @tab encoding supported through external library libaacplus
@item AAC                    @tab  E  @tab  X
    @tab encoding supported through external library libfaac and libvo-aacenc
@@ -905,7 +876,7 @@ following image formats are supported:
@item ADPCM MS IMA           @tab  X  @tab  X
@item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
@item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
-@item ADPCM Nintendo THP  @tab     @tab  X
+@item ADPCM Nintendo Gamecube THP  @tab     @tab  X
@item ADPCM QT IMA           @tab  X  @tab  X
@item ADPCM SEGA CRI ADX     @tab  X  @tab  X
    @tab Used in Sega Dreamcast games.
@@ -983,8 +954,8 @@ following image formats are supported:
@item Musepack SV8           @tab     @tab  X
@item Nellymoser Asao        @tab  X  @tab  X
@item On2 AVC (Audio for Video Codec) @tab     @tab  X
-@item Opus                   @tab  E  @tab  X
-    @tab encoding supported through external library libopus
+@item Opus                   @tab  E  @tab  E
+    @tab supported through external library libopus
@item PCM A-law              @tab  X  @tab  X
@item PCM mu-law             @tab  X  @tab  X
@item PCM signed 8-bit planar  @tab  X  @tab  X
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -1,10 +1,10 @@
 \input texinfo @c -*- texinfo -*-
@documentencoding UTF-8

-@settitle Using Git to develop FFmpeg
+@settitle Using git to develop FFmpeg

@titlepage
-@center @titlefont{Using Git to develop FFmpeg}
+@center @titlefont{Using git to develop FFmpeg}
@end titlepage

@top
@@ -13,9 +13,9 @@

@chapter Introduction

-This document aims in giving some quick references on a set of useful Git
+This document aims in giving some quick references on a set of useful git
 commands. You should always use the extensive and detailed documentation
-provided directly by Git:
+provided directly by git:

@example
 git --help
@@ -32,21 +32,22 @@ man git-<command>
 shows information about the subcommand <command>.

 Additional information could be found on the
-@url{http://gitref.org, Git Reference} website.
+@url{http://gitref.org, Git Reference} website

 For more information about the Git project, visit the
-@url{http://git-scm.com/, Git website}.
+
+@url{http://git-scm.com/, Git website}

 Consult these resources whenever you have problems, they are quite exhaustive.

 What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines to ease the contribution to the project.
+guidelines to ease the contribution to the project

@chapter Basics Usage

-@section Get Git
+@section Get GIT

-You can get Git from @url{http://git-scm.com/}
+You can get git from @url{http://git-scm.com/}
 Most distribution and operating system provide a package for it.


@@ -107,7 +108,7 @@ git add [-A] <filename/dirname>
 git rm [-r] <filename/dirname>
@end example

-Git needs to get notified of all changes you make to your working
+GIT needs to get notified of all changes you make to your working
 directory that makes files appear or disappear.
 Line moves across files are automatically tracked.

@@ -127,8 +128,8 @@ will show all local modifications in your working directory as unified diff.
 git log <filename(s)>
@end example

-You may also use the graphical tools like @command{gitview} or @command{gitk}
-or the web interface available at @url{http://source.ffmpeg.org/}.
+You may also use the graphical tools like gitview or gitk or the web
+interface available at http://source.ffmpeg.org/

@section Checking source tree status

@@ -149,7 +150,6 @@ git diff --check
 to double check your changes before committing them to avoid trouble later
 on. All experienced developers do this on each and every commit, no matter
 how small.
-
 Every one of them has been saved from looking like a fool by this many times.
 It's very easy for stray debug output or cosmetic modifications to slip in,
 please avoid problems through this extra level of scrutiny.
@@ -172,14 +172,14 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
@end example

-Make sure you have told Git your name and email address
+Make sure you have told git your name and email address

@example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
@end example

-Use @option{--global} to set the global configuration for all your Git checkouts.
+Use @var{--global} to set the global configuration for all your git checkouts.

 Git will select the changes to the files for commit. Optionally you can use
 the interactive or the patch mode to select hunk by hunk what should be
@@ -210,7 +210,7 @@ include filenames in log messages, Git provides that information.

 Possibly make the commit message have a terse, descriptive first line, an
 empty line and then a full description. The first line will be used to name
-the patch by @command{git format-patch}.
+the patch by git format-patch.

@section Preparing a patchset

@@ -326,12 +326,10 @@ faulty commit disappear from the history.
@section Pushing changes to remote trees

@example
-git push origin master --dry-run
+git push
@end example

-Will simulate a push of the local master branch to the default remote
-(@var{origin}). And list which branches and ranges or commits would have been
-pushed.
+Will push the changes to the default remote (@var{origin}).
 Git will prevent you from pushing changes if the local and remote trees are
 out of sync. Refer to @ref{Updating the source tree to the latest revision}.

@@ -352,24 +350,23 @@ branches matching the local ones.

@section Finding a specific svn revision

-Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits
+Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
 based on a regular expression. see man gitrevisions

@example
 git show :/'as revision 23456'
@end example

-will show the svn changeset @samp{r23456}. With older Git versions searching in
+will show the svn changeset @var{r23456}. With older git versions searching in
 the @command{git log} output is the easiest option (especially if a pager with
 search capabilities is used).
-
 This commit can be checked out with

@example
 git checkout -b svn_23456 :/'as revision 23456'
@end example

-or for Git < 1.7.1 with
+or for git < 1.7.1 with

@example
 git checkout -b svn_23456 $SHA1
@@ -378,7 +375,7 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.


-@chapter Pre-push checklist
+@chapter pre-push checklist

 Once you have a set of commits that you feel are ready for pushing,
 work through the following checklist to doublecheck everything is in
@@ -389,7 +386,7 @@ Apply your common sense, but if in doubt, err on the side of caution.
 First, make sure that the commits and branches you are going to push
 match what you want pushed and that nothing is missing, extraneous or
 wrong. You can see what will be pushed by running the git push command
-with @option{--dry-run} first. And then inspecting the commits listed with
+with --dry-run first. And then inspecting the commits listed with
@command{git log -p 1234567..987654}. The @command{git status} command
 may help in finding local changes that have been forgotten to be added.

@@ -398,7 +395,7 @@ Next let the code pass through a full run of our testsuite.
@itemize
@item @command{make distclean}
@item @command{/path/to/ffmpeg/configure}
-@item @command{make fate}
+@item @command{make check}
@item if fate fails due to missing samples run @command{make fate-rsync} and retry
@end itemize

@@ -416,5 +413,5 @@ recommended.

@chapter Server Issues

-Contact the project admins at @email{root@@ffmpeg.org} if you have technical
-problems with the Git server.
+Contact the project admins @email{root@@ffmpeg.org} if you have technical
+problems with the GIT server.
--- a/doc/indevs.texi
+++ b/doc/indevs.texi
@@ -51,18 +51,6 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
 For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}

-@subsection Options
-
-@table @option
-
-@item sample_rate
-Set the sample rate in Hz. Default is 48000.
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
@section avfoundation

 AVFoundation input device.
@@ -126,19 +114,6 @@ und the first one in this list is used instead. Available pixel formats are:
 bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
 yuv420p, nv12, yuyv422, gray}

-@item -framerate
-Set the grabbing frame rate. Default is @code{ntsc}, corresponding to a
-frame rate of @code{30000/1001}.
-
-@item -video_size
-Set the video frame size.
-
-@item -capture_cursor
-Capture the mouse pointer. Default is 0.
-
-@item -capture_mouse_clicks
-Capture the screen mouse clicks. Default is 0.
-
@end table

@subsection Examples
@@ -175,36 +150,6 @@ $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

 BSD video input device.

-@subsection Options
-
-@table @option
-
-@item framerate
-Set the frame rate.
-
-@item video_size
-Set the video frame size. Default is @code{vga}.
-
-@item standard
-
-Available values are:
-@table @samp
-@item pal
-
-@item ntsc
-
-@item secam
-
-@item paln
-
-@item palm
-
-@item ntscj
-
-@end table
-
-@end table
-
@section decklink

 The decklink input device provides capture capabilities for Blackmagic
@@ -266,6 +211,18 @@ Capture video clip at 1080i50 10 bit:
 ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
@end example

+@item
+Capture video clip at 720p50 with 32bit audio:
+@example
+ffmpeg -bm_audiodepth 32 -f decklink -i 'UltraStudio Mini Recorder@@14' -acodec copy -vcodec copy output.avi
+@end example
+
+@item
+Capture video clip at 576i50 with 8 audio channels:
+@example
+ffmpeg -bm_channels 8 -f decklink -i 'UltraStudio Mini Recorder@@3' -acodec copy -vcodec copy output.avi
+@end example
+
@end itemize

@section dshow
@@ -318,11 +275,11 @@ If set to @option{true}, print a list of selected device's options
 and exit.

@item video_device_number
-Set video device number for devices with the same name (starts at 0,
+Set video device number for devices with same name (starts at 0,
 defaults to 0).

@item audio_device_number
-Set audio device number for devices with the same name (starts at 0,
+Set audio device number for devices with same name (starts at 0,
 defaults to 0).

@item pixel_format
@@ -472,27 +429,6 @@ $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_numbe

 Linux DV 1394 input device.

-@subsection Options
-
-@table @option
-
-@item framerate
-Set the frame rate. Default is 25.
-
-@item standard
-
-Available values are:
-@table @samp
-@item pal
-
-@item ntsc
-
-@end table
-
-Default value is @code{ntsc}.
-
-@end table
-
@section fbdev

 Linux framebuffer input device.
@@ -505,27 +441,18 @@ console. It is accessed through a file device node, usually
 For more detailed information read the file
 Documentation/fb/framebuffer.txt included in the Linux source tree.

-See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
-
 To record from the framebuffer device @file{/dev/fb0} with
@command{ffmpeg}:
@example
-ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
+ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
@end example

 You can take a single screenshot image with the command:
@example
-ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
+ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
@end example

-@subsection Options
-
-@table @option
-
-@item framerate
-Set the frame rate. Default is 25.
-
-@end table
+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).

@section gdigrab

@@ -711,15 +638,6 @@ $ jack_connect metro:120_bpm ffmpeg:input_1
 For more information read:
@url{http://jackaudio.org/}

-@subsection Options
-
-@table @option
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
@section lavfi

 Libavfilter input virtual device.
@@ -760,9 +678,6 @@ Set the filename of the filtergraph to be read and sent to the other
 filters. Syntax of the filtergraph is the same as the one specified by
 the option @var{graph}.

-@item dumpgraph
-Dump graph to stderr.
-
@end table

@subsection Examples
@@ -964,19 +879,6 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}

-@subsection Options
-
-@table @option
-
-@item sample_rate
-Set the sample rate in Hz. Default is 48000.
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
-
@section pulse

 PulseAudio input device.
@@ -1017,10 +919,6 @@ Specify the number of bytes per frame, by default it is set to 1024.
@item fragment_size
 Specify the minimal buffering fragment in PulseAudio, it will affect the
 audio latency. By default it is unset.
-
-@item wallclock
-Set the initial PTS using the current time. Default is 1.
-
@end table

@subsection Examples
@@ -1056,22 +954,6 @@ ffmpeg -f qtkit -i "default" out.mpg
 ffmpeg -f qtkit -list_devices true -i ""
@end example

-@subsection Options
-
-@table @option
-
-@item frame_rate
-Set frame rate. Default is 30.
-
-@item list_devices
-If set to @code{true}, print a list of devices and exit. Default is
-@code{false}.
-
-@item video_device_index
-Select the video device by index for devices with the same name (starts at 0).
-
-@end table
-
@section sndio

 sndio input device.
@@ -1089,18 +971,6 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example

-@subsection Options
-
-@table @option
-
-@item sample_rate
-Set the sample rate in Hz. Default is 48000.
-
-@item channels
-Set the number of channels. Default is 2.
-
-@end table
-
@section video4linux2, v4l2

 Video4Linux2 input video device.
@@ -1223,10 +1093,6 @@ Force conversion from monotonic to absolute timestamps.
@end table

 Default value is @code{default}.
-
-@item use_libv4l2
-Use libv4l2 (v4l-utils) conversion functions. Default is 0.
-
@end table

@section vfwcap
@@ -1237,19 +1103,6 @@ The filename passed as input is the capture driver number, ranging from
 0 to 9. You may use "list" as filename to print a list of drivers. Any
 other filename will be interpreted as device number 0.

-@subsection Options
-
-@table @option
-
-@item video_size
-Set the video frame size.
-
-@item framerate
-Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a frame rate of @code{30000/1001}.
-
-@end table
-
@section x11grab

 X11 video input device.
--- a/doc/mips.txt
+++ b/doc/mips.txt
@@ -47,16 +47,12 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
      float_dsp_mips.c
      libm_mips.h
-      softfloat_tables.h
 * libavcodec/
      fft_fixed_32.c
      fft_init_table.c
      fft_table.h
      mdct_fixed_32.c
 * libavcodec/mips/
-      aacdec_fixed.c
-      aacsbr_fixed.c
-      aacsbr_template.c
      aaccoder_mips.c
      aacpsy_mips.h
      ac3dsp_mips.c
--- a/doc/multithreading.txt
+++ b/doc/multithreading.txt
@@ -54,7 +54,7 @@ thread.
 If the codec allocates writable tables in its init(), add an init_thread_copy()
 which re-allocates them for other threads.

-Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
+Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.

 If there are inter-frame dependencies, so the codec calls
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -263,62 +263,6 @@ ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
 This example will produce the playlist, @file{out.m3u8}, and segment files:
@file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc.

-@item hls_key_info_file @var{key_info_file}
-Use the information in @var{key_info_file} for segment encryption. The first
-line of @var{key_info_file} specifies the key URI written to the playlist. The
-key URL is used to access the encryption key during playback. The second line
-specifies the path to the key file used to obtain the key during the encryption
-process. The key file is read as a single packed array of 16 octets in binary
-format. The optional third line specifies the initialization vector (IV) as a
-hexadecimal string to be used instead of the segment sequence number (default)
-for encryption. Changes to @var{key_info_file} will result in segment
-encryption with the new key/IV and an entry in the playlist for the new key
-URI/IV.
-
-Key info file format:
-@example
-@var{key URI}
-@var{key file path}
-@var{IV} (optional)
-@end example
-
-Example key URIs:
-@example
-http://server/file.key
-/path/to/file.key
-file.key
-@end example
-
-Example key file paths:
-@example
-file.key
-/path/to/file.key
-@end example
-
-Example IV:
-@example
-0123456789ABCDEF0123456789ABCDEF
-@end example
-
-Key info file example:
-@example
-http://server/file.key
-/path/to/file.key
-0123456789ABCDEF0123456789ABCDEF
-@end example
-
-Example shell script:
-@example
-#!/bin/sh
-BASE_URL=$@{1:-'.'@}
-openssl rand 16 > file.key
-echo $BASE_URL/file.key > file.keyinfo
-echo file.key >> file.keyinfo
-echo $(openssl rand -hex 16) >> file.keyinfo
-ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
-  -hls_key_info_file file.keyinfo out.m3u8
-@end example
-
@item hls_flags single_file
 If this flag is set, the muxer will store all segments in a single MPEG-TS
 file, and will use byte ranges in the playlist. HLS playlists generated with
@@ -667,13 +611,6 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example

-@subsection Audible AAX
-
-Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
-@example
-ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
-@end example
-
@section mp3

 The MP3 muxer writes a raw MP3 stream with the following optional features:
@@ -766,10 +703,6 @@ Set a constant muxrate (default VBR).
@item -pcr_period @var{numer}
 Override the default PCR retransmission time (default 20ms), ignored
 if variable muxrate is selected.
-@item pat_period @var{number}
-Maximal time in seconds between PAT/PMT tables.
-@item sdt_period @var{number}
-Maximal time in seconds between SDT tables.
@item -pes_payload_size @var{number}
 Set minimum PES packet payload in bytes.
@item -mpegts_flags @var{flags}
@@ -821,8 +754,6 @@ Option mpegts_flags may take a set of such flags:
 Reemit PAT/PMT before writing the next packet.
@item latm
 Use LATM packetization for AAC.
-@item pat_pmt_at_frames
-Reemit PAT and PMT at each video frame.
@end table

@subsection Example
@@ -839,21 +770,6 @@ ffmpeg -i file.mpg -c copy \
     -y out.ts
@end example

-@section mxf, mxf_d10
-
-MXF muxer.
-
-@subsection Options
-
-The muxer options are:
-
-@table @option
-@item store_user_comments @var{bool}
-Set if user comments should be stored if available or never.
-IRT D-10 does not allow user comments. The default is thus to write them for
-mxf but not for mxf_d10
-@end table
-
@section null

 Null muxer.
@@ -992,6 +908,13 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
@end table

+@item segment_list_type @var{type}
+Select the listing format.
+@table @option
+@item @var{flat} use a simple flat list of entries.
+@item @var{hls} use a m3u8-like structure.
+@end table
+
@item segment_list_size @var{size}
 Update the list file so that it contains at most @var{size}
 segments. If 0 the list file will contain all the segments. Default
@@ -1001,9 +924,6 @@ value is 0.
 Prepend @var{prefix} to each entry. Useful to generate absolute paths.
 By default no prefix is applied.

-@item segment_list_type @var{type}
-Select the listing format.
-
 The following values are recognized:
@table @samp
@item flat
--- a/doc/platform.texi
+++ b/doc/platform.texi
@@ -175,6 +175,12 @@ Notes:

@itemize

+@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
+You can find out by running @code{which link} to see which @code{link.exe} you
+are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
+in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
+@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
+
@item If you wish to build with zlib support, you will have to grab a compatible
 zlib binary from somewhere, with an MSVC import lib, or if you wish to link
 statically, you can follow the instructions below to build a compatible
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -19,18 +19,6 @@ supported protocols.

 A description of the currently available protocols follows.

-@section async
-
-Asynchronous data filling wrapper for input stream.
-
-Fill data in a background thread, to decouple I/O operation from demux thread.
-
-@example
-async:@var{URL}
-async:http://host/resource
-async:cache:http://host/resource
-@end example
-
@section bluray

 Read BluRay playlist.
@@ -304,8 +292,6 @@ autodetection in the future.
 If set to 1 enables experimental HTTP server. This can be used to send data when
 used as an output option, or read data from a client with HTTP POST when used as
 an input option.
-If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
-in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
@example
 # Server side (sending):
 ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
--- a/doc/scaler.texi
+++ b/doc/scaler.texi
@@ -122,22 +122,6 @@ a_dither).

@end table

-@item alphablend
-Set the alpha blending to use when the input has alpha but the output does not.
-Default value is @samp{none}.
-
-@table @samp
-@item uniform_color
-Blend onto a uniform background color
-
-@item checkerboard
-Blend onto a checkerboard
-
-@item none
-No blending
-
-@end table
-
@end table

@c man end SCALER OPTIONS
--- a/doc/texi2pod.pl
+++ b/doc/texi2pod.pl
@@ -384,7 +384,7 @@ sub postprocess
    # @* is also impossible in .pod; we discard it and any newline that
    # follows it.  Similarly, our macro @gol must be discarded.

-    s/\@anchor\{(?:[^\}]*)\}//g;
+    s/\@anchor{(?:[^\}]*)\}//g;
    s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
    s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
    s/;\s+\@pxref\{(?:[^\}]*)\}//g;
--- a/doc/utils.texi
+++ b/doc/utils.texi
@@ -238,14 +238,6 @@ The following abbreviations are recognized:
 480x320
@item qhd
 960x540
-@item 2kdci
-2048x1080
-@item 4kdci
-4096x2160
-@item uhd2160
-3840x2160
-@item uhd4320
-7680x4320
@end table

@anchor{video rate syntax}
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -49,7 +49,6 @@
 #include "libavutil/parseutils.h"
 #include "libavutil/samplefmt.h"
 #include "libavutil/fifo.h"
-#include "libavutil/internal.h"
 #include "libavutil/intreadwrite.h"
 #include "libavutil/dict.h"
 #include "libavutil/mathematics.h"
@@ -80,10 +79,6 @@
 #include <windows.h>
 #include <psapi.h>
 #endif
-#if HAVE_SETCONSOLECTRLHANDLER
-#include <windows.h>
-#endif
-

 #if HAVE_SYS_SELECT_H
 #include <sys/select.h>
@@ -137,6 +132,8 @@ AVIOContext *progress_avio = NULL;

 static uint8_t *subtitle_out;

+#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"
+
 InputStream **input_streams = NULL;
 int        nb_input_streams = 0;
 InputFile   **input_files   = NULL;
@@ -172,8 +169,8 @@ static int sub2video_get_blank_frame(InputStream *ist)
    AVFrame *frame = ist->sub2video.frame;

    av_frame_unref(frame);
-    ist->sub2video.frame->width  = ist->dec_ctx->width  ? ist->dec_ctx->width  : ist->sub2video.w;
-    ist->sub2video.frame->height = ist->dec_ctx->height ? ist->dec_ctx->height : ist->sub2video.h;
+    ist->sub2video.frame->width  = ist->sub2video.w;
+    ist->sub2video.frame->height = ist->sub2video.h;
    ist->sub2video.frame->format = AV_PIX_FMT_RGB32;
    if ((ret = av_frame_get_buffer(frame, 32)) < 0)
        return ret;
@@ -193,9 +190,7 @@ static void sub2video_copy_rect(uint8_t *dst, int dst_linesize, int w, int h,
        return;
    }
    if (r->x < 0 || r->x + r->w > w || r->y < 0 || r->y + r->h > h) {
-        av_log(NULL, AV_LOG_WARNING, "sub2video: rectangle (%d %d %d %d) overflowing %d %d\n",
-            r->x, r->y, r->w, r->h, w, h
-        );
+        av_log(NULL, AV_LOG_WARNING, "sub2video: rectangle overflowing\n");
        return;
    }

@@ -227,6 +222,7 @@ static void sub2video_push_ref(InputStream *ist, int64_t pts)

 static void sub2video_update(InputStream *ist, AVSubtitle *sub)
 {
+    int w = ist->sub2video.w, h = ist->sub2video.h;
    AVFrame *frame = ist->sub2video.frame;
    int8_t *dst;
    int     dst_linesize;
@@ -254,7 +250,7 @@ static void sub2video_update(InputStream *ist, AVSubtitle *sub)
    dst          = frame->data    [0];
    dst_linesize = frame->linesize[0];
    for (i = 0; i < num_rects; i++)
-        sub2video_copy_rect(dst, dst_linesize, frame->width, frame->height, sub->rects[i]);
+        sub2video_copy_rect(dst, dst_linesize, w, h, sub->rects[i]);
    sub2video_push_ref(ist, pts);
    ist->sub2video.end_pts = end_pts;
 }
@@ -295,7 +291,7 @@ static void sub2video_flush(InputStream *ist)
    if (ist->sub2video.end_pts < INT64_MAX)
        sub2video_update(ist, NULL);
    for (i = 0; i < ist->nb_filters; i++)
-        av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
+        av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0);
 }

 /* end of sub2video hack */
@@ -317,7 +313,6 @@ void term_exit(void)
 static volatile int received_sigterm = 0;
 static volatile int received_nb_signals = 0;
 static volatile int transcode_init_done = 0;
-static volatile int ffmpeg_exited = 0;
 static int main_return_code = 0;

 static void
@@ -326,46 +321,10 @@ sigterm_handler(int sig)
    received_sigterm = sig;
    received_nb_signals++;
    term_exit_sigsafe();
-    if(received_nb_signals > 3) {
-        write(2/*STDERR_FILENO*/, "Received > 3 system signals, hard exiting\n",
-                           strlen("Received > 3 system signals, hard exiting\n"));
-
+    if(received_nb_signals > 3)
        exit(123);
-    }
 }

-#if HAVE_SETCONSOLECTRLHANDLER
-static BOOL WINAPI CtrlHandler(DWORD fdwCtrlType)
-{
-    av_log(NULL, AV_LOG_DEBUG, "\nReceived windows signal %ld\n", fdwCtrlType);
-
-    switch (fdwCtrlType)
-    {
-    case CTRL_C_EVENT:
-    case CTRL_BREAK_EVENT:
-        sigterm_handler(SIGINT);
-        return TRUE;
-
-    case CTRL_CLOSE_EVENT:
-    case CTRL_LOGOFF_EVENT:
-    case CTRL_SHUTDOWN_EVENT:
-        sigterm_handler(SIGTERM);
-        /* Basically, with these 3 events, when we return from this method the
-           process is hard terminated, so stall as long as we need to
-           to try and let the main thread(s) clean up and gracefully terminate
-           (we have at most 5 seconds, but should be done far before that). */
-        while (!ffmpeg_exited) {
-            Sleep(0);
-        }
-        return TRUE;
-
-    default:
-        av_log(NULL, AV_LOG_ERROR, "Received unknown windows signal %ld\n", fdwCtrlType);
-        return FALSE;
-    }
-}
-#endif
-
 void term_init(void)
 {
 #if HAVE_TERMIOS_H
@@ -399,9 +358,6 @@ void term_init(void)
 #ifdef SIGXCPU
    signal(SIGXCPU, sigterm_handler);
 #endif
-#if HAVE_SETCONSOLECTRLHANDLER
-    SetConsoleCtrlHandler((PHANDLER_ROUTINE) CtrlHandler, TRUE);
-#endif
 }

 /* read a key without blocking */
@@ -435,6 +391,10 @@ static int read_key(void)
        is_pipe = !GetConsoleMode(input_handle, &dw);
    }

+    if (stdin->_cnt > 0) {
+        read(0, &ch, 1);
+        return ch;
+    }
    if (is_pipe) {
        /* When running under a GUI, you will end here. */
        if (!PeekNamedPipe(input_handle, NULL, 0, NULL, &nchars, NULL)) {
@@ -469,7 +429,7 @@ static void ffmpeg_cleanup(int ret)

    if (do_benchmark) {
        int maxrss = getmaxrss() / 1024;
-        av_log(NULL, AV_LOG_INFO, "bench: maxrss=%ikB\n", maxrss);
+        printf("bench: maxrss=%ikB\n", maxrss);
    }

    for (i = 0; i < nb_filtergraphs; i++) {
@@ -575,13 +535,12 @@ static void ffmpeg_cleanup(int ret)
    avformat_network_deinit();

    if (received_sigterm) {
-        av_log(NULL, AV_LOG_INFO, "Exiting normally, received signal %d.\n",
+        av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
               (int) received_sigterm);
    } else if (ret && transcode_init_done) {
        av_log(NULL, AV_LOG_INFO, "Conversion failed!\n");
    }
    term_exit();
-    ffmpeg_exited = 1;
 }

 void remove_avoptions(AVDictionary **a, AVDictionary *b)
@@ -618,7 +577,7 @@ static void update_benchmark(const char *fmt, ...)
            va_start(va, fmt);
            vsnprintf(buf, sizeof(buf), fmt, va);
            va_end(va);
-            av_log(NULL, AV_LOG_INFO, "bench: %8"PRIu64" %s \n", t - current_time, buf);
+            printf("bench: %8"PRIu64" %s \n", t - current_time, buf);
        }
        current_time = t;
    }
@@ -640,7 +599,7 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
    int ret;

    if (!ost->st->codec->extradata_size && ost->enc_ctx->extradata_size) {
-        ost->st->codec->extradata = av_mallocz(ost->enc_ctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
+        ost->st->codec->extradata = av_mallocz(ost->enc_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
        if (ost->st->codec->extradata) {
            memcpy(ost->st->codec->extradata, ost->enc_ctx->extradata, ost->enc_ctx->extradata_size);
            ost->st->codec->extradata_size = ost->enc_ctx->extradata_size;
@@ -665,20 +624,6 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
        }
        ost->frame_number++;
    }
-    if (avctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-        int i;
-        uint8_t *sd = av_packet_get_side_data(pkt, AV_PKT_DATA_QUALITY_STATS,
-                                              NULL);
-        ost->quality = sd ? AV_RL32(sd) : -1;
-        ost->pict_type = sd ? sd[4] : AV_PICTURE_TYPE_NONE;
-
-        for (i = 0; i<FF_ARRAY_ELEMS(ost->error); i++) {
-            if (sd && i < sd[5])
-                ost->error[i] = AV_RL64(sd + 8 + 8*i);
-            else
-                ost->error[i] = -1;
-        }
-    }

    if (bsfc)
        av_packet_split_side_data(pkt);
@@ -693,17 +638,11 @@ static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
                                           &new_pkt.data, &new_pkt.size,
                                           pkt->data, pkt->size,
                                           pkt->flags & AV_PKT_FLAG_KEY);
-FF_DISABLE_DEPRECATION_WARNINGS
-        if(a == 0 && new_pkt.data != pkt->data
-#if FF_API_DESTRUCT_PACKET
-           && new_pkt.destruct
-#endif
-           ) {
-FF_ENABLE_DEPRECATION_WARNINGS
-            uint8_t *t = av_malloc(new_pkt.size + AV_INPUT_BUFFER_PADDING_SIZE); //the new should be a subset of the old so cannot overflow
+        if(a == 0 && new_pkt.data != pkt->data && new_pkt.destruct) {
+            uint8_t *t = av_malloc(new_pkt.size + FF_INPUT_BUFFER_PADDING_SIZE); //the new should be a subset of the old so cannot overflow
            if(t) {
                memcpy(t, new_pkt.data, new_pkt.size);
-                memset(t + new_pkt.size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
+                memset(t + new_pkt.size, 0, FF_INPUT_BUFFER_PADDING_SIZE);
                new_pkt.data = t;
                new_pkt.buf = NULL;
                a = 1;
@@ -1131,7 +1070,7 @@ static void do_video_out(AVFormatContext *s,
        int got_packet, forced_keyframe = 0;
        double pts_time;

-        if (enc->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) &&
+        if (enc->flags & (CODEC_FLAG_INTERLACED_DCT|CODEC_FLAG_INTERLACED_ME) &&
            ost->top_field_first >= 0)
            in_picture->top_field_first = !!ost->top_field_first;

@@ -1157,7 +1096,7 @@ static void do_video_out(AVFormatContext *s,
            ost->forced_keyframes_expr_const_values[FKF_T] = pts_time;
            res = av_expr_eval(ost->forced_keyframes_pexpr,
                               ost->forced_keyframes_expr_const_values, NULL);
-            ff_dlog(NULL, "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n",
+            av_dlog(NULL, "force_key_frame: n:%f n_forced:%f prev_forced_n:%f t:%f prev_forced_t:%f -> res:%f\n",
                    ost->forced_keyframes_expr_const_values[FKF_N],
                    ost->forced_keyframes_expr_const_values[FKF_N_FORCED],
                    ost->forced_keyframes_expr_const_values[FKF_PREV_FORCED_N],
@@ -1210,7 +1149,7 @@ static void do_video_out(AVFormatContext *s,
                       av_ts2str(pkt.dts), av_ts2timestr(pkt.dts, &enc->time_base));
            }

-            if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & AV_CODEC_CAP_DELAY))
+            if (pkt.pts == AV_NOPTS_VALUE && !(enc->codec->capabilities & CODEC_CAP_DELAY))
                pkt.pts = ost->sync_opts;

            av_packet_rescale_ts(&pkt, enc->time_base, ost->st->time_base);
@@ -1275,11 +1214,9 @@ static void do_video_stats(OutputStream *ost, int frame_size)
    enc = ost->enc_ctx;
    if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
        frame_number = ost->st->nb_frames;
-        fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number,
-                ost->quality / (float)FF_QP2LAMBDA);
-
-        if (ost->error[0]>=0 && (enc->flags & AV_CODEC_FLAG_PSNR))
-            fprintf(vstats_file, "PSNR= %6.2f ", psnr(ost->error[0] / (enc->width * enc->height * 255.0 * 255.0)));
+        fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number, enc->coded_frame ? enc->coded_frame->quality / (float)FF_QP2LAMBDA : 0);
+        if (enc->coded_frame && (enc->flags&CODEC_FLAG_PSNR))
+            fprintf(vstats_file, "PSNR= %6.2f ", psnr(enc->coded_frame->error[0] / (enc->width * enc->height * 255.0 * 255.0)));

        fprintf(vstats_file,"f_size= %6d ", frame_size);
        /* compute pts value */
@@ -1291,7 +1228,7 @@ static void do_video_stats(OutputStream *ost, int frame_size)
        avg_bitrate = (double)(ost->data_size * 8) / ti1 / 1000.0;
        fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ",
               (double)ost->data_size / 1024, ti1, bitrate, avg_bitrate);
-        fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(ost->pict_type));
+        fprintf(vstats_file, "type= %c\n", enc->coded_frame ? av_get_picture_type_char(enc->coded_frame->pict_type) : 'I');
    }
 }

@@ -1389,7 +1326,7 @@ static int reap_filters(int flush)
                do_video_out(of->ctx, ost, filtered_frame, float_pts);
                break;
            case AVMEDIA_TYPE_AUDIO:
-                if (!(enc->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) &&
+                if (!(enc->codec->capabilities & CODEC_CAP_PARAM_CHANGE) &&
                    enc->channels != av_frame_get_channels(filtered_frame)) {
                    av_log(NULL, AV_LOG_ERROR,
                           "Audio filter graph output is not normalized and encoder does not support parameter changes\n");
@@ -1428,8 +1365,8 @@ static void print_final_stats(int64_t total_size)
        }
        extra_size += ost->enc_ctx->extradata_size;
        data_size  += ost->data_size;
-        if (   (ost->enc_ctx->flags & (AV_CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2))
-            != AV_CODEC_FLAG_PASS1)
+        if (   (ost->enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2))
+            != CODEC_FLAG_PASS1)
            pass1_used = 0;
    }

@@ -1568,9 +1505,8 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
        float q = -1;
        ost = output_streams[i];
        enc = ost->enc_ctx;
-        if (!ost->stream_copy)
-            q = ost->quality / (float) FF_QP2LAMBDA;
-
+        if (!ost->stream_copy && enc->coded_frame)
+            q = enc->coded_frame->quality / (float)FF_QP2LAMBDA;
        if (vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) {
            snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "q=%2.1f ", q);
            av_bprintf(&buf_script, "stream_%d_%d_q=%.1f\n",
@@ -1597,8 +1533,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
                for (j = 0; j < 32; j++)
                    snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%X", (int)lrintf(log2(qp_histogram[j] + 1)));
            }
-
-            if ((enc->flags & AV_CODEC_FLAG_PSNR) && (ost->pict_type != AV_PICTURE_TYPE_NONE || is_last_report)) {
+            if ((enc->flags&CODEC_FLAG_PSNR) && (enc->coded_frame || is_last_report)) {
                int j;
                double error, error_sum = 0;
                double scale, scale_sum = 0;
@@ -1610,7 +1545,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
                        error = enc->error[j];
                        scale = enc->width * enc->height * 255.0 * 255.0 * frame_number;
                    } else {
-                        error = ost->error[j];
+                        error = enc->coded_frame->error[j];
                        scale = enc->width * enc->height * 255.0 * 255.0;
                    }
                    if (j)
@@ -1749,9 +1684,7 @@ static void flush_encoders(void)
                ret = encode(enc, &pkt, NULL, &got_packet);
                update_benchmark("flush %s %d.%d", desc, ost->file_index, ost->index);
                if (ret < 0) {
-                    av_log(NULL, AV_LOG_FATAL, "%s encoding failed: %s\n",
-                           desc,
-                           av_err2str(ret));
+                    av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc);
                    exit_program(1);
                }
                if (ost->logfile && enc->stats_out) {
@@ -1867,22 +1800,17 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p

    opkt.duration = av_rescale_q(pkt->duration, ist->st->time_base, ost->st->time_base);
    opkt.flags    = pkt->flags;
+
    // FIXME remove the following 2 lines they shall be replaced by the bitstream filters
-    if (  ost->st->codec->codec_id != AV_CODEC_ID_H264
-       && ost->st->codec->codec_id != AV_CODEC_ID_MPEG1VIDEO
-       && ost->st->codec->codec_id != AV_CODEC_ID_MPEG2VIDEO
-       && ost->st->codec->codec_id != AV_CODEC_ID_VC1
+    if (  ost->enc_ctx->codec_id != AV_CODEC_ID_H264
+       && ost->enc_ctx->codec_id != AV_CODEC_ID_MPEG1VIDEO
+       && ost->enc_ctx->codec_id != AV_CODEC_ID_MPEG2VIDEO
+       && ost->enc_ctx->codec_id != AV_CODEC_ID_VC1
       ) {
-        int ret = av_parser_change(ost->parser, ost->st->codec,
+        if (av_parser_change(ost->parser, ost->st->codec,
                             &opkt.data, &opkt.size,
                             pkt->data, pkt->size,
-                             pkt->flags & AV_PKT_FLAG_KEY);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_FATAL, "av_parser_change failed: %s\n",
-                   av_err2str(ret));
-            exit_program(1);
-        }
-        if (ret) {
+                             pkt->flags & AV_PKT_FLAG_KEY)) {
            opkt.buf = av_buffer_create(opkt.data, opkt.size, av_buffer_default_free, NULL, 0);
            if (!opkt.buf)
                exit_program(1);
@@ -1893,16 +1821,9 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
    }
    av_copy_packet_side_data(&opkt, pkt);

-    if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO &&
-        ost->st->codec->codec_id == AV_CODEC_ID_RAWVIDEO &&
-        (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
+    if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (of->ctx->oformat->flags & AVFMT_RAWPICTURE)) {
        /* store AVPicture in AVPacket, as expected by the output format */
-        int ret = avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_FATAL, "avpicture_fill failed: %s\n",
-                   av_err2str(ret));
-            exit_program(1);
-        }
+        avpicture_fill(&pict, opkt.data, ost->st->codec->pix_fmt, ost->st->codec->width, ost->st->codec->height);
        opkt.data = (uint8_t *)&pict;
        opkt.size = sizeof(AVPicture);
        opkt.flags |= AV_PKT_FLAG_KEY;
@@ -1959,8 +1880,17 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
    if (ret < 0 && exit_on_error)
        exit_program(1);

-    if (!*got_output || ret < 0)
+    if (!*got_output || ret < 0) {
+        if (!pkt->size) {
+            for (i = 0; i < ist->nb_filters; i++)
+#if 1
+                av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0);
+#else
+                av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
+#endif
+        }
        return ret;
+    }

    ist->samples_decoded += decoded_frame->nb_samples;
    ist->frames_decoded++;
@@ -2084,13 +2014,12 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
        if (ist->dec_ctx->codec_id == AV_CODEC_ID_H264) {
            ist->st->codec->has_b_frames = ist->dec_ctx->has_b_frames;
        } else
-            av_log(ist->dec_ctx, AV_LOG_WARNING,
-                   "has_b_frames is larger in decoder than demuxer %d > %d.\n"
-                   "If you want to help, upload a sample "
-                   "of this file to ftp://upload.ffmpeg.org/incoming/ "
-                   "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)",
-                   ist->dec_ctx->has_b_frames,
-                   ist->st->codec->has_b_frames);
+            av_log_ask_for_sample(
+                ist->dec_ctx,
+                "has_b_frames is larger in decoder than demuxer %d > %d ",
+                ist->dec_ctx->has_b_frames,
+                ist->st->codec->has_b_frames
+            );
    }

    if (*got_output || ret<0)
@@ -2113,8 +2042,17 @@ static int decode_video(InputStream *ist, AVPacket *pkt, int *got_output)
        }
    }

-    if (!*got_output || ret < 0)
+    if (!*got_output || ret < 0) {
+        if (!pkt->size) {
+            for (i = 0; i < ist->nb_filters; i++)
+#if 1
+                av_buffersrc_add_ref(ist->filters[i]->filter, NULL, 0);
+#else
+                av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
+#endif
+        }
        return ret;
+    }

    if(ist->top_field_first>=0)
        decoded_frame->top_field_first = ist->top_field_first;
@@ -2262,17 +2200,6 @@ out:
    return ret;
 }

-static int send_filter_eof(InputStream *ist)
-{
-    int i, ret;
-    for (i = 0; i < ist->nb_filters; i++) {
-        ret = av_buffersrc_add_frame(ist->filters[i]->filter, NULL);
-        if (ret < 0)
-            return ret;
-    }
-    return 0;
-}
-
 /* pkt = NULL means EOF (needed to flush decoder buffers) */
 static int process_input_packet(InputStream *ist, const AVPacket *pkt)
 {
@@ -2320,7 +2247,7 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
        ist->dts = ist->next_dts;

        if (avpkt.size && avpkt.size != pkt->size &&
-            !(ist->dec->capabilities & AV_CODEC_CAP_SUBFRAMES)) {
+            !(ist->dec->capabilities & CODEC_CAP_SUBFRAMES)) {
            av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING,
                   "Multiple frames in a packet from stream %d\n", pkt->stream_index);
            ist->showed_multi_packet_warning = 1;
@@ -2357,13 +2284,8 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
            return -1;
        }

-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
-                   ist->file_index, ist->st->index, av_err2str(ret));
-            if (exit_on_error)
-                exit_program(1);
-            break;
-        }
+        if (ret < 0)
+            return ret;

        avpkt.dts=
        avpkt.pts= AV_NOPTS_VALUE;
@@ -2382,15 +2304,6 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt)
            break;
    }

-    /* after flushing, send an EOF on all the filter inputs attached to the stream */
-    if (!pkt && ist->decoding_needed && !got_output) {
-        int ret = send_filter_eof(ist);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_FATAL, "Error marking filters as finished\n");
-            exit_program(1);
-        }
-    }
-
    /* handle stream copy */
    if (!ist->decoding_needed) {
        ist->dts = ist->next_dts;
@@ -2447,9 +2360,6 @@ static void print_sdp(void)
        }
    }

-    if (!j)
-        goto fail;
-
    av_sdp_create(avc, j, sdp, sizeof(sdp));

    if (!sdp_filename) {
@@ -2465,7 +2375,6 @@ static void print_sdp(void)
        }
    }

-fail:
    av_freep(&avc);
 }

@@ -2586,76 +2495,6 @@ static int compare_int64(const void *a, const void *b)
    return va < vb ? -1 : va > vb ? +1 : 0;
 }

-static int init_output_stream(OutputStream *ost, char *error, int error_len)
-{
-    int ret = 0;
-
-    if (ost->encoding_needed) {
-        AVCodec      *codec = ost->enc;
-        AVCodecContext *dec = NULL;
-        InputStream *ist;
-
-        if ((ist = get_input_stream(ost)))
-            dec = ist->dec_ctx;
-        if (dec && dec->subtitle_header) {
-            /* ASS code assumes this buffer is null terminated so add extra byte. */
-            ost->enc_ctx->subtitle_header = av_mallocz(dec->subtitle_header_size + 1);
-            if (!ost->enc_ctx->subtitle_header)
-                return AVERROR(ENOMEM);
-            memcpy(ost->enc_ctx->subtitle_header, dec->subtitle_header, dec->subtitle_header_size);
-            ost->enc_ctx->subtitle_header_size = dec->subtitle_header_size;
-        }
-        if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0))
-            av_dict_set(&ost->encoder_opts, "threads", "auto", 0);
-        av_dict_set(&ost->encoder_opts, "side_data_only_packets", "1", 0);
-        if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
-            !codec->defaults &&
-            !av_dict_get(ost->encoder_opts, "b", NULL, 0) &&
-            !av_dict_get(ost->encoder_opts, "ab", NULL, 0))
-            av_dict_set(&ost->encoder_opts, "b", "128000", 0);
-
-        if ((ret = avcodec_open2(ost->enc_ctx, codec, &ost->encoder_opts)) < 0) {
-            if (ret == AVERROR_EXPERIMENTAL)
-                abort_codec_experimental(codec, 1);
-            snprintf(error, error_len,
-                     "Error while opening encoder for output stream #%d:%d - "
-                     "maybe incorrect parameters such as bit_rate, rate, width or height",
-                    ost->file_index, ost->index);
-            return ret;
-        }
-        if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
-            !(ost->enc->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
-            av_buffersink_set_frame_size(ost->filter->filter,
-                                            ost->enc_ctx->frame_size);
-        assert_avoptions(ost->encoder_opts);
-        if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000)
-            av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
-                                         " It takes bits/s as argument, not kbits/s\n");
-
-        ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_FATAL,
-                   "Error initializing the output stream codec context.\n");
-            exit_program(1);
-        }
-
-        // copy timebase while removing common factors
-        ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
-        ost->st->codec->codec= ost->enc_ctx->codec;
-    } else {
-        ret = av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts);
-        if (ret < 0) {
-           av_log(NULL, AV_LOG_FATAL,
-                  "Error setting up codec context options.\n");
-           return ret;
-        }
-        // copy timebase while removing common factors
-        ost->st->time_base = av_add_q(ost->st->codec->time_base, (AVRational){0, 1});
-    }
-
-    return ret;
-}
-
 static void parse_forced_key_frames(char *kf, OutputStream *ost,
                                    AVCodecContext *avctx)
 {
@@ -2766,7 +2605,7 @@ static void set_encoder_id(OutputFile *of, OutputStream *ost)
    if (!encoder_string)
        exit_program(1);

-    if (!(format_flags & AVFMT_FLAG_BITEXACT) && !(codec_flags & AV_CODEC_FLAG_BITEXACT))
+    if (!(format_flags & AVFMT_FLAG_BITEXACT) && !(codec_flags & CODEC_FLAG_BITEXACT))
        av_strlcpy(encoder_string, LIBAVCODEC_IDENT " ", encoder_string_len);
    else
        av_strlcpy(encoder_string, "Lavc ", encoder_string_len);
@@ -2807,6 +2646,21 @@ static int transcode_init(void)
                input_streams[j + ifile->ist_index]->start = av_gettime_relative();
    }

+    /* output stream init */
+    for (i = 0; i < nb_output_files; i++) {
+        oc = output_files[i]->ctx;
+        if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) {
+            av_dump_format(oc, i, oc->filename, 1);
+            av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", i);
+            return AVERROR(EINVAL);
+        }
+    }
+
+    /* init complex filtergraphs */
+    for (i = 0; i < nb_filtergraphs; i++)
+        if ((ret = avfilter_graph_config(filtergraphs[i]->graph, NULL)) < 0)
+            return ret;
+
    /* for each output stream, we compute the right encoding parameters */
    for (i = 0; i < nb_output_streams; i++) {
        AVCodecContext *enc_ctx;
@@ -2843,7 +2697,7 @@ static int transcode_init(void)

            av_assert0(ist && !ost->filter);

-            extra_size = (uint64_t)dec_ctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE;
+            extra_size = (uint64_t)dec_ctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE;

            if (extra_size > INT_MAX) {
                return AVERROR(EINVAL);
@@ -2918,7 +2772,7 @@ static int transcode_init(void)
                enc_ctx->time_base = dec_ctx->time_base;
            }

-            if (!ost->frame_rate.num)
+            if (ist && !ost->frame_rate.num)
                ost->frame_rate = ist->framerate;
            if(ost->frame_rate.num)
                enc_ctx->time_base = av_inv_q(ost->frame_rate);
@@ -3015,6 +2869,10 @@ static int transcode_init(void)
                goto dump_format;
            }

+            if (ist)
+                ist->decoding_needed |= DECODING_FOR_OST;
+            ost->encoding_needed = 1;
+
            set_encoder_id(output_files[ost->file_index], ost);

            if (!ost->filter &&
@@ -3143,6 +3001,39 @@ static int transcode_init(void)
                abort();
                break;
            }
+            /* two pass mode */
+            if (enc_ctx->flags & (CODEC_FLAG_PASS1 | CODEC_FLAG_PASS2)) {
+                char logfilename[1024];
+                FILE *f;
+
+                snprintf(logfilename, sizeof(logfilename), "%s-%d.log",
+                         ost->logfile_prefix ? ost->logfile_prefix :
+                                               DEFAULT_PASS_LOGFILENAME_PREFIX,
+                         i);
+                if (!strcmp(ost->enc->name, "libx264")) {
+                    av_dict_set(&ost->encoder_opts, "stats", logfilename, AV_DICT_DONT_OVERWRITE);
+                } else {
+                    if (enc_ctx->flags & CODEC_FLAG_PASS2) {
+                        char  *logbuffer;
+                        size_t logbuffer_size;
+                        if (cmdutils_read_file(logfilename, &logbuffer, &logbuffer_size) < 0) {
+                            av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n",
+                                   logfilename);
+                            exit_program(1);
+                        }
+                        enc_ctx->stats_in = logbuffer;
+                    }
+                    if (enc_ctx->flags & CODEC_FLAG_PASS1) {
+                        f = av_fopen_utf8(logfilename, "wb");
+                        if (!f) {
+                            av_log(NULL, AV_LOG_FATAL, "Cannot write log file '%s' for pass-1 encoding: %s\n",
+                                logfilename, strerror(errno));
+                            exit_program(1);
+                        }
+                        ost->logfile = f;
+                    }
+                }
+            }
        }

        if (ost->disposition) {
@@ -3179,9 +3070,63 @@ static int transcode_init(void)

    /* open each encoder */
    for (i = 0; i < nb_output_streams; i++) {
-        ret = init_output_stream(output_streams[i], error, sizeof(error));
-        if (ret < 0)
-            goto dump_format;
+        ost = output_streams[i];
+        if (ost->encoding_needed) {
+            AVCodec      *codec = ost->enc;
+            AVCodecContext *dec = NULL;
+
+            if ((ist = get_input_stream(ost)))
+                dec = ist->dec_ctx;
+            if (dec && dec->subtitle_header) {
+                /* ASS code assumes this buffer is null terminated so add extra byte. */
+                ost->enc_ctx->subtitle_header = av_mallocz(dec->subtitle_header_size + 1);
+                if (!ost->enc_ctx->subtitle_header) {
+                    ret = AVERROR(ENOMEM);
+                    goto dump_format;
+                }
+                memcpy(ost->enc_ctx->subtitle_header, dec->subtitle_header, dec->subtitle_header_size);
+                ost->enc_ctx->subtitle_header_size = dec->subtitle_header_size;
+            }
+            if (!av_dict_get(ost->encoder_opts, "threads", NULL, 0))
+                av_dict_set(&ost->encoder_opts, "threads", "auto", 0);
+            av_dict_set(&ost->encoder_opts, "side_data_only_packets", "1", 0);
+
+            if ((ret = avcodec_open2(ost->enc_ctx, codec, &ost->encoder_opts)) < 0) {
+                if (ret == AVERROR_EXPERIMENTAL)
+                    abort_codec_experimental(codec, 1);
+                snprintf(error, sizeof(error), "Error while opening encoder for output stream #%d:%d - maybe incorrect parameters such as bit_rate, rate, width or height",
+                        ost->file_index, ost->index);
+                goto dump_format;
+            }
+            if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
+                !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
+                av_buffersink_set_frame_size(ost->filter->filter,
+                                             ost->enc_ctx->frame_size);
+            assert_avoptions(ost->encoder_opts);
+            if (ost->enc_ctx->bit_rate && ost->enc_ctx->bit_rate < 1000)
+                av_log(NULL, AV_LOG_WARNING, "The bitrate parameter is set too low."
+                                             " It takes bits/s as argument, not kbits/s\n");
+
+            ret = avcodec_copy_context(ost->st->codec, ost->enc_ctx);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_FATAL,
+                       "Error initializing the output stream codec context.\n");
+                exit_program(1);
+            }
+
+            // copy timebase while removing common factors
+            ost->st->time_base = av_add_q(ost->enc_ctx->time_base, (AVRational){0, 1});
+            ost->st->codec->codec= ost->enc_ctx->codec;
+        } else {
+            ret = av_opt_set_dict(ost->enc_ctx, &ost->encoder_opts);
+            if (ret < 0) {
+                av_log(NULL, AV_LOG_FATAL,
+                    "Error setting up codec context options.\n");
+                return ret;
+            }
+            // copy timebase while removing common factors
+            ost->st->time_base = av_add_q(ost->st->codec->time_base, (AVRational){0, 1});
+        }
    }

    /* init input streams */
@@ -3451,17 +3396,9 @@ static int check_keyboard_interaction(int64_t cur_time)
            if(!debug) debug = 1;
            while(debug & (FF_DEBUG_DCT_COEFF|FF_DEBUG_VIS_QP|FF_DEBUG_VIS_MB_TYPE)) //unsupported, would just crash
                debug += debug;
-        }else{
-            char buf[32];
-            int k = 0;
-            i = 0;
-            while ((k = read_key()) != '\n' && k != '\r' && i < sizeof(buf)-1)
-                if (k > 0)
-                    buf[i++] = k;
-            buf[i] = 0;
-            if (k <= 0 || sscanf(buf, "%d", &debug)!=1)
+        }else
+            if(scanf("%d", &debug)!=1)
                fprintf(stderr,"error parsing debug value\n");
-        }
        for(i=0;i<nb_input_streams;i++) {
            input_streams[i]->st->codec->debug = debug;
        }
@@ -3539,7 +3476,7 @@ static void free_input_threads(void)
        InputFile *f = input_files[i];
        AVPacket pkt;

-        if (!f || !f->in_thread_queue)
+        if (!f->in_thread_queue)
            continue;
        av_thread_message_queue_set_err_send(f->in_thread_queue, AVERROR_EOF);
        while (av_thread_message_queue_recv(f->in_thread_queue, &pkt, 0) >= 0)
@@ -3842,7 +3779,13 @@ static int process_input(int file_index)

    sub2video_heartbeat(ist, pkt.pts);

-    process_input_packet(ist, &pkt);
+    ret = process_input_packet(ist, &pkt);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Error while decoding stream #%d:%d: %s\n",
+               ist->file_index, ist->st->index, av_err2str(ret));
+        if (exit_on_error)
+            exit_program(1);
+    }

 discard_packet:
    av_free_packet(&pkt);
@@ -4061,7 +4004,6 @@ static int transcode(void)
                av_freep(&ost->apad);
                av_freep(&ost->disposition);
                av_dict_free(&ost->encoder_opts);
-                av_dict_free(&ost->sws_dict);
                av_dict_free(&ost->swr_opts);
                av_dict_free(&ost->resample_opts);
                av_dict_free(&ost->bsf_args);
@@ -4170,7 +4112,7 @@ int main(int argc, char **argv)
        exit_program(1);
    ti = getutime() - ti;
    if (do_benchmark) {
-        av_log(NULL, AV_LOG_INFO, "bench: utime=%0.3fs\n", ti / 1000000.0);
+        printf("bench: utime=%0.3fs\n", ti / 1000000.0);
    }
    av_log(NULL, AV_LOG_DEBUG, "%"PRIu64" frames successfully decoded, %"PRIu64" decoding errors\n",
           decode_error_stat[0], decode_error_stat[1]);
--- a/ffmpeg.h
+++ b/ffmpeg.h
@@ -63,7 +63,6 @@ enum HWAccelID {
    HWACCEL_VDPAU,
    HWACCEL_DXVA2,
    HWACCEL_VDA,
-    HWACCEL_VIDEOTOOLBOX,
 };

 typedef struct HWAccel {
@@ -93,7 +92,6 @@ typedef struct OptionsContext {

    /* input/output options */
    int64_t start_time;
-    int64_t start_time_eof;
    int seek_timestamp;
    const char *format;

@@ -231,7 +229,6 @@ typedef struct OutputFilter {

    /* temporary storage until stream maps are processed */
    AVFilterInOut       *out_tmp;
-    enum AVMediaType     type;
 } OutputFilter;

 typedef struct FilterGraph {
@@ -432,8 +429,8 @@ typedef struct OutputStream {
    char *filters;         ///< filtergraph associated to the -filter option
    char *filters_script;  ///< filtergraph script associated to the -filter_script option

+    int64_t sws_flags;
    AVDictionary *encoder_opts;
-    AVDictionary *sws_dict;
    AVDictionary *swr_opts;
    AVDictionary *resample_opts;
    AVDictionary *bsf_args;
@@ -458,15 +455,6 @@ typedef struct OutputStream {
    // number of frames/samples sent to the encoder
    uint64_t frames_encoded;
    uint64_t samples_encoded;
-
-    /* packet quality factor */
-    int quality;
-
-    /* packet picture type */
-    int pict_type;
-
-    /* frame encode sum of squared error values */
-    int64_t error[4];
 } OutputStream;

 typedef struct OutputFile {
@@ -521,7 +509,6 @@ extern int frame_bits_per_raw_sample;
 extern AVIOContext *progress_avio;
 extern float max_error_rate;
 extern int vdpau_api_ver;
-extern char *videotoolbox_pixfmt;

 extern const AVIOInterruptCB int_cb;

@@ -549,13 +536,11 @@ int configure_filtergraph(FilterGraph *fg);
 int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOut *out);
 int ist_in_filtergraph(FilterGraph *fg, InputStream *ist);
 FilterGraph *init_simple_filtergraph(InputStream *ist, OutputStream *ost);
-int init_complex_filtergraph(FilterGraph *fg);

 int ffmpeg_parse_options(int argc, char **argv);

 int vdpau_init(AVCodecContext *s);
 int dxva2_init(AVCodecContext *s);
 int vda_init(AVCodecContext *s);
-int videotoolbox_init(AVCodecContext *s);

 #endif /* FFMPEG_H */
--- a/ffmpeg_filter.c
+++ b/ffmpeg_filter.c
@@ -85,7 +85,7 @@ void choose_sample_fmt(AVStream *st, AVCodec *codec)
                break;
        }
        if (*p == -1) {
-            if((codec->capabilities & AV_CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
+            if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
                av_log(NULL, AV_LOG_ERROR, "Conversion will not be lossless.\n");
            if(av_get_sample_fmt_name(st->codec->sample_fmt))
            av_log(NULL, AV_LOG_WARNING,
@@ -289,45 +289,6 @@ static void init_input_filter(FilterGraph *fg, AVFilterInOut *in)
    ist->filters[ist->nb_filters - 1] = fg->inputs[fg->nb_inputs - 1];
 }

-int init_complex_filtergraph(FilterGraph *fg)
-{
-    AVFilterInOut *inputs, *outputs, *cur;
-    AVFilterGraph *graph;
-    int ret = 0;
-
-    /* this graph is only used for determining the kinds of inputs
-     * and outputs we have, and is discarded on exit from this function */
-    graph = avfilter_graph_alloc();
-    if (!graph)
-        return AVERROR(ENOMEM);
-
-    ret = avfilter_graph_parse2(graph, fg->graph_desc, &inputs, &outputs);
-    if (ret < 0)
-        goto fail;
-
-    for (cur = inputs; cur; cur = cur->next)
-        init_input_filter(fg, cur);
-
-    for (cur = outputs; cur;) {
-        GROW_ARRAY(fg->outputs, fg->nb_outputs);
-        fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]));
-        if (!fg->outputs[fg->nb_outputs - 1])
-            exit_program(1);
-
-        fg->outputs[fg->nb_outputs - 1]->graph   = fg;
-        fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
-        fg->outputs[fg->nb_outputs - 1]->type    = avfilter_pad_get_type(cur->filter_ctx->output_pads,
-                                                                         cur->pad_idx);
-        cur = cur->next;
-        fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
-    }
-
-fail:
-    avfilter_inout_free(&inputs);
-    avfilter_graph_free(&graph);
-    return ret;
-}
-
 static int insert_trim(int64_t start_time, int64_t duration,
                       AVFilterContext **last_filter, int *pad_idx,
                       const char *filter_name)
@@ -423,17 +384,11 @@ static int configure_output_video_filter(FilterGraph *fg, OutputFilter *ofilter,
    if (codec->width || codec->height) {
        char args[255];
        AVFilterContext *filter;
-        AVDictionaryEntry *e = NULL;

-        snprintf(args, sizeof(args), "%d:%d",
+        snprintf(args, sizeof(args), "%d:%d:0x%X",
                 codec->width,
-                 codec->height);
-
-        while ((e = av_dict_get(ost->sws_dict, "", e,
-                                AV_DICT_IGNORE_SUFFIX))) {
-            av_strlcatf(args, sizeof(args), ":%s=%s", e->key, e->value);
-        }
-
+                 codec->height,
+                 (unsigned)ost->sws_flags);
        snprintf(name, sizeof(name), "scaler for output stream %d:%d",
                 ost->file_index, ost->index);
        if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
@@ -544,7 +499,7 @@ static int configure_output_audio_filter(FilterGraph *fg, OutputFilter *ofilter,
                   av_get_default_channel_layout(ost->audio_channels_mapped));
        for (i = 0; i < ost->audio_channels_mapped; i++)
            if (ost->audio_channels_map[i] != -1)
-                av_bprintf(&pan_buf, "|c%d=c%d", i, ost->audio_channels_map[i]);
+                av_bprintf(&pan_buf, ":c%d=c%d", i, ost->audio_channels_map[i]);

        AUTO_INSERT_FILTER("-map_channel", "pan", pan_buf.str);
        av_bprint_finalize(&pan_buf, NULL);
@@ -647,11 +602,6 @@ int configure_output_filter(FilterGraph *fg, OutputFilter *ofilter, AVFilterInOu
    av_freep(&ofilter->name);
    DESCRIBE_FILTER_LINK(ofilter, out, 0);

-    if (!ofilter->ost) {
-        av_log(NULL, AV_LOG_FATAL, "Filter %s has a unconnected output\n", ofilter->name);
-        exit_program(1);
-    }
-
    switch (avfilter_pad_get_type(out->filter_ctx->output_pads, out->pad_idx)) {
    case AVMEDIA_TYPE_VIDEO: return configure_output_video_filter(fg, ofilter, out);
    case AVMEDIA_TYPE_AUDIO: return configure_output_audio_filter(fg, ofilter, out);
@@ -682,8 +632,8 @@ static int sub2video_prepare(InputStream *ist)
        }
        av_log(avf, AV_LOG_INFO, "sub2video: using %dx%d canvas\n", w, h);
    }
-    ist->sub2video.w = ist->resample_width  = w;
-    ist->sub2video.h = ist->resample_height = h;
+    ist->sub2video.w = ist->dec_ctx->width  = ist->resample_width  = w;
+    ist->sub2video.h = ist->dec_ctx->height = ist->resample_height = h;

    /* rectangles are AV_PIX_FMT_PAL8, but we have no guarantee that the
       palettes for all rectangles are identical or compatible */
@@ -738,7 +688,7 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
             ist->resample_height,
             ist->hwaccel_retrieve_data ? ist->hwaccel_retrieved_pix_fmt : ist->resample_pix_fmt,
             tb.num, tb.den, sar.num, sar.den,
-             SWS_BILINEAR + ((ist->dec_ctx->flags&AV_CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
+             SWS_BILINEAR + ((ist->dec_ctx->flags&CODEC_FLAG_BITEXACT) ? SWS_BITEXACT:0));
    if (fr.num && fr.den)
        av_bprintf(&args, ":frame_rate=%d/%d", fr.num, fr.den);
    snprintf(name, sizeof(name), "graph %d input from stream %d:%d", fg->index,
@@ -954,7 +904,7 @@ static int configure_input_filter(FilterGraph *fg, InputFilter *ifilter,
 int configure_filtergraph(FilterGraph *fg)
 {
    AVFilterInOut *inputs, *outputs, *cur;
-    int ret, i, simple = !fg->graph_desc;
+    int ret, i, init = !fg->graph, simple = !fg->graph_desc;
    const char *graph_desc = simple ? fg->outputs[0]->ost->avfilter :
                                      fg->graph_desc;

@@ -967,13 +917,7 @@ int configure_filtergraph(FilterGraph *fg)
        char args[512];
        AVDictionaryEntry *e = NULL;

-        args[0] = 0;
-        while ((e = av_dict_get(ost->sws_dict, "", e,
-                                AV_DICT_IGNORE_SUFFIX))) {
-            av_strlcatf(args, sizeof(args), "%s=%s:", e->key, e->value);
-        }
-        if (strlen(args))
-            args[strlen(args)-1] = 0;
+        snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
        fg->graph->scale_sws_opts = av_strdup(args);

        args[0] = 0;
@@ -1003,30 +947,14 @@ int configure_filtergraph(FilterGraph *fg)
        return ret;

    if (simple && (!inputs || inputs->next || !outputs || outputs->next)) {
-        const char *num_inputs;
-        const char *num_outputs;
-        if (!outputs) {
-            num_outputs = "0";
-        } else if (outputs->next) {
-            num_outputs = ">1";
-        } else {
-            num_outputs = "1";
-        }
-        if (!inputs) {
-            num_inputs = "0";
-        } else if (inputs->next) {
-            num_inputs = ">1";
-        } else {
-            num_inputs = "1";
-        }
-        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' was expected "
-               "to have exactly 1 input and 1 output."
-               " However, it had %s input(s) and %s output(s)."
-               " Please adjust, or use a complex filtergraph (-filter_complex) instead.\n",
-               graph_desc, num_inputs, num_outputs);
+        av_log(NULL, AV_LOG_ERROR, "Simple filtergraph '%s' does not have "
+               "exactly one input and output.\n", graph_desc);
        return AVERROR(EINVAL);
    }

+    for (cur = inputs; !simple && init && cur; cur = cur->next)
+        init_input_filter(fg, cur);
+
    for (cur = inputs, i = 0; cur; cur = cur->next, i++)
        if ((ret = configure_input_filter(fg, fg->inputs[i], cur)) < 0) {
            avfilter_inout_free(&inputs);
@@ -1035,12 +963,27 @@ int configure_filtergraph(FilterGraph *fg)
        }
    avfilter_inout_free(&inputs);

-    for (cur = outputs, i = 0; cur; cur = cur->next, i++)
-        configure_output_filter(fg, fg->outputs[i], cur);
-    avfilter_inout_free(&outputs);
+    if (!init || simple) {
+        /* we already know the mappings between lavfi outputs and output streams,
+         * so we can finish the setup */
+        for (cur = outputs, i = 0; cur; cur = cur->next, i++)
+            configure_output_filter(fg, fg->outputs[i], cur);
+        avfilter_inout_free(&outputs);

-    if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
-        return ret;
+        if ((ret = avfilter_graph_config(fg->graph, NULL)) < 0)
+            return ret;
+    } else {
+        /* wait until output mappings are processed */
+        for (cur = outputs; cur;) {
+            GROW_ARRAY(fg->outputs, fg->nb_outputs);
+            if (!(fg->outputs[fg->nb_outputs - 1] = av_mallocz(sizeof(*fg->outputs[0]))))
+                exit_program(1);
+            fg->outputs[fg->nb_outputs - 1]->graph   = fg;
+            fg->outputs[fg->nb_outputs - 1]->out_tmp = cur;
+            cur = cur->next;
+            fg->outputs[fg->nb_outputs - 1]->out_tmp->next = NULL;
+        }
+    }

    fg->reconfiguration = 1;

@@ -1048,7 +991,7 @@ int configure_filtergraph(FilterGraph *fg)
        OutputStream *ost = fg->outputs[i]->ost;
        if (ost &&
            ost->enc->type == AVMEDIA_TYPE_AUDIO &&
-            !(ost->enc->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE))
+            !(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
            av_buffersink_set_frame_size(ost->filter->filter,
                                         ost->enc_ctx->frame_size);
    }
--- a/ffmpeg_opt.c
+++ b/ffmpeg_opt.c
@@ -40,9 +40,6 @@
 #include "libavutil/parseutils.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/pixfmt.h"
-#include "libavutil/time_internal.h"
-
-#define DEFAULT_PASS_LOGFILENAME_PREFIX "ffmpeg2pass"

 #define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
 {\
@@ -74,10 +71,7 @@ const HWAccel hwaccels[] = {
    { "dxva2", dxva2_init, HWACCEL_DXVA2, AV_PIX_FMT_DXVA2_VLD },
 #endif
 #if CONFIG_VDA
-    { "vda",   videotoolbox_init,   HWACCEL_VDA,   AV_PIX_FMT_VDA },
-#endif
-#if CONFIG_VIDEOTOOLBOX
-    { "videotoolbox",   videotoolbox_init,   HWACCEL_VIDEOTOOLBOX,   AV_PIX_FMT_VIDEOTOOLBOX },
+    { "vda",   vda_init,   HWACCEL_VDA,   AV_PIX_FMT_VDA },
 #endif
    { 0 },
 };
@@ -159,25 +153,12 @@ static void init_options(OptionsContext *o)
    o->stop_time = INT64_MAX;
    o->mux_max_delay  = 0.7;
    o->start_time     = AV_NOPTS_VALUE;
-    o->start_time_eof = AV_NOPTS_VALUE;
    o->recording_time = INT64_MAX;
    o->limit_filesize = UINT64_MAX;
    o->chapters_input_file = INT_MAX;
    o->accurate_seek  = 1;
 }

-static int show_hwaccels(void *optctx, const char *opt, const char *arg)
-{
-    int i;
-
-    printf("Hardware acceleration methods:\n");
-    for (i = 0; i < FF_ARRAY_ELEMS(hwaccels) - 1; i++) {
-        printf("%s\n", hwaccels[i].name);
-    }
-    printf("\n");
-    return 0;
-}
-
 /* return a copy of the input with the stream specifiers removed from the keys */
 static AVDictionary *strip_specifiers(AVDictionary *dict)
 {
@@ -249,7 +230,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
    int sync_file_idx = -1, sync_stream_idx = 0;
    char *p, *sync;
    char *map;
-    char *allow_unused;

    if (*arg == '-') {
        negative = 1;
@@ -294,8 +274,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
            exit_program(1);
        }
    } else {
-        if (allow_unused = strchr(map, '?'))
-            *allow_unused = 0;
        file_idx = strtol(map, &p, 0);
        if (file_idx >= nb_input_files || file_idx < 0) {
            av_log(NULL, AV_LOG_FATAL, "Invalid input file index: %d.\n", file_idx);
@@ -333,13 +311,8 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
    }

    if (!m) {
-        if (allow_unused) {
-            av_log(NULL, AV_LOG_VERBOSE, "Stream map '%s' matches no streams; ignoring.\n", arg);
-        } else {
-            av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n"
-                                       "To ignore this, add a trailing '?' to the map.\n", arg);
-            exit_program(1);
-        }
+        av_log(NULL, AV_LOG_FATAL, "Stream map '%s' matches no streams.\n", arg);
+        exit_program(1);
    }

    av_freep(&map);
@@ -674,11 +647,9 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
        case AVMEDIA_TYPE_VIDEO:
            if(!ist->dec)
                ist->dec = avcodec_find_decoder(dec->codec_id);
-#if FF_API_EMU_EDGE
            if (av_codec_get_lowres(dec)) {
                dec->flags |= CODEC_FLAG_EMU_EDGE;
            }
-#endif

            ist->resample_height  = ist->dec_ctx->height;
            ist->resample_width   = ist->dec_ctx->width;
@@ -951,12 +922,6 @@ static int open_input_file(OptionsContext *o, const char *filename)
        }
    }

-    if (o->start_time_eof != AV_NOPTS_VALUE) {
-        if (ic->duration>0) {
-            o->start_time = o->start_time_eof + ic->duration;
-        } else
-            av_log(NULL, AV_LOG_WARNING, "Cannot use -sseof, duration of %s not known\n", filename);
-    }
    timestamp = (o->start_time == AV_NOPTS_VALUE) ? 0 : o->start_time;
    /* add the stream start time */
    if (!o->seek_timestamp && ic->start_time != AV_NOPTS_VALUE)
@@ -1244,7 +1209,7 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e

    MATCH_PER_STREAM_OPT(qscale, dbl, qscale, oc, st);
    if (qscale >= 0) {
-        ost->enc_ctx->flags |= AV_CODEC_FLAG_QSCALE;
+        ost->enc_ctx->flags |= CODEC_FLAG_QSCALE;
        ost->enc_ctx->global_quality = FF_QP2LAMBDA * qscale;
    }

@@ -1252,9 +1217,9 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
    ost->disposition = av_strdup(ost->disposition);

    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        ost->enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
+        ost->enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;

-    av_dict_copy(&ost->sws_dict, o->g->sws_dict, 0);
+    av_opt_get_int(o->g->sws_opts, "sws_flags", 0, &ost->sws_flags);

    av_dict_copy(&ost->swr_opts, o->g->swr_opts, 0);
    if (ost->enc && av_get_exact_bits_per_sample(ost->enc->id) == 24)
@@ -1474,17 +1439,17 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
        video_enc->rc_override_count = i;

        if (do_psnr)
-            video_enc->flags|= AV_CODEC_FLAG_PSNR;
+            video_enc->flags|= CODEC_FLAG_PSNR;

        /* two pass mode */
        MATCH_PER_STREAM_OPT(pass, i, do_pass, oc, st);
        if (do_pass) {
            if (do_pass & 1) {
-                video_enc->flags |= AV_CODEC_FLAG_PASS1;
+                video_enc->flags |= CODEC_FLAG_PASS1;
                av_dict_set(&ost->encoder_opts, "flags", "+pass1", AV_DICT_APPEND);
            }
            if (do_pass & 2) {
-                video_enc->flags |= AV_CODEC_FLAG_PASS2;
+                video_enc->flags |= CODEC_FLAG_PASS2;
                av_dict_set(&ost->encoder_opts, "flags", "+pass2", AV_DICT_APPEND);
            }
        }
@@ -1494,40 +1459,6 @@ static OutputStream *new_video_stream(OptionsContext *o, AVFormatContext *oc, in
            !(ost->logfile_prefix = av_strdup(ost->logfile_prefix)))
            exit_program(1);

-        if (do_pass) {
-            char logfilename[1024];
-            FILE *f;
-
-            snprintf(logfilename, sizeof(logfilename), "%s-%d.log",
-                     ost->logfile_prefix ? ost->logfile_prefix :
-                                           DEFAULT_PASS_LOGFILENAME_PREFIX,
-                     i);
-            if (!strcmp(ost->enc->name, "libx264")) {
-                av_dict_set(&ost->encoder_opts, "stats", logfilename, AV_DICT_DONT_OVERWRITE);
-            } else {
-                if (video_enc->flags & AV_CODEC_FLAG_PASS2) {
-                    char  *logbuffer = read_file(logfilename);
-
-                    if (!logbuffer) {
-                        av_log(NULL, AV_LOG_FATAL, "Error reading log file '%s' for pass-2 encoding\n",
-                               logfilename);
-                        exit_program(1);
-                    }
-                    video_enc->stats_in = logbuffer;
-                }
-                if (video_enc->flags & AV_CODEC_FLAG_PASS1) {
-                    f = av_fopen_utf8(logfilename, "wb");
-                    if (!f) {
-                        av_log(NULL, AV_LOG_FATAL,
-                               "Cannot write log file '%s' for pass-1 encoding: %s\n",
-                               logfilename, strerror(errno));
-                        exit_program(1);
-                    }
-                    ost->logfile = f;
-                }
-            }
-        }
-
        MATCH_PER_STREAM_OPT(forced_key_frames, str, ost->forced_keyframes, oc, st);
        if (ost->forced_keyframes)
            ost->forced_keyframes = av_strdup(ost->forced_keyframes);
@@ -1806,7 +1737,8 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
 {
    OutputStream *ost;

-    switch (ofilter->type) {
+    switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads,
+                                  ofilter->out_tmp->pad_idx)) {
    case AVMEDIA_TYPE_VIDEO: ost = new_video_stream(o, oc, -1); break;
    case AVMEDIA_TYPE_AUDIO: ost = new_audio_stream(o, oc, -1); break;
    default:
@@ -1839,19 +1771,11 @@ static void init_output_filter(OutputFilter *ofilter, OptionsContext *o,
        exit_program(1);
    }

-    avfilter_inout_free(&ofilter->out_tmp);
-}
-
-static int init_complex_filters(void)
-{
-    int i, ret = 0;
-
-    for (i = 0; i < nb_filtergraphs; i++) {
-        ret = init_complex_filtergraph(filtergraphs[i]);
-        if (ret < 0)
-            return ret;
+    if (configure_output_filter(ofilter->graph, ofilter, ofilter->out_tmp) < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Error configuring filter.\n");
+        exit_program(1);
    }
-    return 0;
+    avfilter_inout_free(&ofilter->out_tmp);
 }

 static int configure_complex_filters(void)
@@ -1876,6 +1800,10 @@ static int open_output_file(OptionsContext *o, const char *filename)
    AVDictionary *unused_opts = NULL;
    AVDictionaryEntry *e = NULL;

+    if (configure_complex_filters() < 0) {
+        av_log(NULL, AV_LOG_FATAL, "Error configuring filters.\n");
+        exit_program(1);
+    }

    if (o->stop_time != INT64_MAX && o->recording_time != INT64_MAX) {
        o->stop_time = INT64_MAX;
@@ -1930,7 +1858,8 @@ static int open_output_file(OptionsContext *o, const char *filename)
            if (!ofilter->out_tmp || ofilter->out_tmp->name)
                continue;

-            switch (ofilter->type) {
+            switch (avfilter_pad_get_type(ofilter->out_tmp->filter_ctx->output_pads,
+                                          ofilter->out_tmp->pad_idx)) {
            case AVMEDIA_TYPE_VIDEO:    o->video_disable    = 1; break;
            case AVMEDIA_TYPE_AUDIO:    o->audio_disable    = 1; break;
            case AVMEDIA_TYPE_SUBTITLE: o->subtitle_disable = 1; break;
@@ -1989,7 +1918,7 @@ static int open_output_file(OptionsContext *o, const char *filename)
            for (i = 0; i < nb_input_streams; i++) {
                int new_area;
                ist = input_streams[i];
-                new_area = ist->st->codec->width * ist->st->codec->height + 100000000*!!ist->st->codec_info_nb_frames;
+                new_area = ist->st->codec->width * ist->st->codec->height;
                if((qcr!=MKTAG('A', 'P', 'I', 'C')) && (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC))
                    new_area = 1;
                if (ist->st->codec->codec_type == AVMEDIA_TYPE_VIDEO &&
@@ -2006,14 +1935,12 @@ static int open_output_file(OptionsContext *o, const char *filename)

        /* audio: most channels */
        if (!o->audio_disable && av_guess_codec(oc->oformat, NULL, filename, NULL, AVMEDIA_TYPE_AUDIO) != AV_CODEC_ID_NONE) {
-            int best_score = 0, idx = -1;
+            int channels = 0, idx = -1;
            for (i = 0; i < nb_input_streams; i++) {
-                int score;
                ist = input_streams[i];
-                score = ist->st->codec->channels + 100000000*!!ist->st->codec_info_nb_frames;
                if (ist->st->codec->codec_type == AVMEDIA_TYPE_AUDIO &&
-                    score > best_score) {
-                    best_score = score;
+                    ist->st->codec->channels > channels) {
+                    channels = ist->st->codec->channels;
                    idx = i;
                }
            }
@@ -2100,7 +2027,6 @@ loop_end:
                if(o->    data_disable && ist->st->codec->codec_type == AVMEDIA_TYPE_DATA)
                    continue;

-                ost = NULL;
                switch (ist->st->codec->codec_type) {
                case AVMEDIA_TYPE_VIDEO:      ost = new_video_stream     (o, oc, src_idx); break;
                case AVMEDIA_TYPE_AUDIO:      ost = new_audio_stream     (o, oc, src_idx); break;
@@ -2124,9 +2050,6 @@ loop_end:
                        exit_program(1);
                    }
                }
-                if (ost)
-                    ost->sync_ist = input_streams[  input_files[map->sync_file_index]->ist_index
-                                                  + map->sync_stream_index];
            }
        }
    }
@@ -2156,7 +2079,7 @@ loop_end:
        avio_read(pb, attachment, len);

        ost = new_attachment_stream(o, oc, -1);
-        ost->stream_copy               = 1;
+        ost->stream_copy               = 0;
        ost->attachment_filename       = o->attachments[i];
        ost->finished                  = 1;
        ost->st->codec->extradata      = attachment;
@@ -2178,12 +2101,6 @@ loop_end:
                exit_program(1);
    }

-    if (!oc->nb_streams && !(oc->oformat->flags & AVFMT_NOSTREAMS)) {
-        av_dump_format(oc, nb_output_files - 1, oc->filename, 1);
-        av_log(NULL, AV_LOG_ERROR, "Output file #%d does not contain any stream\n", nb_output_files - 1);
-        exit_program(1);
-    }
-
    /* check if all codec options have been used */
    unused_opts = strip_specifiers(o->g->codec_opts);
    for (i = of->ost_index; i < nb_output_streams; i++) {
@@ -2226,17 +2143,6 @@ loop_end:
    }
    av_dict_free(&unused_opts);

-    /* set the encoding/decoding_needed flags */
-    for (i = of->ost_index; i < nb_output_streams; i++) {
-        OutputStream *ost = output_streams[i];
-
-        ost->encoding_needed = !ost->stream_copy;
-        if (ost->encoding_needed && ost->source_index >= 0) {
-            InputStream *ist = input_streams[ost->source_index];
-            ist->decoding_needed |= DECODING_FOR_OST;
-        }
-    }
-
    /* check filename in case of an image number is expected */
    if (oc->oformat->flags & AVFMT_NEEDNUMBER) {
        if (!av_filename_number_test(oc->filename)) {
@@ -2332,7 +2238,6 @@ loop_end:
        char type, *val;
        const char *stream_spec;
        int index = 0, j, ret = 0;
-        char now_time[256];

        val = strchr(o->metadata[i].u.str, '=');
        if (!val) {
@@ -2342,17 +2247,6 @@ loop_end:
        }
        *val++ = 0;

-        if (!strcmp(o->metadata[i].u.str, "creation_time") &&
-            !strcmp(val, "now")) {
-            time_t now = time(0);
-            struct tm *ptm, tmbuf;
-            ptm = localtime_r(&now, &tmbuf);
-            if (ptm) {
-                if (strftime(now_time, sizeof(now_time), "%Y-%m-%d %H:%M:%S", ptm))
-                    val = now_time;
-            }
-        }
-
        parse_meta_type(o->metadata[i].specifier, &type, &index, &stream_spec);
        if (type == 's') {
            for (j = 0; j < oc->nb_streams; j++) {
@@ -2545,10 +2439,8 @@ static int opt_vstats(void *optctx, const char *opt, const char *arg)
    time_t today2 = time(NULL);
    struct tm *today = localtime(&today2);

-    if (!today) { // maybe tomorrow
-        av_log(NULL, AV_LOG_FATAL, "Unable to get current time: %s\n", strerror(errno));
-        exit_program(1);
-    }
+    if (!today)
+        return AVERROR(errno);

    snprintf(filename, sizeof(filename), "vstats_%02d%02d%02d.log", today->tm_hour, today->tm_min,
             today->tm_sec);
@@ -2950,13 +2842,6 @@ int ffmpeg_parse_options(int argc, char **argv)
        goto fail;
    }

-    /* create the complex filtergraphs */
-    ret = init_complex_filters();
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_FATAL, "Error initializing complex filters.\n");
-        goto fail;
-    }
-
    /* open output files */
    ret = open_files(&octx.groups[GROUP_OUTFILE], "output", open_output_file);
    if (ret < 0) {
@@ -2964,13 +2849,6 @@ int ffmpeg_parse_options(int argc, char **argv)
        goto fail;
    }

-    /* configure the complex filtergraphs */
-    ret = configure_complex_filters();
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_FATAL, "Error configuring complex filters.\n");
-        goto fail;
-    }
-
 fail:
    uninit_parse_context(&octx);
    if (ret < 0) {
@@ -3045,9 +2923,6 @@ const OptionDef options[] = {
    { "ss",             HAS_ARG | OPT_TIME | OPT_OFFSET |
                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time) },
        "set the start time offset", "time_off" },
-    { "sseof",          HAS_ARG | OPT_TIME | OPT_OFFSET |
-                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time_eof) },
-        "set the start time offset relative to EOF", "time_off" },
    { "seek_timestamp", HAS_ARG | OPT_INT | OPT_OFFSET |
                        OPT_INPUT,                                   { .off = OFFSET(seek_timestamp) },
        "enable/disable seeking by timestamp with -ss" },
@@ -3085,8 +2960,8 @@ const OptionDef options[] = {
                        OPT_INPUT,                                   { .off = OFFSET(rate_emu) },
        "read input at native frame rate", "" },
    { "target",         HAS_ARG | OPT_PERFILE | OPT_OUTPUT,          { .func_arg = opt_target },
-        "specify target file type (\"vcd\", \"svcd\", \"dvd\", \"dv\" or \"dv50\" "
-        "with optional prefixes \"pal-\", \"ntsc-\" or \"film-\")", "type" },
+        "specify target file type (\"vcd\", \"svcd\", \"dvd\","
+        " \"dv\", \"dv50\", \"pal-vcd\", \"ntsc-svcd\", ...)", "type" },
    { "vsync",          HAS_ARG | OPT_EXPERT,                        { opt_vsync },
        "video sync method", "" },
    { "frame_drop_threshold", HAS_ARG | OPT_FLOAT | OPT_EXPERT,      { &frame_drop_threshold },
@@ -3248,15 +3123,10 @@ const OptionDef options[] = {
        "use HW accelerated decoding", "hwaccel name" },
    { "hwaccel_device",   OPT_VIDEO | OPT_STRING | HAS_ARG | OPT_EXPERT |
                          OPT_SPEC | OPT_INPUT,                                  { .off = OFFSET(hwaccel_devices) },
-        "select a device for HW acceleration", "devicename" },
+        "select a device for HW acceleration" "devicename" },
 #if HAVE_VDPAU_X11
    { "vdpau_api_ver", HAS_ARG | OPT_INT | OPT_EXPERT, { &vdpau_api_ver }, "" },
 #endif
-#if CONFIG_VDA || CONFIG_VIDEOTOOLBOX
-    { "videotoolbox_pixfmt", HAS_ARG | OPT_STRING | OPT_EXPERT, { &videotoolbox_pixfmt}, "" },
-#endif
-    { "hwaccels",         OPT_EXIT,                                              { .func_arg = show_hwaccels },
-        "show available HW acceleration methods" },
    { "autorotate",       HAS_ARG | OPT_BOOL | OPT_SPEC |
                          OPT_EXPERT | OPT_INPUT,                                { .off = OFFSET(autorotate) },
        "automatically insert correct rotate filters" },
--- a/ffmpeg_vda.c
+++ b/ffmpeg_vda.c
@@ -0,0 +1,136 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavcodec/avcodec.h"
+#include "libavcodec/vda.h"
+#include "libavutil/imgutils.h"
+
+#include "ffmpeg.h"
+
+typedef struct VDAContext {
+    AVFrame *tmp_frame;
+} VDAContext;
+
+static int vda_retrieve_data(AVCodecContext *s, AVFrame *frame)
+{
+    InputStream *ist = s->opaque;
+    VDAContext  *vda = ist->hwaccel_ctx;
+    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
+    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
+    CVReturn err;
+    uint8_t *data[4] = { 0 };
+    int linesize[4] = { 0 };
+    int planes, ret, i;
+
+    av_frame_unref(vda->tmp_frame);
+
+    switch (pixel_format) {
+    case kCVPixelFormatType_420YpCbCr8Planar: vda->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
+    case kCVPixelFormatType_422YpCbCr8:       vda->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
+    default:
+        av_log(NULL, AV_LOG_ERROR,
+               "Unsupported pixel format: %u\n", pixel_format);
+        return AVERROR(ENOSYS);
+    }
+
+    vda->tmp_frame->width  = frame->width;
+    vda->tmp_frame->height = frame->height;
+    ret = av_frame_get_buffer(vda->tmp_frame, 32);
+    if (ret < 0)
+        return ret;
+
+    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+    if (err != kCVReturnSuccess) {
+        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
+        return AVERROR_UNKNOWN;
+    }
+
+    if (CVPixelBufferIsPlanar(pixbuf)) {
+
+        planes = CVPixelBufferGetPlaneCount(pixbuf);
+        for (i = 0; i < planes; i++) {
+            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
+            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
+        }
+    } else {
+        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
+        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
+    }
+
+    av_image_copy(vda->tmp_frame->data, vda->tmp_frame->linesize,
+                  (const uint8_t **)data, linesize, vda->tmp_frame->format,
+                  frame->width, frame->height);
+
+    ret = av_frame_copy_props(vda->tmp_frame, frame);
+    CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
+
+    if (ret < 0)
+        return ret;
+
+    av_frame_unref(frame);
+    av_frame_move_ref(frame, vda->tmp_frame);
+
+    return 0;
+}
+
+static void vda_uninit(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    VDAContext  *vda = ist->hwaccel_ctx;
+
+    ist->hwaccel_uninit        = NULL;
+    ist->hwaccel_retrieve_data = NULL;
+
+    av_frame_free(&vda->tmp_frame);
+
+    av_vda_default_free(s);
+    av_freep(&ist->hwaccel_ctx);
+}
+
+int vda_init(AVCodecContext *s)
+{
+    InputStream *ist = s->opaque;
+    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
+    VDAContext *vda;
+    int ret;
+
+    vda = av_mallocz(sizeof(*vda));
+    if (!vda)
+        return AVERROR(ENOMEM);
+
+    ist->hwaccel_ctx           = vda;
+    ist->hwaccel_uninit        = vda_uninit;
+    ist->hwaccel_retrieve_data = vda_retrieve_data;
+
+    vda->tmp_frame = av_frame_alloc();
+    if (!vda->tmp_frame) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    ret = av_vda_default_init(s);
+    if (ret < 0) {
+        av_log(NULL, loglevel, "Error creating VDA decoder.\n");
+        goto fail;
+    }
+
+    return 0;
+fail:
+    vda_uninit(s);
+    return ret;
+}
--- a/ffmpeg_vdpau.c
+++ b/ffmpeg_vdpau.c
@@ -289,8 +289,7 @@ do {

        s->hwaccel_context = vdpau_ctx;
    } else
-    if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address,
-                              AV_HWACCEL_FLAG_IGNORE_LEVEL))
+    if (av_vdpau_bind_context(s, ctx->device, ctx->get_proc_address, 0))
        goto fail;

    ctx->get_information_string(&vendor);
--- a/ffmpeg_videotoolbox.c
+++ b/ffmpeg_videotoolbox.c
@@ -1,187 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <CoreServices/CoreServices.h>
-
-#include "config.h"
-#include "libavcodec/avcodec.h"
-#if CONFIG_VDA
-#  include "libavcodec/vda.h"
-#endif
-#if CONFIG_VIDEOTOOLBOX
-#  include "libavcodec/videotoolbox.h"
-#endif
-#include "libavutil/imgutils.h"
-#include "ffmpeg.h"
-
-typedef struct VTContext {
-    AVFrame *tmp_frame;
-} VTContext;
-
-char *videotoolbox_pixfmt;
-
-static int videotoolbox_retrieve_data(AVCodecContext *s, AVFrame *frame)
-{
-    InputStream *ist = s->opaque;
-    VTContext  *vt = ist->hwaccel_ctx;
-    CVPixelBufferRef pixbuf = (CVPixelBufferRef)frame->data[3];
-    OSType pixel_format = CVPixelBufferGetPixelFormatType(pixbuf);
-    CVReturn err;
-    uint8_t *data[4] = { 0 };
-    int linesize[4] = { 0 };
-    int planes, ret, i;
-    char codec_str[32];
-
-    av_frame_unref(vt->tmp_frame);
-
-    switch (pixel_format) {
-    case kCVPixelFormatType_420YpCbCr8Planar: vt->tmp_frame->format = AV_PIX_FMT_YUV420P; break;
-    case kCVPixelFormatType_422YpCbCr8:       vt->tmp_frame->format = AV_PIX_FMT_UYVY422; break;
-    case kCVPixelFormatType_32BGRA:           vt->tmp_frame->format = AV_PIX_FMT_BGRA; break;
-#ifdef kCFCoreFoundationVersionNumber10_7
-    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: vt->tmp_frame->format = AV_PIX_FMT_NV12; break;
-#endif
-    default:
-        av_get_codec_tag_string(codec_str, sizeof(codec_str), s->codec_tag);
-        av_log(NULL, AV_LOG_ERROR,
-               "%s: Unsupported pixel format: %s\n", codec_str, videotoolbox_pixfmt);
-        return AVERROR(ENOSYS);
-    }
-
-    vt->tmp_frame->width  = frame->width;
-    vt->tmp_frame->height = frame->height;
-    ret = av_frame_get_buffer(vt->tmp_frame, 32);
-    if (ret < 0)
-        return ret;
-
-    err = CVPixelBufferLockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
-    if (err != kCVReturnSuccess) {
-        av_log(NULL, AV_LOG_ERROR, "Error locking the pixel buffer.\n");
-        return AVERROR_UNKNOWN;
-    }
-
-    if (CVPixelBufferIsPlanar(pixbuf)) {
-
-        planes = CVPixelBufferGetPlaneCount(pixbuf);
-        for (i = 0; i < planes; i++) {
-            data[i]     = CVPixelBufferGetBaseAddressOfPlane(pixbuf, i);
-            linesize[i] = CVPixelBufferGetBytesPerRowOfPlane(pixbuf, i);
-        }
-    } else {
-        data[0] = CVPixelBufferGetBaseAddress(pixbuf);
-        linesize[0] = CVPixelBufferGetBytesPerRow(pixbuf);
-    }
-
-    av_image_copy(vt->tmp_frame->data, vt->tmp_frame->linesize,
-                  (const uint8_t **)data, linesize, vt->tmp_frame->format,
-                  frame->width, frame->height);
-
-    ret = av_frame_copy_props(vt->tmp_frame, frame);
-    CVPixelBufferUnlockBaseAddress(pixbuf, kCVPixelBufferLock_ReadOnly);
-    if (ret < 0)
-        return ret;
-
-    av_frame_unref(frame);
-    av_frame_move_ref(frame, vt->tmp_frame);
-
-    return 0;
-}
-
-static void videotoolbox_uninit(AVCodecContext *s)
-{
-    InputStream *ist = s->opaque;
-    VTContext  *vt = ist->hwaccel_ctx;
-
-    ist->hwaccel_uninit        = NULL;
-    ist->hwaccel_retrieve_data = NULL;
-
-    av_frame_free(&vt->tmp_frame);
-
-    if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
-#if CONFIG_VIDEOTOOLBOX
-        av_videotoolbox_default_free(s);
-#endif
-    } else {
-#if CONFIG_VDA
-        av_vda_default_free(s);
-#endif
-    }
-    av_freep(&ist->hwaccel_ctx);
-}
-
-int videotoolbox_init(AVCodecContext *s)
-{
-    InputStream *ist = s->opaque;
-    int loglevel = (ist->hwaccel_id == HWACCEL_AUTO) ? AV_LOG_VERBOSE : AV_LOG_ERROR;
-    int ret = 0;
-    VTContext *vt;
-
-    vt = av_mallocz(sizeof(*vt));
-    if (!vt)
-        return AVERROR(ENOMEM);
-
-    ist->hwaccel_ctx           = vt;
-    ist->hwaccel_uninit        = videotoolbox_uninit;
-    ist->hwaccel_retrieve_data = videotoolbox_retrieve_data;
-
-    vt->tmp_frame = av_frame_alloc();
-    if (!vt->tmp_frame) {
-        ret = AVERROR(ENOMEM);
-        goto fail;
-    }
-
-    if (ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX) {
-#if CONFIG_VIDEOTOOLBOX
-        if (!videotoolbox_pixfmt) {
-            ret = av_videotoolbox_default_init(s);
-        } else {
-            AVVideotoolboxContext *vtctx = av_videotoolbox_alloc_context();
-            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
-                                                               videotoolbox_pixfmt,
-                                                               kCFStringEncodingUTF8);
-            vtctx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
-            ret = av_videotoolbox_default_init2(s, vtctx);
-            CFRelease(pixfmt_str);
-        }
-#endif
-    } else {
-#if CONFIG_VDA
-        if (!videotoolbox_pixfmt) {
-            ret = av_vda_default_init(s);
-        } else {
-            AVVDAContext *vdactx = av_vda_alloc_context();
-            CFStringRef pixfmt_str = CFStringCreateWithCString(kCFAllocatorDefault,
-                                                               videotoolbox_pixfmt,
-                                                               kCFStringEncodingUTF8);
-            vdactx->cv_pix_fmt_type = UTGetOSTypeFromString(pixfmt_str);
-            ret = av_vda_default_init2(s, vdactx);
-            CFRelease(pixfmt_str);
-        }
-#endif
-    }
-    if (ret < 0) {
-        av_log(NULL, loglevel,
-               "Error creating %s decoder.\n", ist->hwaccel_id == HWACCEL_VIDEOTOOLBOX ? "Videotoolbox" : "VDA");
-        goto fail;
-    }
-
-    return 0;
-fail:
-    videotoolbox_uninit(s);
-    return ret;
-}
--- a/ffplay.c
+++ b/ffplay.c
@@ -31,6 +31,7 @@
 #include <stdint.h>

 #include "libavutil/avstring.h"
+#include "libavutil/colorspace.h"
 #include "libavutil/eval.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/pixdesc.h"
@@ -65,9 +66,7 @@ const char program_name[] = "ffplay";
 const int program_birth_year = 2003;

 #define MAX_QUEUE_SIZE (15 * 1024 * 1024)
-#define MIN_FRAMES 25
-#define EXTERNAL_CLOCK_MIN_FRAMES 2
-#define EXTERNAL_CLOCK_MAX_FRAMES 10
+#define MIN_FRAMES 5

 /* Minimum SDL audio buffer size, in samples. */
 #define SDL_AUDIO_MIN_BUFFER_SIZE 512
@@ -103,7 +102,7 @@ const int program_birth_year = 2003;

 #define CURSOR_HIDE_DELAY 1000000

-static unsigned sws_flags = SWS_BICUBIC;
+static int64_t sws_flags = SWS_BICUBIC;

 typedef struct MyAVPacketList {
    AVPacket pkt;
@@ -224,9 +223,6 @@ typedef struct VideoState {
    Decoder viddec;
    Decoder subdec;

-    int viddec_width;
-    int viddec_height;
-
    int audio_stream;

    int av_sync_type;
@@ -282,7 +278,6 @@ typedef struct VideoState {
 #if !CONFIG_AVFILTER
    struct SwsContext *img_convert_ctx;
 #endif
-    struct SwsContext *sub_convert_ctx;
    SDL_Rect last_display_rect;
    int eof;

@@ -834,50 +829,229 @@ static void fill_border(int xleft, int ytop, int width, int height, int x, int y
 #define ALPHA_BLEND(a, oldp, newp, s)\
 ((((oldp << s) * (255 - (a))) + (newp * (a))) / (255 << s))

+#define RGBA_IN(r, g, b, a, s)\
+{\
+    unsigned int v = ((const uint32_t *)(s))[0];\
+    a = (v >> 24) & 0xff;\
+    r = (v >> 16) & 0xff;\
+    g = (v >> 8) & 0xff;\
+    b = v & 0xff;\
+}
+
+#define YUVA_IN(y, u, v, a, s, pal)\
+{\
+    unsigned int val = ((const uint32_t *)(pal))[*(const uint8_t*)(s)];\
+    a = (val >> 24) & 0xff;\
+    y = (val >> 16) & 0xff;\
+    u = (val >> 8) & 0xff;\
+    v = val & 0xff;\
+}
+
+#define YUVA_OUT(d, y, u, v, a)\
+{\
+    ((uint32_t *)(d))[0] = (a << 24) | (y << 16) | (u << 8) | v;\
+}


 #define BPP 1

 static void blend_subrect(AVPicture *dst, const AVSubtitleRect *rect, int imgw, int imgh)
 {
-    int x, y, Y, U, V, A;
+    int wrap, wrap3, width2, skip2;
+    int y, u, v, a, u1, v1, a1, w, h;
    uint8_t *lum, *cb, *cr;
+    const uint8_t *p;
+    const uint32_t *pal;
    int dstx, dsty, dstw, dsth;
-    const AVPicture *src = &rect->pict;

    dstw = av_clip(rect->w, 0, imgw);
    dsth = av_clip(rect->h, 0, imgh);
    dstx = av_clip(rect->x, 0, imgw - dstw);
    dsty = av_clip(rect->y, 0, imgh - dsth);
-    lum = dst->data[0] + dstx + dsty * dst->linesize[0];
-    cb  = dst->data[1] + dstx/2 + (dsty >> 1) * dst->linesize[1];
-    cr  = dst->data[2] + dstx/2 + (dsty >> 1) * dst->linesize[2];
+    lum = dst->data[0] + dsty * dst->linesize[0];
+    cb  = dst->data[1] + (dsty >> 1) * dst->linesize[1];
+    cr  = dst->data[2] + (dsty >> 1) * dst->linesize[2];

-    for (y = 0; y<dsth; y++) {
-        for (x = 0; x<dstw; x++) {
-            Y = src->data[0][x + y*src->linesize[0]];
-            A = src->data[3][x + y*src->linesize[3]];
-            lum[0] = ALPHA_BLEND(A, lum[0], Y, 0);
-            lum++;
-        }
-        lum += dst->linesize[0] - dstw;
-    }
+    width2 = ((dstw + 1) >> 1) + (dstx & ~dstw & 1);
+    skip2 = dstx >> 1;
+    wrap = dst->linesize[0];
+    wrap3 = rect->pict.linesize[0];
+    p = rect->pict.data[0];
+    pal = (const uint32_t *)rect->pict.data[1];  /* Now in YCrCb! */

-    for (y = 0; y<dsth/2; y++) {
-        for (x = 0; x<dstw/2; x++) {
-            U = src->data[1][x + y*src->linesize[1]];
-            V = src->data[2][x + y*src->linesize[2]];
-            A = src->data[3][2*x     +  2*y   *src->linesize[3]]
-              + src->data[3][2*x + 1 +  2*y   *src->linesize[3]]
-              + src->data[3][2*x + 1 + (2*y+1)*src->linesize[3]]
-              + src->data[3][2*x     + (2*y+1)*src->linesize[3]];
-            cb[0] = ALPHA_BLEND(A>>2, cb[0], U, 0);
-            cr[0] = ALPHA_BLEND(A>>2, cr[0], V, 0);
+    if (dsty & 1) {
+        lum += dstx;
+        cb += skip2;
+        cr += skip2;
+
+        if (dstx & 1) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
            cb++;
            cr++;
+            lum++;
+            p += BPP;
+        }
+        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+            cb++;
+            cr++;
+            p += 2 * BPP;
+            lum += 2;
+        }
+        if (w) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
+            p++;
+            lum++;
+        }
+        p += wrap3 - dstw * BPP;
+        lum += wrap - dstw - dstx;
+        cb += dst->linesize[1] - width2 - skip2;
+        cr += dst->linesize[2] - width2 - skip2;
+    }
+    for (h = dsth - (dsty & 1); h >= 2; h -= 2) {
+        lum += dstx;
+        cb += skip2;
+        cr += skip2;
+
+        if (dstx & 1) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            p += wrap3;
+            lum += wrap;
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+            cb++;
+            cr++;
+            p += -wrap3 + BPP;
+            lum += -wrap + 1;
+        }
+        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+            p += wrap3;
+            lum += wrap;
+
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 2);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 2);
+
+            cb++;
+            cr++;
+            p += -wrap3 + 2 * BPP;
+            lum += -wrap + 2;
+        }
+        if (w) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            p += wrap3;
+            lum += wrap;
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u1, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v1, 1);
+            cb++;
+            cr++;
+            p += -wrap3 + BPP;
+            lum += -wrap + 1;
+        }
+        p += wrap3 + (wrap3 - dstw * BPP);
+        lum += wrap + (wrap - dstw - dstx);
+        cb += dst->linesize[1] - width2 - skip2;
+        cr += dst->linesize[2] - width2 - skip2;
+    }
+    /* handle odd height */
+    if (h) {
+        lum += dstx;
+        cb += skip2;
+        cr += skip2;
+
+        if (dstx & 1) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
+            cb++;
+            cr++;
+            lum++;
+            p += BPP;
+        }
+        for (w = dstw - (dstx & 1); w >= 2; w -= 2) {
+            YUVA_IN(y, u, v, a, p, pal);
+            u1 = u;
+            v1 = v;
+            a1 = a;
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+
+            YUVA_IN(y, u, v, a, p + BPP, pal);
+            u1 += u;
+            v1 += v;
+            a1 += a;
+            lum[1] = ALPHA_BLEND(a, lum[1], y, 0);
+            cb[0] = ALPHA_BLEND(a1 >> 2, cb[0], u, 1);
+            cr[0] = ALPHA_BLEND(a1 >> 2, cr[0], v, 1);
+            cb++;
+            cr++;
+            p += 2 * BPP;
+            lum += 2;
+        }
+        if (w) {
+            YUVA_IN(y, u, v, a, p, pal);
+            lum[0] = ALPHA_BLEND(a, lum[0], y, 0);
+            cb[0] = ALPHA_BLEND(a >> 2, cb[0], u, 0);
+            cr[0] = ALPHA_BLEND(a >> 2, cr[0], v, 0);
        }
-        cb += dst->linesize[1] - dstw/2;
-        cr += dst->linesize[2] - dstw/2;
    }
 }

@@ -1132,7 +1306,6 @@ static void stream_close(VideoState *is)
 #if !CONFIG_AVFILTER
    sws_freeContext(is->img_convert_ctx);
 #endif
-    sws_freeContext(is->sub_convert_ctx);
    av_free(is);
 }

@@ -1302,11 +1475,11 @@ static double get_master_clock(VideoState *is)
 }

 static void check_external_clock_speed(VideoState *is) {
-   if (is->video_stream >= 0 && is->videoq.nb_packets <= EXTERNAL_CLOCK_MIN_FRAMES ||
-       is->audio_stream >= 0 && is->audioq.nb_packets <= EXTERNAL_CLOCK_MIN_FRAMES) {
+   if (is->video_stream >= 0 && is->videoq.nb_packets <= MIN_FRAMES / 2 ||
+       is->audio_stream >= 0 && is->audioq.nb_packets <= MIN_FRAMES / 2) {
       set_clock_speed(&is->extclk, FFMAX(EXTERNAL_CLOCK_SPEED_MIN, is->extclk.speed - EXTERNAL_CLOCK_SPEED_STEP));
-   } else if ((is->video_stream < 0 || is->videoq.nb_packets > EXTERNAL_CLOCK_MAX_FRAMES) &&
-              (is->audio_stream < 0 || is->audioq.nb_packets > EXTERNAL_CLOCK_MAX_FRAMES)) {
+   } else if ((is->video_stream < 0 || is->videoq.nb_packets > MIN_FRAMES * 2) &&
+              (is->audio_stream < 0 || is->audioq.nb_packets > MIN_FRAMES * 2)) {
       set_clock_speed(&is->extclk, FFMIN(EXTERNAL_CLOCK_SPEED_MAX, is->extclk.speed + EXTERNAL_CLOCK_SPEED_STEP));
   } else {
       double speed = is->extclk.speed;
@@ -1678,18 +1851,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
        av_picture_copy(&pict, (AVPicture *)src_frame,
                        src_frame->format, vp->width, vp->height);
 #else
-        {
-            AVDictionaryEntry *e = av_dict_get(sws_dict, "sws_flags", NULL, 0);
-            if (e) {
-                const AVClass *class = sws_get_class();
-                const AVOption    *o = av_opt_find(&class, "sws_flags", NULL, 0,
-                                                   AV_OPT_SEARCH_FAKE_OBJ);
-                int ret = av_opt_eval_flags(&class, o, e->value, &sws_flags);
-                if (ret < 0)
-                    exit(1);
-            }
-        }
-
+        av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
        is->img_convert_ctx = sws_getCachedContext(is->img_convert_ctx,
            vp->width, vp->height, src_frame->format, vp->width, vp->height,
            AV_PIX_FMT_YUV420P, sws_flags, NULL, NULL, NULL);
@@ -1731,9 +1893,6 @@ static int get_video_frame(VideoState *is, AVFrame *frame)

        frame->sample_aspect_ratio = av_guess_sample_aspect_ratio(is->ic, is->video_st, frame);

-        is->viddec_width  = frame->width;
-        is->viddec_height = frame->height;
-
        if (framedrop>0 || (framedrop && get_master_sync_type(is) != AV_SYNC_VIDEO_MASTER)) {
            if (frame->pts != AV_NOPTS_VALUE) {
                double diff = dpts - get_master_clock(is);
@@ -1799,23 +1958,15 @@ fail:
 static int configure_video_filters(AVFilterGraph *graph, VideoState *is, const char *vfilters, AVFrame *frame)
 {
    static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_YUV420P, AV_PIX_FMT_NONE };
-    char sws_flags_str[512] = "";
+    char sws_flags_str[128];
    char buffersrc_args[256];
    int ret;
    AVFilterContext *filt_src = NULL, *filt_out = NULL, *last_filter = NULL;
    AVCodecContext *codec = is->video_st->codec;
    AVRational fr = av_guess_frame_rate(is->ic, is->video_st, NULL);
-    AVDictionaryEntry *e = NULL;
-
-    while ((e = av_dict_get(sws_dict, "", e, AV_DICT_IGNORE_SUFFIX))) {
-        if (!strcmp(e->key, "sws_flags")) {
-            av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", "flags", e->value);
-        } else
-            av_strlcatf(sws_flags_str, sizeof(sws_flags_str), "%s=%s:", e->key, e->value);
-    }
-    if (strlen(sws_flags_str))
-        sws_flags_str[strlen(sws_flags_str)-1] = '\0';

+    av_opt_get_int(sws_opts, "sws_flags", 0, &sws_flags);
+    snprintf(sws_flags_str, sizeof(sws_flags_str), "flags=%"PRId64, sws_flags);
    graph->scale_sws_opts = av_strdup(sws_flags_str);

    snprintf(buffersrc_args, sizeof(buffersrc_args),
@@ -2177,7 +2328,8 @@ static int subtitle_thread(void *arg)
    Frame *sp;
    int got_subtitle;
    double pts;
-    int i;
+    int i, j;
+    int r, g, b, y, u, v, a;

    for (;;) {
        if (!(sp = frame_queue_peek_writable(&is->subpq)))
@@ -2196,41 +2348,14 @@ static int subtitle_thread(void *arg)

            for (i = 0; i < sp->sub.num_rects; i++)
            {
-                int in_w = sp->sub.rects[i]->w;
-                int in_h = sp->sub.rects[i]->h;
-                int subw = is->subdec.avctx->width  ? is->subdec.avctx->width  : is->viddec_width;
-                int subh = is->subdec.avctx->height ? is->subdec.avctx->height : is->viddec_height;
-                int out_w = is->viddec_width  ? in_w * is->viddec_width  / subw : in_w;
-                int out_h = is->viddec_height ? in_h * is->viddec_height / subh : in_h;
-                AVPicture newpic;
-
-                //can not use avpicture_alloc as it is not compatible with avsubtitle_free()
-                av_image_fill_linesizes(newpic.linesize, AV_PIX_FMT_YUVA420P, out_w);
-                newpic.data[0] = av_malloc(newpic.linesize[0] * out_h);
-                newpic.data[3] = av_malloc(newpic.linesize[3] * out_h);
-                newpic.data[1] = av_malloc(newpic.linesize[1] * ((out_h+1)/2));
-                newpic.data[2] = av_malloc(newpic.linesize[2] * ((out_h+1)/2));
-
-                is->sub_convert_ctx = sws_getCachedContext(is->sub_convert_ctx,
-                    in_w, in_h, AV_PIX_FMT_PAL8, out_w, out_h,
-                    AV_PIX_FMT_YUVA420P, sws_flags, NULL, NULL, NULL);
-                if (!is->sub_convert_ctx || !newpic.data[0] || !newpic.data[3] ||
-                    !newpic.data[1] || !newpic.data[2]
-                ) {
-                    av_log(NULL, AV_LOG_FATAL, "Cannot initialize the sub conversion context\n");
-                    exit(1);
+                for (j = 0; j < sp->sub.rects[i]->nb_colors; j++)
+                {
+                    RGBA_IN(r, g, b, a, (uint32_t*)sp->sub.rects[i]->pict.data[1] + j);
+                    y = RGB_TO_Y_CCIR(r, g, b);
+                    u = RGB_TO_U_CCIR(r, g, b, 0);
+                    v = RGB_TO_V_CCIR(r, g, b, 0);
+                    YUVA_OUT((uint32_t*)sp->sub.rects[i]->pict.data[1] + j, y, u, v, a);
                }
-                sws_scale(is->sub_convert_ctx,
-                          (void*)sp->sub.rects[i]->pict.data, sp->sub.rects[i]->pict.linesize,
-                          0, in_h, newpic.data, newpic.linesize);
-
-                av_free(sp->sub.rects[i]->pict.data[0]);
-                av_free(sp->sub.rects[i]->pict.data[1]);
-                sp->sub.rects[i]->pict = newpic;
-                sp->sub.rects[i]->w = out_w;
-                sp->sub.rects[i]->h = out_h;
-                sp->sub.rects[i]->x = sp->sub.rects[i]->x * out_w / in_w;
-                sp->sub.rects[i]->y = sp->sub.rects[i]->y * out_h / in_h;
            }

            /* now we can update the picture count */
@@ -2323,13 +2448,6 @@ static int audio_decode_frame(VideoState *is)
        return -1;

    do {
-#if defined(_WIN32)
-        while (frame_queue_nb_remaining(&is->sampq) == 0) {
-            if ((av_gettime_relative() - audio_callback_time) > 1000000LL * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec / 2)
-                return -1;
-            av_usleep (1000);
-        }
-#endif
        if (!(af = frame_queue_peek_readable(&is->sampq)))
            return -1;
        frame_queue_next(&is->sampq);
@@ -2577,15 +2695,10 @@ static int stream_component_open(VideoState *is, int stream_index)
    }
    av_codec_set_lowres(avctx, stream_lowres);

-#if FF_API_EMU_EDGE
    if(stream_lowres) avctx->flags |= CODEC_FLAG_EMU_EDGE;
-#endif
-    if (fast)
-        avctx->flags2 |= AV_CODEC_FLAG2_FAST;
-#if FF_API_EMU_EDGE
-    if(codec->capabilities & AV_CODEC_CAP_DR1)
+    if (fast)   avctx->flags2 |= CODEC_FLAG2_FAST;
+    if(codec->capabilities & CODEC_CAP_DR1)
        avctx->flags |= CODEC_FLAG_EMU_EDGE;
-#endif

    opts = filter_codec_opts(codec_opts, avctx->codec_id, ic, ic->streams[stream_index], codec);
    if (!av_dict_get(opts, "threads", NULL, 0))
@@ -2658,9 +2771,6 @@ static int stream_component_open(VideoState *is, int stream_index)
        is->video_stream = stream_index;
        is->video_st = ic->streams[stream_index];

-        is->viddec_width  = avctx->width;
-        is->viddec_height = avctx->height;
-
        decoder_init(&is->viddec, avctx, &is->videoq, is->continue_read_thread);
        decoder_start(&is->viddec, video_thread, is);
        is->queue_attachments_req = 1;
--- a/ffprobe.c
+++ b/ffprobe.c
@@ -2831,9 +2831,6 @@ static int opt_show_format_entry(void *optctx, const char *opt, const char *arg)
    char *buf = av_asprintf("format=%s", arg);
    int ret;

-    if (!buf)
-        return AVERROR(ENOMEM);
-
    av_log(NULL, AV_LOG_WARNING,
           "Option '%s' is deprecated, use '-show_entries format=%s' instead\n",
           opt, arg);
--- a/ffserver.c
+++ b/ffserver.c
@@ -31,7 +31,7 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include "libavformat/avformat.h"
-/* FIXME: those are internal headers, ffserver _really_ shouldn't use them */
+// FIXME those are internal headers, ffserver _really_ shouldn't use them
 #include "libavformat/ffm.h"
 #include "libavformat/network.h"
 #include "libavformat/os_support.h"
@@ -209,7 +209,6 @@ static void close_connection(HTTPContext *c);

 /* HTTP handling */
 static int handle_connection(HTTPContext *c);
-static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream);
 static void compute_status(HTTPContext *c);
 static int open_input_stream(HTTPContext *c, const char *info);
 static int http_parse_request(HTTPContext *c);
@@ -251,8 +250,7 @@ static unsigned int nb_connections;

 static uint64_t current_bandwidth;

-/* Making this global saves on passing it around everywhere */
-static int64_t cur_time;
+static int64_t cur_time;           // Making this global saves on passing it around everywhere

 static AVLFG random_state;

@@ -316,12 +314,12 @@ static char *ctime1(char *buf2, int buf_size)
 static void http_vlog(const char *fmt, va_list vargs)
 {
    static int print_prefix = 1;
-    char buf[32];

    if (!logfile)
        return;

    if (print_prefix) {
+        char buf[32];
        ctime1(buf, sizeof(buf));
        fprintf(logfile, "%s ", buf);
    }
@@ -506,7 +504,8 @@ static void start_multicast(void)
        random1 = av_lfg_get(&random_state);

        /* open the RTP connection */
-        snprintf(session_id, sizeof(session_id), "%08x%08x", random0, random1);
+        snprintf(session_id, sizeof(session_id), "%08x%08x",
+                 random0, random1);

        /* choose a port if none given */
        if (stream->multicast_port == 0) {
@@ -631,8 +630,9 @@ static int http_server(void)
                    poll_entry++;
                } else {
                    /* when ffserver is doing the timing, we work by
-                     * looking at which packet needs to be sent every
-                     * 10 ms (one tick wait XXX: 10 ms assumed) */
+                       looking at which packet needs to be sent every
+                       10 ms */
+                    /* one tick wait XXX: 10 ms assumed */
                    if (delay > 10)
                        delay = 10;
                }
@@ -655,7 +655,7 @@ static int http_server(void)
        }

        /* wait for an event on one connection. We poll at least every
-         * second to handle timeouts */
+           second to handle timeouts */
        do {
            ret = poll(poll_table, poll_entry - poll_table, delay);
            if (ret < 0 && ff_neterrno() != AVERROR(EAGAIN) &&
@@ -703,9 +703,13 @@ static void start_wait_request(HTTPContext *c, int is_rtsp)
    c->buffer_ptr = c->buffer;
    c->buffer_end = c->buffer + c->buffer_size - 1; /* leave room for '\0' */

-    c->state = is_rtsp ? RTSPSTATE_WAIT_REQUEST : HTTPSTATE_WAIT_REQUEST;
-    c->timeout = cur_time +
-                 (is_rtsp ? RTSP_REQUEST_TIMEOUT : HTTP_REQUEST_TIMEOUT);
+    if (is_rtsp) {
+        c->timeout = cur_time + RTSP_REQUEST_TIMEOUT;
+        c->state = RTSPSTATE_WAIT_REQUEST;
+    } else {
+        c->timeout = cur_time + HTTP_REQUEST_TIMEOUT;
+        c->state = HTTPSTATE_WAIT_REQUEST;
+    }
 }

 static void http_send_too_busy_reply(int fd)
@@ -783,6 +787,7 @@ static void close_connection(HTTPContext *c)
    HTTPContext **cp, *c1;
    int i, nb_streams;
    AVFormatContext *ctx;
+    URLContext *h;
    AVStream *st;

    /* remove connection from list */
@@ -827,7 +832,9 @@ static void close_connection(HTTPContext *c)
            av_freep(&ctx->streams[0]);
            av_freep(&ctx);
        }
-        ffurl_close(c->rtp_handles[i]);
+        h = c->rtp_handles[i];
+        if (h)
+            ffurl_close(h);
    }

    ctx = &c->fmt_ctx;
@@ -896,11 +903,11 @@ static int handle_connection(HTTPContext *c)
        if ((ptr >= c->buffer + 2 && !memcmp(ptr-2, "\n\n", 2)) ||
            (ptr >= c->buffer + 4 && !memcmp(ptr-4, "\r\n\r\n", 4))) {
            /* request found : parse it and reply */
-            if (c->state == HTTPSTATE_WAIT_REQUEST)
+            if (c->state == HTTPSTATE_WAIT_REQUEST) {
                ret = http_parse_request(c);
-            else
+            } else {
                ret = rtsp_parse_request(c);
-
+            }
            if (ret < 0)
                return -1;
        } else if (ptr >= c->buffer_end) {
@@ -945,8 +952,8 @@ static int handle_connection(HTTPContext *c)
    case HTTPSTATE_SEND_DATA_HEADER:
    case HTTPSTATE_SEND_DATA_TRAILER:
        /* for packetized output, we consider we can always write (the
-         * input streams set the speed). It may be better to verify
-         * that we do not rely too much on the kernel queues */
+           input streams set the speed). It may be better to verify
+           that we do not rely too much on the kernel queues */
        if (!c->is_packetized) {
            if (c->poll_entry->revents & (POLLERR | POLLHUP))
                return -1;
@@ -1159,10 +1166,8 @@ static int modify_current_stream(HTTPContext *c, char *rates)
                break;
        }

-        if (c->switch_feed_streams[i] >= 0 &&
-            c->switch_feed_streams[i] != c->feed_streams[i]) {
+        if (c->switch_feed_streams[i] >= 0 && c->switch_feed_streams[i] != c->feed_streams[i])
            action_required = 1;
-        }
    }

    return action_required;
@@ -1266,17 +1271,17 @@ static int validate_acl(FFServerStream *stream, HTTPContext *c)

    if (stream->dynamic_acl[0]) {
        acl = parse_dynamic_acl(stream, c);
+
        ret = validate_acl_list(acl, c);
+
        free_acl_list(acl);
    }

    return ret;
 }

-/**
- * compute the real filename of a file by matching it without its
- * extensions to all the stream's filenames
- */
+/* compute the real filename of a file by matching it without its
+   extensions to all the stream's filenames */
 static void compute_real_filename(char *filename, int max_size)
 {
    char file1[1024];
@@ -1394,7 +1399,7 @@ static int http_parse_request(HTTPContext *c)
        compute_real_filename(filename, sizeof(filename) - 1);
    }

-    /* "redirect" request to index.html */
+    // "redirect" / request to index.html
    if (!strlen(filename))
        av_strlcpy(filename, "index.html", sizeof(filename) - 1);

@@ -1733,9 +1738,8 @@ static int http_parse_request(HTTPContext *c)
    return 0;
 send_status:
    compute_status(c);
-    /* horrible: we use this value to avoid
-     * going to the send data state */
-    c->http_error = 200;
+    c->http_error = 200; /* horrible : we use this value to avoid
+                            going to the send data state */
    c->state = HTTPSTATE_SEND_HEADER;
    return 0;
 }
@@ -1750,52 +1754,6 @@ static void fmt_bytecount(AVIOContext *pb, int64_t count)
    avio_printf(pb, "%"PRId64"%c", count, *s);
 }

-static inline void print_stream_params(AVIOContext *pb, FFServerStream *stream)
-{
-    int i, stream_no;
-    const char *type = "unknown";
-    char parameters[64];
-    AVStream *st;
-    AVCodec *codec;
-
-    stream_no = stream->nb_streams;
-
-    avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
-                    "type<th>kbits/s<th align=left>codec<th align=left>"
-                    "Parameters\n");
-
-    for (i = 0; i < stream_no; i++) {
-        st = stream->streams[i];
-        codec = avcodec_find_encoder(st->codec->codec_id);
-
-        parameters[0] = 0;
-
-        switch(st->codec->codec_type) {
-        case AVMEDIA_TYPE_AUDIO:
-            type = "audio";
-            snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
-                     st->codec->channels, st->codec->sample_rate);
-            break;
-        case AVMEDIA_TYPE_VIDEO:
-            type = "video";
-            snprintf(parameters, sizeof(parameters),
-                     "%dx%d, q=%d-%d, fps=%d", st->codec->width,
-                     st->codec->height, st->codec->qmin, st->codec->qmax,
-                     st->codec->time_base.den / st->codec->time_base.num);
-            break;
-        default:
-            abort();
-        }
-
-        avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d"
-                        "<td>%s<td>%s\n",
-                    i, type, st->codec->bit_rate/1000,
-                    codec ? codec->name : "", parameters);
-     }
-
-     avio_printf(pb, "</table>\n");
-}
-
 static void compute_status(HTTPContext *c)
 {
    HTTPContext *c1;
@@ -1846,8 +1804,8 @@ static void compute_status(HTTPContext *c)
                strcpy(eosf - 3, ".ram");
            else if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
                /* generate a sample RTSP director if
-                 * unicast. Generate an SDP redirector if
-                 * multicast */
+                   unicast. Generate an SDP redirector if
+                   multicast */
                eosf = strrchr(sfilename, '.');
                if (!eosf)
                    eosf = sfilename + strlen(sfilename);
@@ -1936,7 +1894,7 @@ static void compute_status(HTTPContext *c)

        avio_printf(pb, "<h2>Feed %s</h2>", stream->filename);
        if (stream->pid) {
-            avio_printf(pb, "Running as pid %"PRId64".\n", (int64_t) stream->pid);
+            avio_printf(pb, "Running as pid %d.\n", stream->pid);

 #if defined(linux)
            {
@@ -1945,8 +1903,8 @@ static void compute_status(HTTPContext *c)

                /* This is somewhat linux specific I guess */
                snprintf(ps_cmd, sizeof(ps_cmd),
-                         "ps -o \"%%cpu,cputime\" --no-headers %"PRId64"",
-                         (int64_t) stream->pid);
+                         "ps -o \"%%cpu,cputime\" --no-headers %d",
+                         stream->pid);

                 pid_stat = popen(ps_cmd, "r");
                 if (pid_stat) {
@@ -1966,7 +1924,42 @@ static void compute_status(HTTPContext *c)
            avio_printf(pb, "<p>");
        }

-        print_stream_params(pb, stream);
+        avio_printf(pb, "<table cellspacing=0 cellpadding=4><tr><th>Stream<th>"
+                        "type<th>kbits/s<th align=left>codec<th align=left>"
+                        "Parameters\n");
+
+        for (i = 0; i < stream->nb_streams; i++) {
+            AVStream *st = stream->streams[i];
+            AVCodec *codec = avcodec_find_encoder(st->codec->codec_id);
+            const char *type = "unknown";
+            char parameters[64];
+
+            parameters[0] = 0;
+
+            switch(st->codec->codec_type) {
+            case AVMEDIA_TYPE_AUDIO:
+                type = "audio";
+                snprintf(parameters, sizeof(parameters), "%d channel(s), %d Hz",
+                         st->codec->channels, st->codec->sample_rate);
+                break;
+            case AVMEDIA_TYPE_VIDEO:
+                type = "video";
+                snprintf(parameters, sizeof(parameters),
+                         "%dx%d, q=%d-%d, fps=%d", st->codec->width,
+                         st->codec->height, st->codec->qmin, st->codec->qmax,
+                         st->codec->time_base.den / st->codec->time_base.num);
+                break;
+            default:
+                abort();
+            }
+
+            avio_printf(pb, "<tr><td align=right>%d<td>%s<td align=right>%d"
+                            "<td>%s<td>%s\n",
+                        i, type, st->codec->bit_rate/1000,
+                        codec ? codec->name : "", parameters);
+        }
+
+        avio_printf(pb, "</table>\n");
        stream = stream->next;
    }

@@ -2118,7 +2111,8 @@ static int64_t get_server_clock(HTTPContext *c)
    return (cur_time - c->start_time) * 1000;
 }

-/* return the estimated time (in us) at which the current packet must be sent */
+/* return the estimated time at which the current packet must be sent
+   (in us) */
 static int64_t get_packet_send_clock(HTTPContext *c)
 {
    int bytes_left, bytes_sent, frame_bytes;
@@ -2126,10 +2120,11 @@ static int64_t get_packet_send_clock(HTTPContext *c)
    frame_bytes = c->cur_frame_bytes;
    if (frame_bytes <= 0)
        return c->cur_pts;
-
-    bytes_left = c->buffer_end - c->buffer_ptr;
-    bytes_sent = frame_bytes - bytes_left;
-    return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
+    else {
+        bytes_left = c->buffer_end - c->buffer_ptr;
+        bytes_sent = frame_bytes - bytes_left;
+        return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
+    }
 }


@@ -2156,8 +2151,7 @@ static int http_prepare_data(HTTPContext *c)
            AVStream *src;
            c->fmt_ctx.streams[i] = av_mallocz(sizeof(AVStream));

-            /* if file or feed, then just take streams from FFServerStream
-             * struct */
+            /* if file or feed, then just take streams from FFServerStream struct */
            if (!c->stream->feed ||
                c->stream->feed == c->stream)
                src = c->stream->streams[i];
@@ -2222,23 +2216,23 @@ static int http_prepare_data(HTTPContext *c)
            if (ret < 0) {
                if (c->stream->feed) {
                    /* if coming from feed, it means we reached the end of the
-                     * ffm file, so must wait for more data */
+                       ffm file, so must wait for more data */
                    c->state = HTTPSTATE_WAIT_FEED;
                    return 1; /* state changed */
-                }
-                if (ret == AVERROR(EAGAIN)) {
+                } else if (ret == AVERROR(EAGAIN)) {
                    /* input not ready, come back later */
                    return 0;
-                }
-                if (c->stream->loop) {
-                    avformat_close_input(&c->fmt_in);
-                    if (open_input_stream(c, "") < 0)
-                        goto no_loop;
-                    goto redo;
                } else {
+                    if (c->stream->loop) {
+                        avformat_close_input(&c->fmt_in);
+                        if (open_input_stream(c, "") < 0)
+                            goto no_loop;
+                        goto redo;
+                    } else {
                    no_loop:
                        /* must send trailer now because EOF or error */
                        c->state = HTTPSTATE_SEND_DATA_TRAILER;
+                    }
                }
            } else {
                int source_index = pkt.stream_index;
@@ -2309,9 +2303,9 @@ static int http_prepare_data(HTTPContext *c)
                            max_packet_size = c->rtp_handles[c->packet_stream_index]->max_packet_size;
                        ret = ffio_open_dyn_packet_buf(&ctx->pb,
                                                       max_packet_size);
-                    } else
+                    } else {
                        ret = avio_open_dyn_buf(&ctx->pb);
-
+                    }
                    if (ret < 0) {
                        /* XXX: potential leak */
                        return -1;
@@ -2374,8 +2368,7 @@ static int http_prepare_data(HTTPContext *c)

 /* should convert the format at the same time */
 /* send data starting at c->buffer_ptr to the output connection
- * (either UDP or TCP)
- */
+ * (either UDP or TCP) */
 static int http_send_data(HTTPContext *c)
 {
    int len, ret;
@@ -2456,8 +2449,8 @@ static int http_send_data(HTTPContext *c)
                        rtsp_c->packet_buffer_ptr += len;
                    if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
                        /* if we could not send all the data, we will
-                         * send it later, so a new state is needed to
-                         * "lock" the RTSP TCP connection */
+                           send it later, so a new state is needed to
+                           "lock" the RTSP TCP connection */
                        rtsp_c->state = RTSPSTATE_SEND_PACKET;
                        break;
                    } else
@@ -2541,8 +2534,9 @@ static int http_start_receive_data(HTTPContext *c)
            http_log("Error reading write index from feed file '%s': %s\n",
                     c->stream->feed_filename, strerror(errno));
            return ret;
+        } else {
+            c->stream->feed_write_index = ret;
        }
-        c->stream->feed_write_index = ret;
    }

    c->stream->feed_write_index = FFMAX(ffm_read_write_index(fd),
@@ -2584,11 +2578,12 @@ static int http_receive_data(HTTPContext *c)
                goto fail;
            c->buffer_ptr = c->buffer;
            break;
-        } else if (++loop_run > 10)
+        } else if (++loop_run > 10) {
            /* no chunk header, abort */
            goto fail;
-        else
+        } else {
            c->buffer_ptr++;
+        }
    }

    if (c->buffer_end > c->buffer_ptr) {
@@ -2621,7 +2616,7 @@ static int http_receive_data(HTTPContext *c)
    if (c->buffer_ptr >= c->buffer_end) {
        FFServerStream *feed = c->stream;
        /* a packet has been received : write it in the store, except
-         * if header */
+           if header */
        if (c->data_count > FFM_PACKET_SIZE) {
            /* XXX: use llseek or url_seek
             * XXX: Should probably fail? */
@@ -2827,10 +2822,10 @@ static int rtsp_parse_request(HTTPContext *c)
 the_end:
    len = avio_close_dyn_buf(c->pb, &c->pb_buffer);
    c->pb = NULL; /* safety */
-    if (len < 0)
+    if (len < 0) {
        /* XXX: cannot do more */
        return -1;
-
+    }
    c->buffer_ptr = c->pb_buffer;
    c->buffer_end = c->pb_buffer + len;
    c->state = RTSPSTATE_SEND_REPLY;
@@ -2849,9 +2844,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
    *pbuffer = NULL;

    avc =  avformat_alloc_context();
-    if (!avc || !rtp_format)
+    if (!avc || !rtp_format) {
        return -1;
-
+    }
    avc->oformat = rtp_format;
    av_dict_set(&avc->metadata, "title",
                entry ? entry->value : "No Title", 0);
@@ -2860,8 +2855,9 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,
        snprintf(avc->filename, 1024, "rtp://%s:%d?multicast=1?ttl=%d",
                 inet_ntoa(stream->multicast_ip),
                 stream->multicast_port, stream->multicast_ttl);
-    } else
+    } else {
        snprintf(avc->filename, 1024, "rtp://0.0.0.0");
+    }

    avc->streams = av_malloc_array(avc->nb_streams, sizeof(*avc->streams));
    if (!avc->streams)
@@ -2891,7 +2887,7 @@ static int prepare_sdp_description(FFServerStream *stream, uint8_t **pbuffer,

 static void rtsp_cmd_options(HTTPContext *c, const char *url)
 {
-    /* rtsp_reply_header(c, RTSP_STATUS_OK); */
+//    rtsp_reply_header(c, RTSP_STATUS_OK);
    avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
    avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
    avio_printf(c->pb, "Public: %s\r\n",
@@ -3058,7 +3054,7 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
    }

    /* test if stream is OK (test needed because several SETUP needs
-     * to be done for a given file) */
+       to be done for a given file) */
    if (rtp_c->stream != stream) {
        rtsp_reply_error(c, RTSP_STATUS_SERVICE);
        return;
@@ -3119,10 +3115,8 @@ static void rtsp_cmd_setup(HTTPContext *c, const char *url,
 }


-/**
- * find an RTP connection by using the session ID. Check consistency
- * with filename
- */
+/* find an RTP connection by using the session ID. Check consistency
+   with filename */
 static HTTPContext *find_rtp_session_with_url(const char *url,
                                              const char *session_id)
 {
@@ -3145,10 +3139,10 @@ static HTTPContext *find_rtp_session_with_url(const char *url,
    for(s=0; s<rtp_c->stream->nb_streams; ++s) {
      snprintf(buf, sizeof(buf), "%s/streamid=%d",
        rtp_c->stream->filename, s);
-      if(!strncmp(path, buf, sizeof(buf)))
-        /* XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE
-         * if nb_streams>1? */
+      if(!strncmp(path, buf, sizeof(buf))) {
+    // XXX: Should we reply with RTSP_STATUS_ONLY_AGGREGATE if nb_streams>1?
        return rtp_c;
+      }
    }
    len = strlen(path);
    if (len > 0 && path[len - 1] == '/' &&
@@ -3226,7 +3220,7 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
    const char *proto_str;

    /* XXX: should output a warning page when coming
-     * close to the connection limit */
+       close to the connection limit */
    if (nb_connections >= config.nb_max_connections)
        goto fail;

@@ -3281,11 +3275,9 @@ static HTTPContext *rtp_new_connection(struct sockaddr_in *from_addr,
    return NULL;
 }

-/**
- * add a new RTP stream in an RTP connection (used in RTSP SETUP
- * command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
- * used.
- */
+/* add a new RTP stream in an RTP connection (used in RTSP SETUP
+   command). If RTP/TCP protocol is used, TCP connection 'rtsp_c' is
+   used. */
 static int rtp_new_av_stream(HTTPContext *c,
                             int stream_index, struct sockaddr_in *dest_addr,
                             HTTPContext *rtsp_c)
@@ -3363,10 +3355,10 @@ static int rtp_new_av_stream(HTTPContext *c,

    /* normally, no packets should be output here, but the packet size may
     * be checked */
-    if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0)
+    if (ffio_open_dyn_packet_buf(&ctx->pb, max_packet_size) < 0) {
        /* XXX: close stream */
        goto fail;
-
+    }
    if (avformat_write_header(ctx, NULL) < 0) {
    fail:
        if (h)
@@ -3403,12 +3395,12 @@ static AVStream *add_av_stream1(FFServerStream *stream,
            return NULL;
        }
        avcodec_copy_context(fst->codec, codec);
-    } else
+    } else {
        /* live streams must use the actual feed's codec since it may be
         * updated later to carry extradata needed by them.
         */
        fst->codec = codec;
-
+    }
    fst->priv_data = av_mallocz(sizeof(FeedData));
    fst->index = stream->nb_streams;
    avpriv_set_pts_info(fst, 33, 1, 90000);
@@ -3510,7 +3502,7 @@ static void extract_mpeg4_header(AVFormatContext *infile)
                if (p[0] == 0x00 && p[1] == 0x00 &&
                    p[2] == 0x01 && p[3] == 0xb6) {
                    size = p - pkt.data;
-                    st->codec->extradata = av_mallocz(size + AV_INPUT_BUFFER_PADDING_SIZE);
+                    st->codec->extradata = av_mallocz(size + FF_INPUT_BUFFER_PADDING_SIZE);
                    st->codec->extradata_size = size;
                    memcpy(st->codec->extradata, pkt.data, size);
                    break;
@@ -3540,7 +3532,7 @@ static void build_file_streams(void)
            /* open stream */
            if (stream->fmt && !strcmp(stream->fmt->name, "rtp")) {
                /* specific case : if transport stream output to RTP,
-                 * we use a raw transport stream reader */
+                   we use a raw transport stream reader */
                av_dict_set(&stream->in_opts, "mpeg2ts_compute_pcr", "1", 0);
            }

@@ -3562,7 +3554,7 @@ static void build_file_streams(void)
                remove_stream(stream);
            } else {
                /* find all the AVStreams inside and reference them in
-                 * 'stream' */
+                   'stream' */
                if (avformat_find_stream_info(infile, NULL) < 0) {
                    http_log("Could not find codec parameters from '%s'\n",
                             stream->feed_filename);
@@ -3589,17 +3581,16 @@ static void build_feed_streams(void)
    /* gather all streams */
    for(stream = config.first_stream; stream; stream = stream->next) {
        feed = stream->feed;
-        if (!feed)
-            continue;
-
-        if (stream->is_feed) {
-            for(i=0;i<stream->nb_streams;i++)
-                stream->feed_streams[i] = i;
-        } else {
-            /* we handle a stream coming from a feed */
-            for(i=0;i<stream->nb_streams;i++)
-                stream->feed_streams[i] = add_av_stream(feed,
-                                                        stream->streams[i]);
+        if (feed) {
+            if (stream->is_feed) {
+                for(i=0;i<stream->nb_streams;i++)
+                    stream->feed_streams[i] = i;
+            } else {
+                /* we handle a stream coming from a feed */
+                for(i=0;i<stream->nb_streams;i++)
+                    stream->feed_streams[i] = add_av_stream(feed,
+                                                            stream->streams[i]);
+            }
        }
    }

@@ -3766,24 +3757,24 @@ static void compute_bandwidth(void)
 static void handle_child_exit(int sig)
 {
    pid_t pid;
-    int status, uptime;
+    int status;

    while ((pid = waitpid(-1, &status, WNOHANG)) > 0) {
        FFServerStream *feed;

        for (feed = config.first_feed; feed; feed = feed->next) {
-            if (feed->pid != pid)
-                continue;
+            if (feed->pid == pid) {
+                int uptime = time(0) - feed->pid_start;

-            uptime = time(0) - feed->pid_start;
-            feed->pid = 0;
-            fprintf(stderr,
-                    "%s: Pid %"PRId64" exited with status %d after %d seconds\n",
-                    feed->filename, (int64_t) pid, status, uptime);
+                feed->pid = 0;
+                fprintf(stderr,
+                        "%s: Pid %d exited with status %d after %d seconds\n",
+                        feed->filename, pid, status, uptime);

-            if (uptime < 30)
-                /* Turn off any more restarts */
-                ffserver_free_child_args(&feed->child_argv);
+                if (uptime < 30)
+                    /* Turn off any more restarts */
+                    ffserver_free_child_args(&feed->child_argv);
+            }
        }
    }

--- a/ffserver_config.c
+++ b/ffserver_config.c
@@ -230,9 +230,9 @@ static void add_codec(FFServerStream *stream, AVCodecContext *av,
    /* compute default parameters */
    switch(av->codec_type) {
    case AVMEDIA_TYPE_AUDIO:
-        if (!av_dict_get(recommended, "b", NULL, 0)) {
+        if (!av_dict_get(recommended, "ab", NULL, 0)) {
            av->bit_rate = 64000;
-            av_dict_set_int(&recommended, "b", av->bit_rate, 0);
+            av_dict_set_int(&recommended, "ab", av->bit_rate, 0);
            WARNING("Setting default value for audio bit rate = %d. "
                    "Use NoDefaults to disable it.\n",
                    av->bit_rate);
@@ -923,7 +923,7 @@ static int ffserver_parse_config_stream(FFServerConfig *config, const char *cmd,
        ffserver_get_arg(arg, sizeof(arg), p);
        ffserver_set_float_param(&f, arg, 1000, -FLT_MAX, FLT_MAX, config,
                "Invalid %s: '%s'\n", cmd, arg);
-        if (ffserver_save_avoption_int("b", (int64_t)lrintf(f),
+        if (ffserver_save_avoption_int("ab", (int64_t)lrintf(f),
                                       AV_OPT_FLAG_AUDIO_PARAM, config) < 0)
            goto nomem;
    } else if (!av_strcasecmp(cmd, "AudioChannels")) {
--- a/libavcodec/012v.c
+++ b/libavcodec/012v.c
@@ -151,5 +151,5 @@ AVCodec ff_zero12v_decoder = {
    .id             = AV_CODEC_ID_012V,
    .init           = zero12v_decode_init,
    .decode         = zero12v_decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = CODEC_CAP_DR1,
 };
--- a/libavcodec/4xm.c
+++ b/libavcodec/4xm.c
@@ -559,7 +559,7 @@ static inline void idct_put(FourXContext *f, int x, int y)
        idct(block[i]);
    }

-    if (!(f->avctx->flags & AV_CODEC_FLAG_GRAY)) {
+    if (!(f->avctx->flags & CODEC_FLAG_GRAY)) {
        for (i = 4; i < 6; i++)
            idct(block[i]);
    }
@@ -883,11 +883,11 @@ static int decode_frame(AVCodecContext *avctx, void *data,
        }
        cfrm = &f->cfrm[i];

-        if (data_size > UINT_MAX -  cfrm->size - AV_INPUT_BUFFER_PADDING_SIZE)
+        if (data_size > UINT_MAX -  cfrm->size - FF_INPUT_BUFFER_PADDING_SIZE)
            return AVERROR_INVALIDDATA;

        cfrm->data = av_fast_realloc(cfrm->data, &cfrm->allocated_size,
-                                     cfrm->size + data_size + AV_INPUT_BUFFER_PADDING_SIZE);
+                                     cfrm->size + data_size + FF_INPUT_BUFFER_PADDING_SIZE);
        // explicit check needed as memcpy below might not catch a NULL
        if (!cfrm->data) {
            av_log(f->avctx, AV_LOG_ERROR, "realloc failure\n");
@@ -1026,5 +1026,5 @@ AVCodec ff_fourxm_decoder = {
    .init           = decode_init,
    .close          = decode_end,
    .decode         = decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = CODEC_CAP_DR1,
 };
--- a/libavcodec/8bps.c
+++ b/libavcodec/8bps.c
@@ -184,5 +184,5 @@ AVCodec ff_eightbps_decoder = {
    .priv_data_size = sizeof(EightBpsContext),
    .init           = decode_init,
    .decode         = decode_frame,
-    .capabilities   = AV_CODEC_CAP_DR1,
+    .capabilities   = CODEC_CAP_DR1,
 };
--- a/libavcodec/8svx.c
+++ b/libavcodec/8svx.c
@@ -194,7 +194,7 @@ AVCodec ff_eightsvx_fib_decoder = {
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
-  .capabilities   = AV_CODEC_CAP_DR1,
+  .capabilities   = CODEC_CAP_DR1,
  .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                    AV_SAMPLE_FMT_NONE },
 };
@@ -209,7 +209,7 @@ AVCodec ff_eightsvx_exp_decoder = {
  .init           = eightsvx_decode_init,
  .decode         = eightsvx_decode_frame,
  .close          = eightsvx_decode_close,
-  .capabilities   = AV_CODEC_CAP_DR1,
+  .capabilities   = CODEC_CAP_DR1,
  .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
                                                    AV_SAMPLE_FMT_NONE },
 };
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -13,7 +13,6 @@ HEADERS = avcodec.h                                                     \
          vda.h                                                         \
          vdpau.h                                                       \
          version.h                                                     \
-          videotoolbox.h                                                \
          vorbis_parser.h                                               \
          xvmc.h                                                        \

@@ -57,7 +56,6 @@ FFT-OBJS-$(CONFIG_HARDCODED_TABLES)    += cos_tables.o cos_fixed_tables.o
 OBJS-$(CONFIG_FFT)                     += avfft.o fft_fixed.o fft_float.o \
                                          fft_fixed_32.o fft_init_table.o \
                                          $(FFT-OBJS-yes)
-OBJS-$(CONFIG_FLACDSP)                 += flacdsp.o
 OBJS-$(CONFIG_FMTCONVERT)              += fmtconvert.o
 OBJS-$(CONFIG_GOLOMB)                  += golomb.o
 OBJS-$(CONFIG_H263DSP)                 += h263dsp.o
@@ -73,7 +71,6 @@ OBJS-$(CONFIG_IDCTDSP)                 += idctdsp.o simple_idct.o jrevdct.o
 OBJS-$(CONFIG_IIRFILTER)               += iirfilter.o
 OBJS-$(CONFIG_IMDCT15)                 += imdct15.o
 OBJS-$(CONFIG_INTRAX8)                 += intrax8.o intrax8dsp.o
-OBJS-$(CONFIG_IVIDSP)                  += ivi_dsp.o
 OBJS-$(CONFIG_JPEGTABLES)              += jpegtables.o
 OBJS-$(CONFIG_LIBXVID)                 += libxvid_rc.o
 OBJS-$(CONFIG_LLAUDDSP)                += lossless_audiodsp.o
@@ -91,11 +88,10 @@ OBJS-$(CONFIG_MPEGAUDIODSP)            += mpegaudiodsp.o                \
                                          mpegaudiodsp_float.o
 OBJS-$(CONFIG_MPEGVIDEO)               += mpegvideo.o mpegvideodsp.o rl.o \
                                          mpegvideo_motion.o mpegutils.o \
-                                          mpegvideodata.o mpegpicture.o
+                                          mpegvideodata.o
 OBJS-$(CONFIG_MPEGVIDEOENC)            += mpegvideo_enc.o mpeg12data.o  \
                                          motion_est.o ratecontrol.o    \
                                          mpegvideoencdsp.o
-OBJS-$(CONFIG_MSS34DSP)                += mss34dsp.o
 OBJS-$(CONFIG_NVENC)                   += nvenc.o
 OBJS-$(CONFIG_PIXBLOCKDSP)             += pixblockdsp.o
 OBJS-$(CONFIG_QPELDSP)                 += qpeldsp.o
@@ -105,36 +101,23 @@ OBJS-$(CONFIG_QSVENC)                  += qsvenc.o
 OBJS-$(CONFIG_RANGECODER)              += rangecoder.o
 RDFT-OBJS-$(CONFIG_HARDCODED_TABLES)   += sin_tables.o
 OBJS-$(CONFIG_RDFT)                    += rdft.o $(RDFT-OBJS-yes)
-OBJS-$(CONFIG_RV34DSP)                 += rv34dsp.o
-OBJS-$(CONFIG_SHARED)                  += log2_tab.o reverse.o
-OBJS-$(CONFIG_SINEWIN)                 += sinewin.o sinewin_fixed.o
-OBJS-$(CONFIG_SNAPPY)                  += snappy.o
+OBJS-$(CONFIG_SHARED)                  += log2_tab.o
+OBJS-$(CONFIG_SINEWIN)                 += sinewin.o
 OBJS-$(CONFIG_STARTCODE)               += startcode.o
-OBJS-$(CONFIG_TEXTUREDSP)              += texturedsp.o
-OBJS-$(CONFIG_TEXTUREDSPENC)           += texturedspenc.o
 OBJS-$(CONFIG_TPELDSP)                 += tpeldsp.o
 OBJS-$(CONFIG_VIDEODSP)                += videodsp.o
 OBJS-$(CONFIG_VP3DSP)                  += vp3dsp.o
-OBJS-$(CONFIG_VP56DSP)                 += vp56dsp.o
-OBJS-$(CONFIG_VP8DSP)                  += vp8dsp.o
 OBJS-$(CONFIG_WMA_FREQS)               += wma_freqs.o
-OBJS-$(CONFIG_WMV2DSP)                 += wmv2dsp.o

 # decoders/encoders
 OBJS-$(CONFIG_ZERO12V_DECODER)         += 012v.o
 OBJS-$(CONFIG_A64MULTI_ENCODER)        += a64multienc.o elbg.o
 OBJS-$(CONFIG_A64MULTI5_ENCODER)       += a64multienc.o elbg.o
-OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps_float.o \
+OBJS-$(CONFIG_AAC_DECODER)             += aacdec.o aactab.o aacsbr.o aacps.o \
                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp.o aacpsdsp_float.o
-OBJS-$(CONFIG_AAC_FIXED_DECODER)       += aacdec_fixed.o aactab.o aacsbr_fixed.o aacps_fixed.o \
-                                          aacadtsdec.o mpeg4audio.o kbdwin.o \
-                                          sbrdsp_fixed.o aacpsdsp_fixed.o
-OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o aacenctab.o    \
+                                          sbrdsp.o aacpsdsp.o
+OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o    \
                                          aacpsy.o aactab.o      \
-                                          aacenc_is.o \
-                                          aacenc_tns.o \
-                                          aacenc_pred.o \
                                          psymodel.o mpeg4audio.o kbdwin.o
 OBJS-$(CONFIG_AASC_DECODER)            += aasc.o msrledec.o
 OBJS-$(CONFIG_AC3_DECODER)             += ac3dec_float.o ac3dec_data.o ac3.o kbdwin.o
@@ -219,7 +202,6 @@ OBJS-$(CONFIG_DCA_DECODER)             += dcadec.o dca.o dcadsp.o      \
                                          dcadata.o dca_exss.o         \
                                          dca_xll.o synth_filter.o
 OBJS-$(CONFIG_DCA_ENCODER)             += dcaenc.o dca.o dcadata.o
-OBJS-$(CONFIG_DDS_DECODER)             += dds.o
 OBJS-$(CONFIG_DIRAC_DECODER)           += diracdec.o dirac.o diracdsp.o \
                                          dirac_arith.o mpeg12data.o dirac_dwt.o
 OBJS-$(CONFIG_DFA_DECODER)             += dfa.o
@@ -261,8 +243,8 @@ OBJS-$(CONFIG_FFV1_DECODER)            += ffv1dec.o ffv1.o
 OBJS-$(CONFIG_FFV1_ENCODER)            += ffv1enc.o ffv1.o
 OBJS-$(CONFIG_FFWAVESYNTH_DECODER)     += ffwavesynth.o
 OBJS-$(CONFIG_FIC_DECODER)             += fic.o
-OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o
-OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o vorbis_data.o
+OBJS-$(CONFIG_FLAC_DECODER)            += flacdec.o flacdata.o flac.o flacdsp.o
+OBJS-$(CONFIG_FLAC_ENCODER)            += flacenc.o flacdata.o flac.o flacdsp.o vorbis_data.o
 OBJS-$(CONFIG_FLASHSV_DECODER)         += flashsv.o
 OBJS-$(CONFIG_FLASHSV_ENCODER)         += flashsvenc.o
 OBJS-$(CONFIG_FLASHSV2_ENCODER)        += flashsv2enc.o
@@ -271,7 +253,7 @@ OBJS-$(CONFIG_FLIC_DECODER)            += flicvideo.o
 OBJS-$(CONFIG_FOURXM_DECODER)          += 4xm.o
 OBJS-$(CONFIG_FRAPS_DECODER)           += fraps.o
 OBJS-$(CONFIG_FRWU_DECODER)            += frwu.o
-OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o elsdec.o
+OBJS-$(CONFIG_G2M_DECODER)             += g2meet.o
 OBJS-$(CONFIG_G723_1_DECODER)          += g723_1.o acelp_vectors.o \
                                          celp_filters.o celp_math.o
 OBJS-$(CONFIG_G723_1_ENCODER)          += g723_1.o acelp_vectors.o celp_math.o
@@ -284,24 +266,20 @@ OBJS-$(CONFIG_H261_DECODER)            += h261dec.o h261data.o h261.o
 OBJS-$(CONFIG_H261_ENCODER)            += h261enc.o h261data.o h261.o
 OBJS-$(CONFIG_H263_DECODER)            += h263dec.o h263.o ituh263dec.o        \
                                          mpeg4video.o mpeg4videodec.o flvdec.o\
-                                          intelh263dec.o h263data.o
+                                          intelh263dec.o
 OBJS-$(CONFIG_H263_ENCODER)            += mpeg4videoenc.o mpeg4video.o  \
-                                          h263.o h263data.o ituh263enc.o flvenc.o
+                                          h263.o ituh263enc.o flvenc.o
 OBJS-$(CONFIG_H264_DECODER)            += h264.o h264_cabac.o h264_cavlc.o \
                                          h264_direct.o h264_loopfilter.o  \
                                          h264_mb.o h264_picture.o h264_ps.o \
                                          h264_refs.o h264_sei.o h264_slice.o
 OBJS-$(CONFIG_H264_MMAL_DECODER)       += mmaldec.o
 OBJS-$(CONFIG_H264_VDA_DECODER)        += vda_h264_dec.o
-OBJS-$(CONFIG_H264_QSV_DECODER)        += qsvdec_h2645.o
+OBJS-$(CONFIG_H264_QSV_DECODER)        += qsvdec_h264.o
 OBJS-$(CONFIG_H264_QSV_ENCODER)        += qsvenc_h264.o
-OBJS-$(CONFIG_HAP_DECODER)             += hapdec.o hap.o
-OBJS-$(CONFIG_HAP_ENCODER)             += hapenc.o hap.o
 OBJS-$(CONFIG_HEVC_DECODER)            += hevc.o hevc_mvs.o hevc_ps.o hevc_sei.o \
                                          hevc_cabac.o hevc_refs.o hevcpred.o    \
-                                          hevcdsp.o hevc_filter.o hevc_parse.o hevc_data.o
-OBJS-$(CONFIG_HEVC_QSV_DECODER)        += qsvdec_h2645.o
-OBJS-$(CONFIG_HEVC_QSV_ENCODER)        += qsvenc_hevc.o hevc_ps_enc.o hevc_parse.o
+                                          hevcdsp.o hevc_filter.o
 OBJS-$(CONFIG_HNM4_VIDEO_DECODER)      += hnm4video.o
 OBJS-$(CONFIG_HQ_HQA_DECODER)          += hq_hqa.o hq_hqadata.o hq_hqadsp.o \
                                          canopus.o
@@ -315,8 +293,8 @@ OBJS-$(CONFIG_IFF_ILBM_DECODER)        += iff.o
 OBJS-$(CONFIG_IMC_DECODER)             += imc.o
 OBJS-$(CONFIG_INDEO2_DECODER)          += indeo2.o
 OBJS-$(CONFIG_INDEO3_DECODER)          += indeo3.o
-OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi.o
-OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi.o
+OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi.o ivi_dsp.o
+OBJS-$(CONFIG_INDEO5_DECODER)          += indeo5.o ivi.o ivi_dsp.o
 OBJS-$(CONFIG_INTERPLAY_DPCM_DECODER)  += dpcm.o
 OBJS-$(CONFIG_INTERPLAY_VIDEO_DECODER) += interplayvideo.o
 OBJS-$(CONFIG_JACOSUB_DECODER)         += jacosubdec.o ass.o
@@ -368,8 +346,6 @@ OBJS-$(CONFIG_MPEG1VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG1VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
 OBJS-$(CONFIG_MPEG2VIDEO_DECODER)      += mpeg12dec.o mpeg12.o mpeg12data.o
 OBJS-$(CONFIG_MPEG2VIDEO_ENCODER)      += mpeg12enc.o mpeg12.o
-OBJS-$(CONFIG_MPEG2_QSV_DECODER)       += qsvdec_mpeg2.o
-OBJS-$(CONFIG_MPEG2_QSV_ENCODER)       += qsvenc_mpeg2.o
 OBJS-$(CONFIG_MPEG4_DECODER)           += xvididct.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
 OBJS-$(CONFIG_MSMPEG4V1_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
@@ -378,13 +354,13 @@ OBJS-$(CONFIG_MSMPEG4V2_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V3_DECODER)       += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSMPEG4V3_ENCODER)       += msmpeg4enc.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_MSRLE_DECODER)           += msrle.o msrledec.o
-OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o
+OBJS-$(CONFIG_MSA1_DECODER)            += mss3.o mss34dsp.o
 OBJS-$(CONFIG_MSS1_DECODER)            += mss1.o mss12.o
 OBJS-$(CONFIG_MSS2_DECODER)            += mss2.o mss12.o mss2dsp.o
 OBJS-$(CONFIG_MSVIDEO1_DECODER)        += msvideo1.o
 OBJS-$(CONFIG_MSVIDEO1_ENCODER)        += msvideo1enc.o elbg.o
 OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
-OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o
+OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o mss34dsp.o
 OBJS-$(CONFIG_MVC1_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MVC2_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o
@@ -448,8 +424,8 @@ OBJS-$(CONFIG_RV10_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV10_ENCODER)            += rv10enc.o
 OBJS-$(CONFIG_RV20_DECODER)            += rv10.o
 OBJS-$(CONFIG_RV20_ENCODER)            += rv20enc.o
-OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o
-OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv40dsp.o
+OBJS-$(CONFIG_RV30_DECODER)            += rv30.o rv34.o rv30dsp.o rv34dsp.o
+OBJS-$(CONFIG_RV40_DECODER)            += rv40.o rv34.o rv34dsp.o rv40dsp.o
 OBJS-$(CONFIG_SAMI_DECODER)            += samidec.o ass.o
 OBJS-$(CONFIG_S302M_DECODER)           += s302m.o
 OBJS-$(CONFIG_S302M_ENCODER)           += s302menc.o
@@ -506,7 +482,7 @@ OBJS-$(CONFIG_TSCC2_DECODER)           += tscc2.o
 OBJS-$(CONFIG_TTA_DECODER)             += tta.o ttadata.o ttadsp.o
 OBJS-$(CONFIG_TTA_ENCODER)             += ttaenc.o ttadata.o
 OBJS-$(CONFIG_TWINVQ_DECODER)          += twinvqdec.o twinvq.o
-OBJS-$(CONFIG_TXD_DECODER)             += txd.o
+OBJS-$(CONFIG_TXD_DECODER)             += txd.o s3tc.o
 OBJS-$(CONFIG_ULTI_DECODER)            += ulti.o
 OBJS-$(CONFIG_UTVIDEO_DECODER)         += utvideodec.o utvideo.o
 OBJS-$(CONFIG_UTVIDEO_ENCODER)         += utvideoenc.o utvideo.o
@@ -526,7 +502,6 @@ OBJS-$(CONFIG_VC1_DECODER)             += vc1dec.o vc1_block.o vc1_loopfilter.o
                                          vc1dsp.o \
                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o \
                                          wmv2dsp.o
-OBJS-$(CONFIG_VC1_QSV_DECODER)         += qsvdec_vc1.o
 OBJS-$(CONFIG_VCR1_DECODER)            += vcr1.o
 OBJS-$(CONFIG_VMDAUDIO_DECODER)        += vmdaudio.o
 OBJS-$(CONFIG_VMDVIDEO_DECODER)        += vmdvideo.o
@@ -536,11 +511,12 @@ OBJS-$(CONFIG_VORBIS_DECODER)          += vorbisdec.o vorbisdsp.o vorbis.o \
 OBJS-$(CONFIG_VORBIS_ENCODER)          += vorbisenc.o vorbis.o \
                                          vorbis_data.o
 OBJS-$(CONFIG_VP3_DECODER)             += vp3.o
-OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56rac.o
-OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o \
+OBJS-$(CONFIG_VP5_DECODER)             += vp5.o vp56.o vp56data.o vp56dsp.o \
+                                          vp56rac.o
+OBJS-$(CONFIG_VP6_DECODER)             += vp6.o vp56.o vp56data.o vp56dsp.o \
                                          vp6dsp.o vp56rac.o
-OBJS-$(CONFIG_VP7_DECODER)             += vp8.o vp56rac.o
-OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp56rac.o
+OBJS-$(CONFIG_VP7_DECODER)             += vp8.o vp8dsp.o vp56rac.o
+OBJS-$(CONFIG_VP8_DECODER)             += vp8.o vp8dsp.o vp56rac.o
 OBJS-$(CONFIG_VP9_DECODER)             += vp9.o vp9dsp.o vp56rac.o vp9dsp_8bpp.o \
                                          vp9dsp_10bpp.o vp9dsp_12bpp.o
 OBJS-$(CONFIG_VPLAYER_DECODER)         += textdec.o ass.o
@@ -562,9 +538,9 @@ OBJS-$(CONFIG_WMAVOICE_DECODER)        += wmavoice.o \
                                          acelp_vectors.o acelp_filters.o
 OBJS-$(CONFIG_WMV1_DECODER)            += msmpeg4dec.o msmpeg4.o msmpeg4data.o
 OBJS-$(CONFIG_WMV1_ENCODER)            += msmpeg4enc.o
-OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o \
+OBJS-$(CONFIG_WMV2_DECODER)            += wmv2dec.o wmv2.o wmv2dsp.o \
                                          msmpeg4dec.o msmpeg4.o msmpeg4data.o
-OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o \
+OBJS-$(CONFIG_WMV2_ENCODER)            += wmv2enc.o wmv2.o wmv2dsp.o \
                                          msmpeg4.o msmpeg4enc.o msmpeg4data.o
 OBJS-$(CONFIG_WNV1_DECODER)            += wnv1.o
 OBJS-$(CONFIG_WS_SND1_DECODER)         += ws-snd1.o
@@ -700,35 +676,27 @@ OBJS-$(CONFIG_VIMA_DECODER)               += vima.o adpcm_data.o
 OBJS-$(CONFIG_D3D11VA)                    += dxva2.o
 OBJS-$(CONFIG_DXVA2)                      += dxva2.o
 OBJS-$(CONFIG_VAAPI)                      += vaapi.o
-OBJS-$(CONFIG_VDA)                        += vda.o videotoolbox.o
-OBJS-$(CONFIG_VIDEOTOOLBOX)               += videotoolbox.o
+OBJS-$(CONFIG_VDA)                        += vda.o
 OBJS-$(CONFIG_VDPAU)                      += vdpau.o

 OBJS-$(CONFIG_H263_VAAPI_HWACCEL)         += vaapi_mpeg4.o
 OBJS-$(CONFIG_H263_VDPAU_HWACCEL)         += vdpau_mpeg4.o
-OBJS-$(CONFIG_H263_VIDEOTOOLBOX_HWACCEL)  += videotoolbox.o
 OBJS-$(CONFIG_H264_D3D11VA_HWACCEL)       += dxva2_h264.o
 OBJS-$(CONFIG_H264_DXVA2_HWACCEL)         += dxva2_h264.o
 OBJS-$(CONFIG_H264_VAAPI_HWACCEL)         += vaapi_h264.o
 OBJS-$(CONFIG_H264_VDA_HWACCEL)           += vda_h264.o
 OBJS-$(CONFIG_H264_VDPAU_HWACCEL)         += vdpau_h264.o
-OBJS-$(CONFIG_H264_VIDEOTOOLBOX_HWACCEL)  += videotoolbox.o
 OBJS-$(CONFIG_HEVC_D3D11VA_HWACCEL)       += dxva2_hevc.o
 OBJS-$(CONFIG_HEVC_DXVA2_HWACCEL)         += dxva2_hevc.o
-OBJS-$(CONFIG_HEVC_VAAPI_HWACCEL)         += vaapi_hevc.o
-OBJS-$(CONFIG_HEVC_VDPAU_HWACCEL)         += vdpau_hevc.o
 OBJS-$(CONFIG_MPEG1_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG1_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_MPEG1_XVMC_HWACCEL)         += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG2_D3D11VA_HWACCEL)      += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_DXVA2_HWACCEL)        += dxva2_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VAAPI_HWACCEL)        += vaapi_mpeg2.o
 OBJS-$(CONFIG_MPEG2_VDPAU_HWACCEL)        += vdpau_mpeg12.o
-OBJS-$(CONFIG_MPEG2_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_MPEG2_XVMC_HWACCEL)         += mpegvideo_xvmc.o
 OBJS-$(CONFIG_MPEG4_VAAPI_HWACCEL)        += vaapi_mpeg4.o
 OBJS-$(CONFIG_MPEG4_VDPAU_HWACCEL)        += vdpau_mpeg4.o
-OBJS-$(CONFIG_MPEG4_VIDEOTOOLBOX_HWACCEL) += videotoolbox.o
 OBJS-$(CONFIG_VC1_D3D11VA_HWACCEL)        += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_DXVA2_HWACCEL)          += dxva2_vc1.o
 OBJS-$(CONFIG_VC1_VAAPI_HWACCEL)          += vaapi_vc1.o
@@ -789,7 +757,6 @@ OBJS-$(CONFIG_LIBGSM_MS_DECODER)          += libgsmdec.o
 OBJS-$(CONFIG_LIBGSM_MS_ENCODER)          += libgsmenc.o
 OBJS-$(CONFIG_LIBILBC_DECODER)            += libilbc.o
 OBJS-$(CONFIG_LIBILBC_ENCODER)            += libilbc.o
-OBJS-$(CONFIG_LIBKVAZAAR_ENCODER)         += libkvazaar.o
 OBJS-$(CONFIG_LIBMP3LAME_ENCODER)         += libmp3lame.o mpegaudiodecheader.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER)  += libopencore-amr.o
 OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER)  += libopencore-amr.o
@@ -825,7 +792,6 @@ OBJS-$(CONFIG_LIBVPX_VP9_ENCODER)         += libvpxenc.o libvpx.o
 OBJS-$(CONFIG_LIBWAVPACK_ENCODER)         += libwavpackenc.o
 OBJS-$(CONFIG_LIBWEBP_ENCODER)            += libwebpenc_common.o libwebpenc.o
 OBJS-$(CONFIG_LIBWEBP_ANIM_ENCODER)       += libwebpenc_common.o libwebpenc_animencoder.o
-OBJS-$(CONFIG_LIBX262_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX265_ENCODER)            += libx265.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
@@ -851,17 +817,16 @@ OBJS-$(CONFIG_DVD_NAV_PARSER)          += dvd_nav_parser.o
 OBJS-$(CONFIG_DVDSUB_PARSER)           += dvdsub_parser.o
 OBJS-$(CONFIG_FLAC_PARSER)             += flac_parser.o flacdata.o flac.o \
                                          vorbis_data.o
-OBJS-$(CONFIG_G729_PARSER)             += g729_parser.o
 OBJS-$(CONFIG_GSM_PARSER)              += gsm_parser.o
 OBJS-$(CONFIG_H261_PARSER)             += h261_parser.o
 OBJS-$(CONFIG_H263_PARSER)             += h263_parser.o
 OBJS-$(CONFIG_H264_PARSER)             += h264_parser.o
-OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o hevc_parse.o hevc_ps.o hevc_data.o
+OBJS-$(CONFIG_HEVC_PARSER)             += hevc_parser.o
 OBJS-$(CONFIG_MJPEG_PARSER)            += mjpeg_parser.o
 OBJS-$(CONFIG_MLP_PARSER)              += mlp_parser.o mlp.o
 OBJS-$(CONFIG_MPEG4VIDEO_PARSER)       += mpeg4video_parser.o h263.o \
                                          mpeg4videodec.o mpeg4video.o \
-                                          ituh263dec.o h263dec.o h263data.o
+                                          ituh263dec.o h263dec.o
 OBJS-$(CONFIG_PNG_PARSER)              += png_parser.o
 OBJS-$(CONFIG_MPEGAUDIO_PARSER)        += mpegaudio_parser.o \
                                          mpegaudiodecheader.o mpegaudiodata.o
@@ -884,7 +849,6 @@ OBJS-$(CONFIG_AAC_ADTSTOASC_BSF)          += aac_adtstoasc_bsf.o aacadtsdec.o \
 OBJS-$(CONFIG_CHOMP_BSF)                  += chomp_bsf.o
 OBJS-$(CONFIG_DUMP_EXTRADATA_BSF)         += dump_extradata_bsf.o
 OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF)       += h264_mp4toannexb_bsf.o
-OBJS-$(CONFIG_HEVC_MP4TOANNEXB_BSF)       += hevc_mp4toannexb_bsf.o
 OBJS-$(CONFIG_IMX_DUMP_HEADER_BSF)        += imx_dump_header_bsf.o
 OBJS-$(CONFIG_MJPEG2JPEG_BSF)             += mjpeg2jpeg_bsf.o
 OBJS-$(CONFIG_MJPEGA_DUMP_HEADER_BSF)     += mjpega_dump_header_bsf.o
@@ -909,6 +873,7 @@ SKIPHEADERS                            += %_tablegen.h                  \
                                          %_tables.h                    \
                                          aac_tablegen_decl.h           \
                                          fft-internal.h                \
+                                          libutvideo.h                  \
                                          old_codec_ids.h               \
                                          tableprint.h                  \
                                          tableprint_vlc.h              \
@@ -918,22 +883,21 @@ SKIPHEADERS-$(CONFIG_D3D11VA)          += d3d11va.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_DXVA2)            += dxva2.h dxva2_internal.h
 SKIPHEADERS-$(CONFIG_LIBSCHROEDINGER)  += libschroedinger.h
 SKIPHEADERS-$(CONFIG_LIBUTVIDEO)       += libutvideo.h
-SKIPHEADERS-$(CONFIG_LIBWEBP_ENCODER)  += libwebpenc_common.h
 SKIPHEADERS-$(CONFIG_QSV)              += qsv.h qsv_internal.h
 SKIPHEADERS-$(CONFIG_QSVDEC)           += qsvdec.h
 SKIPHEADERS-$(CONFIG_QSVENC)           += qsvenc.h
 SKIPHEADERS-$(CONFIG_XVMC)             += xvmc.h
 SKIPHEADERS-$(CONFIG_VAAPI)            += vaapi_internal.h
-SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_vt_internal.h
+SKIPHEADERS-$(CONFIG_VDA)              += vda.h vda_internal.h
 SKIPHEADERS-$(CONFIG_VDPAU)            += vdpau.h vdpau_internal.h
-SKIPHEADERS-$(CONFIG_VIDEOTOOLBOX)     += videotoolbox.h vda_vt_internal.h

 TESTPROGS = imgconvert                                                  \
-            jpeg2000dwt                                                 \
            mathops                                                    \
            options                                                     \
            avfft                                                       \

+TESTPROGS += api-flac
+
 TESTPROGS-$(CONFIG_CABAC)                 += cabac
 TESTPROGS-$(CONFIG_FFT)                   += fft fft-fixed fft-fixed32
 TESTPROGS-$(CONFIG_IDCTDSP)               += dct
@@ -949,12 +913,9 @@ TOOLS = fourcc2pixfmt

 HOSTPROGS = aac_tablegen                                                \
            aacps_tablegen                                              \
-            aacps_fixed_tablegen                                        \
            aacsbr_tablegen                                             \
-            aacsbr_fixed_tablegen                                       \
            cabac_tablegen                                              \
            cbrt_tablegen                                               \
-            cbrt_fixed_tablegen                                         \
            cos_tablegen                                                \
            dsd_tablegen                                                \
            dv_tablegen                                                 \
@@ -963,7 +924,6 @@ HOSTPROGS = aac_tablegen                                                \
            pcm_tablegen                                                \
            qdm2_tablegen                                               \
            sinewin_tablegen                                            \
-            sinewin_fixed_tablegen                                      \

 CLEANFILES = *_tables.c *_tables.h *_tablegen$(HOSTEXESUF)

@@ -982,9 +942,8 @@ else
 $(SUBDIR)%_tablegen$(HOSTEXESUF): HOSTCFLAGS += -DCONFIG_SMALL=0
 endif

-GEN_HEADERS = cabac_tables.h cbrt_tables.h cbrt_fixed_tables.h aacps_tables.h aacps_fixed_tables.h aacsbr_tables.h \
-              aacsbr_fixed_tables.h aac_tables.h dsd_tables.h dv_tables.h     \
-              sinewin_tables.h sinewin_fixed_tables.h mpegaudio_tables.h motionpixels_tables.h \
+GEN_HEADERS = cabac_tables.h cbrt_tables.h aacps_tables.h aacsbr_tables.h aac_tables.h dsd_tables.h dv_tables.h     \
+              sinewin_tables.h mpegaudio_tables.h motionpixels_tables.h \
              pcm_tables.h qdm2_tables.h
 GEN_HEADERS := $(addprefix $(SUBDIR), $(GEN_HEADERS))

@@ -993,18 +952,13 @@ $(GEN_HEADERS): $(SUBDIR)%_tables.h: $(SUBDIR)%_tablegen$(HOSTEXESUF)

 ifdef CONFIG_HARDCODED_TABLES
 $(SUBDIR)aacdec.o: $(SUBDIR)cbrt_tables.h
-$(SUBDIR)aacdec_fixed.o: $(SUBDIR)cbrt_fixed_tables.h
-$(SUBDIR)aacps_float.o: $(SUBDIR)aacps_tables.h
-$(SUBDIR)aacps_fixed.o: $(SUBDIR)aacps_fixed_tables.h
+$(SUBDIR)aacps.o: $(SUBDIR)aacps_tables.h
 $(SUBDIR)aacsbr.o: $(SUBDIR)aacsbr_tables.h
-$(SUBDIR)aacsbr_fixed.o: $(SUBDIR)aacsbr_fixed_tables.h
 $(SUBDIR)aactab.o: $(SUBDIR)aac_tables.h
-$(SUBDIR)aactab_fixed.o: $(SUBDIR)aac_fixed_tables.h
 $(SUBDIR)cabac.o: $(SUBDIR)cabac_tables.h
 $(SUBDIR)dsddec.o: $(SUBDIR)dsd_tables.h
 $(SUBDIR)dvenc.o: $(SUBDIR)dv_tables.h
 $(SUBDIR)sinewin.o: $(SUBDIR)sinewin_tables.h
-$(SUBDIR)sinewin_fixed.o: $(SUBDIR)sinewin_fixed_tables.h
 $(SUBDIR)mpegaudiodec_fixed.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)mpegaudiodec_float.o: $(SUBDIR)mpegaudio_tables.h
 $(SUBDIR)motionpixels.o: $(SUBDIR)motionpixels_tables.h
--- a/libavcodec/a64multienc.c
+++ b/libavcodec/a64multienc.c
@@ -66,8 +66,7 @@ static const int mc_colors[5]={0x0,0xb,0xc,0xf,0x1};
 //static const int mc_colors[5]={0x0,0x8,0xa,0xf,0x7};
 //static const int mc_colors[5]={0x0,0x9,0x8,0xa,0x3};

-static void to_meta_with_crop(AVCodecContext *avctx,
-                              const AVFrame *p, int *dest)
+static void to_meta_with_crop(AVCodecContext *avctx, const AVFrame *p, int *dest)
 {
    int blockx, blocky, x, y;
    int luma = 0;
@@ -235,7 +234,7 @@ static av_cold int a64multi_encode_init(AVCodecContext *avctx)
    }

    /* set up extradata */
-    if (!(avctx->extradata = av_mallocz(8 * 4 + AV_INPUT_BUFFER_PADDING_SIZE))) {
+    if (!(avctx->extradata = av_mallocz(8 * 4 + FF_INPUT_BUFFER_PADDING_SIZE))) {
        av_log(avctx, AV_LOG_ERROR, "Failed to allocate memory for extradata.\n");
        return AVERROR(ENOMEM);
    }
@@ -328,7 +327,7 @@ static int a64multi_encode_frame(AVCodecContext *avctx, AVPacket *pkt,
        /* any frames to encode? */
        if (c->mc_lifetime) {
            int alloc_size = charset_size + c->mc_lifetime*(screen_size + colram_size);
-            if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size, 0)) < 0)
+            if ((ret = ff_alloc_packet2(avctx, pkt, alloc_size)) < 0)
                return ret;
            buf = pkt->data;

@@ -406,7 +405,7 @@ AVCodec ff_a64multi_encoder = {
    .encode2        = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
-    .capabilities   = AV_CODEC_CAP_DELAY,
+    .capabilities   = CODEC_CAP_DELAY,
 };
 #endif
 #if CONFIG_A64MULTI5_ENCODER
@@ -420,6 +419,6 @@ AVCodec ff_a64multi5_encoder = {
    .encode2        = a64multi_encode_frame,
    .close          = a64multi_close_encoder,
    .pix_fmts       = (const enum AVPixelFormat[]) {AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE},
-    .capabilities   = AV_CODEC_CAP_DELAY,
+    .capabilities   = CODEC_CAP_DELAY,
 };
 #endif
--- a/libavcodec/aac.h
+++ b/libavcodec/aac.h
@@ -30,14 +30,9 @@
 #ifndef AVCODEC_AAC_H
 #define AVCODEC_AAC_H

-
-#include "aac_defines.h"
 #include "libavutil/float_dsp.h"
-#include "libavutil/fixed_dsp.h"
 #include "avcodec.h"
-#if !USE_FIXED
 #include "imdct15.h"
-#endif
 #include "fft.h"
 #include "mpeg4audio.h"
 #include "sbr.h"
@@ -50,8 +45,6 @@
 #define TNS_MAX_ORDER 20
 #define MAX_LTP_LONG_SFB 40

-#define CLIP_AVOIDANCE_FACTOR 0.95f
-
 enum RawDataBlockType {
    TYPE_SCE,
    TYPE_CPE,
@@ -83,10 +76,9 @@ enum BandType {
    ZERO_BT        = 0,     ///< Scalefactors and spectral data are all zero.
    FIRST_PAIR_BT  = 5,     ///< This and later band types encode two values (rather than four) with one code word.
    ESC_BT         = 11,    ///< Spectral data are coded with an escape sequence.
-    RESERVED_BT    = 12,    ///< Band types following are encoded differently from others.
    NOISE_BT       = 13,    ///< Spectral data are scaled white noise not coded in the bitstream.
-    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions (out of phase).
-    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions (in phase).
+    INTENSITY_BT2  = 14,    ///< Scalefactor data are intensity stereo positions.
+    INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
 };

 #define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
@@ -133,14 +125,12 @@ typedef struct OutputConfiguration {
 * Predictor State
 */
 typedef struct PredictorState {
-    AAC_FLOAT cor0;
-    AAC_FLOAT cor1;
-    AAC_FLOAT var0;
-    AAC_FLOAT var1;
-    AAC_FLOAT r0;
-    AAC_FLOAT r1;
-    AAC_FLOAT k1;
-    AAC_FLOAT x_est;
+    float cor0;
+    float cor1;
+    float var0;
+    float var1;
+    float r0;
+    float r1;
 } PredictorState;

 #define MAX_PREDICTORS 672
@@ -161,7 +151,7 @@ typedef struct PredictorState {
 typedef struct LongTermPrediction {
    int8_t present;
    int16_t lag;
-    INTFLOAT coef;
+    float coef;
    int8_t used[MAX_LTP_LONG_SFB];
 } LongTermPrediction;

@@ -183,10 +173,7 @@ typedef struct IndividualChannelStream {
    int predictor_present;
    int predictor_initialized;
    int predictor_reset_group;
-    int predictor_reset_count[31];  ///< used by encoder to count prediction resets
    uint8_t prediction_used[41];
-    uint8_t window_clipping[8]; ///< set if a certain window is near clipping
-    float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
 } IndividualChannelStream;

 /**
@@ -198,8 +185,7 @@ typedef struct TemporalNoiseShaping {
    int length[8][4];
    int direction[8][4];
    int order[8][4];
-    int coef_idx[8][4][TNS_MAX_ORDER];
-    INTFLOAT coef[8][4][TNS_MAX_ORDER];
+    float coef[8][4][TNS_MAX_ORDER];
 } TemporalNoiseShaping;

 /**
@@ -236,7 +222,7 @@ typedef struct ChannelCoupling {
    int ch_select[8];      /**< [0] shared list of gains; [1] list of gains for right channel;
                            *   [2] list of gains for left channel; [3] lists of gains for both channels
                            */
-    INTFLOAT gain[16][120];
+    float gain[16][120];
 } ChannelCoupling;

 /**
@@ -247,21 +233,17 @@ typedef struct SingleChannelElement {
    TemporalNoiseShaping tns;
    Pulse pulse;
    enum BandType band_type[128];                   ///< band types
-    enum BandType band_alt[128];                    ///< alternative band type (used by encoder)
    int band_type_run_end[120];                     ///< band type run end points
-    INTFLOAT sf[120];                               ///< scalefactors
+    float sf[120];                                  ///< scalefactors
    int sf_idx[128];                                ///< scalefactor indices (used by encoder)
    uint8_t zeroes[128];                            ///< band is not coded (used by encoder)
-    float  is_ener[128];                            ///< Intensity stereo pos (used by encoder)
-    float pns_ener[128];                            ///< Noise energy values (used by encoder)
-    DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024];   ///< coefficients for IMDCT, pristine
-    DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024];    ///< coefficients for IMDCT, maybe processed
-    DECLARE_ALIGNED(32, INTFLOAT, saved)[1536];     ///< overlap
-    DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048];   ///< PCM output buffer
-    DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
-    DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
+    DECLARE_ALIGNED(32, float,   pcoeffs)[1024];    ///< coefficients for IMDCT, pristine
+    DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT, maybe processed
+    DECLARE_ALIGNED(32, float,   saved)[1536];      ///< overlap
+    DECLARE_ALIGNED(32, float,   ret_buf)[2048];    ///< PCM output buffer
+    DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
    PredictorState predictor_state[MAX_PREDICTORS];
-    INTFLOAT *ret;                                  ///< PCM output
+    float *ret;                                     ///< PCM output
 } SingleChannelElement;

 /**
@@ -272,9 +254,7 @@ typedef struct ChannelElement {
    // CPE specific
    int common_window;        ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
    int     ms_mode;          ///< Signals mid/side stereo flags coding mode (used by encoder)
-    uint8_t is_mode;          ///< Set if any bands have been encoded using intensity stereo (used by encoder)
    uint8_t ms_mask[128];     ///< Set if mid/side stereo is used for each scalefactor window band
-    uint8_t is_mask[128];     ///< Set if intensity stereo is used (used by encoder)
    // shared
    SingleChannelElement ch[2];
    // CCE specific
@@ -308,7 +288,7 @@ struct AACContext {
     * (We do not want to have these on the stack.)
     * @{
     */
-    DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
+    DECLARE_ALIGNED(32, float, buf_mdct)[1024];
    /** @} */

    /**
@@ -319,12 +299,8 @@ struct AACContext {
    FFTContext mdct_small;
    FFTContext mdct_ld;
    FFTContext mdct_ltp;
-#if USE_FIXED
-    AVFixedDSPContext *fdsp;
-#else
    IMDCT15Context *mdct480;
    AVFloatDSPContext *fdsp;
-#endif /* USE_FIXED */
    int random_state;
    /** @} */

@@ -344,7 +320,7 @@ struct AACContext {
    int dmono_mode;      ///< 0->not dmono, 1->use first channel, 2->use second channel
    /** @} */

-    DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
+    DECLARE_ALIGNED(32, float, temp)[128];

    OutputConfiguration oc[2];
    int warned_num_aac_frames;
@@ -352,13 +328,11 @@ struct AACContext {
    /* aacdec functions pointers */
    void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
    void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
-    void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
+    void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
                      IndividualChannelStream *ics, int decode);
-    void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
-                                   INTFLOAT *in, IndividualChannelStream *ics);
+    void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
+                                   float *in, IndividualChannelStream *ics);
    void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
-    void (*vector_pow43)(int *coefs, int len);
-    void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);

 };

--- a/libavcodec/aac_adtstoasc_bsf.c
+++ b/libavcodec/aac_adtstoasc_bsf.c
@@ -89,7 +89,7 @@ static int aac_adtstoasc_filter(AVBitStreamFilterContext *bsfc,
        }
        av_free(avctx->extradata);
        avctx->extradata_size = 2 + pce_size;
-        avctx->extradata = av_mallocz(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
+        avctx->extradata = av_mallocz(avctx->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE);
        if (!avctx->extradata) {
            avctx->extradata_size = 0;
            return AVERROR(ENOMEM);
--- a/libavcodec/aac_defines.h
+++ b/libavcodec/aac_defines.h
@@ -1,114 +0,0 @@
-/*
- * AAC defines
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_AAC_DEFINES_H
-#define AVCODEC_AAC_DEFINES_H
-
-#ifndef USE_FIXED
-#define USE_FIXED 0
-#endif
-
-#if USE_FIXED
-
-#include "libavutil/softfloat.h"
-
-#define FFT_FLOAT    0
-#define FFT_FIXED_32 1
-
-#define AAC_RENAME(x)       x ## _fixed
-#define AAC_RENAME_32(x)    x ## _fixed_32
-#define INTFLOAT int
-#define INT64FLOAT          int64_t
-#define SHORTFLOAT int16_t
-#define AAC_FLOAT SoftFloat
-#define AAC_SIGNE           int
-#define FIXR(a)             ((int)((a) * 1 + 0.5))
-#define FIXR10(a)           ((int)((a) * 1024.0 + 0.5))
-#define Q23(a)              (int)((a) * 8388608.0 + 0.5)
-#define Q30(x)              (int)((x)*1073741824.0 + 0.5)
-#define Q31(x)              (int)((x)*2147483648.0 + 0.5)
-#define RANGE15(x)          x
-#define GET_GAIN(x, y)      (-(y) << (x)) + 1024
-#define AAC_MUL16(x, y)     (int)(((int64_t)(x) * (y) + 0x8000) >> 16)
-#define AAC_MUL26(x, y)     (int)(((int64_t)(x) * (y) + 0x2000000) >> 26)
-#define AAC_MUL30(x, y)     (int)(((int64_t)(x) * (y) + 0x20000000) >> 30)
-#define AAC_MUL31(x, y)     (int)(((int64_t)(x) * (y) + 0x40000000) >> 31)
-#define AAC_MADD28(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
-                                      ((int64_t)(a) * (b)) + \
-                                        0x8000000) >> 28)
-#define AAC_MADD30(x, y, a, b) (int)((((int64_t)(x) * (y)) + \
-                                      ((int64_t)(a) * (b)) + \
-                                        0x20000000) >> 30)
-#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
-                                                     ((int64_t)(a) * (b)) + \
-                                                     ((int64_t)(c) * (d)) + \
-                                                     ((int64_t)(e) * (f)) + \
-                                                       0x20000000) >> 30)
-#define AAC_MSUB30(x, y, a, b) (int)((((int64_t)(x) * (y)) - \
-                                      ((int64_t)(a) * (b)) + \
-                                        0x20000000) >> 30)
-#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) (int)((((int64_t)(x) * (y)) + \
-                                                     ((int64_t)(a) * (b)) - \
-                                                     ((int64_t)(c) * (d)) - \
-                                                     ((int64_t)(e) * (f)) + \
-                                                       0x20000000) >> 30)
-#define AAC_MSUB31_V3(x, y, z)    (int)((((int64_t)(x) * (z)) - \
-                                      ((int64_t)(y) * (z)) + \
-                                        0x40000000) >> 31)
-#define AAC_HALF_SUM(x, y)  (x) >> 1 + (y) >> 1
-#define AAC_SRA_R(x, y)     (int)(((x) + (1 << ((y) - 1))) >> (y))
-
-#else
-
-#define FFT_FLOAT    1
-#define FFT_FIXED_32 0
-
-#define AAC_RENAME(x)       x
-#define AAC_RENAME_32(x)    x
-#define INTFLOAT float
-#define INT64FLOAT          float
-#define SHORTFLOAT float
-#define AAC_FLOAT float
-#define AAC_SIGNE           unsigned
-#define FIXR(x)             ((float)(x))
-#define FIXR10(x)           ((float)(x))
-#define Q23(x)              x
-#define Q30(x)              x
-#define Q31(x)              x
-#define RANGE15(x)          (32768.0 * (x))
-#define GET_GAIN(x, y)      powf((x), -(y))
-#define AAC_MUL16(x, y)     ((x) * (y))
-#define AAC_MUL26(x, y)     ((x) * (y))
-#define AAC_MUL30(x, y)     ((x) * (y))
-#define AAC_MUL31(x, y)     ((x) * (y))
-#define AAC_MADD28(x, y, a, b) ((x) * (y) + (a) * (b))
-#define AAC_MADD30(x, y, a, b) ((x) * (y) + (a) * (b))
-#define AAC_MADD30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) + \
-                                               (c) * (d) + (e) * (f))
-#define AAC_MSUB30(x, y, a, b) ((x) * (y) - (a) * (b))
-#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f) ((x) * (y) + (a) * (b) - \
-                                               (c) * (d) - (e) * (f))
-#define AAC_MSUB31_V3(x, y, z)    ((x) - (y)) * (z)
-#define AAC_HALF_SUM(x, y)  ((x) + (y)) * 0.5f
-#define AAC_SRA_R(x, y)     (x)
-
-#endif /* USE_FIXED */
-
-#endif /* AVCODEC_AAC_DEFINES_H */
--- a/libavcodec/aac_parser.c
+++ b/libavcodec/aac_parser.c
@@ -34,7 +34,7 @@ static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
    int size;
    union {
        uint64_t u64;
-        uint8_t  u8[8 + AV_INPUT_BUFFER_PADDING_SIZE];
+        uint8_t  u8[8 + FF_INPUT_BUFFER_PADDING_SIZE];
    } tmp;

    tmp.u64 = av_be2ne64(state);
--- a/libavcodec/aac_tablegen.c
+++ b/libavcodec/aac_tablegen.c
@@ -33,7 +33,5 @@ int main(void)

    WRITE_ARRAY("const", float, ff_aac_pow2sf_tab);

-    WRITE_ARRAY("const", float, ff_aac_pow34sf_tab);
-
    return 0;
 }
--- a/libavcodec/aac_tablegen.h
+++ b/libavcodec/aac_tablegen.h
@@ -30,15 +30,12 @@
 #else
 #include "libavutil/mathematics.h"
 float ff_aac_pow2sf_tab[428];
-float ff_aac_pow34sf_tab[428];

 av_cold void ff_aac_tableinit(void)
 {
    int i;
-    for (i = 0; i < 428; i++) {
+    for (i = 0; i < 428; i++)
        ff_aac_pow2sf_tab[i] = pow(2, (i - POW_SF2_ZERO) / 4.0);
-        ff_aac_pow34sf_tab[i] = pow(ff_aac_pow2sf_tab[i], 3.0/4.0);
-    }
 }
 #endif /* CONFIG_HARDCODED_TABLES */

--- a/libavcodec/aac_tablegen_decl.h
+++ b/libavcodec/aac_tablegen_decl.h
@@ -28,11 +28,9 @@
 #if CONFIG_HARDCODED_TABLES
 #define ff_aac_tableinit()
 extern const float ff_aac_pow2sf_tab[428];
-extern const float ff_aac_pow34sf_tab[428];
 #else
 void ff_aac_tableinit(void);
 extern       float ff_aac_pow2sf_tab[428];
-extern       float ff_aac_pow34sf_tab[428];
 #endif /* CONFIG_HARDCODED_TABLES */

 #endif /* AVCODEC_AAC_TABLEGEN_DECL_H */
--- a/libavcodec/aaccoder.c
+++ b/libavcodec/aaccoder.c
@@ -39,26 +39,290 @@
 #include "aac.h"
 #include "aacenc.h"
 #include "aactab.h"
-#include "aacenctab.h"
-#include "aacenc_utils.h"
-#include "aacenc_quantization.h"
-#include "aac_tablegen_decl.h"
-
-#include "aacenc_is.h"
-#include "aacenc_tns.h"
-#include "aacenc_pred.h"

 /** Frequency in Hz for lower limit of noise substitution **/
-#define NOISE_LOW_LIMIT 4500
+#define NOISE_LOW_LIMIT 4000

-/* Energy spread threshold value below which no PNS is used, this corresponds to
- * typically around 17Khz, after which PNS usage decays ending at 19Khz */
-#define NOISE_SPREAD_THRESHOLD 0.5f
+/** Total number of usable codebooks **/
+#define CB_TOT 13

-/* This constant gets divided by lambda to return ~1.65 which when multiplied
- * by the band->threshold and compared to band->energy is the boundary between
- * excessive PNS and little PNS usage. */
-#define NOISE_LAMBDA_NUMERATOR 252.1f
+/** bits needed to code codebook run value for long windows */
+static const uint8_t run_value_bits_long[64] = {
+     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
+     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
+    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
+    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
+};
+
+/** bits needed to code codebook run value for short windows */
+static const uint8_t run_value_bits_short[16] = {
+    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
+};
+
+static const uint8_t * const run_value_bits[2] = {
+    run_value_bits_long, run_value_bits_short
+};
+
+/** Map to convert values from BandCodingPath index to a codebook index **/
+static const uint8_t aac_cb_out_map[CB_TOT]  = {0,1,2,3,4,5,6,7,8,9,10,11,13};
+/** Inverse map to convert from codebooks to BandCodingPath indices **/
+static const uint8_t aac_cb_in_map[CB_TOT+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12};
+
+/**
+ * Quantize one coefficient.
+ * @return absolute value of the quantized coefficient
+ * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
+ */
+static av_always_inline int quant(float coef, const float Q)
+{
+    float a = coef * Q;
+    return sqrtf(a * sqrtf(a)) + 0.4054;
+}
+
+static void quantize_bands(int *out, const float *in, const float *scaled,
+                           int size, float Q34, int is_signed, int maxval)
+{
+    int i;
+    double qc;
+    for (i = 0; i < size; i++) {
+        qc = scaled[i] * Q34;
+        out[i] = (int)FFMIN(qc + 0.4054, (double)maxval);
+        if (is_signed && in[i] < 0.0f) {
+            out[i] = -out[i];
+        }
+    }
+}
+
+static void abs_pow34_v(float *out, const float *in, const int size)
+{
+#ifndef USE_REALLY_FULL_SEARCH
+    int i;
+    for (i = 0; i < size; i++) {
+        float a = fabsf(in[i]);
+        out[i] = sqrtf(a * sqrtf(a));
+    }
+#endif /* USE_REALLY_FULL_SEARCH */
+}
+
+static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
+static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
+
+/**
+ * Calculate rate distortion cost for quantizing with given codebook
+ *
+ * @return quantization distortion
+ */
+static av_always_inline float quantize_and_encode_band_cost_template(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits, int BT_ZERO, int BT_UNSIGNED,
+                                int BT_PAIR, int BT_ESC, int BT_NOISE)
+{
+    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
+    const float Q   = ff_aac_pow2sf_tab [q_idx];
+    const float Q34 = ff_aac_pow34sf_tab[q_idx];
+    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
+    const float CLIPPED_ESCAPE = 165140.0f*IQ;
+    int i, j;
+    float cost = 0;
+    const int dim = BT_PAIR ? 2 : 4;
+    int resbits = 0;
+    int off;
+
+    if (BT_ZERO) {
+        for (i = 0; i < size; i++)
+            cost += in[i]*in[i];
+        if (bits)
+            *bits = 0;
+        return cost * lambda;
+    }
+    if (BT_NOISE) {
+        for (i = 0; i < size; i++)
+            cost += in[i]*in[i];
+        if (bits)
+            *bits = 0;
+        return cost * lambda;
+    }
+    if (!scaled) {
+        abs_pow34_v(s->scoefs, in, size);
+        scaled = s->scoefs;
+    }
+    quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, aac_cb_maxval[cb]);
+    if (BT_UNSIGNED) {
+        off = 0;
+    } else {
+        off = aac_cb_maxval[cb];
+    }
+    for (i = 0; i < size; i += dim) {
+        const float *vec;
+        int *quants = s->qcoefs + i;
+        int curidx = 0;
+        int curbits;
+        float rd = 0.0f;
+        for (j = 0; j < dim; j++) {
+            curidx *= aac_cb_range[cb];
+            curidx += quants[j] + off;
+        }
+        curbits =  ff_aac_spectral_bits[cb-1][curidx];
+        vec     = &ff_aac_codebook_vectors[cb-1][curidx*dim];
+        if (BT_UNSIGNED) {
+            for (j = 0; j < dim; j++) {
+                float t = fabsf(in[i+j]);
+                float di;
+                if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow
+                    if (t >= CLIPPED_ESCAPE) {
+                        di = t - CLIPPED_ESCAPE;
+                        curbits += 21;
+                    } else {
+                        int c = av_clip_uintp2(quant(t, Q), 13);
+                        di = t - c*cbrtf(c)*IQ;
+                        curbits += av_log2(c)*2 - 4 + 1;
+                    }
+                } else {
+                    di = t - vec[j]*IQ;
+                }
+                if (vec[j] != 0.0f)
+                    curbits++;
+                rd += di*di;
+            }
+        } else {
+            for (j = 0; j < dim; j++) {
+                float di = in[i+j] - vec[j]*IQ;
+                rd += di*di;
+            }
+        }
+        cost    += rd * lambda + curbits;
+        resbits += curbits;
+        if (cost >= uplim)
+            return uplim;
+        if (pb) {
+            put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]);
+            if (BT_UNSIGNED)
+                for (j = 0; j < dim; j++)
+                    if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f)
+                        put_bits(pb, 1, in[i+j] < 0.0f);
+            if (BT_ESC) {
+                for (j = 0; j < 2; j++) {
+                    if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) {
+                        int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q), 13);
+                        int len = av_log2(coef);
+
+                        put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
+                        put_sbits(pb, len, coef);
+                    }
+                }
+            }
+        }
+    }
+
+    if (bits)
+        *bits = resbits;
+    return cost;
+}
+
+static float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb,
+                                                const float *in, const float *scaled,
+                                                int size, int scale_idx, int cb,
+                                                const float lambda, const float uplim,
+                                                int *bits) {
+    av_assert0(0);
+    return 0.0f;
+}
+
+#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE) \
+static float quantize_and_encode_band_cost_ ## NAME(                                    \
+                                struct AACEncContext *s,                                \
+                                PutBitContext *pb, const float *in,                     \
+                                const float *scaled, int size, int scale_idx,           \
+                                int cb, const float lambda, const float uplim,          \
+                                int *bits) {                                            \
+    return quantize_and_encode_band_cost_template(                                      \
+                                s, pb, in, scaled, size, scale_idx,                     \
+                                BT_ESC ? ESC_BT : cb, lambda, uplim, bits,              \
+                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE);       \
+}
+
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1, 0)
+QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1)
+
+static float (*const quantize_and_encode_band_cost_arr[])(
+                                struct AACEncContext *s,
+                                PutBitContext *pb, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits) = {
+    quantize_and_encode_band_cost_ZERO,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_SQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_UQUAD,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_SPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_UPAIR,
+    quantize_and_encode_band_cost_ESC,
+    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
+    quantize_and_encode_band_cost_NOISE,
+};
+
+#define quantize_and_encode_band_cost(                                  \
+                                s, pb, in, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits)                    \
+    quantize_and_encode_band_cost_arr[cb](                              \
+                                s, pb, in, scaled, size, scale_idx, cb, \
+                                lambda, uplim, bits)
+
+static float quantize_band_cost(struct AACEncContext *s, const float *in,
+                                const float *scaled, int size, int scale_idx,
+                                int cb, const float lambda, const float uplim,
+                                int *bits)
+{
+    return quantize_and_encode_band_cost(s, NULL, in, scaled, size, scale_idx,
+                                         cb, lambda, uplim, bits);
+}
+
+static void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
+                                     const float *in, int size, int scale_idx,
+                                     int cb, const float lambda)
+{
+    quantize_and_encode_band_cost(s, pb, in, NULL, size, scale_idx, cb, lambda,
+                                  INFINITY, NULL);
+}
+
+static float find_max_val(int group_len, int swb_size, const float *scaled) {
+    float maxval = 0.0f;
+    int w2, i;
+    for (w2 = 0; w2 < group_len; w2++) {
+        for (i = 0; i < swb_size; i++) {
+            maxval = FFMAX(maxval, scaled[w2*128+i]);
+        }
+    }
+    return maxval;
+}
+
+static int find_min_book(float maxval, int sf) {
+    float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
+    float Q34 = sqrtf(Q * sqrtf(Q));
+    int qmaxval, cb;
+    qmaxval = maxval * Q34 + 0.4054f;
+    if      (qmaxval ==  0) cb = 0;
+    else if (qmaxval ==  1) cb = 1;
+    else if (qmaxval ==  2) cb = 3;
+    else if (qmaxval <=  4) cb = 5;
+    else if (qmaxval <=  7) cb = 7;
+    else if (qmaxval <= 12) cb = 9;
+    else                    cb = 11;
+    return cb;
+}

 /**
 * structure used in optimal codebook search
@@ -75,7 +339,7 @@ typedef struct BandCodingPath {
 static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][CB_TOT_ALL];
+    BandCodingPath path[120][CB_TOT];
    int w, swb, cb, start, size;
    int i, j;
    const int max_sfb  = sce->ics.max_sfb;
@@ -88,7 +352,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce

    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
    start = win*128;
-    for (cb = 0; cb < CB_TOT_ALL; cb++) {
+    for (cb = 0; cb < CB_TOT; cb++) {
        path[0][cb].cost     = 0.0f;
        path[0][cb].prev_idx = -1;
        path[0][cb].run      = 0;
@@ -96,7 +360,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
    for (swb = 0; swb < max_sfb; swb++) {
        size = sce->ics.swb_sizes[swb];
        if (sce->zeroes[win*16 + swb]) {
-            for (cb = 0; cb < CB_TOT_ALL; cb++) {
+            for (cb = 0; cb < CB_TOT; cb++) {
                path[swb+1][cb].prev_idx = cb;
                path[swb+1][cb].cost     = path[swb][cb].cost;
                path[swb+1][cb].run      = path[swb][cb].run + 1;
@@ -106,22 +370,15 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
            int mincb = next_mincb;
            next_minrd = INFINITY;
            next_mincb = 0;
-            for (cb = 0; cb < CB_TOT_ALL; cb++) {
+            for (cb = 0; cb < CB_TOT; cb++) {
                float cost_stay_here, cost_get_here;
                float rd = 0.0f;
-                if (cb >= 12 && sce->band_type[win*16+swb] < aac_cb_out_map[cb] ||
-                    cb  < aac_cb_in_map[sce->band_type[win*16+swb]] && sce->band_type[win*16+swb] > aac_cb_out_map[cb]) {
-                    path[swb+1][cb].prev_idx = -1;
-                    path[swb+1][cb].cost     = INFINITY;
-                    path[swb+1][cb].run      = path[swb][cb].run + 1;
-                    continue;
-                }
                for (w = 0; w < group_len; w++) {
                    FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(win+w)*16+swb];
-                    rd += quantize_band_cost(s, &sce->coeffs[start + w*128],
-                                             &s->scoefs[start + w*128], size,
+                    rd += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                             s->scoefs + start + w*128, size,
                                             sce->sf_idx[(win+w)*16+swb], aac_cb_out_map[cb],
-                                             lambda / band->threshold, INFINITY, NULL, 0);
+                                             lambda / band->threshold, INFINITY, NULL);
                }
                cost_stay_here = path[swb][cb].cost + rd;
                cost_get_here  = minrd              + rd + run_bits + 4;
@@ -149,12 +406,11 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
    //convert resulting path from backward-linked list
    stack_len = 0;
    idx       = 0;
-    for (cb = 1; cb < CB_TOT_ALL; cb++)
+    for (cb = 1; cb < CB_TOT; cb++)
        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
            idx = cb;
    ppos = max_sfb;
    while (ppos > 0) {
-        av_assert1(idx >= 0);
        cb = idx;
        stackrun[stack_len] = path[ppos][cb].run;
        stackcb [stack_len] = cb;
@@ -185,7 +441,7 @@ static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce
 static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                                  int win, int group_len, const float lambda)
 {
-    BandCodingPath path[120][CB_TOT_ALL];
+    BandCodingPath path[120][CB_TOT];
    int w, swb, cb, start, size;
    int i, j;
    const int max_sfb  = sce->ics.max_sfb;
@@ -198,7 +454,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,

    abs_pow34_v(s->scoefs, sce->coeffs, 1024);
    start = win*128;
-    for (cb = 0; cb < CB_TOT_ALL; cb++) {
+    for (cb = 0; cb < CB_TOT; cb++) {
        path[0][cb].cost     = run_bits+4;
        path[0][cb].prev_idx = -1;
        path[0][cb].run      = 0;
@@ -222,7 +478,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
            }
            next_minbits = path[swb+1][0].cost;
            next_mincb = 0;
-            for (cb = 1; cb < CB_TOT_ALL; cb++) {
+            for (cb = 1; cb < CB_TOT; cb++) {
                path[swb+1][cb].cost = 61450;
                path[swb+1][cb].prev_idx = -1;
                path[swb+1][cb].run = 0;
@@ -239,21 +495,21 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
                path[swb+1][cb].prev_idx = -1;
                path[swb+1][cb].run = 0;
            }
-            for (cb = startcb; cb < CB_TOT_ALL; cb++) {
+            for (cb = startcb; cb < CB_TOT; cb++) {
                float cost_stay_here, cost_get_here;
                float bits = 0.0f;
-                if (cb >= 12 && sce->band_type[win*16+swb] != aac_cb_out_map[cb]) {
+                if (cb == 12 && sce->band_type[win*16+swb] != NOISE_BT) {
                    path[swb+1][cb].cost = 61450;
                    path[swb+1][cb].prev_idx = -1;
                    path[swb+1][cb].run = 0;
                    continue;
                }
                for (w = 0; w < group_len; w++) {
-                    bits += quantize_band_cost(s, &sce->coeffs[start + w*128],
-                                               &s->scoefs[start + w*128], size,
-                                               sce->sf_idx[win*16+swb],
+                    bits += quantize_band_cost(s, sce->coeffs + start + w*128,
+                                               s->scoefs + start + w*128, size,
+                                               sce->sf_idx[(win+w)*16+swb],
                                               aac_cb_out_map[cb],
-                                               0, INFINITY, NULL, 0);
+                                               0, INFINITY, NULL);
                }
                cost_stay_here = path[swb][cb].cost + bits;
                cost_get_here  = minbits            + bits + run_bits + 4;
@@ -281,7 +537,7 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
    //convert resulting path from backward-linked list
    stack_len = 0;
    idx       = 0;
-    for (cb = 1; cb < CB_TOT_ALL; cb++)
+    for (cb = 1; cb < CB_TOT; cb++)
        if (path[max_sfb][cb].cost < path[max_sfb][idx].cost)
            idx = cb;
    ppos = max_sfb;
@@ -314,6 +570,16 @@ static void codebook_trellis_rate(AACEncContext *s, SingleChannelElement *sce,
    }
 }

+/** Return the minimum scalefactor where the quantized coef does not clip. */
+static av_always_inline uint8_t coef2minsf(float coef) {
+    return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
+/** Return the maximum scalefactor where the quantized coef is not zero. */
+static av_always_inline uint8_t coef2maxsf(float coef) {
+    return av_clip_uint8(log2f(coef)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
+}
+
 typedef struct TrellisPath {
    float cost;
    int prev;
@@ -322,43 +588,6 @@ typedef struct TrellisPath {
 #define TRELLIS_STAGES 121
 #define TRELLIS_STATES (SCALE_MAX_DIFF+1)

-static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce)
-{
-    int w, g, start = 0;
-    int minscaler_n = sce->sf_idx[0], minscaler_i = sce->sf_idx[0];
-    int bands = 0;
-
-    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        start = 0;
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
-                sce->sf_idx[w*16+g] = av_clip(ceilf(log2f(sce->is_ener[w*16+g])*2), -155, 100);
-                minscaler_i = FFMIN(minscaler_i, sce->sf_idx[w*16+g]);
-                bands++;
-            } else if (sce->band_type[w*16+g] == NOISE_BT) {
-                sce->sf_idx[w*16+g] = av_clip(4+log2f(sce->pns_ener[w*16+g])*2, -100, 155);
-                minscaler_n = FFMIN(minscaler_n, sce->sf_idx[w*16+g]);
-                bands++;
-            }
-            start += sce->ics.swb_sizes[g];
-        }
-    }
-
-    if (!bands)
-        return;
-
-    /* Clip the scalefactor indices */
-    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) {
-                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_i, minscaler_i + SCALE_MAX_DIFF);
-            } else if (sce->band_type[w*16+g] == NOISE_BT) {
-                sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
-            }
-        }
-    }
-}
-
 static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
                                       SingleChannelElement *sce,
                                       const float lambda)
@@ -424,7 +653,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        start = w*128;
        for (g = 0; g < sce->ics.num_swb; g++) {
-            const float *coefs = &sce->coeffs[start];
+            const float *coefs = sce->coeffs + start;
            float qmin, qmax;
            int nz = 0;

@@ -463,7 +692,7 @@ static void search_for_quantizers_anmr(AVCodecContext *avctx, AACEncContext *s,
                    for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                        FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
                        dist += quantize_band_cost(s, coefs + w2*128, s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
-                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL, 0);
+                                                   q + q0, cb, lambda / band->threshold, INFINITY, NULL);
                    }
                    minrd = FFMIN(minrd, dist);

@@ -519,9 +748,11 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
 {
    int start = 0, i, w, w2, g;
    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->channels * (lambda / 120.f);
+    const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
    float dists[128] = { 0 }, uplims[128] = { 0 };
    float maxvals[128];
-    int fflag, minscaler;
+    int noise_sf[128] = { 0 };
+    int fflag, minscaler, minscaler_n;
    int its  = 0;
    int allz = 0;
    float minthr = INFINITY;
@@ -532,12 +763,13 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
    //XXX: some heuristic to determine initial quantizers will reduce search time
    //determine zero bands and upper limits
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
+        start = 0;
        for (g = 0;  g < sce->ics.num_swb; g++) {
            int nz = 0;
            float uplim = 0.0f, energy = 0.0f;
            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
-                uplim  += band->threshold;
+                uplim += band->threshold;
                energy += band->energy;
                if (band->energy <= band->threshold || band->threshold == 0.0f) {
                    sce->zeroes[(w+w2)*16+g] = 1;
@@ -546,10 +778,18 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                nz = 1;
            }
            uplims[w*16+g] = uplim *512;
+            if (s->options.pns && start*freq_mult > NOISE_LOW_LIMIT && energy < uplim * 1.2f) {
+                noise_sf[w*16+g] = av_clip(4+FFMIN(log2f(energy)*2,255), -100, 155);
+                sce->band_type[w*16+g] = NOISE_BT;
+                nz= 1;
+            } else { /** Band type will be determined by the twoloop algorithm */
+                sce->band_type[w*16+g] = 0;
+            }
            sce->zeroes[w*16+g] = !nz;
            if (nz)
                minthr = FFMIN(minthr, uplim);
            allz |= nz;
+            start += sce->ics.swb_sizes[g];
        }
    }
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
@@ -580,6 +820,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
    do {
        int tbits, qstep;
        minscaler = sce->sf_idx[0];
+        minscaler_n = sce->sf_idx[0];
        //inner loop - quantize spectrum to fit into given number of bits
        qstep = its ? 1 : 32;
        do {
@@ -588,13 +829,17 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
            for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
                start = w*128;
                for (g = 0;  g < sce->ics.num_swb; g++) {
-                    const float *coefs = &sce->coeffs[start];
-                    const float *scaled = &s->scoefs[start];
+                    const float *coefs = sce->coeffs + start;
+                    const float *scaled = s->scoefs + start;
                    int bits = 0;
                    int cb;
                    float dist = 0.0f;

-                    if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
+                    if (sce->band_type[w*16+g] == NOISE_BT) {
+                        minscaler_n = FFMIN(minscaler_n, noise_sf[w*16+g]);
+                        start += sce->ics.swb_sizes[g];
+                        continue;
+                    } else if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) {
                        start += sce->ics.swb_sizes[g];
                        continue;
                    }
@@ -609,8 +854,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
                                                   cb,
                                                   1.0f,
                                                   INFINITY,
-                                                   &b,
-                                                   0);
+                                                   &b);
                        bits += b;
                    }
                    dists[w*16+g] = dist - bits;
@@ -639,9 +883,16 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
        fflag = 0;
        minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF);

+        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
+            for (g = 0; g < sce->ics.num_swb; g++)
+                if (sce->band_type[w*16+g] == NOISE_BT)
+                    sce->sf_idx[w*16+g] = av_clip(noise_sf[w*16+g], minscaler_n, minscaler_n + SCALE_MAX_DIFF);
+
        for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
            for (g = 0; g < sce->ics.num_swb; g++) {
                int prevsc = sce->sf_idx[w*16+g];
+                if (sce->band_type[w*16+g] == NOISE_BT)
+                    continue;
                if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) {
                    if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1))
                        sce->sf_idx[w*16+g]--;
@@ -684,7 +935,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
        }
    } else {
        for (w = 0; w < 8; w++) {
-            const float *coeffs = &sce->coeffs[w*128];
+            const float *coeffs = sce->coeffs + w*128;
            curband = start = 0;
            for (i = 0; i < 128; i++) {
                if (i - start >= sce->ics.swb_sizes[curband]) {
@@ -709,7 +960,7 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        start = w*128;
        for (g = 0; g < sce->ics.num_swb; g++) {
-            float *coefs   = &sce->coeffs[start];
+            float *coefs   = sce->coeffs + start;
            const int size = sce->ics.swb_sizes[g];
            int start2 = start, end2 = start + size, peakpos = start;
            float maxval = -1, thr = 0.0f, t;
@@ -750,8 +1001,8 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
        start = w*128;
        for (g = 0;  g < sce->ics.num_swb; g++) {
-            const float *coefs  = &sce->coeffs[start];
-            const float *scaled = &s->scoefs[start];
+            const float *coefs  = sce->coeffs + start;
+            const float *scaled = s->scoefs   + start;
            const int size      = sce->ics.swb_sizes[g];
            int scf, prev_scf, step;
            int min_scf = -1, max_scf = 256;
@@ -776,12 +1027,11 @@ static void search_for_quantizers_faac(AVCodecContext *avctx, AACEncContext *s,
                                               ESC_BT,
                                               lambda,
                                               INFINITY,
-                                               &b,
-                                               0);
+                                               &b);
                    dist -= b;
                }
                dist *= 1.0f / 512.0f / lambda;
-                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512], ROUND_STANDARD);
+                quant_max = quant(maxq[w*16+g], ff_aac_pow2sf_tab[POW_SF2_ZERO - scf + SCALE_ONE_POS - SCALE_DIV_512]);
                if (quant_max >= 8191) { // too much, return to the previous quantizer
                    sce->sf_idx[w*16+g] = prev_scf;
                    break;
@@ -861,50 +1111,17 @@ static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s,
                sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
 }

-static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
-{
-    int start = 0, w, w2, g;
-    const float lambda = s->lambda;
-    const float freq_mult = avctx->sample_rate/(1024.0f/sce->ics.num_windows)/2.0f;
-    const float spread_threshold = NOISE_SPREAD_THRESHOLD*(lambda/120.f);
-    const float thr_mult = NOISE_LAMBDA_NUMERATOR/lambda;
-
-    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        start = 0;
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (start*freq_mult > NOISE_LOW_LIMIT*(lambda/170.0f)) {
-                float energy = 0.0f, threshold = 0.0f, spread = 0.0f;
-                for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
-                    FFPsyBand *band = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-                    energy += band->energy;
-                    threshold += band->threshold;
-                    spread += band->spread;
-                }
-                if (spread > spread_threshold*sce->ics.group_len[w] &&
-                    ((sce->zeroes[w*16+g] && energy >= threshold) ||
-                    energy < threshold*thr_mult*sce->ics.group_len[w])) {
-                    sce->band_type[w*16+g] = NOISE_BT;
-                    sce->pns_ener[w*16+g] = energy / sce->ics.group_len[w];
-                    sce->zeroes[w*16+g] = 0;
-                }
-            }
-            start += sce->ics.swb_sizes[g];
-        }
-    }
-}
-
-static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
+static void search_for_ms(AACEncContext *s, ChannelElement *cpe,
+                          const float lambda)
 {
    int start = 0, i, w, w2, g;
    float M[128], S[128];
    float *L34 = s->scoefs, *R34 = s->scoefs + 128, *M34 = s->scoefs + 128*2, *S34 = s->scoefs + 128*3;
-    const float lambda = s->lambda;
    SingleChannelElement *sce0 = &cpe->ch[0];
    SingleChannelElement *sce1 = &cpe->ch[1];
    if (!cpe->common_window)
        return;
    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
-        start = 0;
        for (g = 0;  g < sce0->ics.num_swb; g++) {
            if (!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]) {
                float dist1 = 0.0f, dist2 = 0.0f;
@@ -914,39 +1131,39 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
                    float minthr = FFMIN(band0->threshold, band1->threshold);
                    float maxthr = FFMAX(band0->threshold, band1->threshold);
                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                        M[i] = (sce0->coeffs[start+(w+w2)*128+i]
-                              + sce1->coeffs[start+(w+w2)*128+i]) * 0.5;
+                        M[i] = (sce0->pcoeffs[start+w2*128+i]
+                              + sce1->pcoeffs[start+w2*128+i]) * 0.5;
                        S[i] =  M[i]
-                              - sce1->coeffs[start+(w+w2)*128+i];
+                              - sce1->pcoeffs[start+w2*128+i];
                    }
-                    abs_pow34_v(L34, sce0->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
-                    abs_pow34_v(R34, sce1->coeffs+start+(w+w2)*128, sce0->ics.swb_sizes[g]);
+                    abs_pow34_v(L34, sce0->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
+                    abs_pow34_v(R34, sce1->coeffs+start+w2*128, sce0->ics.swb_sizes[g]);
                    abs_pow34_v(M34, M,                         sce0->ics.swb_sizes[g]);
                    abs_pow34_v(S34, S,                         sce0->ics.swb_sizes[g]);
-                    dist1 += quantize_band_cost(s, &sce0->coeffs[start + (w+w2)*128],
+                    dist1 += quantize_band_cost(s, sce0->coeffs + start + w2*128,
                                                L34,
                                                sce0->ics.swb_sizes[g],
                                                sce0->sf_idx[(w+w2)*16+g],
                                                sce0->band_type[(w+w2)*16+g],
-                                                lambda / band0->threshold, INFINITY, NULL, 0);
-                    dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
+                                                lambda / band0->threshold, INFINITY, NULL);
+                    dist1 += quantize_band_cost(s, sce1->coeffs + start + w2*128,
                                                R34,
                                                sce1->ics.swb_sizes[g],
                                                sce1->sf_idx[(w+w2)*16+g],
                                                sce1->band_type[(w+w2)*16+g],
-                                                lambda / band1->threshold, INFINITY, NULL, 0);
+                                                lambda / band1->threshold, INFINITY, NULL);
                    dist2 += quantize_band_cost(s, M,
                                                M34,
                                                sce0->ics.swb_sizes[g],
                                                sce0->sf_idx[(w+w2)*16+g],
                                                sce0->band_type[(w+w2)*16+g],
-                                                lambda / maxthr, INFINITY, NULL, 0);
+                                                lambda / maxthr, INFINITY, NULL);
                    dist2 += quantize_band_cost(s, S,
                                                S34,
                                                sce1->ics.swb_sizes[g],
                                                sce1->sf_idx[(w+w2)*16+g],
                                                sce1->band_type[(w+w2)*16+g],
-                                                lambda / minthr, INFINITY, NULL, 0);
+                                                lambda / minthr, INFINITY, NULL);
                }
                cpe->ms_mask[w*16+g] = dist2 < dist1;
            }
@@ -960,64 +1177,24 @@ AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
        search_for_quantizers_faac,
        encode_window_bands_info,
        quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
        search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
    },
    [AAC_CODER_ANMR] = {
        search_for_quantizers_anmr,
        encode_window_bands_info,
        quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
        search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
    },
    [AAC_CODER_TWOLOOP] = {
        search_for_quantizers_twoloop,
        codebook_trellis_rate,
        quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
        search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
    },
    [AAC_CODER_FAST] = {
        search_for_quantizers_fast,
        encode_window_bands_info,
        quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_encode_main_pred,
-        ff_aac_adjust_common_prediction,
-        ff_aac_apply_main_pred,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        search_for_pns,
-        ff_aac_search_for_tns,
        search_for_ms,
-        ff_aac_search_for_is,
-        ff_aac_search_for_pred,
    },
 };
--- a/libavcodec/aacdec.c
+++ b/libavcodec/aacdec.c
--- a/libavcodec/aacdec_fixed.c
+++ b/libavcodec/aacdec_fixed.c
@@ -1,443 +0,0 @@
-/*
- * Copyright (c) 2013
- *      MIPS Technologies, Inc., California.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
- * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
- *    contributors may be used to endorse or promote products derived from
- *    this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
- * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
- * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
- * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
- * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
- * SUCH DAMAGE.
- *
- * AAC decoder fixed-point implementation
- *
- * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
- * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC decoder
- * @author Oded Shimon  ( ods15 ods15 dyndns org )
- * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
- *
- * Fixed point implementation
- * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
- */
-
-#define FFT_FLOAT 0
-#define FFT_FIXED_32 1
-#define USE_FIXED 1
-
-#include "libavutil/fixed_dsp.h"
-#include "libavutil/opt.h"
-#include "avcodec.h"
-#include "internal.h"
-#include "get_bits.h"
-#include "fft.h"
-#include "lpc.h"
-#include "kbdwin.h"
-#include "sinewin.h"
-
-#include "aac.h"
-#include "aactab.h"
-#include "aacdectab.h"
-#include "cbrt_tablegen.h"
-#include "sbr.h"
-#include "aacsbr.h"
-#include "mpeg4audio.h"
-#include "aacadtsdec.h"
-#include "libavutil/intfloat.h"
-
-#include <math.h>
-#include <string.h>
-
-static av_always_inline void reset_predict_state(PredictorState *ps)
-{
-    ps->r0.mant   = 0;
-    ps->r0.exp   = 0;
-    ps->r1.mant   = 0;
-    ps->r1.exp   = 0;
-    ps->cor0.mant = 0;
-    ps->cor0.exp = 0;
-    ps->cor1.mant = 0;
-    ps->cor1.exp = 0;
-    ps->var0.mant = 0x20000000;
-    ps->var0.exp = 1;
-    ps->var1.mant = 0x20000000;
-    ps->var1.exp = 1;
-}
-
-static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) };  // 2^0, 2^0.25, 2^0.5, 2^0.75
-
-static inline int *DEC_SPAIR(int *dst, unsigned idx)
-{
-    dst[0] = (idx & 15) - 4;
-    dst[1] = (idx >> 4 & 15) - 4;
-
-    return dst + 2;
-}
-
-static inline int *DEC_SQUAD(int *dst, unsigned idx)
-{
-    dst[0] = (idx & 3) - 1;
-    dst[1] = (idx >> 2 & 3) - 1;
-    dst[2] = (idx >> 4 & 3) - 1;
-    dst[3] = (idx >> 6 & 3) - 1;
-
-    return dst + 4;
-}
-
-static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
-{
-    dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
-    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) << 1));
-
-    return dst + 2;
-}
-
-static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
-{
-    unsigned nz = idx >> 12;
-
-    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) << 1));
-    sign <<= nz & 1;
-    nz >>= 1;
-    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) << 1));
-    sign <<= nz & 1;
-    nz >>= 1;
-    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) << 1));
-    sign <<= nz & 1;
-    nz >>= 1;
-    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) << 1));
-
-    return dst + 4;
-}
-
-static void vector_pow43(int *coefs, int len)
-{
-    int i, coef;
-
-    for (i=0; i<len; i++) {
-        coef = coefs[i];
-        if (coef < 0)
-            coef = -(int)cbrt_tab[-coef];
-        else
-            coef = (int)cbrt_tab[coef];
-        coefs[i] = coef;
-    }
-}
-
-static void subband_scale(int *dst, int *src, int scale, int offset, int len)
-{
-    int ssign = scale < 0 ? -1 : 1;
-    int s = FFABS(scale);
-    unsigned int round;
-    int i, out, c = exp2tab[s & 3];
-
-    s = offset - (s >> 2);
-
-    if (s > 0) {
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)(((int64_t)src[i] * c) >> 32);
-            dst[i] = ((int)(out+round) >> s) * ssign;
-        }
-    }
-    else {
-        s = s + 32;
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
-            dst[i] = out * ssign;
-        }
-    }
-}
-
-static void noise_scale(int *coefs, int scale, int band_energy, int len)
-{
-    int ssign = scale < 0 ? -1 : 1;
-    int s = FFABS(scale);
-    unsigned int round;
-    int i, out, c = exp2tab[s & 3];
-    int nlz = 0;
-
-    while (band_energy > 0x7fff) {
-        band_energy >>= 1;
-        nlz++;
-    }
-    c /= band_energy;
-    s = 21 + nlz - (s >> 2);
-
-    if (s > 0) {
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)(((int64_t)coefs[i] * c) >> 32);
-            coefs[i] = ((int)(out+round) >> s) * ssign;
-        }
-    }
-    else {
-        s = s + 32;
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
-            coefs[i] = out * ssign;
-        }
-    }
-}
-
-static av_always_inline SoftFloat flt16_round(SoftFloat pf)
-{
-    SoftFloat tmp;
-    int s;
-
-    tmp.exp = pf.exp;
-    s = pf.mant >> 31;
-    tmp.mant = (pf.mant ^ s) - s;
-    tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
-    tmp.mant = (tmp.mant ^ s) - s;
-
-    return tmp;
-}
-
-static av_always_inline SoftFloat flt16_even(SoftFloat pf)
-{
-    SoftFloat tmp;
-    int s;
-
-    tmp.exp = pf.exp;
-    s = pf.mant >> 31;
-    tmp.mant = (pf.mant ^ s) - s;
-    tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
-    tmp.mant = (tmp.mant ^ s) - s;
-
-    return tmp;
-}
-
-static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
-{
-    SoftFloat pun;
-    int s;
-
-    pun.exp = pf.exp;
-    s = pf.mant >> 31;
-    pun.mant = (pf.mant ^ s) - s;
-    pun.mant = pun.mant & 0xFFC00000U;
-    pun.mant = (pun.mant ^ s) - s;
-
-    return pun;
-}
-
-static av_always_inline void predict(PredictorState *ps, int *coef,
-                                     int output_enable)
-{
-    const SoftFloat a     = { 1023410176, 0 };  // 61.0 / 64
-    const SoftFloat alpha = {  973078528, 0 };  // 29.0 / 32
-    SoftFloat e0, e1;
-    SoftFloat pv;
-    SoftFloat k1, k2;
-    SoftFloat   r0 = ps->r0,     r1 = ps->r1;
-    SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
-    SoftFloat var0 = ps->var0, var1 = ps->var1;
-    SoftFloat tmp;
-
-    if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
-        k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
-    }
-    else {
-        k1.mant = 0;
-        k1.exp = 0;
-    }
-
-    if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
-        k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
-    }
-    else {
-        k2.mant = 0;
-        k2.exp = 0;
-    }
-
-    tmp = av_mul_sf(k1, r0);
-    pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
-    if (output_enable) {
-        int shift = 28 - pv.exp;
-
-        if (shift < 31)
-            *coef += (pv.mant + (1 << (shift - 1))) >> shift;
-    }
-
-    e0 = av_int2sf(*coef, 2);
-    e1 = av_sub_sf(e0, tmp);
-
-    ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
-    tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
-    tmp.exp--;
-    ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
-    ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
-    tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
-    tmp.exp--;
-    ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
-
-    ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
-    ps->r0 = flt16_trunc(av_mul_sf(a, e0));
-}
-
-
-static const int cce_scale_fixed[8] = {
-    Q30(1.0),          //2^(0/8)
-    Q30(1.0905077327), //2^(1/8)
-    Q30(1.1892071150), //2^(2/8)
-    Q30(1.2968395547), //2^(3/8)
-    Q30(1.4142135624), //2^(4/8)
-    Q30(1.5422108254), //2^(5/8)
-    Q30(1.6817928305), //2^(6/8)
-    Q30(1.8340080864), //2^(7/8)
-};
-
-/**
- * Apply dependent channel coupling (applied before IMDCT).
- *
- * @param   index   index into coupling gain array
- */
-static void apply_dependent_coupling_fixed(AACContext *ac,
-                                     SingleChannelElement *target,
-                                     ChannelElement *cce, int index)
-{
-    IndividualChannelStream *ics = &cce->ch[0].ics;
-    const uint16_t *offsets = ics->swb_offset;
-    int *dest = target->coeffs;
-    const int *src = cce->ch[0].coeffs;
-    int g, i, group, k, idx = 0;
-    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
-        av_log(ac->avctx, AV_LOG_ERROR,
-               "Dependent coupling is not supported together with LTP\n");
-        return;
-    }
-    for (g = 0; g < ics->num_window_groups; g++) {
-        for (i = 0; i < ics->max_sfb; i++, idx++) {
-            if (cce->ch[0].band_type[idx] != ZERO_BT) {
-                const int gain = cce->coup.gain[index][idx];
-                int shift, round, c, tmp;
-
-                if (gain < 0) {
-                    c = -cce_scale_fixed[-gain & 7];
-                    shift = (-gain-1024) >> 3;
-                }
-                else {
-                    c = cce_scale_fixed[gain & 7];
-                    shift = (gain-1024) >> 3;
-                }
-
-                if (shift < 0) {
-                    shift = -shift;
-                    round = 1 << (shift - 1);
-
-                    for (group = 0; group < ics->group_len[g]; group++) {
-                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
-                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
-                                       (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += (tmp + round) >> shift;
-                        }
-                    }
-                }
-                else {
-                    for (group = 0; group < ics->group_len[g]; group++) {
-                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
-                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
-                                        (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += tmp << shift;
-                        }
-                    }
-                }
-            }
-        }
-        dest += ics->group_len[g] * 128;
-        src  += ics->group_len[g] * 128;
-    }
-}
-
-/**
- * Apply independent channel coupling (applied after IMDCT).
- *
- * @param   index   index into coupling gain array
- */
-static void apply_independent_coupling_fixed(AACContext *ac,
-                                       SingleChannelElement *target,
-                                       ChannelElement *cce, int index)
-{
-    int i, c, shift, round, tmp;
-    const int gain = cce->coup.gain[index][0];
-    const int *src = cce->ch[0].ret;
-    int *dest = target->ret;
-    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
-
-    c = cce_scale_fixed[gain & 7];
-    shift = (gain-1024) >> 3;
-    if (shift < 0) {
-        shift = -shift;
-        round = 1 << (shift - 1);
-
-        for (i = 0; i < len; i++) {
-            tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
-            dest[i] += (tmp + round) >> shift;
-        }
-    }
-    else {
-      for (i = 0; i < len; i++) {
-          tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
-          dest[i] += tmp << shift;
-      }
-    }
-}
-
-#include "aacdec_template.c"
-
-AVCodec ff_aac_fixed_decoder = {
-    .name            = "aac_fixed",
-    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
-    .type            = AVMEDIA_TYPE_AUDIO,
-    .id              = AV_CODEC_ID_AAC,
-    .priv_data_size  = sizeof(AACContext),
-    .init            = aac_decode_init,
-    .close           = aac_decode_close,
-    .decode          = aac_decode_frame,
-    .sample_fmts     = (const enum AVSampleFormat[]) {
-        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
-    },
-    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
-    .channel_layouts = aac_channel_layout,
-    .flush = flush,
-};
--- a/libavcodec/aacdec_template.c
+++ b/libavcodec/aacdec_template.c
--- a/libavcodec/aacdectab.h
+++ b/libavcodec/aacdectab.h
@@ -38,11 +38,46 @@
 /* @name ltp_coef
 * Table of the LTP coefficients
 */
-static const INTFLOAT ltp_coef[8] = {
-    Q30(0.570829f), Q30(0.696616f), Q30(0.813004f), Q30(0.911304f),
-    Q30(0.984900f), Q30(1.067894f), Q30(1.194601f), Q30(1.369533f),
+static const float ltp_coef[8] = {
+    0.570829, 0.696616, 0.813004, 0.911304,
+    0.984900, 1.067894, 1.194601, 1.369533,
 };

+/* @name tns_tmp2_map
+ * Tables of the tmp2[] arrays of LPC coefficients used for TNS.
+ * The suffix _M_N[] indicate the values of coef_compress and coef_res
+ * respectively.
+ * @{
+ */
+static const float tns_tmp2_map_1_3[4] = {
+     0.00000000, -0.43388373,  0.64278758,  0.34202015,
+};
+
+static const float tns_tmp2_map_0_3[8] = {
+     0.00000000, -0.43388373, -0.78183150, -0.97492790,
+     0.98480773,  0.86602539,  0.64278758,  0.34202015,
+};
+
+static const float tns_tmp2_map_1_4[8] = {
+     0.00000000, -0.20791170, -0.40673664, -0.58778524,
+     0.67369562,  0.52643216,  0.36124167,  0.18374951,
+};
+
+static const float tns_tmp2_map_0_4[16] = {
+     0.00000000, -0.20791170, -0.40673664, -0.58778524,
+    -0.74314481, -0.86602539, -0.95105654, -0.99452192,
+     0.99573416,  0.96182561,  0.89516330,  0.79801720,
+     0.67369562,  0.52643216,  0.36124167,  0.18374951,
+};
+
+static const float * const tns_tmp2_map[4] = {
+    tns_tmp2_map_0_3,
+    tns_tmp2_map_0_4,
+    tns_tmp2_map_1_3,
+    tns_tmp2_map_1_4
+};
+// @}
+
 static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 4, 5, 0, 5, 0 };

 static const uint8_t aac_channel_layout_map[16][5][3] = {
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -27,6 +27,7 @@
 /***********************************
 *              TODOs:
 * add sane pulse detection
+ * add temporal noise shaping
 ***********************************/

 #include "libavutil/float_dsp.h"
@@ -41,11 +42,127 @@
 #include "aac.h"
 #include "aactab.h"
 #include "aacenc.h"
-#include "aacenctab.h"
-#include "aacenc_utils.h"

 #include "psymodel.h"

+#define AAC_MAX_CHANNELS 6
+
+#define ERROR_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
+        return AVERROR(EINVAL); \
+    }
+
+#define WARN_IF(cond, ...) \
+    if (cond) { \
+        av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
+    }
+
+float ff_aac_pow34sf_tab[428];
+
+static const uint8_t swb_size_1024_96[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
+    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_64[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
+    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
+    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
+};
+
+static const uint8_t swb_size_1024_48[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
+    96
+};
+
+static const uint8_t swb_size_1024_32[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
+};
+
+static const uint8_t swb_size_1024_24[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
+    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_16[] = {
+    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
+    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
+    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
+};
+
+static const uint8_t swb_size_1024_8[] = {
+    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
+    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
+    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
+};
+
+static const uint8_t *swb_size_1024[] = {
+    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
+    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
+    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
+    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
+    swb_size_1024_8
+};
+
+static const uint8_t swb_size_128_96[] = {
+    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
+};
+
+static const uint8_t swb_size_128_48[] = {
+    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
+};
+
+static const uint8_t swb_size_128_24[] = {
+    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
+};
+
+static const uint8_t swb_size_128_16[] = {
+    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
+};
+
+static const uint8_t swb_size_128_8[] = {
+    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
+};
+
+static const uint8_t *swb_size_128[] = {
+    /* the last entry on the following row is swb_size_128_64 but is a
+       duplicate of swb_size_128_96 */
+    swb_size_128_96, swb_size_128_96, swb_size_128_96,
+    swb_size_128_48, swb_size_128_48, swb_size_128_48,
+    swb_size_128_24, swb_size_128_24, swb_size_128_16,
+    swb_size_128_16, swb_size_128_16, swb_size_128_8,
+    swb_size_128_8
+};
+
+/** default channel configurations */
+static const uint8_t aac_chan_configs[6][5] = {
+ {1, TYPE_SCE},                               // 1 channel  - single channel element
+ {1, TYPE_CPE},                               // 2 channels - channel pair
+ {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
+ {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
+ {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
+ {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
+};
+
+/**
+ * Table to remap channels from libavcodec's default order to AAC order.
+ */
+static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
+    { 0 },
+    { 0, 1 },
+    { 2, 0, 1 },
+    { 2, 0, 1, 3 },
+    { 2, 0, 1, 3, 4 },
+    { 2, 0, 1, 4, 5, 3 },
+};
+
 /**
 * Make AAC audio config object.
 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
@@ -56,7 +173,7 @@ static void put_audio_specific_config(AVCodecContext *avctx)
    AACEncContext *s = avctx->priv_data;

    init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
-    put_bits(&pb, 5, s->profile+1); //profile
+    put_bits(&pb, 5, 2); //object type - AAC-LC
    put_bits(&pb, 4, s->samplerate_index); //sample rate index
    put_bits(&pb, 4, s->channels);
    //GASpecificConfig
@@ -148,7 +265,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
        s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
    else
        for (i = 0; i < 1024; i += 128)
-            s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
+            s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
    memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
    memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
 }
@@ -166,7 +283,7 @@ static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
    put_bits(&s->pb, 1, info->use_kb_window[0]);
    if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
        put_bits(&s->pb, 6, info->max_sfb);
-        put_bits(&s->pb, 1, !!info->predictor_present);
+        put_bits(&s->pb, 1, 0);            // no prediction
    } else {
        put_bits(&s->pb, 4, info->max_sfb);
        for (w = 1; w < 8; w++)
@@ -195,14 +312,26 @@ static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
 static void adjust_frame_information(ChannelElement *cpe, int chans)
 {
    int i, w, w2, g, ch;
-    int maxsfb, cmaxsfb;
+    int start, maxsfb, cmaxsfb;

    for (ch = 0; ch < chans; ch++) {
        IndividualChannelStream *ics = &cpe->ch[ch].ics;
+        start = 0;
        maxsfb = 0;
        cpe->ch[ch].pulse.num_pulse = 0;
        for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-            for (w2 =  0; w2 < ics->group_len[w]; w2++) {
+            for (w2 = 0; w2 < ics->group_len[w]; w2++) {
+                start = (w+w2) * 128;
+                for (g = 0; g < ics->num_swb; g++) {
+                    //apply M/S
+                    if (cpe->common_window && !ch && cpe->ms_mask[w*16 + g]) {
+                        for (i = 0; i < ics->swb_sizes[g]; i++) {
+                            cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f;
+                            cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i];
+                        }
+                    }
+                    start += ics->swb_sizes[g];
+                }
                for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
                    ;
                maxsfb = FFMAX(maxsfb, cmaxsfb);
@@ -242,59 +371,6 @@ static void adjust_frame_information(ChannelElement *cpe, int chans)
    }
 }

-static void apply_intensity_stereo(ChannelElement *cpe)
-{
-    int w, w2, g, i;
-    IndividualChannelStream *ics = &cpe->ch[0].ics;
-    if (!cpe->common_window)
-        return;
-    for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-        for (w2 =  0; w2 < ics->group_len[w]; w2++) {
-            int start = (w+w2) * 128;
-            for (g = 0; g < ics->num_swb; g++) {
-                int p  = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
-                float scale = cpe->ch[0].is_ener[w*16+g];
-                if (!cpe->is_mask[w*16 + g]) {
-                    start += ics->swb_sizes[g];
-                    continue;
-                }
-                for (i = 0; i < ics->swb_sizes[g]; i++) {
-                    float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
-                    cpe->ch[0].coeffs[start+i] = sum;
-                    cpe->ch[1].coeffs[start+i] = 0.0f;
-                }
-                start += ics->swb_sizes[g];
-            }
-        }
-    }
-}
-
-static void apply_mid_side_stereo(ChannelElement *cpe)
-{
-    int w, w2, g, i;
-    IndividualChannelStream *ics = &cpe->ch[0].ics;
-    if (!cpe->common_window)
-        return;
-    for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-        for (w2 =  0; w2 < ics->group_len[w]; w2++) {
-            int start = (w+w2) * 128;
-            for (g = 0; g < ics->num_swb; g++) {
-                if (!cpe->ms_mask[w*16 + g]) {
-                    start += ics->swb_sizes[g];
-                    continue;
-                }
-                for (i = 0; i < ics->swb_sizes[g]; i++) {
-                    float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
-                    float R = L - cpe->ch[1].coeffs[start+i];
-                    cpe->ch[0].coeffs[start+i] = L;
-                    cpe->ch[1].coeffs[start+i] = R;
-                }
-                start += ics->swb_sizes[g];
-            }
-        }
-    }
-}
-
 /**
 * Encode scalefactor band coding type.
 */
@@ -302,9 +378,6 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 {
    int w;

-    if (s->coder->set_special_band_scalefactors)
-        s->coder->set_special_band_scalefactors(s, sce);
-
    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
        s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
 }
@@ -316,7 +389,7 @@ static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                                 SingleChannelElement *sce)
 {
    int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
-    int off_is = 0, noise_flag = 1;
+    int noise_flag = 1;
    int i, w;

    for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
@@ -329,10 +402,6 @@ static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                        put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
                        continue;
                    }
-                } else if (sce->band_type[w*16 + i] == INTENSITY_BT  ||
-                           sce->band_type[w*16 + i] == INTENSITY_BT2) {
-                    diff = sce->sf_idx[w*16 + i] - off_is;
-                    off_is = sce->sf_idx[w*16 + i];
                } else {
                    diff = sce->sf_idx[w*16 + i] - off_sf;
                    off_sf = sce->sf_idx[w*16 + i];
@@ -378,40 +447,17 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
                start += sce->ics.swb_sizes[i];
                continue;
            }
-            for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
-                s->coder->quantize_and_encode_band(s, &s->pb,
-                                                   &sce->coeffs[start + w2*128],
-                                                   NULL, sce->ics.swb_sizes[i],
+            for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
+                s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
+                                                   sce->ics.swb_sizes[i],
                                                   sce->sf_idx[w*16 + i],
                                                   sce->band_type[w*16 + i],
-                                                   s->lambda,
-                                                   sce->ics.window_clipping[w]);
-            }
+                                                   s->lambda);
            start += sce->ics.swb_sizes[i];
        }
    }
 }

-/**
- * Downscale spectral coefficients for near-clipping windows to avoid artifacts
- */
-static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
-{
-    int start, i, j, w;
-
-    if (sce->ics.clip_avoidance_factor < 1.0f) {
-        for (w = 0; w < sce->ics.num_windows; w++) {
-            start = 0;
-            for (i = 0; i < sce->ics.max_sfb; i++) {
-                float *swb_coeffs = &sce->coeffs[start + w*128];
-                for (j = 0; j < sce->ics.swb_sizes[i]; j++)
-                    swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
-                start += sce->ics.swb_sizes[i];
-            }
-        }
-    }
-}
-
 /**
 * Encode one channel of audio data.
 */
@@ -420,17 +466,12 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
                                     int common_window)
 {
    put_bits(&s->pb, 8, sce->sf_idx[0]);
-    if (!common_window) {
+    if (!common_window)
        put_ics_info(s, &sce->ics);
-        if (s->coder->encode_main_pred)
-            s->coder->encode_main_pred(s, sce);
-    }
    encode_band_info(s, sce);
    encode_scale_factors(avctx, s, sce);
    encode_pulses(s, &sce->pulse);
-    put_bits(&s->pb, 1, !!sce->tns.present);
-    if (s->coder->encode_tns_info)
-        s->coder->encode_tns_info(s, sce);
+    put_bits(&s->pb, 1, 0); //tns
    put_bits(&s->pb, 1, 0); //ssr
    encode_spectral_coeffs(s, sce);
    return 0;
@@ -488,9 +529,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
    AACEncContext *s = avctx->priv_data;
    float **samples = s->planar_samples, *samples2, *la, *overlap;
    ChannelElement *cpe;
-    SingleChannelElement *sce;
-    int i, ch, w, chans, tag, start_ch, ret;
-    int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
+    int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0;
    int chan_el_counter[4];
    FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

@@ -519,7 +558,6 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
        for (ch = 0; ch < chans; ch++) {
            IndividualChannelStream *ics = &cpe->ch[ch].ics;
            int cur_channel = start_ch + ch;
-            float clip_avoidance_factor;
            overlap  = &samples[cur_channel][0];
            samples2 = overlap + 1024;
            la       = samples2 + (448+64);
@@ -547,46 +585,25 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            ics->num_windows        = wi[ch].num_windows;
            ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
            ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
-            ics->swb_offset         = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
-                                        ff_swb_offset_128 [s->samplerate_index]:
-                                        ff_swb_offset_1024[s->samplerate_index];
-            ics->tns_max_bands      = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
-                                        ff_tns_max_bands_128 [s->samplerate_index]:
-                                        ff_tns_max_bands_1024[s->samplerate_index];
-            clip_avoidance_factor = 0.0f;
            for (w = 0; w < ics->num_windows; w++)
                ics->group_len[w] = wi[ch].grouping[w];
-            for (w = 0; w < ics->num_windows; w++) {
-                if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
-                    ics->window_clipping[w] = 1;
-                    clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
-                } else {
-                    ics->window_clipping[w] = 0;
-                }
-            }
-            if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
-                ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
-            } else {
-                ics->clip_avoidance_factor = 1.0f;
-            }

            apply_window_and_mdct(s, &cpe->ch[ch], overlap);
            if (isnan(cpe->ch->coeffs[0])) {
                av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
                return AVERROR(EINVAL);
            }
-            avoid_clipping(s, &cpe->ch[ch]);
        }
        start_ch += chans;
    }
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
+    if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
        return ret;
    do {
        int frame_bits;

        init_put_bits(&s->pb, avpkt->data, avpkt->size);

-        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
+        if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
            put_bitstream_info(s, LIBAVCODEC_IDENT);
        start_ch = 0;
        memset(chan_el_counter, 0, sizeof(chan_el_counter));
@@ -596,26 +613,16 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            tag      = s->chan_map[i+1];
            chans    = tag == TYPE_CPE ? 2 : 1;
            cpe      = &s->cpe[i];
-            cpe->common_window = 0;
-            memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
-            memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
            put_bits(&s->pb, 3, tag);
            put_bits(&s->pb, 4, chan_el_counter[tag]++);
-            for (ch = 0; ch < chans; ch++) {
-                sce = &cpe->ch[ch];
-                coeffs[ch] = sce->coeffs;
-                sce->ics.predictor_present = 0;
-                memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
-                memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
-                for (w = 0; w < 128; w++)
-                    if (sce->band_type[w] > RESERVED_BT)
-                        sce->band_type[w] = 0;
-            }
+            for (ch = 0; ch < chans; ch++)
+                coeffs[ch] = cpe->ch[ch].coeffs;
            s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
            for (ch = 0; ch < chans; ch++) {
                s->cur_channel = start_ch + ch;
                s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
            }
+            cpe->common_window = 0;
            if (chans > 1
                && wi[0].window_type[0] == wi[1].window_type[0]
                && wi[0].window_shape   == wi[1].window_shape) {
@@ -628,59 +635,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                    }
                }
            }
-            for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
-                sce = &cpe->ch[ch];
-                s->cur_channel = start_ch + ch;
-                if (s->options.pns && s->coder->search_for_pns)
-                    s->coder->search_for_pns(s, avctx, sce);
-                if (s->options.tns && s->coder->search_for_tns)
-                    s->coder->search_for_tns(s, sce);
-                if (s->options.tns && s->coder->apply_tns_filt)
-                    s->coder->apply_tns_filt(s, sce);
-                if (sce->tns.present)
-                    tns_mode = 1;
-            }
            s->cur_channel = start_ch;
-            if (s->options.intensity_stereo) { /* Intensity Stereo */
-                if (s->coder->search_for_is)
-                    s->coder->search_for_is(s, avctx, cpe);
-                if (cpe->is_mode) is_mode = 1;
-                apply_intensity_stereo(cpe);
-            }
-            if (s->options.pred) { /* Prediction */
-                for (ch = 0; ch < chans; ch++) {
-                    sce = &cpe->ch[ch];
-                    s->cur_channel = start_ch + ch;
-                    if (s->options.pred && s->coder->search_for_pred)
-                        s->coder->search_for_pred(s, sce);
-                    if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
+            if (s->options.stereo_mode && cpe->common_window) {
+                if (s->options.stereo_mode > 0) {
+                    IndividualChannelStream *ics = &cpe->ch[0].ics;
+                    for (w = 0; w < ics->num_windows; w += ics->group_len[w])
+                        for (g = 0;  g < ics->num_swb; g++)
+                            cpe->ms_mask[w*16+g] = 1;
+                } else if (s->coder->search_for_ms) {
+                    s->coder->search_for_ms(s, cpe, s->lambda);
                }
-                if (s->coder->adjust_common_prediction)
-                    s->coder->adjust_common_prediction(s, cpe);
-                for (ch = 0; ch < chans; ch++) {
-                    sce = &cpe->ch[ch];
-                    s->cur_channel = start_ch + ch;
-                    if (s->options.pred && s->coder->apply_main_pred)
-                        s->coder->apply_main_pred(s, sce);
-                }
-                s->cur_channel = start_ch;
-            }
-            if (s->options.stereo_mode) { /* Mid/Side stereo */
-                if (s->options.stereo_mode == -1 && s->coder->search_for_ms)
-                    s->coder->search_for_ms(s, cpe);
-                else if (cpe->common_window)
-                    memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
-                for (w = 0; w < 128; w++)
-                    cpe->ms_mask[w] = cpe->is_mask[w] ? 0 : cpe->ms_mask[w];
-                apply_mid_side_stereo(cpe);
            }
            adjust_frame_information(cpe, chans);
            if (chans == 2) {
                put_bits(&s->pb, 1, cpe->common_window);
                if (cpe->common_window) {
                    put_ics_info(s, &cpe->ch[0].ics);
-                    if (s->coder->encode_main_pred)
-                        s->coder->encode_main_pred(s, &cpe->ch[0]);
                    encode_ms_info(&s->pb, cpe);
                    if (cpe->ms_mode) ms_mode = 1;
                }
@@ -697,7 +667,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
            s->psy.bitres.bits = frame_bits / s->channels;
            break;
        }
-        if (is_mode || ms_mode || tns_mode || pred_mode) {
+        if (ms_mode) {
            for (i = 0; i < s->chan_map[0]; i++) {
                // Must restore coeffs
                chans = tag == TYPE_CPE ? 2 : 1;
@@ -716,7 +686,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
    avctx->frame_bits = put_bits_count(&s->pb);

    // rate control stuff
-    if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) {
+    if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
        float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
        s->lambda *= ratio;
        s->lambda = FFMIN(s->lambda, 65536.f);
@@ -740,7 +710,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
    ff_mdct_end(&s->mdct1024);
    ff_mdct_end(&s->mdct128);
    ff_psy_end(&s->psy);
-    ff_lpc_end(&s->lpc);
    if (s->psypp)
        ff_psy_preprocess_end(s->psypp);
    av_freep(&s->buffer.samples);
@@ -754,7 +723,7 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
    int ret = 0;

-    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
    if (!s->fdsp)
        return AVERROR(ENOMEM);

@@ -777,7 +746,7 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
    int ch;
    FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
    FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
-    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
+    FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);

    for(ch = 0; ch < s->channels; ch++)
        s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
@@ -803,29 +772,16 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)

    s->channels = avctx->channels;

-    ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
+    ERROR_IF(i == 16
+                || i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))
+                || i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),
             "Unsupported sample rate %d\n", avctx->sample_rate);
    ERROR_IF(s->channels > AAC_MAX_CHANNELS,
             "Unsupported number of channels: %d\n", s->channels);
+    ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
+             "Unsupported profile %d\n", avctx->profile);
    WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
             "Too many bits per frame requested, clamping to max\n");
-    if (avctx->profile == FF_PROFILE_AAC_MAIN) {
-        s->options.pred = 1;
-    } else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
-                avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
-        s->profile = 0; /* Main */
-        WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
-    } else if (avctx->profile == FF_PROFILE_AAC_LOW ||
-               avctx->profile == FF_PROFILE_UNKNOWN) {
-        s->profile = 1; /* Low */
-    } else {
-        ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
-    }
-
-    if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
-        s->options.intensity_stereo = 0;
-        s->options.pns = 0;
-    }

    avctx->bit_rate = (int)FFMIN(
        6144 * s->channels / 1024.0 * avctx->sample_rate,
@@ -844,8 +800,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
    avctx->extradata_size = 5;
    put_audio_specific_config(avctx);

-    sizes[0]   = ff_aac_swb_size_1024[i];
-    sizes[1]   = ff_aac_swb_size_128[i];
+    sizes[0]   = swb_size_1024[i];
+    sizes[1]   = swb_size_128[i];
    lengths[0] = ff_aac_num_swb_1024[i];
    lengths[1] = ff_aac_num_swb_128[i];
    for (i = 0; i < s->chan_map[0]; i++)
@@ -855,7 +811,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
        goto fail;
    s->psypp = ff_psy_preprocess_init(avctx);
    s->coder = &ff_aac_coders[s->options.aac_coder];
-    ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);

    if (HAVE_MIPSDSPR1)
        ff_aac_coder_init_mips(s);
@@ -864,6 +819,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)

    ff_aac_tableinit();

+    for (i = 0; i < 428; i++)
+        ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
+
    avctx->initial_padding = 1024;
    ff_af_queue_init(avctx, &s->afq);

@@ -879,23 +837,14 @@ static const AVOption aacenc_options[] = {
        {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
        {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
        {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
-    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
+    {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
        {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
        {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
        {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
        {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
-    {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "aac_pns"},
-        {"disable",  "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-        {"enable",   "Enable perceptual noise substitution",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
-    {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
-        {"disable",  "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
-        {"enable",   "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
-    {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_tns"},
-        {"disable",  "Disable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
-        {"enable",   "Enable temporal noise shaping", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_tns"},
-    {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pred"},
-        {"disable",  "Disable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
-        {"enable",   "Enable AAC-Main prediction", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pred"},
+    {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
+        {"disable",  "Disable PNS", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
+        {"enable",   "Enable PNS (Proof of concept)",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
    {NULL}
 };

@@ -906,6 +855,13 @@ static const AVClass aacenc_class = {
    LIBAVUTIL_VERSION_INT,
 };

+/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
+ * failures */
+static const int mpeg4audio_sample_rates[16] = {
+    96000, 88200, 64000, 48000, 44100, 32000,
+    24000, 22050, 16000, 12000, 11025, 8000, 7350
+};
+
 AVCodec ff_aac_encoder = {
    .name           = "aac",
    .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
@@ -916,8 +872,8 @@ AVCodec ff_aac_encoder = {
    .encode2        = aac_encode_frame,
    .close          = aac_encode_end,
    .supported_samplerates = mpeg4audio_sample_rates,
-    .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
-                      AV_CODEC_CAP_EXPERIMENTAL,
+    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
+                      CODEC_CAP_EXPERIMENTAL,
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
                                                     AV_SAMPLE_FMT_NONE },
    .priv_class     = &aacenc_class,
--- a/libavcodec/aacenc.h
+++ b/libavcodec/aacenc.h
@@ -30,8 +30,6 @@
 #include "audio_frame_queue.h"
 #include "psymodel.h"

-#include "lpc.h"
-
 typedef enum AACCoder {
    AAC_CODER_FAAC = 0,
    AAC_CODER_ANMR,
@@ -45,9 +43,6 @@ typedef struct AACEncOptions {
    int stereo_mode;
    int aac_coder;
    int pns;
-    int tns;
-    int pred;
-    int intensity_stereo;
 } AACEncOptions;

 struct AACEncContext;
@@ -57,19 +52,9 @@ typedef struct AACCoefficientsEncoder {
                                  SingleChannelElement *sce, const float lambda);
    void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
                                     int win, int group_len, const float lambda);
-    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
-                                     int scale_idx, int cb, const float lambda, int rtz);
-    void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*adjust_common_prediction)(struct AACEncContext *s, ChannelElement *cpe);
-    void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
-    void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
-    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
-    void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
-    void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
+    void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
+                                     int scale_idx, int cb, const float lambda);
+    void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
 } AACCoefficientsEncoder;

 extern AACCoefficientsEncoder ff_aac_coders[];
@@ -86,8 +71,6 @@ typedef struct AACEncContext {
    AVFloatDSPContext *fdsp;
    float *planar_samples[6];                    ///< saved preprocessed input

-    int profile;                                 ///< copied from avctx
-    LPCContext lpc;                              ///< used by TNS
    int samplerate_index;                        ///< MPEG-4 samplerate index
    int channels;                                ///< channel count
    const uint8_t *chan_map;                     ///< channel configuration map
@@ -108,6 +91,8 @@ typedef struct AACEncContext {
    } buffer;
 } AACEncContext;

+extern float ff_aac_pow34sf_tab[428];
+
 void ff_aac_coder_init_mips(AACEncContext *c);

 #endif /* AVCODEC_AACENC_H */
--- a/libavcodec/aacenc_is.c
+++ b/libavcodec/aacenc_is.c
@@ -1,136 +0,0 @@
-/*
- * AAC encoder intensity stereo
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder Intensity Stereo
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#include "aacenc.h"
-#include "aacenc_utils.h"
-#include "aacenc_is.h"
-#include "aacenc_quantization.h"
-
-struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
-                                         int start, int w, int g, float ener0,
-                                         float ener1, float ener01,
-                                         int use_pcoeffs, int phase)
-{
-    int i, w2;
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    float *L = use_pcoeffs ? sce0->pcoeffs : sce0->coeffs;
-    float *R = use_pcoeffs ? sce1->pcoeffs : sce1->coeffs;
-    float *L34 = &s->scoefs[256*0], *R34 = &s->scoefs[256*1];
-    float *IS  = &s->scoefs[256*2], *I34 = &s->scoefs[256*3];
-    float dist1 = 0.0f, dist2 = 0.0f;
-    struct AACISError is_error = {0};
-
-    for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
-        FFPsyBand *band0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-        FFPsyBand *band1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
-        int is_band_type, is_sf_idx = FFMAX(1, sce0->sf_idx[(w+w2)*16+g]-4);
-        float e01_34 = phase*pow(ener1/ener0, 3.0/4.0);
-        float maxval, dist_spec_err = 0.0f;
-        float minthr = FFMIN(band0->threshold, band1->threshold);
-        for (i = 0; i < sce0->ics.swb_sizes[g]; i++)
-            IS[i] = (L[start+(w+w2)*128+i] + phase*R[start+(w+w2)*128+i])*sqrt(ener0/ener01);
-        abs_pow34_v(L34, &L[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
-        abs_pow34_v(R34, &R[start+(w+w2)*128], sce0->ics.swb_sizes[g]);
-        abs_pow34_v(I34, IS,                   sce0->ics.swb_sizes[g]);
-        maxval = find_max_val(1, sce0->ics.swb_sizes[g], I34);
-        is_band_type = find_min_book(maxval, is_sf_idx);
-        dist1 += quantize_band_cost(s, &L[start + (w+w2)*128], L34,
-                                    sce0->ics.swb_sizes[g],
-                                    sce0->sf_idx[(w+w2)*16+g],
-                                    sce0->band_type[(w+w2)*16+g],
-                                    s->lambda / band0->threshold, INFINITY, NULL, 0);
-        dist1 += quantize_band_cost(s, &R[start + (w+w2)*128], R34,
-                                    sce1->ics.swb_sizes[g],
-                                    sce1->sf_idx[(w+w2)*16+g],
-                                    sce1->band_type[(w+w2)*16+g],
-                                    s->lambda / band1->threshold, INFINITY, NULL, 0);
-        dist2 += quantize_band_cost(s, IS, I34, sce0->ics.swb_sizes[g],
-                                    is_sf_idx, is_band_type,
-                                    s->lambda / minthr, INFINITY, NULL, 0);
-        for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-            dist_spec_err += (L34[i] - I34[i])*(L34[i] - I34[i]);
-            dist_spec_err += (R34[i] - I34[i]*e01_34)*(R34[i] - I34[i]*e01_34);
-        }
-        dist_spec_err *= s->lambda / minthr;
-        dist2 += dist_spec_err;
-    }
-
-    is_error.pass = dist2 <= dist1;
-    is_error.phase = phase;
-    is_error.error = fabsf(dist1 - dist2);
-    is_error.dist1 = dist1;
-    is_error.dist2 = dist2;
-
-    return is_error;
-}
-
-void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
-{
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    int start = 0, count = 0, w, w2, g, i;
-    const float freq_mult = avctx->sample_rate/(1024.0f/sce0->ics.num_windows)/2.0f;
-
-    if (!cpe->common_window)
-        return;
-
-    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
-        start = 0;
-        for (g = 0;  g < sce0->ics.num_swb; g++) {
-            if (start*freq_mult > INT_STEREO_LOW_LIMIT*(s->lambda/170.0f) &&
-                cpe->ch[0].band_type[w*16+g] != NOISE_BT && !cpe->ch[0].zeroes[w*16+g] &&
-                cpe->ch[1].band_type[w*16+g] != NOISE_BT && !cpe->ch[1].zeroes[w*16+g]) {
-                float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
-                struct AACISError ph_err1, ph_err2, *erf;
-                for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
-                    for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                        float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
-                        float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
-                        ener0  += coef0*coef0;
-                        ener1  += coef1*coef1;
-                        ener01 += (coef0 + coef1)*(coef0 + coef1);
-                    }
-                }
-                ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                                 ener0, ener1, ener01, 0, -1);
-                ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                                 ener0, ener1, ener01, 0, +1);
-                erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
-                if (erf->pass) {
-                    cpe->is_mask[w*16+g] = 1;
-                    cpe->ch[0].is_ener[w*16+g] = sqrt(ener0/ener01);
-                    cpe->ch[1].is_ener[w*16+g] = ener0/ener1;
-                    cpe->ch[1].band_type[w*16+g] = erf->phase ? INTENSITY_BT : INTENSITY_BT2;
-                    count++;
-                }
-            }
-            start += sce0->ics.swb_sizes[g];
-        }
-    }
-    cpe->is_mode = !!count;
-}
--- a/libavcodec/aacenc_is.h
+++ b/libavcodec/aacenc_is.h
@@ -1,50 +0,0 @@
-/*
- * AAC encoder intensity stereo
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder Intensity Stereo
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_IS_H
-#define AVCODEC_AACENC_IS_H
-
-#include "aacenc.h"
-
-/** Frequency in Hz for lower limit of intensity stereo **/
-#define INT_STEREO_LOW_LIMIT 6100
-
-struct AACISError {
-    int pass;    /* 1 if dist2 <= dist1  */
-    int phase;   /* -1 or +1             */
-    float error; /* fabs(dist1 - dist2)  */
-    float dist1; /* From original coeffs */
-    float dist2; /* From IS'd coeffs     */
-};
-
-struct AACISError ff_aac_is_encoding_err(AACEncContext *s, ChannelElement *cpe,
-                                         int start, int w, int g, float ener0,
-                                         float ener1, float ener01,
-                                         int use_pcoeffs, int phase);
-void ff_aac_search_for_is(AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
-
-#endif /* AVCODEC_AACENC_IS_H */
--- a/libavcodec/aacenc_pred.c
+++ b/libavcodec/aacenc_pred.c
@@ -1,342 +0,0 @@
-/*
- * AAC encoder main-type prediction
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder Intensity Stereo
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#include "aactab.h"
-#include "aacenc_pred.h"
-#include "aacenc_utils.h"
-#include "aacenc_is.h"            /* <- Needed for common window distortions */
-#include "aacenc_quantization.h"
-
-#define RESTORE_PRED(sce, sfb) \
-        if (sce->ics.prediction_used[sfb]) {\
-            sce->ics.prediction_used[sfb] = 0;\
-            sce->band_type[sfb] = sce->band_alt[sfb];\
-        }
-
-static inline float flt16_round(float pf)
-{
-    union av_intfloat32 tmp;
-    tmp.f = pf;
-    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
-    return tmp.f;
-}
-
-static inline float flt16_even(float pf)
-{
-    union av_intfloat32 tmp;
-    tmp.f = pf;
-    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
-    return tmp.f;
-}
-
-static inline float flt16_trunc(float pf)
-{
-    union av_intfloat32 pun;
-    pun.f = pf;
-    pun.i &= 0xFFFF0000U;
-    return pun.f;
-}
-
-static inline void predict(PredictorState *ps, float *coef, float *rcoef, int set)
-{
-    float k2;
-    const float a     = 0.953125; // 61.0 / 64
-    const float alpha = 0.90625;  // 29.0 / 32
-    const float   k1 = ps->k1;
-    const float   r0 = ps->r0,     r1 = ps->r1;
-    const float cor0 = ps->cor0, cor1 = ps->cor1;
-    const float var0 = ps->var0, var1 = ps->var1;
-    const float e0 = *coef - ps->x_est;
-    const float e1 = e0 - k1 * r0;
-
-    if (set)
-        *coef = e0;
-
-    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
-    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
-    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
-    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
-    ps->r1   = flt16_trunc(a * (r0 - k1 * e0));
-    ps->r0   = flt16_trunc(a * e0);
-
-    /* Prediction for next frame */
-    ps->k1   = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
-    k2       = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
-    *rcoef   = ps->x_est = flt16_round(ps->k1*ps->r0 + k2*ps->r1);
-}
-
-static inline void reset_predict_state(PredictorState *ps)
-{
-    ps->r0    = 0.0f;
-    ps->r1    = 0.0f;
-    ps->k1    = 0.0f;
-    ps->cor0  = 0.0f;
-    ps->cor1  = 0.0f;
-    ps->var0  = 1.0f;
-    ps->var1  = 1.0f;
-    ps->x_est = 0.0f;
-}
-
-static inline void reset_all_predictors(PredictorState *ps)
-{
-    int i;
-    for (i = 0; i < MAX_PREDICTORS; i++)
-        reset_predict_state(&ps[i]);
-}
-
-static inline void reset_predictor_group(SingleChannelElement *sce, int group_num)
-{
-    int i;
-    PredictorState *ps = sce->predictor_state;
-    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
-        reset_predict_state(&ps[i]);
-}
-
-void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce)
-{
-    int sfb, k;
-    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-
-    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
-        for (sfb = 0; sfb < pmax; sfb++) {
-            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
-                predict(&sce->predictor_state[k], &sce->coeffs[k], &sce->prcoeffs[k],
-                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
-            }
-        }
-        if (sce->ics.predictor_reset_group) {
-            reset_predictor_group(sce, sce->ics.predictor_reset_group);
-        }
-    } else {
-        reset_all_predictors(sce->predictor_state);
-    }
-}
-
-/* If inc = 0 you can check if this returns 0 to see if you can reset freely */
-static inline int update_counters(IndividualChannelStream *ics, int inc)
-{
-    int i;
-    for (i = 1; i < 31; i++) {
-        ics->predictor_reset_count[i] += inc;
-        if (ics->predictor_reset_count[i] > PRED_RESET_FRAME_MIN)
-            return i; /* Reset this immediately */
-    }
-    return 0;
-}
-
-void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe)
-{
-    int start, w, w2, g, i, count = 0;
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    const int pmax0 = FFMIN(sce0->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-    const int pmax1 = FFMIN(sce1->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-    const int pmax  = FFMIN(pmax0, pmax1);
-
-    if (!cpe->common_window ||
-        sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
-        sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
-        return;
-
-    for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
-        start = 0;
-        for (g = 0; g < sce0->ics.num_swb; g++) {
-            int sfb = w*16+g;
-            int sum = sce0->ics.prediction_used[sfb] + sce1->ics.prediction_used[sfb];
-            float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
-            struct AACISError ph_err1, ph_err2, *erf;
-            if (sfb < PRED_SFB_START || sfb > pmax || sum != 2) {
-                RESTORE_PRED(sce0, sfb);
-                RESTORE_PRED(sce1, sfb);
-                start += sce0->ics.swb_sizes[g];
-                continue;
-            }
-            for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
-                for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
-                    float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
-                    float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
-                    ener0  += coef0*coef0;
-                    ener1  += coef1*coef1;
-                    ener01 += (coef0 + coef1)*(coef0 + coef1);
-                }
-            }
-            ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                             ener0, ener1, ener01, 1, -1);
-            ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
-                                             ener0, ener1, ener01, 1, +1);
-            erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
-            if (erf->pass) {
-                sce0->ics.prediction_used[sfb] = 1;
-                sce1->ics.prediction_used[sfb] = 1;
-                count++;
-            } else {
-                RESTORE_PRED(sce0, sfb);
-                RESTORE_PRED(sce1, sfb);
-            }
-            start += sce0->ics.swb_sizes[g];
-        }
-    }
-
-    sce1->ics.predictor_present = sce0->ics.predictor_present = !!count;
-}
-
-static void update_pred_resets(SingleChannelElement *sce)
-{
-    int i, max_group_id_c, max_frame = 0;
-    float avg_frame = 0.0f;
-    IndividualChannelStream *ics = &sce->ics;
-
-    /* Update the counters and immediately update any frame behind schedule */
-    if ((ics->predictor_reset_group = update_counters(&sce->ics, 1)))
-        return;
-
-    for (i = 1; i < 31; i++) {
-        /* Count-based */
-        if (ics->predictor_reset_count[i] > max_frame) {
-            max_group_id_c = i;
-            max_frame = ics->predictor_reset_count[i];
-        }
-        avg_frame = (ics->predictor_reset_count[i] + avg_frame)/2;
-    }
-
-    if (max_frame > PRED_RESET_MIN) {
-        ics->predictor_reset_group = max_group_id_c;
-    } else {
-        ics->predictor_reset_group = 0;
-    }
-}
-
-void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce)
-{
-    int sfb, i, count = 0, cost_coeffs = 0, cost_pred = 0;
-    const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-    float *O34  = &s->scoefs[128*0], *P34 = &s->scoefs[128*1];
-    float *SENT = &s->scoefs[128*2], *S34 = &s->scoefs[128*3];
-    float *QERR = &s->scoefs[128*4];
-
-    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-        sce->ics.predictor_present = 0;
-        return;
-    }
-
-    if (!sce->ics.predictor_initialized) {
-        reset_all_predictors(sce->predictor_state);
-        sce->ics.predictor_initialized = 1;
-        memcpy(sce->prcoeffs, sce->coeffs, 1024*sizeof(float));
-        for (i = 1; i < 31; i++)
-            sce->ics.predictor_reset_count[i] = i;
-    }
-
-    update_pred_resets(sce);
-    memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
-
-    for (sfb = PRED_SFB_START; sfb < pmax; sfb++) {
-        int cost1, cost2, cb_p;
-        float dist1, dist2, dist_spec_err = 0.0f;
-        const int cb_n = sce->band_type[sfb];
-        const int start_coef = sce->ics.swb_offset[sfb];
-        const int num_coeffs = sce->ics.swb_offset[sfb + 1] - start_coef;
-        const FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[sfb];
-
-        if (start_coef + num_coeffs > MAX_PREDICTORS)
-            continue;
-
-        /* Normal coefficients */
-        abs_pow34_v(O34, &sce->coeffs[start_coef], num_coeffs);
-        dist1 = quantize_and_encode_band_cost(s, NULL, &sce->coeffs[start_coef], NULL,
-                                              O34, num_coeffs, sce->sf_idx[sfb],
-                                              cb_n, s->lambda / band->threshold, INFINITY, &cost1, 0);
-        cost_coeffs += cost1;
-
-        /* Encoded coefficients - needed for #bits, band type and quant. error */
-        for (i = 0; i < num_coeffs; i++)
-            SENT[i] = sce->coeffs[start_coef + i] - sce->prcoeffs[start_coef + i];
-        abs_pow34_v(S34, SENT, num_coeffs);
-        if (cb_n < RESERVED_BT)
-            cb_p = find_min_book(find_max_val(1, num_coeffs, S34), sce->sf_idx[sfb]);
-        else
-            cb_p = cb_n;
-        quantize_and_encode_band_cost(s, NULL, SENT, QERR, S34, num_coeffs,
-                                      sce->sf_idx[sfb], cb_p, s->lambda / band->threshold, INFINITY,
-                                      &cost2, 0);
-
-        /* Reconstructed coefficients - needed for distortion measurements */
-        for (i = 0; i < num_coeffs; i++)
-            sce->prcoeffs[start_coef + i] += QERR[i] != 0.0f ? (sce->prcoeffs[start_coef + i] - QERR[i]) : 0.0f;
-        abs_pow34_v(P34, &sce->prcoeffs[start_coef], num_coeffs);
-        if (cb_n < RESERVED_BT)
-            cb_p = find_min_book(find_max_val(1, num_coeffs, P34), sce->sf_idx[sfb]);
-        else
-            cb_p = cb_n;
-        dist2 = quantize_and_encode_band_cost(s, NULL, &sce->prcoeffs[start_coef], NULL,
-                                              P34, num_coeffs, sce->sf_idx[sfb],
-                                              cb_p, s->lambda / band->threshold, INFINITY, NULL, 0);
-        for (i = 0; i < num_coeffs; i++)
-            dist_spec_err += (O34[i] - P34[i])*(O34[i] - P34[i]);
-        dist_spec_err *= s->lambda / band->threshold;
-        dist2 += dist_spec_err;
-
-        if (dist2 <= dist1 && cb_p <= cb_n) {
-            cost_pred += cost2;
-            sce->ics.prediction_used[sfb] = 1;
-            sce->band_alt[sfb]  = cb_n;
-            sce->band_type[sfb] = cb_p;
-            count++;
-        } else {
-            cost_pred += cost1;
-            sce->band_alt[sfb] = cb_p;
-        }
-    }
-
-    if (count && cost_coeffs < cost_pred) {
-        count = 0;
-        for (sfb = PRED_SFB_START; sfb < pmax; sfb++)
-            RESTORE_PRED(sce, sfb);
-        memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
-    }
-
-    sce->ics.predictor_present = !!count;
-}
-
-/**
- * Encoder predictors data.
- */
-void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce)
-{
-    int sfb;
-    IndividualChannelStream *ics = &sce->ics;
-    const int pmax = FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
-
-    if (!ics->predictor_present)
-        return;
-
-    put_bits(&s->pb, 1, !!ics->predictor_reset_group);
-    if (ics->predictor_reset_group)
-        put_bits(&s->pb, 5, ics->predictor_reset_group);
-    for (sfb = 0; sfb < pmax; sfb++)
-        put_bits(&s->pb, 1, ics->prediction_used[sfb]);
-}
--- a/libavcodec/aacenc_pred.h
+++ b/libavcodec/aacenc_pred.h
@@ -1,47 +0,0 @@
-/*
- * AAC encoder main-type prediction
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder main prediction
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_PRED_H
-#define AVCODEC_AACENC_PRED_H
-
-#include "aacenc.h"
-
-/* Every predictor group needs to get reset at least once in this many frames */
-#define PRED_RESET_FRAME_MIN 240
-
-/* Any frame with less than this amount of frames since last reset is ok */
-#define PRED_RESET_MIN 64
-
-/* Raise to filter any low frequency artifacts due to prediction */
-#define PRED_SFB_START 10
-
-void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce);
-void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe);
-void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce);
-void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce);
-
-#endif /* AVCODEC_AACENC_PRED_H */
--- a/libavcodec/aacenc_quantization.h
+++ b/libavcodec/aacenc_quantization.h
@@ -1,260 +0,0 @@
-/*
- * AAC encoder intensity stereo
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder quantizer
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_QUANTIZATION_H
-#define AVCODEC_AACENC_QUANTIZATION_H
-
-#include "aactab.h"
-#include "aacenc.h"
-#include "aacenctab.h"
-#include "aacenc_utils.h"
-
-/**
- * Calculate rate distortion cost for quantizing with given codebook
- *
- * @return quantization distortion
- */
-static av_always_inline float quantize_and_encode_band_cost_template(
-                                struct AACEncContext *s,
-                                PutBitContext *pb, const float *in, float *out,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits, int BT_ZERO, int BT_UNSIGNED,
-                                int BT_PAIR, int BT_ESC, int BT_NOISE, int BT_STEREO,
-                                const float ROUNDING)
-{
-    const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512;
-    const float Q   = ff_aac_pow2sf_tab [q_idx];
-    const float Q34 = ff_aac_pow34sf_tab[q_idx];
-    const float IQ  = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
-    const float CLIPPED_ESCAPE = 165140.0f*IQ;
-    int i, j;
-    float cost = 0;
-    const int dim = BT_PAIR ? 2 : 4;
-    int resbits = 0;
-    int off;
-
-    if (BT_ZERO || BT_NOISE || BT_STEREO) {
-        for (i = 0; i < size; i++)
-            cost += in[i]*in[i];
-        if (bits)
-            *bits = 0;
-        if (out) {
-            for (i = 0; i < size; i += dim)
-                for (j = 0; j < dim; j++)
-                    out[i+j] = 0.0f;
-        }
-        return cost * lambda;
-    }
-    if (!scaled) {
-        abs_pow34_v(s->scoefs, in, size);
-        scaled = s->scoefs;
-    }
-    quantize_bands(s->qcoefs, in, scaled, size, Q34, !BT_UNSIGNED, aac_cb_maxval[cb], ROUNDING);
-    if (BT_UNSIGNED) {
-        off = 0;
-    } else {
-        off = aac_cb_maxval[cb];
-    }
-    for (i = 0; i < size; i += dim) {
-        const float *vec;
-        int *quants = s->qcoefs + i;
-        int curidx = 0;
-        int curbits;
-        float quantized, rd = 0.0f;
-        for (j = 0; j < dim; j++) {
-            curidx *= aac_cb_range[cb];
-            curidx += quants[j] + off;
-        }
-        curbits =  ff_aac_spectral_bits[cb-1][curidx];
-        vec     = &ff_aac_codebook_vectors[cb-1][curidx*dim];
-        if (BT_UNSIGNED) {
-            for (j = 0; j < dim; j++) {
-                float t = fabsf(in[i+j]);
-                float di;
-                if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow
-                    if (t >= CLIPPED_ESCAPE) {
-                        quantized = CLIPPED_ESCAPE;
-                        curbits += 21;
-                    } else {
-                        int c = av_clip_uintp2(quant(t, Q, ROUNDING), 13);
-                        quantized = c*cbrtf(c)*IQ;
-                        curbits += av_log2(c)*2 - 4 + 1;
-                    }
-                } else {
-                    quantized = vec[j]*IQ;
-                }
-                di = t - quantized;
-                if (out)
-                    out[i+j] = in[i+j] >= 0 ? quantized : -quantized;
-                if (vec[j] != 0.0f)
-                    curbits++;
-                rd += di*di;
-            }
-        } else {
-            for (j = 0; j < dim; j++) {
-                quantized = vec[j]*IQ;
-                if (out)
-                    out[i+j] = quantized;
-                rd += (in[i+j] - quantized)*(in[i+j] - quantized);
-            }
-        }
-        cost    += rd * lambda + curbits;
-        resbits += curbits;
-        if (cost >= uplim)
-            return uplim;
-        if (pb) {
-            put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]);
-            if (BT_UNSIGNED)
-                for (j = 0; j < dim; j++)
-                    if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f)
-                        put_bits(pb, 1, in[i+j] < 0.0f);
-            if (BT_ESC) {
-                for (j = 0; j < 2; j++) {
-                    if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) {
-                        int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q, ROUNDING), 13);
-                        int len = av_log2(coef);
-
-                        put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
-                        put_sbits(pb, len, coef);
-                    }
-                }
-            }
-        }
-    }
-
-    if (bits)
-        *bits = resbits;
-    return cost;
-}
-
-static inline float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb,
-                                                const float *in, float *quant, const float *scaled,
-                                                int size, int scale_idx, int cb,
-                                                const float lambda, const float uplim,
-                                                int *bits) {
-    av_assert0(0);
-    return 0.0f;
-}
-
-#define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, ROUNDING) \
-static float quantize_and_encode_band_cost_ ## NAME(                                         \
-                                struct AACEncContext *s,                                     \
-                                PutBitContext *pb, const float *in, float *quant,            \
-                                const float *scaled, int size, int scale_idx,                \
-                                int cb, const float lambda, const float uplim,               \
-                                int *bits) {                                                 \
-    return quantize_and_encode_band_cost_template(                                           \
-                                s, pb, in, quant, scaled, size, scale_idx,                   \
-                                BT_ESC ? ESC_BT : cb, lambda, uplim, bits,                   \
-                                BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO,  \
-                                ROUNDING);                                                   \
-}
-
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO,  1, 0, 0, 0, 0, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC,   0, 1, 1, 1, 0, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC_RTZ, 0, 1, 1, 1, 0, 0, ROUND_TO_ZERO)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0, ROUND_STANDARD)
-QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1, ROUND_STANDARD)
-
-static float (*const quantize_and_encode_band_cost_arr[])(
-                                struct AACEncContext *s,
-                                PutBitContext *pb, const float *in, float *quant,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits) = {
-    quantize_and_encode_band_cost_ZERO,
-    quantize_and_encode_band_cost_SQUAD,
-    quantize_and_encode_band_cost_SQUAD,
-    quantize_and_encode_band_cost_UQUAD,
-    quantize_and_encode_band_cost_UQUAD,
-    quantize_and_encode_band_cost_SPAIR,
-    quantize_and_encode_band_cost_SPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_ESC,
-    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
-    quantize_and_encode_band_cost_NOISE,
-    quantize_and_encode_band_cost_STEREO,
-    quantize_and_encode_band_cost_STEREO,
-};
-
-static float (*const quantize_and_encode_band_cost_rtz_arr[])(
-                                struct AACEncContext *s,
-                                PutBitContext *pb, const float *in, float *quant,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits) = {
-    quantize_and_encode_band_cost_ZERO,
-    quantize_and_encode_band_cost_SQUAD,
-    quantize_and_encode_band_cost_SQUAD,
-    quantize_and_encode_band_cost_UQUAD,
-    quantize_and_encode_band_cost_UQUAD,
-    quantize_and_encode_band_cost_SPAIR,
-    quantize_and_encode_band_cost_SPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_UPAIR,
-    quantize_and_encode_band_cost_ESC_RTZ,
-    quantize_and_encode_band_cost_NONE,     /* CB 12 doesn't exist */
-    quantize_and_encode_band_cost_NOISE,
-    quantize_and_encode_band_cost_STEREO,
-    quantize_and_encode_band_cost_STEREO,
-};
-
-#define quantize_and_encode_band_cost(                                  \
-                                s, pb, in, quant, scaled, size, scale_idx, cb, \
-                                lambda, uplim, bits, rtz)               \
-    ((rtz) ? quantize_and_encode_band_cost_rtz_arr : quantize_and_encode_band_cost_arr)[cb]( \
-                                s, pb, in, quant, scaled, size, scale_idx, cb, \
-                                lambda, uplim, bits)
-
-static inline float quantize_band_cost(struct AACEncContext *s, const float *in,
-                                const float *scaled, int size, int scale_idx,
-                                int cb, const float lambda, const float uplim,
-                                int *bits, int rtz)
-{
-    return quantize_and_encode_band_cost(s, NULL, in, NULL, scaled, size, scale_idx,
-                                         cb, lambda, uplim, bits, rtz);
-}
-
-static inline void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb,
-                                            const float *in, float *out, int size, int scale_idx,
-                                            int cb, const float lambda, int rtz)
-{
-    quantize_and_encode_band_cost(s, pb, in, out, NULL, size, scale_idx, cb, lambda,
-                                  INFINITY, NULL, rtz);
-}
-
-#endif /* AVCODEC_AACENC_QUANTIZATION_H */
--- a/libavcodec/aacenc_tns.c
+++ b/libavcodec/aacenc_tns.c
@@ -1,194 +0,0 @@
-/*
- * AAC encoder TNS
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder temporal noise shaping
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#include "aacenc.h"
-#include "aacenc_tns.h"
-#include "aactab.h"
-#include "aacenc_utils.h"
-#include "aacenc_quantization.h"
-
-/**
- * Encode TNS data.
- * Coefficient compression saves a single bit per coefficient.
- */
-void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
-{
-    uint8_t u_coef;
-    const uint8_t coef_res = TNS_Q_BITS == 4;
-    int i, w, filt, coef_len, coef_compress = 0;
-    const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
-    TemporalNoiseShaping *tns = &sce->tns;
-
-    if (!sce->tns.present)
-        return;
-
-    for (i = 0; i < sce->ics.num_windows; i++) {
-        put_bits(&s->pb, 2 - is8, sce->tns.n_filt[i]);
-        if (tns->n_filt[i]) {
-            put_bits(&s->pb, 1, coef_res);
-            for (filt = 0; filt < tns->n_filt[i]; filt++) {
-                put_bits(&s->pb, 6 - 2 * is8, tns->length[i][filt]);
-                put_bits(&s->pb, 5 - 2 * is8, tns->order[i][filt]);
-                if (tns->order[i][filt]) {
-                    put_bits(&s->pb, 1, !!tns->direction[i][filt]);
-                    put_bits(&s->pb, 1, !!coef_compress);
-                    coef_len = coef_res + 3 - coef_compress;
-                    for (w = 0; w < tns->order[i][filt]; w++) {
-                        u_coef = (tns->coef_idx[i][filt][w])&(~(~0<<coef_len));
-                        put_bits(&s->pb, coef_len, u_coef);
-                    }
-                }
-            }
-        }
-    }
-}
-
-static inline void quantize_coefs(double *coef, int *idx, float *lpc, int order)
-{
-    int i;
-    uint8_t u_coef;
-    const float *quant_arr = tns_tmp2_map[TNS_Q_BITS == 4];
-    const double iqfac_p = ((1 << (TNS_Q_BITS-1)) - 0.5)/(M_PI/2.0);
-    const double iqfac_m = ((1 << (TNS_Q_BITS-1)) + 0.5)/(M_PI/2.0);
-    for (i = 0; i < order; i++) {
-        idx[i] = ceilf(asin(coef[i])*((coef[i] >= 0) ? iqfac_p : iqfac_m));
-        u_coef = (idx[i])&(~(~0<<TNS_Q_BITS));
-        lpc[i] = quant_arr[u_coef];
-    }
-}
-
-/* Apply TNS filter */
-void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
-{
-    TemporalNoiseShaping *tns = &sce->tns;
-    IndividualChannelStream *ics = &sce->ics;
-    int w, filt, m, i, top, order, bottom, start, end, size, inc;
-    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
-    float lpc[TNS_MAX_ORDER];
-
-    for (w = 0; w < ics->num_windows; w++) {
-        bottom = ics->num_swb;
-        for (filt = 0; filt < tns->n_filt[w]; filt++) {
-            top    = bottom;
-            bottom = FFMAX(0, top - tns->length[w][filt]);
-            order  = tns->order[w][filt];
-            if (order == 0)
-                continue;
-
-            // tns_decode_coef
-            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
-
-            start = ics->swb_offset[FFMIN(bottom, mmm)];
-            end   = ics->swb_offset[FFMIN(   top, mmm)];
-            if ((size = end - start) <= 0)
-                continue;
-            if (tns->direction[w][filt]) {
-                inc = -1;
-                start = end - 1;
-            } else {
-                inc = 1;
-            }
-            start += w * 128;
-
-            // ar filter
-            for (m = 0; m < size; m++, start += inc)
-                for (i = 1; i <= FFMIN(m, order); i++)
-                    sce->coeffs[start] += lpc[i-1]*sce->pcoeffs[start - i*inc];
-        }
-    }
-}
-
-void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
-{
-    TemporalNoiseShaping *tns = &sce->tns;
-    int w, w2, g, count = 0;
-    const int mmm = FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb);
-    const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
-    const int order = is8 ? 7 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
-
-    int sfb_start = av_clip(tns_min_sfb[is8][s->samplerate_index], 0, mmm);
-    int sfb_end   = av_clip(sce->ics.num_swb, 0, mmm);
-
-    for (w = 0; w < sce->ics.num_windows; w++) {
-        float e_ratio = 0.0f, threshold = 0.0f, spread = 0.0f, en[2] = {0.0, 0.0f};
-        double gain = 0.0f, coefs[MAX_LPC_ORDER] = {0};
-        int coef_start = w*sce->ics.num_swb + sce->ics.swb_offset[sfb_start];
-        int coef_len = sce->ics.swb_offset[sfb_end] - sce->ics.swb_offset[sfb_start];
-
-        for (g = 0;  g < sce->ics.num_swb; g++) {
-            if (w*16+g < sfb_start || w*16+g > sfb_end)
-                continue;
-            for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
-                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
-                if ((w+w2)*16+g > sfb_start + ((sfb_end - sfb_start)/2))
-                    en[1] += band->energy;
-                else
-                    en[0] += band->energy;
-                threshold += band->threshold;
-                spread += band->spread;
-            }
-        }
-
-        if (coef_len <= 0 || (sfb_end - sfb_start) <= 0)
-            continue;
-        else
-            e_ratio = en[0]/en[1];
-
-        /* LPC */
-        gain = ff_lpc_calc_ref_coefs_f(&s->lpc, &sce->coeffs[coef_start],
-                                       coef_len, order, coefs);
-
-        if (gain > TNS_GAIN_THRESHOLD_LOW && gain < TNS_GAIN_THRESHOLD_HIGH &&
-            (en[0]+en[1]) > TNS_GAIN_THRESHOLD_LOW*threshold &&
-            spread < TNS_SPREAD_THRESHOLD && order) {
-            if (is8 || order < 2 || (e_ratio > TNS_E_RATIO_LOW && e_ratio < TNS_E_RATIO_HIGH)) {
-                tns->n_filt[w] = 1;
-                for (g = 0; g < tns->n_filt[w]; g++) {
-                    tns->length[w][g] = sfb_end - sfb_start;
-                    tns->direction[w][g] = en[0] < en[1];
-                    tns->order[w][g] = order;
-                    quantize_coefs(coefs, tns->coef_idx[w][g], tns->coef[w][g],
-                                   order);
-                }
-            } else {  /* 2 filters due to energy disbalance */
-                tns->n_filt[w] = 2;
-                for (g = 0; g < tns->n_filt[w]; g++) {
-                    tns->direction[w][g] = en[g] < en[!g];
-                    tns->order[w][g] = !g ? order/2 : order - tns->order[w][g-1];
-                    tns->length[w][g] = !g ? (sfb_end - sfb_start)/2 : \
-                                    (sfb_end - sfb_start) - tns->length[w][g-1];
-                    quantize_coefs(&coefs[!g ? 0 : order - tns->order[w][g-1]],
-                                   tns->coef_idx[w][g], tns->coef[w][g],
-                                   tns->order[w][g]);
-                }
-            }
-            count++;
-        }
-    }
-
-    sce->tns.present = !!count;
-}
--- a/libavcodec/aacenc_tns.h
+++ b/libavcodec/aacenc_tns.h
@@ -1,52 +0,0 @@
-/*
- * AAC encoder TNS
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder temporal noise shaping
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_TNS_H
-#define AVCODEC_AACENC_TNS_H
-
-#include "aacenc.h"
-
-/* Could be set to 3 to save an additional bit at the cost of little quality */
-#define TNS_Q_BITS 4
-
-/* TNS will only be used if the LPC gain is within these margins */
-#define TNS_GAIN_THRESHOLD_LOW  1.395f
-#define TNS_GAIN_THRESHOLD_HIGH 11.19f
-
-/* If the energy ratio between the low SFBs vs the high SFBs is not between
- * those two values, use 2 filters instead */
-#define TNS_E_RATIO_LOW  0.77
-#define TNS_E_RATIO_HIGH 1.23
-
-/* Do not use TNS if the psy band spread is below this value */
-#define TNS_SPREAD_THRESHOLD 37.081512f
-
-void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce);
-void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce);
-void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce);
-
-#endif /* AVCODEC_AACENC_TNS_H */
--- a/libavcodec/aacenc_utils.h
+++ b/libavcodec/aacenc_utils.h
@@ -1,143 +0,0 @@
-/*
- * AAC encoder utilities
- * Copyright (C) 2015 Rostislav Pehlivanov
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder utilities
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENC_UTILS_H
-#define AVCODEC_AACENC_UTILS_H
-
-#include "aac.h"
-#include "aac_tablegen_decl.h"
-#include "aacenctab.h"
-
-#define ROUND_STANDARD 0.4054f
-#define ROUND_TO_ZERO 0.1054f
-#define C_QUANT 0.4054f
-
-static inline void abs_pow34_v(float *out, const float *in, const int size)
-{
-    int i;
-    for (i = 0; i < size; i++) {
-        float a = fabsf(in[i]);
-        out[i] = sqrtf(a * sqrtf(a));
-    }
-}
-
-/**
- * Quantize one coefficient.
- * @return absolute value of the quantized coefficient
- * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
- */
-static inline int quant(float coef, const float Q, const float rounding)
-{
-    float a = coef * Q;
-    return sqrtf(a * sqrtf(a)) + rounding;
-}
-
-static inline void quantize_bands(int *out, const float *in, const float *scaled,
-                                  int size, float Q34, int is_signed, int maxval,
-                                  const float rounding)
-{
-    int i;
-    double qc;
-    for (i = 0; i < size; i++) {
-        qc = scaled[i] * Q34;
-        out[i] = (int)FFMIN(qc + rounding, (double)maxval);
-        if (is_signed && in[i] < 0.0f) {
-            out[i] = -out[i];
-        }
-    }
-}
-
-static inline float find_max_val(int group_len, int swb_size, const float *scaled)
-{
-    float maxval = 0.0f;
-    int w2, i;
-    for (w2 = 0; w2 < group_len; w2++) {
-        for (i = 0; i < swb_size; i++) {
-            maxval = FFMAX(maxval, scaled[w2*128+i]);
-        }
-    }
-    return maxval;
-}
-
-static inline int find_min_book(float maxval, int sf)
-{
-    float Q = ff_aac_pow2sf_tab[POW_SF2_ZERO - sf + SCALE_ONE_POS - SCALE_DIV_512];
-    float Q34 = sqrtf(Q * sqrtf(Q));
-    int qmaxval, cb;
-    qmaxval = maxval * Q34 + C_QUANT;
-    if      (qmaxval ==  0) cb = 0;
-    else if (qmaxval ==  1) cb = 1;
-    else if (qmaxval ==  2) cb = 3;
-    else if (qmaxval <=  4) cb = 5;
-    else if (qmaxval <=  7) cb = 7;
-    else if (qmaxval <= 12) cb = 9;
-    else                    cb = 11;
-    return cb;
-}
-
-/** Return the minimum scalefactor where the quantized coef does not clip. */
-static inline uint8_t coef2minsf(float coef)
-{
-    return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
-}
-
-/** Return the maximum scalefactor where the quantized coef is not zero. */
-static inline uint8_t coef2maxsf(float coef)
-{
-    return av_clip_uint8(log2f(coef)*4 +  6 + SCALE_ONE_POS - SCALE_DIV_512);
-}
-
-/*
- * Returns the closest possible index to an array of float values, given a value.
- */
-static inline int quant_array_idx(const float val, const float *arr, const int num)
-{
-    int i, index = 0;
-    float quant_min_err = INFINITY;
-    for (i = 0; i < num; i++) {
-        float error = (val - arr[i])*(val - arr[i]);
-        if (error < quant_min_err) {
-            quant_min_err = error;
-            index = i;
-        }
-    }
-    return index;
-}
-
-#define ERROR_IF(cond, ...) \
-    if (cond) { \
-        av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
-        return AVERROR(EINVAL); \
-    }
-
-#define WARN_IF(cond, ...) \
-    if (cond) { \
-        av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
-    }
-
-
-#endif /* AVCODEC_AACENC_UTILS_H */
--- a/libavcodec/aacenctab.c
+++ b/libavcodec/aacenctab.c
@@ -1,108 +0,0 @@
-/*
- * AAC encoder data
- * Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "aacenctab.h"
-
-static const uint8_t swb_size_128_96[] = {
-    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
-};
-
-static const uint8_t swb_size_128_64[] = {
-    4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
-};
-
-static const uint8_t swb_size_128_48[] = {
-    4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
-};
-
-static const uint8_t swb_size_128_24[] = {
-    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
-};
-
-static const uint8_t swb_size_128_16[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
-};
-
-static const uint8_t swb_size_128_8[] = {
-    4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
-};
-
-static const uint8_t swb_size_1024_96[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
-    64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_64[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
-    12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
-    40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
-};
-
-static const uint8_t swb_size_1024_48[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
-    96
-};
-
-static const uint8_t swb_size_1024_32[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
-};
-
-static const uint8_t swb_size_1024_24[] = {
-    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
-    32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_16[] = {
-    8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
-    12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
-    32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
-};
-
-static const uint8_t swb_size_1024_8[] = {
-    12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
-    16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
-    32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
-};
-
-const uint8_t *ff_aac_swb_size_128[] = {
-    swb_size_128_96, swb_size_128_96, swb_size_128_64,
-    swb_size_128_48, swb_size_128_48, swb_size_128_48,
-    swb_size_128_24, swb_size_128_24, swb_size_128_16,
-    swb_size_128_16, swb_size_128_16, swb_size_128_8,
-    swb_size_128_8
-};
-
-const uint8_t *ff_aac_swb_size_1024[] = {
-    swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
-    swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
-    swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
-    swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
-    swb_size_1024_8
-};
-
-const int ff_aac_swb_size_128_len  = FF_ARRAY_ELEMS(ff_aac_swb_size_128);
-const int ff_aac_swb_size_1024_len = FF_ARRAY_ELEMS(ff_aac_swb_size_1024);
--- a/libavcodec/aacenctab.h
+++ b/libavcodec/aacenctab.h
@@ -1,113 +0,0 @@
-/*
- * AAC encoder data
- * Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker gmail com )
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AAC encoder data
- * @author Rostislav Pehlivanov ( atomnuker gmail com )
- */
-
-#ifndef AVCODEC_AACENCTAB_H
-#define AVCODEC_AACENCTAB_H
-
-#include "aac.h"
-
-/** Total number of usable codebooks **/
-#define CB_TOT 12
-
-/** Total number of codebooks, including special ones **/
-#define CB_TOT_ALL 15
-
-#define AAC_MAX_CHANNELS 6
-
-extern const uint8_t *ff_aac_swb_size_1024[];
-extern const int      ff_aac_swb_size_1024_len;
-extern const uint8_t *ff_aac_swb_size_128[];
-extern const int      ff_aac_swb_size_128_len;
-
-/** default channel configurations */
-static const uint8_t aac_chan_configs[6][5] = {
-    {1, TYPE_SCE},                               // 1 channel  - single channel element
-    {1, TYPE_CPE},                               // 2 channels - channel pair
-    {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
-    {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
-    {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
-    {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
-};
-
-/**
- * Table to remap channels from libavcodec's default order to AAC order.
- */
-static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
-    { 0 },
-    { 0, 1 },
-    { 2, 0, 1 },
-    { 2, 0, 1, 3 },
-    { 2, 0, 1, 3, 4 },
-    { 2, 0, 1, 4, 5, 3 },
-};
-
-/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
- * failures */
-static const int mpeg4audio_sample_rates[16] = {
-    96000, 88200, 64000, 48000, 44100, 32000,
-    24000, 22050, 16000, 12000, 11025, 8000, 7350
-};
-
-/** bits needed to code codebook run value for long windows */
-static const uint8_t run_value_bits_long[64] = {
-     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
-     5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
-    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
-    10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
-};
-
-/** bits needed to code codebook run value for short windows */
-static const uint8_t run_value_bits_short[16] = {
-    3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
-};
-
-/* TNS starting SFBs for long and short windows */
-static const uint8_t tns_min_sfb_short[16] = {
-    2, 2, 2, 3, 3, 4, 6, 6, 8, 10, 10, 12, 12, 12, 12, 12
-};
-
-static const uint8_t tns_min_sfb_long[16] = {
-    12, 13, 15, 16, 17, 20, 25, 26, 24, 28, 30, 31, 31, 31, 31, 31
-};
-
-static const uint8_t * const tns_min_sfb[2] = {
-    tns_min_sfb_long, tns_min_sfb_short
-};
-
-static const uint8_t * const run_value_bits[2] = {
-    run_value_bits_long, run_value_bits_short
-};
-
-/** Map to convert values from BandCodingPath index to a codebook index **/
-static const uint8_t aac_cb_out_map[CB_TOT_ALL]  = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
-/** Inverse map to convert from codebooks to BandCodingPath indices **/
-static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12,13,14};
-
-static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
-static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
-
-#endif /* AVCODEC_AACENCTAB_H */
--- a/libavcodec/aacps.c
+++ b/libavcodec/aacps.c
@@ -17,23 +17,16 @@
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Note: Rounding-to-nearest used unless otherwise stated
- *
 */

 #include <stdint.h>
 #include "libavutil/common.h"
+#include "libavutil/internal.h"
 #include "libavutil/mathematics.h"
 #include "avcodec.h"
 #include "get_bits.h"
 #include "aacps.h"
-#if USE_FIXED
-#include "aacps_fixed_tablegen.h"
-#else
-#include "libavutil/internal.h"
 #include "aacps_tablegen.h"
-#endif /* USE_FIXED */
 #include "aacpsdata.c"

 #define PS_BASELINE 0  ///< Operate in Baseline PS mode
@@ -155,7 +148,7 @@ static void ipdopd_reset(int8_t *ipd_hist, int8_t *opd_hist)
    }
 }

-int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
+int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb_host, PSContext *ps, int bits_left)
 {
    int e;
    int bit_count_start = get_bits_count(gb_host);
@@ -309,41 +302,35 @@ err:

 /** Split one subband into 2 subsubbands with a symmetric real filter.
 * The filter must have its non-center even coefficients equal to zero. */
-static void hybrid2_re(INTFLOAT (*in)[2], INTFLOAT (*out)[32][2], const INTFLOAT filter[8], int len, int reverse)
+static void hybrid2_re(float (*in)[2], float (*out)[32][2], const float filter[8], int len, int reverse)
 {
    int i, j;
    for (i = 0; i < len; i++, in++) {
-        INT64FLOAT re_in = AAC_MUL31(filter[6], in[6][0]); //real inphase
-        INT64FLOAT re_op = 0.0f;                          //real out of phase
-        INT64FLOAT im_in = AAC_MUL31(filter[6], in[6][1]); //imag inphase
-        INT64FLOAT im_op = 0.0f;                          //imag out of phase
+        float re_in = filter[6] * in[6][0];          //real inphase
+        float re_op = 0.0f;                          //real out of phase
+        float im_in = filter[6] * in[6][1];          //imag inphase
+        float im_op = 0.0f;                          //imag out of phase
        for (j = 0; j < 6; j += 2) {
-            re_op += (INT64FLOAT)filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
-            im_op += (INT64FLOAT)filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
+            re_op += filter[j+1] * (in[j+1][0] + in[12-j-1][0]);
+            im_op += filter[j+1] * (in[j+1][1] + in[12-j-1][1]);
        }
-
-#if USE_FIXED
-        re_op = (re_op + 0x40000000) >> 31;
-        im_op = (im_op + 0x40000000) >> 31;
-#endif /* USE_FIXED */
-
-        out[ reverse][i][0] = (INTFLOAT)(re_in + re_op);
-        out[ reverse][i][1] = (INTFLOAT)(im_in + im_op);
-        out[!reverse][i][0] = (INTFLOAT)(re_in - re_op);
-        out[!reverse][i][1] = (INTFLOAT)(im_in - im_op);
+        out[ reverse][i][0] = re_in + re_op;
+        out[ reverse][i][1] = im_in + im_op;
+        out[!reverse][i][0] = re_in - re_op;
+        out[!reverse][i][1] = im_in - im_op;
    }
 }

 /** Split one subband into 6 subsubbands with a complex filter */
-static void hybrid6_cx(PSDSPContext *dsp, INTFLOAT (*in)[2], INTFLOAT (*out)[32][2],
-                       TABLE_CONST INTFLOAT (*filter)[8][2], int len)
+static void hybrid6_cx(PSDSPContext *dsp, float (*in)[2], float (*out)[32][2],
+                       TABLE_CONST float (*filter)[8][2], int len)
 {
    int i;
    int N = 8;
-    LOCAL_ALIGNED_16(INTFLOAT, temp, [8], [2]);
+    LOCAL_ALIGNED_16(float, temp, [8], [2]);

    for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(temp, in, (const INTFLOAT (*)[8][2]) filter, 1, N);
+        dsp->hybrid_analysis(temp, in, (const float (*)[8][2]) filter, 1, N);
        out[0][i][0] = temp[6][0];
        out[0][i][1] = temp[6][1];
        out[1][i][0] = temp[7][0];
@@ -360,18 +347,18 @@ static void hybrid6_cx(PSDSPContext *dsp, INTFLOAT (*in)[2], INTFLOAT (*out)[32]
 }

 static void hybrid4_8_12_cx(PSDSPContext *dsp,
-                            INTFLOAT (*in)[2], INTFLOAT (*out)[32][2],
-                            TABLE_CONST INTFLOAT (*filter)[8][2], int N, int len)
+                            float (*in)[2], float (*out)[32][2],
+                            TABLE_CONST float (*filter)[8][2], int N, int len)
 {
    int i;

    for (i = 0; i < len; i++, in++) {
-        dsp->hybrid_analysis(out[0] + i, in, (const INTFLOAT (*)[8][2]) filter, 32, N);
+        dsp->hybrid_analysis(out[0] + i, in, (const float (*)[8][2]) filter, 32, N);
    }
 }

-static void hybrid_analysis(PSDSPContext *dsp, INTFLOAT out[91][32][2],
-                            INTFLOAT in[5][44][2], INTFLOAT L[2][38][64],
+static void hybrid_analysis(PSDSPContext *dsp, float out[91][32][2],
+                            float in[5][44][2], float L[2][38][64],
                            int is34, int len)
 {
    int i, j;
@@ -400,8 +387,8 @@ static void hybrid_analysis(PSDSPContext *dsp, INTFLOAT out[91][32][2],
    }
 }

-static void hybrid_synthesis(PSDSPContext *dsp, INTFLOAT out[2][38][64],
-                             INTFLOAT in[91][32][2], int is34, int len)
+static void hybrid_synthesis(PSDSPContext *dsp, float out[2][38][64],
+                             float in[91][32][2], int is34, int len)
 {
    int i, n;
    if (is34) {
@@ -442,7 +429,7 @@ static void hybrid_synthesis(PSDSPContext *dsp, INTFLOAT out[2][38][64],
 }

 /// All-pass filter decay slope
-#define DECAY_SLOPE      Q30(0.05f)
+#define DECAY_SLOPE      0.05f
 /// Number of frequency bands that can be addressed by the parameter index, b(k)
 static const int   NR_PAR_BANDS[]      = { 20, 34 };
 static const int   NR_IPDOPD_BANDS[]   = { 11, 17 };
@@ -496,43 +483,28 @@ static void map_idx_34_to_20(int8_t *par_mapped, const int8_t *par, int full)
    }
 }

-static void map_val_34_to_20(INTFLOAT par[PS_MAX_NR_IIDICC])
+static void map_val_34_to_20(float par[PS_MAX_NR_IIDICC])
 {
-#if USE_FIXED
-    par[ 0] = (int)(((int64_t)(par[ 0] + (par[ 1]>>1)) * 1431655765 + \
-                      0x40000000) >> 31);
-    par[ 1] = (int)(((int64_t)((par[ 1]>>1) + par[ 2]) * 1431655765 + \
-                      0x40000000) >> 31);
-    par[ 2] = (int)(((int64_t)(par[ 3] + (par[ 4]>>1)) * 1431655765 + \
-                      0x40000000) >> 31);
-    par[ 3] = (int)(((int64_t)((par[ 4]>>1) + par[ 5]) * 1431655765 + \
-                      0x40000000) >> 31);
-#else
    par[ 0] = (2*par[ 0] +   par[ 1]) * 0.33333333f;
    par[ 1] = (  par[ 1] + 2*par[ 2]) * 0.33333333f;
    par[ 2] = (2*par[ 3] +   par[ 4]) * 0.33333333f;
    par[ 3] = (  par[ 4] + 2*par[ 5]) * 0.33333333f;
-#endif /* USE_FIXED */
-    par[ 4] = AAC_HALF_SUM(par[ 6], par[ 7]);
-    par[ 5] = AAC_HALF_SUM(par[ 8], par[ 9]);
+    par[ 4] = (  par[ 6] +   par[ 7]) * 0.5f;
+    par[ 5] = (  par[ 8] +   par[ 9]) * 0.5f;
    par[ 6] =    par[10];
    par[ 7] =    par[11];
-    par[ 8] = AAC_HALF_SUM(par[12], par[13]);
-    par[ 9] = AAC_HALF_SUM(par[14], par[15]);
+    par[ 8] = (  par[12] +   par[13]) * 0.5f;
+    par[ 9] = (  par[14] +   par[15]) * 0.5f;
    par[10] =    par[16];
    par[11] =    par[17];
    par[12] =    par[18];
    par[13] =    par[19];
-    par[14] = AAC_HALF_SUM(par[20], par[21]);
-    par[15] = AAC_HALF_SUM(par[22], par[23]);
-    par[16] = AAC_HALF_SUM(par[24], par[25]);
-    par[17] = AAC_HALF_SUM(par[26], par[27]);
-#if USE_FIXED
-    par[18] = (((par[28]+2)>>2) + ((par[29]+2)>>2) + ((par[30]+2)>>2) + ((par[31]+2)>>2));
-#else
+    par[14] = (  par[20] +   par[21]) * 0.5f;
+    par[15] = (  par[22] +   par[23]) * 0.5f;
+    par[16] = (  par[24] +   par[25]) * 0.5f;
+    par[17] = (  par[26] +   par[27]) * 0.5f;
    par[18] = (  par[28] +   par[29] +   par[30] +   par[31]) * 0.25f;
-#endif /* USE_FIXED */
-    par[19] = AAC_HALF_SUM(par[32], par[33]);
+    par[19] = (  par[32] +   par[33]) * 0.5f;
 }

 static void map_idx_10_to_34(int8_t *par_mapped, const int8_t *par, int full)
@@ -617,7 +589,7 @@ static void map_idx_20_to_34(int8_t *par_mapped, const int8_t *par, int full)
    par_mapped[ 0] =  par[ 0];
 }

-static void map_val_20_to_34(INTFLOAT par[PS_MAX_NR_IIDICC])
+static void map_val_20_to_34(float par[PS_MAX_NR_IIDICC])
 {
    par[33] =  par[19];
    par[32] =  par[19];
@@ -648,29 +620,27 @@ static void map_val_20_to_34(INTFLOAT par[PS_MAX_NR_IIDICC])
    par[ 7] =  par[ 4];
    par[ 6] =  par[ 4];
    par[ 5] =  par[ 3];
-    par[ 4] = AAC_HALF_SUM(par[ 2], par[ 3]);
+    par[ 4] = (par[ 2] + par[ 3]) * 0.5f;
    par[ 3] =  par[ 2];
    par[ 2] =  par[ 1];
-    par[ 1] = AAC_HALF_SUM(par[ 0], par[ 1]);
+    par[ 1] = (par[ 0] + par[ 1]) * 0.5f;
 }

-static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT (*s)[32][2], int is34)
+static void decorrelation(PSContext *ps, float (*out)[32][2], const float (*s)[32][2], int is34)
 {
-    LOCAL_ALIGNED_16(INTFLOAT, power, [34], [PS_QMF_TIME_SLOTS]);
-    LOCAL_ALIGNED_16(INTFLOAT, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
-    INTFLOAT *peak_decay_nrg = ps->peak_decay_nrg;
-    INTFLOAT *power_smooth = ps->power_smooth;
-    INTFLOAT *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
-    INTFLOAT (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
-    INTFLOAT (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
-#if !USE_FIXED
+    LOCAL_ALIGNED_16(float, power, [34], [PS_QMF_TIME_SLOTS]);
+    LOCAL_ALIGNED_16(float, transient_gain, [34], [PS_QMF_TIME_SLOTS]);
+    float *peak_decay_nrg = ps->peak_decay_nrg;
+    float *power_smooth = ps->power_smooth;
+    float *peak_decay_diff_smooth = ps->peak_decay_diff_smooth;
+    float (*delay)[PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2] = ps->delay;
+    float (*ap_delay)[PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2] = ps->ap_delay;
+    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
+    const float peak_decay_factor = 0.76592833836465f;
    const float transient_impact  = 1.5f;
    const float a_smooth          = 0.25f; ///< Smoothing coefficient
-#endif /* USE_FIXED */
-    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
    int i, k, m, n;
    int n0 = 0, nL = 32;
-    const INTFLOAT peak_decay_factor = Q31(0.76592833836465f);

    memset(power, 0, 34 * sizeof(*power));

@@ -688,33 +658,6 @@ static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT
    }

    //Transient detection
-#if USE_FIXED
-    for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
-        for (n = n0; n < nL; n++) {
-            int decayed_peak;
-            int denom;
-
-            decayed_peak = (int)(((int64_t)peak_decay_factor * \
-                                           peak_decay_nrg[i] + 0x40000000) >> 31);
-            peak_decay_nrg[i] = FFMAX(decayed_peak, power[i][n]);
-            power_smooth[i] += (power[i][n] - power_smooth[i] + 2) >> 2;
-            peak_decay_diff_smooth[i] += (peak_decay_nrg[i] - power[i][n] - \
-                                          peak_decay_diff_smooth[i] + 2) >> 2;
-            denom = peak_decay_diff_smooth[i] + (peak_decay_diff_smooth[i] >> 1);
-            if (denom > power_smooth[i]) {
-              int p = power_smooth[i];
-              while (denom < 0x40000000) {
-                denom <<= 1;
-                p <<= 1;
-              }
-              transient_gain[i][n] = p / (denom >> 16);
-            }
-            else {
-              transient_gain[i][n] = 1 << 16;
-            }
-        }
-    }
-#else
    for (i = 0; i < NR_PAR_BANDS[is34]; i++) {
        for (n = n0; n < nL; n++) {
            float decayed_peak = peak_decay_factor * peak_decay_nrg[i];
@@ -728,7 +671,6 @@ static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT
        }
    }

-#endif /* USE_FIXED */
    //Decorrelation and transient reduction
    //                         PS_AP_LINKS - 1
    //                               -----
@@ -739,22 +681,8 @@ static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT
    //d[k][z] (out) = transient_gain_mapped[k][z] * H[k][z] * s[k][z]
    for (k = 0; k < NR_ALLPASS_BANDS[is34]; k++) {
        int b = k_to_i[k];
-#if USE_FIXED
-        int g_decay_slope;
-
-        if (k - DECAY_CUTOFF[is34] <= 0) {
-          g_decay_slope = 1 << 30;
-        }
-        else if (k - DECAY_CUTOFF[is34] >= 20) {
-          g_decay_slope = 0;
-        }
-        else {
-          g_decay_slope = (1 << 30) - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
-        }
-#else
        float g_decay_slope = 1.f - DECAY_SLOPE * (k - DECAY_CUTOFF[is34]);
        g_decay_slope = av_clipf(g_decay_slope, 0.f, 1.f);
-#endif /* USE_FIXED */
        memcpy(delay[k], delay[k]+nL, PS_MAX_DELAY*sizeof(delay[k][0]));
        memcpy(delay[k]+PS_MAX_DELAY, s[k], numQMFSlots*sizeof(delay[k][0]));
        for (m = 0; m < PS_AP_LINKS; m++) {
@@ -762,7 +690,7 @@ static void decorrelation(PSContext *ps, INTFLOAT (*out)[32][2], const INTFLOAT
        }
        ps->dsp.decorrelate(out[k], delay[k] + PS_MAX_DELAY - 2, ap_delay[k],
                            phi_fract[is34][k],
-                            (const INTFLOAT (*)[2]) Q_fract_allpass[is34][k],
+                            (const float (*)[2]) Q_fract_allpass[is34][k],
                            transient_gain[b], g_decay_slope, nL - n0);
    }
    for (; k < SHORT_DELAY_BAND[is34]; k++) {
@@ -821,14 +749,14 @@ static void remap20(int8_t (**p_par_mapped)[PS_MAX_NR_IIDICC],
    }
 }

-static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)[32][2], int is34)
+static void stereo_processing(PSContext *ps, float (*l)[32][2], float (*r)[32][2], int is34)
 {
    int e, b, k;

-    INTFLOAT (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
-    INTFLOAT (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
-    INTFLOAT (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
-    INTFLOAT (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
+    float (*H11)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H11;
+    float (*H12)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H12;
+    float (*H21)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H21;
+    float (*H22)[PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC] = ps->H22;
    int8_t *opd_hist = ps->opd_hist;
    int8_t *ipd_hist = ps->ipd_hist;
    int8_t iid_mapped_buf[PS_MAX_NUM_ENV][PS_MAX_NR_IIDICC];
@@ -840,7 +768,7 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
    int8_t (*ipd_mapped)[PS_MAX_NR_IIDICC] = ipd_mapped_buf;
    int8_t (*opd_mapped)[PS_MAX_NR_IIDICC] = opd_mapped_buf;
    const int8_t *k_to_i = is34 ? k_to_i_34 : k_to_i_20;
-    TABLE_CONST INTFLOAT (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;
+    TABLE_CONST float (*H_LUT)[8][4] = (PS_BASELINE || ps->icc_mode < 3) ? HA : HB;

    //Remapping
    if (ps->num_env_old) {
@@ -895,7 +823,7 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
    //Mixing
    for (e = 0; e < ps->num_env; e++) {
        for (b = 0; b < NR_PAR_BANDS[is34]; b++) {
-            INTFLOAT h11, h12, h21, h22;
+            float h11, h12, h21, h22;
            h11 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][0];
            h12 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][1];
            h21 = H_LUT[iid_mapped[e][b] + 7 + 23 * ps->iid_quant][icc_mapped[e][b]][2];
@@ -904,27 +832,27 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
            if (!PS_BASELINE && ps->enable_ipdopd && b < NR_IPDOPD_BANDS[is34]) {
                //The spec say says to only run this smoother when enable_ipdopd
                //is set but the reference decoder appears to run it constantly
-                INTFLOAT h11i, h12i, h21i, h22i;
-                INTFLOAT ipd_adj_re, ipd_adj_im;
+                float h11i, h12i, h21i, h22i;
+                float ipd_adj_re, ipd_adj_im;
                int opd_idx = opd_hist[b] * 8 + opd_mapped[e][b];
                int ipd_idx = ipd_hist[b] * 8 + ipd_mapped[e][b];
-                INTFLOAT opd_re = pd_re_smooth[opd_idx];
-                INTFLOAT opd_im = pd_im_smooth[opd_idx];
-                INTFLOAT ipd_re = pd_re_smooth[ipd_idx];
-                INTFLOAT ipd_im = pd_im_smooth[ipd_idx];
+                float opd_re = pd_re_smooth[opd_idx];
+                float opd_im = pd_im_smooth[opd_idx];
+                float ipd_re = pd_re_smooth[ipd_idx];
+                float ipd_im = pd_im_smooth[ipd_idx];
                opd_hist[b] = opd_idx & 0x3F;
                ipd_hist[b] = ipd_idx & 0x3F;

-                ipd_adj_re = AAC_MADD30(opd_re, ipd_re, opd_im, ipd_im);
-                ipd_adj_im = AAC_MSUB30(opd_im, ipd_re, opd_re, ipd_im);
-                h11i = AAC_MUL30(h11,  opd_im);
-                h11  = AAC_MUL30(h11,  opd_re);
-                h12i = AAC_MUL30(h12,  ipd_adj_im);
-                h12  = AAC_MUL30(h12,  ipd_adj_re);
-                h21i = AAC_MUL30(h21,  opd_im);
-                h21  = AAC_MUL30(h21,  opd_re);
-                h22i = AAC_MUL30(h22,  ipd_adj_im);
-                h22  = AAC_MUL30(h22,  ipd_adj_re);
+                ipd_adj_re = opd_re*ipd_re + opd_im*ipd_im;
+                ipd_adj_im = opd_im*ipd_re - opd_re*ipd_im;
+                h11i = h11 * opd_im;
+                h11  = h11 * opd_re;
+                h12i = h12 * ipd_adj_im;
+                h12  = h12 * ipd_adj_re;
+                h21i = h21 * opd_im;
+                h21  = h21 * opd_re;
+                h22i = h22 * ipd_adj_im;
+                h22  = h22 * ipd_adj_re;
                H11[1][e+1][b] = h11i;
                H12[1][e+1][b] = h12i;
                H21[1][e+1][b] = h21i;
@@ -936,14 +864,11 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
            H22[0][e+1][b] = h22;
        }
        for (k = 0; k < NR_BANDS[is34]; k++) {
-            LOCAL_ALIGNED_16(INTFLOAT, h, [2], [4]);
-            LOCAL_ALIGNED_16(INTFLOAT, h_step, [2], [4]);
+            float h[2][4];
+            float h_step[2][4];
            int start = ps->border_position[e];
            int stop  = ps->border_position[e+1];
-            INTFLOAT width = Q30(1.f) / (stop - start);
-#if USE_FIXED
-            width <<= 1;
-#endif
+            float width = 1.f / (stop - start);
            b = k_to_i[k];
            h[0][0] = H11[0][e][b];
            h[0][1] = H12[0][e][b];
@@ -964,15 +889,15 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
            }
            }
            //Interpolation
-            h_step[0][0] = AAC_MSUB31_V3(H11[0][e+1][b], h[0][0], width);
-            h_step[0][1] = AAC_MSUB31_V3(H12[0][e+1][b], h[0][1], width);
-            h_step[0][2] = AAC_MSUB31_V3(H21[0][e+1][b], h[0][2], width);
-            h_step[0][3] = AAC_MSUB31_V3(H22[0][e+1][b], h[0][3], width);
+            h_step[0][0] = (H11[0][e+1][b] - h[0][0]) * width;
+            h_step[0][1] = (H12[0][e+1][b] - h[0][1]) * width;
+            h_step[0][2] = (H21[0][e+1][b] - h[0][2]) * width;
+            h_step[0][3] = (H22[0][e+1][b] - h[0][3]) * width;
            if (!PS_BASELINE && ps->enable_ipdopd) {
-                h_step[1][0] = AAC_MSUB31_V3(H11[1][e+1][b], h[1][0], width);
-                h_step[1][1] = AAC_MSUB31_V3(H12[1][e+1][b], h[1][1], width);
-                h_step[1][2] = AAC_MSUB31_V3(H21[1][e+1][b], h[1][2], width);
-                h_step[1][3] = AAC_MSUB31_V3(H22[1][e+1][b], h[1][3], width);
+                h_step[1][0] = (H11[1][e+1][b] - h[1][0]) * width;
+                h_step[1][1] = (H12[1][e+1][b] - h[1][1]) * width;
+                h_step[1][2] = (H21[1][e+1][b] - h[1][2]) * width;
+                h_step[1][3] = (H22[1][e+1][b] - h[1][3]) * width;
            }
            ps->dsp.stereo_interpolate[!PS_BASELINE && ps->enable_ipdopd](
                l[k] + start + 1, r[k] + start + 1,
@@ -981,10 +906,10 @@ static void stereo_processing(PSContext *ps, INTFLOAT (*l)[32][2], INTFLOAT (*r)
    }
 }

-int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top)
+int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top)
 {
-    INTFLOAT (*Lbuf)[32][2] = ps->Lbuf;
-    INTFLOAT (*Rbuf)[32][2] = ps->Rbuf;
+    float (*Lbuf)[32][2] = ps->Lbuf;
+    float (*Rbuf)[32][2] = ps->Rbuf;
    const int len = 32;
    int is34 = ps->is34bands;

@@ -994,7 +919,7 @@ int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][
        memset(ps->ap_delay + top, 0, (NR_ALLPASS_BANDS[is34] - top)*sizeof(ps->ap_delay[0]));

    hybrid_analysis(&ps->dsp, Lbuf, ps->in_buf, L, is34, len);
-    decorrelation(ps, Rbuf, (const INTFLOAT (*)[32][2]) Lbuf, is34);
+    decorrelation(ps, Rbuf, (const float (*)[32][2]) Lbuf, is34);
    stereo_processing(ps, Lbuf, Rbuf, is34);
    hybrid_synthesis(&ps->dsp, L, Lbuf, is34, len);
    hybrid_synthesis(&ps->dsp, R, Rbuf, is34, len);
@@ -1011,7 +936,7 @@ int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][
 #define PS_VLC_ROW(name) \
    { name ## _codes, name ## _bits, sizeof(name ## _codes), sizeof(name ## _codes[0]) }

-av_cold void AAC_RENAME(ff_ps_init)(void) {
+av_cold void ff_ps_init(void) {
    // Syntax initialization
    static const struct {
        const void *ps_codes, *ps_bits;
@@ -1043,7 +968,7 @@ av_cold void AAC_RENAME(ff_ps_init)(void) {
    ps_tableinit();
 }

-av_cold void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps)
+av_cold void ff_ps_ctx_init(PSContext *ps)
 {
-    AAC_RENAME(ff_psdsp_init)(&ps->dsp);
+    ff_psdsp_init(&ps->dsp);
 }
--- a/libavcodec/aacps.h
+++ b/libavcodec/aacps.h
@@ -61,26 +61,26 @@ typedef struct PSContext {
    int    is34bands;
    int    is34bands_old;

-    DECLARE_ALIGNED(16, INTFLOAT, in_buf)[5][44][2];
-    DECLARE_ALIGNED(16, INTFLOAT, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
-    DECLARE_ALIGNED(16, INTFLOAT, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
-    DECLARE_ALIGNED(16, INTFLOAT, peak_decay_nrg)[34];
-    DECLARE_ALIGNED(16, INTFLOAT, power_smooth)[34];
-    DECLARE_ALIGNED(16, INTFLOAT, peak_decay_diff_smooth)[34];
-    DECLARE_ALIGNED(16, INTFLOAT, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, INTFLOAT, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, INTFLOAT, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, INTFLOAT, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
-    DECLARE_ALIGNED(16, INTFLOAT, Lbuf)[91][32][2];
-    DECLARE_ALIGNED(16, INTFLOAT, Rbuf)[91][32][2];
+    DECLARE_ALIGNED(16, float, in_buf)[5][44][2];
+    DECLARE_ALIGNED(16, float, delay)[PS_MAX_SSB][PS_QMF_TIME_SLOTS + PS_MAX_DELAY][2];
+    DECLARE_ALIGNED(16, float, ap_delay)[PS_MAX_AP_BANDS][PS_AP_LINKS][PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2];
+    DECLARE_ALIGNED(16, float, peak_decay_nrg)[34];
+    DECLARE_ALIGNED(16, float, power_smooth)[34];
+    DECLARE_ALIGNED(16, float, peak_decay_diff_smooth)[34];
+    DECLARE_ALIGNED(16, float, H11)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, float, H12)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, float, H21)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, float, H22)[2][PS_MAX_NUM_ENV+1][PS_MAX_NR_IIDICC];
+    DECLARE_ALIGNED(16, float, Lbuf)[91][32][2];
+    DECLARE_ALIGNED(16, float, Rbuf)[91][32][2];
    int8_t opd_hist[PS_MAX_NR_IIDICC];
    int8_t ipd_hist[PS_MAX_NR_IIDICC];
    PSDSPContext dsp;
 } PSContext;

-void AAC_RENAME(ff_ps_init)(void);
-void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps);
-int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
-int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top);
+void ff_ps_init(void);
+void ff_ps_ctx_init(PSContext *ps);
+int ff_ps_read_data(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left);
+int ff_ps_apply(AVCodecContext *avctx, PSContext *ps, float L[2][38][64], float R[2][38][64], int top);

 #endif /* AVCODEC_PS_H */
--- a/libavcodec/aacps_fixed.c
+++ b/libavcodec/aacps_fixed.c
@@ -1,24 +0,0 @@
-/*
- * MPEG-4 Parametric Stereo decoding functions
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#define USE_FIXED 1
-
-#include "aacps.c"
--- a/libavcodec/aacps_fixed_tablegen.c
+++ b/libavcodec/aacps_fixed_tablegen.c
@@ -1,24 +0,0 @@
-/*
- * Generate a header file for hardcoded Parametric Stereo tables
- *
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#define USE_FIXED 1
-#include "aacps_tablegen_template.c"
--- a/libavcodec/aacps_fixed_tablegen.h
+++ b/libavcodec/aacps_fixed_tablegen.h
@@ -1,403 +0,0 @@
-/*
- * Header file for hardcoded Parametric Stereo tables
- *
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- *
- * Note: Rounding-to-nearest used unless otherwise stated
- *
- */
-
-#ifndef AACPS_FIXED_TABLEGEN_H
-#define AACPS_FIXED_TABLEGEN_H
-
-#include <math.h>
-#include <stdint.h>
-
-#if CONFIG_HARDCODED_TABLES
-#define ps_tableinit()
-#define TABLE_CONST const
-#include "libavcodec/aacps_fixed_tables.h"
-#else
-#include "libavutil/common.h"
-#include "libavutil/mathematics.h"
-#include "libavutil/mem.h"
-
-#include "aac_defines.h"
-#include "libavutil/softfloat.h"
-#define NR_ALLPASS_BANDS20 30
-#define NR_ALLPASS_BANDS34 50
-#define PS_AP_LINKS 3
-#define TABLE_CONST
-static int pd_re_smooth[8*8*8];
-static int pd_im_smooth[8*8*8];
-static int HA[46][8][4];
-static int HB[46][8][4];
-static DECLARE_ALIGNED(16, int, f20_0_8) [ 8][8][2];
-static DECLARE_ALIGNED(16, int, f34_0_12)[12][8][2];
-static DECLARE_ALIGNED(16, int, f34_1_8) [ 8][8][2];
-static DECLARE_ALIGNED(16, int, f34_2_4) [ 4][8][2];
-static TABLE_CONST DECLARE_ALIGNED(16, int, Q_fract_allpass)[2][50][3][2];
-static DECLARE_ALIGNED(16, int, phi_fract)[2][50][2];
-
-static const int g0_Q8[] = {
-    Q31(0.00746082949812f), Q31(0.02270420949825f), Q31(0.04546865930473f), Q31(0.07266113929591f),
-    Q31(0.09885108575264f), Q31(0.11793710567217f), Q31(0.125f)
-};
-
-static const int g0_Q12[] = {
-    Q31(0.04081179924692f), Q31(0.03812810994926f), Q31(0.05144908135699f), Q31(0.06399831151592f),
-    Q31(0.07428313801106f), Q31(0.08100347892914f), Q31(0.08333333333333f)
-};
-
-static const int g1_Q8[] = {
-    Q31(0.01565675600122f), Q31(0.03752716391991f), Q31(0.05417891378782f), Q31(0.08417044116767f),
-    Q31(0.10307344158036f), Q31(0.12222452249753f), Q31(0.125f)
-};
-
-static const int g2_Q4[] = {
-    Q31(-0.05908211155639f), Q31(-0.04871498374946f), Q31(0.0f),   Q31(0.07778723915851f),
-    Q31( 0.16486303567403f), Q31( 0.23279856662996f), Q31(0.25f)
-};
-
-static const int sintbl_4[4]   = {           0,  1073741824,           0, -1073741824 };
-static const int costbl_4[4]   = {  1073741824,           0, -1073741824,           0 };
-static const int sintbl_8[8]   = {           0,   759250125,  1073741824,   759250125,
-                                             0,  -759250125, -1073741824,  -759250125 };
-static const int costbl_8[8]   = {  1073741824,   759250125,           0,  -759250125,
-                                   -1073741824,  -759250125,           0,   759250125 };
-static const int sintbl_12[12] = {           0,   536870912,   929887697,  1073741824,
-                                     929887697,   536870912,           0,  -536870912,
-                                    -929887697, -1073741824,  -929887697,  -536870912 };
-static const int costbl_12[12] = {  1073741824,   929887697,   536870912,           0,
-                                    -536870912,  -929887697, -1073741824,  -929887697,
-                                    -536870912,           0,   536870912,   929887697 };
-
-static void make_filters_from_proto(int (*filter)[8][2], const int *proto, int bands)
-{
-
-    const int *sinptr, *cosptr;
-    int s, c, sinhalf, coshalf;
-    int q, n;
-
-    if (bands == 4) {
-        sinptr = sintbl_4;
-        cosptr = costbl_4;
-        sinhalf = 759250125;
-        coshalf = 759250125;
-    } else if (bands == 8) {
-        sinptr = sintbl_8;
-        cosptr = costbl_8;
-        sinhalf = 410903207;
-        coshalf = 992008094;
-    } else {
-        sinptr = sintbl_12;
-        cosptr = costbl_12;
-        sinhalf = 277904834;
-        coshalf = 1037154959;
-    }
-
-    for (q = 0; q < bands; q++) {
-        for (n = 0; n < 7; n++) {
-            int theta = (q*(n-6) + (n>>1) - 3) % bands;
-
-            if (theta < 0)
-                theta += bands;
-            s = sinptr[theta];
-            c = cosptr[theta];
-
-            if (n & 1) {
-                theta = (int)(((int64_t)c * coshalf - (int64_t)s * sinhalf + 0x20000000) >> 30);
-                s = (int)(((int64_t)s * coshalf + (int64_t)c * sinhalf + 0x20000000) >> 30);
-                c = theta;
-            }
-            filter[q][n][0] = (int)(((int64_t)proto[n] * c + 0x20000000) >> 30);
-            filter[q][n][1] = -(int)(((int64_t)proto[n] * s + 0x20000000) >> 30);
-        }
-    }
-}
-
-static void ps_tableinit(void)
-{
-    static const int ipdopd_sin[] = { Q30(0), Q30(M_SQRT1_2), Q30(1), Q30( M_SQRT1_2), Q30( 0), Q30(-M_SQRT1_2), Q30(-1), Q30(-M_SQRT1_2) };
-    static const int ipdopd_cos[] = { Q30(1), Q30(M_SQRT1_2), Q30(0), Q30(-M_SQRT1_2), Q30(-1), Q30(-M_SQRT1_2), Q30( 0), Q30( M_SQRT1_2) };
-    int pd0, pd1, pd2;
-    int idx;
-
-    static const int alpha_tab[] =
-    {
-      Q30(1.5146213770f/M_PI), Q30(1.5181334019f/M_PI), Q30(1.5234849453f/M_PI), Q30(1.5369486809f/M_PI), Q30(1.5500687361f/M_PI), Q30(1.5679757595f/M_PI),
-      Q30(1.4455626011f/M_PI), Q30(1.4531552792f/M_PI), Q30(1.4648091793f/M_PI), Q30(1.4945238829f/M_PI), Q30(1.5239057541f/M_PI), Q30(1.5644006729f/M_PI),
-      Q30(1.3738563061f/M_PI), Q30(1.3851221800f/M_PI), Q30(1.4026404619f/M_PI), Q30(1.4484288692f/M_PI), Q30(1.4949874878f/M_PI), Q30(1.5604078770f/M_PI),
-      Q30(1.2645189762f/M_PI), Q30(1.2796478271f/M_PI), Q30(1.3038636446f/M_PI), Q30(1.3710125685f/M_PI), Q30(1.4443849325f/M_PI), Q30(1.5532352924f/M_PI),
-      Q30(1.1507037878f/M_PI), Q30(1.1669205427f/M_PI), Q30(1.1938756704f/M_PI), Q30(1.2754167318f/M_PI), Q30(1.3761177063f/M_PI), Q30(1.5429240465f/M_PI),
-      Q30(1.0079245567f/M_PI), Q30(1.0208238363f/M_PI), Q30(1.0433073044f/M_PI), Q30(1.1208510399f/M_PI), Q30(1.2424604893f/M_PI), Q30(1.5185726881f/M_PI),
-      Q30(0.8995233774f/M_PI), Q30(0.9069069624f/M_PI), Q30(0.9201194048f/M_PI), Q30(0.9698365927f/M_PI), Q30(1.0671583414f/M_PI), Q30(1.4647934437f/M_PI),
-      Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI),
-      Q30(0.6712729335f/M_PI), Q30(0.6638893485f/M_PI), Q30(0.6506769061f/M_PI), Q30(0.6009597182f/M_PI), Q30(0.5036380291f/M_PI), Q30(0.1060028747f/M_PI),
-      Q30(0.5628717542f/M_PI), Q30(0.5499725342f/M_PI), Q30(0.5274890065f/M_PI), Q30(0.4499453008f/M_PI), Q30(0.3283358216f/M_PI), Q30(0.0522236861f/M_PI),
-      Q30(0.4200925827f/M_PI), Q30(0.4038758278f/M_PI), Q30(0.3769206405f/M_PI), Q30(0.2953795493f/M_PI), Q30(0.1946786791f/M_PI), Q30(0.0278722942f/M_PI),
-      Q30(0.3062773645f/M_PI), Q30(0.2911485136f/M_PI), Q30(0.2669326365f/M_PI), Q30(0.1997837722f/M_PI), Q30(0.1264114529f/M_PI), Q30(0.0175609849f/M_PI),
-      Q30(0.1969399750f/M_PI), Q30(0.1856741160f/M_PI), Q30(0.1681558639f/M_PI), Q30(0.1223674342f/M_PI), Q30(0.0758088827f/M_PI), Q30(0.0103884479f/M_PI),
-      Q30(0.1252337098f/M_PI), Q30(0.1176410317f/M_PI), Q30(0.1059871912f/M_PI), Q30(0.0762724727f/M_PI), Q30(0.0468905345f/M_PI), Q30(0.0063956482f/M_PI),
-      Q30(0.0561749674f/M_PI), Q30(0.0526629239f/M_PI), Q30(0.0473113805f/M_PI), Q30(0.0338476151f/M_PI), Q30(0.0207276177f/M_PI), Q30(0.0028205961f/M_PI),
-      Q30(1.5676341057f/M_PI), Q30(1.5678333044f/M_PI), Q30(1.5681363344f/M_PI), Q30(1.5688960552f/M_PI), Q30(1.5696337223f/M_PI), Q30(1.5706381798f/M_PI),
-      Q30(1.5651730299f/M_PI), Q30(1.5655272007f/M_PI), Q30(1.5660660267f/M_PI), Q30(1.5674170256f/M_PI), Q30(1.5687289238f/M_PI), Q30(1.5705151558f/M_PI),
-      Q30(1.5607966185f/M_PI), Q30(1.5614265203f/M_PI), Q30(1.5623844862f/M_PI), Q30(1.5647867918f/M_PI), Q30(1.5671195984f/M_PI), Q30(1.5702962875f/M_PI),
-      Q30(1.5530153513f/M_PI), Q30(1.5541347265f/M_PI), Q30(1.5558375120f/M_PI), Q30(1.5601085424f/M_PI), Q30(1.5642569065f/M_PI), Q30(1.5699069500f/M_PI),
-      Q30(1.5391840935f/M_PI), Q30(1.5411708355f/M_PI), Q30(1.5441943407f/M_PI), Q30(1.5517836809f/M_PI), Q30(1.5591609478f/M_PI), Q30(1.5692136288f/M_PI),
-      Q30(1.5146213770f/M_PI), Q30(1.5181334019f/M_PI), Q30(1.5234849453f/M_PI), Q30(1.5369486809f/M_PI), Q30(1.5500687361f/M_PI), Q30(1.5679757595f/M_PI),
-      Q30(1.4915299416f/M_PI), Q30(1.4964480400f/M_PI), Q30(1.5039558411f/M_PI), Q30(1.5229074955f/M_PI), Q30(1.5414420366f/M_PI), Q30(1.5667995214f/M_PI),
-      Q30(1.4590617418f/M_PI), Q30(1.4658898115f/M_PI), Q30(1.4763505459f/M_PI), Q30(1.5029321909f/M_PI), Q30(1.5291173458f/M_PI), Q30(1.5651149750f/M_PI),
-      Q30(1.4136143923f/M_PI), Q30(1.4229322672f/M_PI), Q30(1.4373078346f/M_PI), Q30(1.4743183851f/M_PI), Q30(1.5113102198f/M_PI), Q30(1.5626684427f/M_PI),
-      Q30(1.3505556583f/M_PI), Q30(1.3628427982f/M_PI), Q30(1.3820509911f/M_PI), Q30(1.4327841997f/M_PI), Q30(1.4850014448f/M_PI), Q30(1.5590143204f/M_PI),
-      Q30(1.2645189762f/M_PI), Q30(1.2796478271f/M_PI), Q30(1.3038636446f/M_PI), Q30(1.3710125685f/M_PI), Q30(1.4443849325f/M_PI), Q30(1.5532352924f/M_PI),
-      Q30(1.1919227839f/M_PI), Q30(1.2081253529f/M_PI), Q30(1.2346779108f/M_PI), Q30(1.3123005629f/M_PI), Q30(1.4034168720f/M_PI), Q30(1.5471596718f/M_PI),
-      Q30(1.1061993837f/M_PI), Q30(1.1219338179f/M_PI), Q30(1.1484941244f/M_PI), Q30(1.2320860624f/M_PI), Q30(1.3421301842f/M_PI), Q30(1.5373806953f/M_PI),
-      Q30(1.0079245567f/M_PI), Q30(1.0208238363f/M_PI), Q30(1.0433073044f/M_PI), Q30(1.1208510399f/M_PI), Q30(1.2424604893f/M_PI), Q30(1.5185726881f/M_PI),
-      Q30(0.8995233774f/M_PI), Q30(0.9069069624f/M_PI), Q30(0.9201194048f/M_PI), Q30(0.9698365927f/M_PI), Q30(1.0671583414f/M_PI), Q30(1.4647934437f/M_PI),
-      Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI), Q30(0.7853981853f/M_PI),
-      Q30(0.6712729335f/M_PI), Q30(0.6638893485f/M_PI), Q30(0.6506769061f/M_PI), Q30(0.6009597182f/M_PI), Q30(0.5036380291f/M_PI), Q30(0.1060028747f/M_PI),
-      Q30(0.5628717542f/M_PI), Q30(0.5499725342f/M_PI), Q30(0.5274890065f/M_PI), Q30(0.4499453008f/M_PI), Q30(0.3283358216f/M_PI), Q30(0.0522236861f/M_PI),
-      Q30(0.4645969570f/M_PI), Q30(0.4488625824f/M_PI), Q30(0.4223022461f/M_PI), Q30(0.3387103081f/M_PI), Q30(0.2286661267f/M_PI), Q30(0.0334156826f/M_PI),
-      Q30(0.3788735867f/M_PI), Q30(0.3626709878f/M_PI), Q30(0.3361184299f/M_PI), Q30(0.2584958076f/M_PI), Q30(0.1673794836f/M_PI), Q30(0.0236366931f/M_PI),
-      Q30(0.3062773645f/M_PI), Q30(0.2911485136f/M_PI), Q30(0.2669326365f/M_PI), Q30(0.1997837722f/M_PI), Q30(0.1264114529f/M_PI), Q30(0.0175609849f/M_PI),
-      Q30(0.2202406377f/M_PI), Q30(0.2079535723f/M_PI), Q30(0.1887452900f/M_PI), Q30(0.1380121708f/M_PI), Q30(0.0857949182f/M_PI), Q30(0.0117820343f/M_PI),
-      Q30(0.1571819335f/M_PI), Q30(0.1478640437f/M_PI), Q30(0.1334884763f/M_PI), Q30(0.0964778885f/M_PI), Q30(0.0594860613f/M_PI), Q30(0.0081279324f/M_PI),
-      Q30(0.1117345318f/M_PI), Q30(0.1049065739f/M_PI), Q30(0.0944457650f/M_PI), Q30(0.0678641573f/M_PI), Q30(0.0416790098f/M_PI), Q30(0.0056813755f/M_PI),
-      Q30(0.0792663917f/M_PI), Q30(0.0743482932f/M_PI), Q30(0.0668405443f/M_PI), Q30(0.0478888862f/M_PI), Q30(0.0293543357f/M_PI), Q30(0.0039967746f/M_PI),
-      Q30(0.0561749674f/M_PI), Q30(0.0526629239f/M_PI), Q30(0.0473113805f/M_PI), Q30(0.0338476151f/M_PI), Q30(0.0207276177f/M_PI), Q30(0.0028205961f/M_PI),
-      Q30(0.0316122435f/M_PI), Q30(0.0296254847f/M_PI), Q30(0.0266019460f/M_PI), Q30(0.0190126132f/M_PI), Q30(0.0116353342f/M_PI), Q30(0.0015827164f/M_PI),
-      Q30(0.0177809205f/M_PI), Q30(0.0166615788f/M_PI), Q30(0.0149587989f/M_PI), Q30(0.0106877899f/M_PI), Q30(0.0065393616f/M_PI), Q30(0.0008894200f/M_PI),
-      Q30(0.0099996664f/M_PI), Q30(0.0093698399f/M_PI), Q30(0.0084118480f/M_PI), Q30(0.0060095116f/M_PI), Q30(0.0036767013f/M_PI), Q30(0.0005000498f/M_PI),
-      Q30(0.0056233541f/M_PI), Q30(0.0052691097f/M_PI), Q30(0.0047303112f/M_PI), Q30(0.0033792770f/M_PI), Q30(0.0020674451f/M_PI), Q30(0.0002811795f/M_PI),
-      Q30(0.0031622672f/M_PI), Q30(0.0029630491f/M_PI), Q30(0.0026600463f/M_PI), Q30(0.0019002859f/M_PI), Q30(0.0011625893f/M_PI), Q30(0.0001581155f/M_PI)
-    };
-
-    static const int gamma_tab[] =
-    {
-      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0433459543f/M_PI), Q30(0.0672172382f/M_PI), Q30(0.0997167900f/M_PI), Q30(0.1162951663f/M_PI), Q30(0.1250736862f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0672341362f/M_PI), Q30(0.1045235619f/M_PI), Q30(0.1558904350f/M_PI), Q30(0.1824723780f/M_PI), Q30(0.1966800541f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1315985769f/M_PI), Q30(0.2072522491f/M_PI), Q30(0.3188187480f/M_PI), Q30(0.3825501204f/M_PI), Q30(0.4193951190f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1784276664f/M_PI), Q30(0.2856673002f/M_PI), Q30(0.4630723596f/M_PI), Q30(0.5971632004f/M_PI), Q30(0.7603877187f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1315985769f/M_PI), Q30(0.2072522491f/M_PI), Q30(0.3188187480f/M_PI), Q30(0.3825501204f/M_PI), Q30(0.4193951190f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0672341362f/M_PI), Q30(0.1045235619f/M_PI), Q30(0.1558904350f/M_PI), Q30(0.1824723780f/M_PI), Q30(0.1966800541f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0433459543f/M_PI), Q30(0.0672172382f/M_PI), Q30(0.0997167900f/M_PI), Q30(0.1162951663f/M_PI), Q30(0.1250736862f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0011053939f/M_PI), Q30(0.0017089852f/M_PI), Q30(0.0025254129f/M_PI), Q30(0.0029398468f/M_PI), Q30(0.0031597170f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0019607407f/M_PI), Q30(0.0030395309f/M_PI), Q30(0.0044951206f/M_PI), Q30(0.0052305623f/M_PI), Q30(0.0056152637f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0034913034f/M_PI), Q30(0.0054070661f/M_PI), Q30(0.0079917293f/M_PI), Q30(0.0092999367f/M_PI), Q30(0.0099875759f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0062100487f/M_PI), Q30(0.0096135242f/M_PI), Q30(0.0142110568f/M_PI), Q30(0.0165348612f/M_PI), Q30(0.0177587029f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0110366223f/M_PI), Q30(0.0170863140f/M_PI), Q30(0.0252620988f/M_PI), Q30(0.0293955617f/M_PI), Q30(0.0315726399f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0275881495f/M_PI), Q30(0.0427365713f/M_PI), Q30(0.0632618815f/M_PI), Q30(0.0736731067f/M_PI), Q30(0.0791663304f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0387469754f/M_PI), Q30(0.0600636788f/M_PI), Q30(0.0890387669f/M_PI), Q30(0.1037906483f/M_PI), Q30(0.1115923747f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0541138873f/M_PI), Q30(0.0839984417f/M_PI), Q30(0.1248718798f/M_PI), Q30(0.1458375156f/M_PI), Q30(0.1569785923f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0747506917f/M_PI), Q30(0.1163287833f/M_PI), Q30(0.1738867164f/M_PI), Q30(0.2038587779f/M_PI), Q30(0.2199459076f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1212290376f/M_PI), Q30(0.1903949380f/M_PI), Q30(0.2907958031f/M_PI), Q30(0.3466993868f/M_PI), Q30(0.3782821596f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1418247074f/M_PI), Q30(0.2240308374f/M_PI), Q30(0.3474813402f/M_PI), Q30(0.4202919006f/M_PI), Q30(0.4637607038f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1784276664f/M_PI), Q30(0.2856673002f/M_PI), Q30(0.4630723596f/M_PI), Q30(0.5971632004f/M_PI), Q30(0.7603877187f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1736015975f/M_PI), Q30(0.2773745656f/M_PI), Q30(0.4461984038f/M_PI), Q30(0.5666890144f/M_PI), Q30(0.6686112881f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1603866369f/M_PI), Q30(0.2549437582f/M_PI), Q30(0.4029446840f/M_PI), Q30(0.4980689585f/M_PI), Q30(0.5615641475f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1418247074f/M_PI), Q30(0.2240308374f/M_PI), Q30(0.3474813402f/M_PI), Q30(0.4202919006f/M_PI), Q30(0.4637607038f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1212290376f/M_PI), Q30(0.1903949380f/M_PI), Q30(0.2907958031f/M_PI), Q30(0.3466993868f/M_PI), Q30(0.3782821596f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.1011129096f/M_PI), Q30(0.1580764502f/M_PI), Q30(0.2387557179f/M_PI), Q30(0.2820728719f/M_PI), Q30(0.3058380187f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0747506917f/M_PI), Q30(0.1163287833f/M_PI), Q30(0.1738867164f/M_PI), Q30(0.2038587779f/M_PI), Q30(0.2199459076f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0541138873f/M_PI), Q30(0.0839984417f/M_PI), Q30(0.1248718798f/M_PI), Q30(0.1458375156f/M_PI), Q30(0.1569785923f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0387469754f/M_PI), Q30(0.0600636788f/M_PI), Q30(0.0890387669f/M_PI), Q30(0.1037906483f/M_PI), Q30(0.1115923747f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0275881495f/M_PI), Q30(0.0427365713f/M_PI), Q30(0.0632618815f/M_PI), Q30(0.0736731067f/M_PI), Q30(0.0791663304f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0195873566f/M_PI), Q30(0.0303316917f/M_PI), Q30(0.0448668823f/M_PI), Q30(0.0522258915f/M_PI), Q30(0.0561044961f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0110366223f/M_PI), Q30(0.0170863140f/M_PI), Q30(0.0252620988f/M_PI), Q30(0.0293955617f/M_PI), Q30(0.0315726399f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0062100487f/M_PI), Q30(0.0096135242f/M_PI), Q30(0.0142110568f/M_PI), Q30(0.0165348612f/M_PI), Q30(0.0177587029f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0034913034f/M_PI), Q30(0.0054070661f/M_PI), Q30(0.0079917293f/M_PI), Q30(0.0092999367f/M_PI), Q30(0.0099875759f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0019607407f/M_PI), Q30(0.0030395309f/M_PI), Q30(0.0044951206f/M_PI), Q30(0.0052305623f/M_PI), Q30(0.0056152637f/M_PI),
-      Q30(0.0000000000f/M_PI), Q30(0.0011053939f/M_PI), Q30(0.0017089852f/M_PI), Q30(0.0025254129f/M_PI), Q30(0.0029398468f/M_PI), Q30(0.0031597170f/M_PI)
-    };
-
-    static const int iid_par_dequant_c1[] = {
-        //iid_par_dequant_default
-        Q30(1.41198278375959f), Q30(1.40313815268360f), Q30(1.38687670404960f), Q30(1.34839972492648f),
-        Q30(1.29124937110028f), Q30(1.19603741667993f), Q30(1.10737240362323f), Q30(1),
-        Q30(0.87961716655242f), Q30(0.75464859232732f), Q30(0.57677990744575f), Q30(0.42640143271122f),
-        Q30(0.27671828230984f), Q30(0.17664462766713f), Q30(0.07940162697653f),
-        //iid_par_dequant_fine
-        Q30(1.41420649135832f), Q30(1.41419120222364f), Q30(1.41414285699784f), Q30(1.41399000859438f),
-        Q30(1.41350698548044f), Q30(1.41198278375959f), Q30(1.40977302262355f), Q30(1.40539479488545f),
-        Q30(1.39677960498402f), Q30(1.38005309967827f), Q30(1.34839972492648f), Q30(1.31392017367631f),
-        Q30(1.26431008149654f), Q30(1.19603741667993f), Q30(1.10737240362323f), Q30(1),
-        Q30(0.87961716655242f), Q30(0.75464859232732f), Q30(0.63365607219232f), Q30(0.52308104267543f),
-        Q30(0.42640143271122f), Q30(0.30895540465965f), Q30(0.22137464873077f), Q30(0.15768788954414f),
-        Q30(0.11198225164225f), Q30(0.07940162697653f), Q30(0.04469901562677f), Q30(0.02514469318284f),
-        Q30(0.01414142856998f), Q30(0.00795258154731f), Q30(0.00447211359449f),
-    };
-
-    static const int acos_icc_invq[] = {
-        Q31(0), Q31(0.178427635f/M_PI), Q31(0.28566733f/M_PI), Q31(0.46307236f/M_PI), Q31(0.59716315f/M_PI), Q31(0.78539816f/M_PI), Q31(1.10030855f/M_PI), Q31(1.57079633f/M_PI)
-    };
-    int iid, icc;
-
-    int k, m;
-    static const int8_t f_center_20[] = {
-        -3, -1, 1, 3, 5, 7, 10, 14, 18, 22,
-    };
-    static const int32_t f_center_34[] = {
-      Q31(  2/768.0),Q31(  6/768.0),Q31(10/768.0),Q31(14/768.0),Q31( 18/768.0),Q31( 22/768.0),Q31( 26/768.0),Q31(30/768.0),
-      Q31( 34/768.0),Q31(-10/768.0),Q31(-6/768.0),Q31(-2/768.0),Q31( 51/768.0),Q31( 57/768.0),Q31( 15/768.0),Q31(21/768.0),
-      Q31( 27/768.0),Q31( 33/768.0),Q31(39/768.0),Q31(45/768.0),Q31( 54/768.0),Q31( 66/768.0),Q31( 78/768.0),Q31(42/768.0),
-      Q31(102/768.0),Q31( 66/768.0),Q31(78/768.0),Q31(90/768.0),Q31(102/768.0),Q31(114/768.0),Q31(126/768.0),Q31(90/768.0)
-    };
-    static const int fractional_delay_links[] = { Q31(0.43f), Q31(0.75f), Q31(0.347f) };
-    const int fractional_delay_gain = Q31(0.39f);
-
-    for (pd0 = 0; pd0 < 8; pd0++) {
-        int pd0_re = (ipdopd_cos[pd0]+2)>>2;
-        int pd0_im = (ipdopd_sin[pd0]+2)>>2;
-        for (pd1 = 0; pd1 < 8; pd1++) {
-            int pd1_re = ipdopd_cos[pd1] >> 1;
-            int pd1_im = ipdopd_sin[pd1] >> 1;
-            for (pd2 = 0; pd2 < 8; pd2++) {
-                int shift, round;
-                int pd2_re = ipdopd_cos[pd2];
-                int pd2_im = ipdopd_sin[pd2];
-                int re_smooth = pd0_re + pd1_re + pd2_re;
-                int im_smooth = pd0_im + pd1_im + pd2_im;
-
-                SoftFloat pd_mag = av_int2sf(((ipdopd_cos[(pd0-pd1)&7]+8)>>4) + ((ipdopd_cos[(pd0-pd2)&7]+4)>>3) +
-                                               ((ipdopd_cos[(pd1-pd2)&7]+2)>>2) + 0x15000000, 28);
-                pd_mag = av_div_sf(FLOAT_1, av_sqrt_sf(pd_mag));
-                shift = 30 - pd_mag.exp;
-                round = 1 << (shift-1);
-                pd_re_smooth[pd0*64+pd1*8+pd2] = (int)(((int64_t)re_smooth * pd_mag.mant + round) >> shift);
-                pd_im_smooth[pd0*64+pd1*8+pd2] = (int)(((int64_t)im_smooth * pd_mag.mant + round) >> shift);
-            }
-        }
-    }
-
-    idx = 0;
-    for (iid = 0; iid < 46; iid++) {
-        int c1, c2;
-
-        c1 = iid_par_dequant_c1[iid];
-        if (iid < 15)
-          c2 = iid_par_dequant_c1[14-iid];
-        else
-          c2 = iid_par_dequant_c1[60-iid];
-
-        for (icc = 0; icc < 8; icc++) {
-            /*if (PS_BASELINE || ps->icc_mode < 3)*/{
-                int alpha, beta;
-                int ca, sa, cb, sb;
-
-                alpha = acos_icc_invq[icc];
-                beta = (int)(((int64_t)alpha * 1518500250 + 0x40000000) >> 31);
-                alpha >>= 1;
-                beta = (int)(((int64_t)beta * (c1 - c2) + 0x40000000) >> 31);
-                av_sincos_sf(beta + alpha, &sa, &ca);
-                av_sincos_sf(beta - alpha, &sb, &cb);
-
-                HA[iid][icc][0] = (int)(((int64_t)c2 * ca + 0x20000000) >> 30);
-                HA[iid][icc][1] = (int)(((int64_t)c1 * cb + 0x20000000) >> 30);
-                HA[iid][icc][2] = (int)(((int64_t)c2 * sa + 0x20000000) >> 30);
-                HA[iid][icc][3] = (int)(((int64_t)c1 * sb + 0x20000000) >> 30);
-            } /* else */ {
-                int alpha_int, gamma_int;
-                int alpha_c_int, alpha_s_int, gamma_c_int, gamma_s_int;
-
-                alpha_int = alpha_tab[idx];
-                gamma_int = gamma_tab[idx];
-
-                av_sincos_sf(alpha_int, &alpha_s_int, &alpha_c_int);
-                av_sincos_sf(gamma_int, &gamma_s_int, &gamma_c_int);
-
-                alpha_c_int = (int)(((int64_t)alpha_c_int * 1518500250 + 0x20000000) >> 30);
-                alpha_s_int = (int)(((int64_t)alpha_s_int * 1518500250 + 0x20000000) >> 30);
-
-                HB[iid][icc][0] = (int)(((int64_t)alpha_c_int * gamma_c_int + 0x20000000) >> 30);
-                HB[iid][icc][1] = (int)(((int64_t)alpha_s_int * gamma_c_int + 0x20000000) >> 30);
-                HB[iid][icc][2] = -(int)(((int64_t)alpha_s_int * gamma_s_int + 0x20000000) >> 30);
-                HB[iid][icc][3] = (int)(((int64_t)alpha_c_int * gamma_s_int + 0x20000000) >> 30);
-            }
-
-            if (icc < 5 || icc > 6)
-              idx++;
-        }
-    }
-
-    for (k = 0; k < NR_ALLPASS_BANDS20; k++) {
-        int theta;
-        int64_t f_center;
-        int c, s;
-
-        if (k < FF_ARRAY_ELEMS(f_center_20))
-          f_center = f_center_20[k];
-        else
-          f_center = (k << 3) - 52;
-
-        for (m = 0; m < PS_AP_LINKS; m++) {
-            theta = (int)(((int64_t)fractional_delay_links[m] * f_center + 8) >> 4);
-            av_sincos_sf(-theta, &s, &c);
-            Q_fract_allpass[0][k][m][0] = c;
-            Q_fract_allpass[0][k][m][1] = s;
-        }
-
-        theta = (int)(((int64_t)fractional_delay_gain * f_center + 8) >> 4);
-        av_sincos_sf(-theta, &s, &c);
-        phi_fract[0][k][0] = c;
-        phi_fract[0][k][1] = s;
-    }
-
-    for (k = 0; k < NR_ALLPASS_BANDS34; k++) {
-        int theta, f_center;
-        int c, s;
-
-        if (k < FF_ARRAY_ELEMS(f_center_34))
-            f_center = f_center_34[k];
-        else
-            f_center = ((int64_t)k << 26) - (53 << 25);
-
-        for (m = 0; m < PS_AP_LINKS; m++) {
-            theta = (int)(((int64_t)fractional_delay_links[m] * f_center + 0x10000000) >> 27);
-            av_sincos_sf(-theta, &s, &c);
-            Q_fract_allpass[1][k][m][0] = c;
-            Q_fract_allpass[1][k][m][1] = s;
-        }
-
-        theta = (int)(((int64_t)fractional_delay_gain * f_center + 0x10000000) >> 27);
-        av_sincos_sf(-theta, &s, &c);
-        phi_fract[1][k][0] = c;
-        phi_fract[1][k][1] = s;
-    }
-
-    make_filters_from_proto(f20_0_8,  g0_Q8,   8);
-    make_filters_from_proto(f34_0_12, g0_Q12, 12);
-    make_filters_from_proto(f34_1_8,  g1_Q8,   8);
-    make_filters_from_proto(f34_2_4,  g2_Q4,   4);
-}
-#endif /* CONFIG_HARDCODED_TABLES */
-
-#endif /* AACPS_FIXED_TABLEGEN_H */
--- a/libavcodec/aacps_float.c
+++ b/libavcodec/aacps_float.c
@@ -1,24 +0,0 @@
-/*
- * MPEG-4 Parametric Stereo decoding functions
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#define USE_FIXED 0
-
-#include "aacps.c"
--- a/libavcodec/aacps_tablegen.c
+++ b/libavcodec/aacps_tablegen.c
@@ -20,5 +20,74 @@
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

-#define USE_FIXED 0
-#include "aacps_tablegen_template.c"
+#include <stdlib.h>
+#define CONFIG_HARDCODED_TABLES 0
+#include "aacps_tablegen.h"
+#include "tableprint.h"
+
+void write_float_3d_array (const void *p, int b, int c, int d)
+{
+    int i;
+    const float *f = p;
+    for (i = 0; i < b; i++) {
+        printf("{\n");
+        write_float_2d_array(f, c, d);
+        printf("},\n");
+        f += c * d;
+    }
+}
+
+void write_float_4d_array (const void *p, int a, int b, int c, int d)
+{
+    int i;
+    const float *f = p;
+    for (i = 0; i < a; i++) {
+        printf("{\n");
+        write_float_3d_array(f, b, c, d);
+        printf("},\n");
+        f += b * c * d;
+    }
+}
+
+int main(void)
+{
+    ps_tableinit();
+
+    write_fileheader();
+
+    printf("static const float pd_re_smooth[8*8*8] = {\n");
+    write_float_array(pd_re_smooth, 8*8*8);
+    printf("};\n");
+    printf("static const float pd_im_smooth[8*8*8] = {\n");
+    write_float_array(pd_im_smooth, 8*8*8);
+    printf("};\n");
+
+    printf("static const float HA[46][8][4] = {\n");
+    write_float_3d_array(HA, 46, 8, 4);
+    printf("};\n");
+    printf("static const float HB[46][8][4] = {\n");
+    write_float_3d_array(HB, 46, 8, 4);
+    printf("};\n");
+
+    printf("static const DECLARE_ALIGNED(16, float, f20_0_8)[8][8][2] = {\n");
+    write_float_3d_array(f20_0_8, 8, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, f34_0_12)[12][8][2] = {\n");
+    write_float_3d_array(f34_0_12, 12, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, f34_1_8)[8][8][2] = {\n");
+    write_float_3d_array(f34_1_8, 8, 8, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, f34_2_4)[4][8][2] = {\n");
+    write_float_3d_array(f34_2_4, 4, 8, 2);
+    printf("};\n");
+
+    printf("static const DECLARE_ALIGNED(16, float, Q_fract_allpass)[2][50][3][2] = {\n");
+    write_float_4d_array(Q_fract_allpass, 2, 50, 3, 2);
+    printf("};\n");
+    printf("static const DECLARE_ALIGNED(16, float, phi_fract)[2][50][2] = {\n");
+    write_float_3d_array(phi_fract, 2, 50, 2);
+    printf("};\n");
+
+    return 0;
+}
--- a/libavcodec/aacps_tablegen_template.c
+++ b/libavcodec/aacps_tablegen_template.c
@@ -1,107 +0,0 @@
-/*
- * Generate a header file for hardcoded Parametric Stereo tables
- *
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <stdlib.h>
-#define CONFIG_HARDCODED_TABLES 0
-#include "aac_defines.h"
-
-#if USE_FIXED
-#define TYPE_NAME "int32_t"
-#define INT32FLOAT int32_t
-#define ARRAY_RENAME(x) write_int32_t_ ## x
-#define ARRAY_URENAME(x) write_uint32_t_ ## x
-#include "aacps_fixed_tablegen.h"
-#else
-#define TYPE_NAME "float"
-#define INT32FLOAT float
-#define ARRAY_RENAME(x) write_float_ ## x
-#define ARRAY_URENAME(x) write_float_ ## x
-#include "aacps_tablegen.h"
-#endif /* USE_FIXED */
-#include "tableprint.h"
-
-void ARRAY_RENAME(3d_array) (const void *p, int b, int c, int d)
-{
-    int i;
-    const INT32FLOAT *f = p;
-    for (i = 0; i < b; i++) {
-        printf("{\n");
-        ARRAY_URENAME(2d_array)(f, c, d);
-        printf("},\n");
-        f += c * d;
-    }
-}
-
-void ARRAY_RENAME(4d_array) (const void *p, int a, int b, int c, int d)
-{
-    int i;
-    const INT32FLOAT *f = p;
-    for (i = 0; i < a; i++) {
-        printf("{\n");
-        ARRAY_RENAME(3d_array)(f, b, c, d);
-        printf("},\n");
-        f += b * c * d;
-    }
-}
-
-int main(void)
-{
-    ps_tableinit();
-
-    write_fileheader();
-
-    printf("static const %s pd_re_smooth[8*8*8] = {\n", TYPE_NAME);
-    ARRAY_RENAME(array)(pd_re_smooth, 8*8*8);
-    printf("};\n");
-    printf("static const %s pd_im_smooth[8*8*8] = {\n", TYPE_NAME);
-    ARRAY_RENAME(array)(pd_im_smooth, 8*8*8);
-    printf("};\n");
-
-    printf("static const %s HA[46][8][4] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(HA, 46, 8, 4);
-    printf("};\n");
-    printf("static const %s HB[46][8][4] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(HB, 46, 8, 4);
-    printf("};\n");
-
-    printf("static const DECLARE_ALIGNED(16, %s, f20_0_8)[8][8][2] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(f20_0_8, 8, 8, 2);
-    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, %s, f34_0_12)[12][8][2] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(f34_0_12, 12, 8, 2);
-    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, %s, f34_1_8)[8][8][2] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(f34_1_8, 8, 8, 2);
-    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, %s, f34_2_4)[4][8][2] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(f34_2_4, 4, 8, 2);
-    printf("};\n");
-
-    printf("static const DECLARE_ALIGNED(16, %s, Q_fract_allpass)[2][50][3][2] = {\n", TYPE_NAME);
-    ARRAY_RENAME(4d_array)(Q_fract_allpass, 2, 50, 3, 2);
-    printf("};\n");
-    printf("static const DECLARE_ALIGNED(16, %s, phi_fract)[2][50][2] = {\n", TYPE_NAME);
-    ARRAY_RENAME(3d_array)(phi_fract, 2, 50, 2);
-    printf("};\n");
-
-    return 0;
-}
--- a/libavcodec/aacpsdata.c
+++ b/libavcodec/aacpsdata.c
@@ -157,7 +157,7 @@ static const int8_t k_to_i_34[] = {
    33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33, 33
 };

-static const INTFLOAT g1_Q2[] = {
-    Q31(0.0f),  Q31(0.01899487526049f), Q31(0.0f), Q31(-0.07293139167538f),
-    Q31(0.0f),  Q31(0.30596630545168f), Q31(0.5f)
+static const float g1_Q2[] = {
+    0.0f,  0.01899487526049f, 0.0f, -0.07293139167538f,
+    0.0f,  0.30596630545168f, 0.5f
 };
--- a/libavcodec/aacpsdsp.c
+++ b/libavcodec/aacpsdsp.c
@@ -0,0 +1,216 @@
+/*
+ * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+#include "libavutil/attributes.h"
+#include "aacpsdsp.h"
+
+static void ps_add_squares_c(float *dst, const float (*src)[2], int n)
+{
+    int i;
+    for (i = 0; i < n; i++)
+        dst[i] += src[i][0] * src[i][0] + src[i][1] * src[i][1];
+}
+
+static void ps_mul_pair_single_c(float (*dst)[2], float (*src0)[2], float *src1,
+                                 int n)
+{
+    int i;
+    for (i = 0; i < n; i++) {
+        dst[i][0] = src0[i][0] * src1[i];
+        dst[i][1] = src0[i][1] * src1[i];
+    }
+}
+
+static void ps_hybrid_analysis_c(float (*out)[2], float (*in)[2],
+                                 const float (*filter)[8][2],
+                                 int stride, int n)
+{
+    int i, j;
+
+    for (i = 0; i < n; i++) {
+        float sum_re = filter[i][6][0] * in[6][0];
+        float sum_im = filter[i][6][0] * in[6][1];
+
+        for (j = 0; j < 6; j++) {
+            float in0_re = in[j][0];
+            float in0_im = in[j][1];
+            float in1_re = in[12-j][0];
+            float in1_im = in[12-j][1];
+            sum_re += filter[i][j][0] * (in0_re + in1_re) -
+                      filter[i][j][1] * (in0_im - in1_im);
+            sum_im += filter[i][j][0] * (in0_im + in1_im) +
+                      filter[i][j][1] * (in0_re - in1_re);
+        }
+        out[i * stride][0] = sum_re;
+        out[i * stride][1] = sum_im;
+    }
+}
+
+static void ps_hybrid_analysis_ileave_c(float (*out)[32][2], float L[2][38][64],
+                                        int i, int len)
+{
+    int j;
+
+    for (; i < 64; i++) {
+        for (j = 0; j < len; j++) {
+            out[i][j][0] = L[0][j][i];
+            out[i][j][1] = L[1][j][i];
+        }
+    }
+}
+
+static void ps_hybrid_synthesis_deint_c(float out[2][38][64],
+                                        float (*in)[32][2],
+                                        int i, int len)
+{
+    int n;
+
+    for (; i < 64; i++) {
+        for (n = 0; n < len; n++) {
+            out[0][n][i] = in[i][n][0];
+            out[1][n][i] = in[i][n][1];
+        }
+    }
+}
+
+static void ps_decorrelate_c(float (*out)[2], float (*delay)[2],
+                             float (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
+                             const float phi_fract[2], const float (*Q_fract)[2],
+                             const float *transient_gain,
+                             float g_decay_slope,
+                             int len)
+{
+    static const float a[] = { 0.65143905753106f,
+                               0.56471812200776f,
+                               0.48954165955695f };
+    float ag[PS_AP_LINKS];
+    int m, n;
+
+    for (m = 0; m < PS_AP_LINKS; m++)
+        ag[m] = a[m] * g_decay_slope;
+
+    for (n = 0; n < len; n++) {
+        float in_re = delay[n][0] * phi_fract[0] - delay[n][1] * phi_fract[1];
+        float in_im = delay[n][0] * phi_fract[1] + delay[n][1] * phi_fract[0];
+        for (m = 0; m < PS_AP_LINKS; m++) {
+            float a_re                = ag[m] * in_re;
+            float a_im                = ag[m] * in_im;
+            float link_delay_re       = ap_delay[m][n+2-m][0];
+            float link_delay_im       = ap_delay[m][n+2-m][1];
+            float fractional_delay_re = Q_fract[m][0];
+            float fractional_delay_im = Q_fract[m][1];
+            float apd_re = in_re;
+            float apd_im = in_im;
+            in_re = link_delay_re * fractional_delay_re -
+                    link_delay_im * fractional_delay_im - a_re;
+            in_im = link_delay_re * fractional_delay_im +
+                    link_delay_im * fractional_delay_re - a_im;
+            ap_delay[m][n+5][0] = apd_re + ag[m] * in_re;
+            ap_delay[m][n+5][1] = apd_im + ag[m] * in_im;
+        }
+        out[n][0] = transient_gain[n] * in_re;
+        out[n][1] = transient_gain[n] * in_im;
+    }
+}
+
+static void ps_stereo_interpolate_c(float (*l)[2], float (*r)[2],
+                                    float h[2][4], float h_step[2][4],
+                                    int len)
+{
+    float h0 = h[0][0];
+    float h1 = h[0][1];
+    float h2 = h[0][2];
+    float h3 = h[0][3];
+    float hs0 = h_step[0][0];
+    float hs1 = h_step[0][1];
+    float hs2 = h_step[0][2];
+    float hs3 = h_step[0][3];
+    int n;
+
+    for (n = 0; n < len; n++) {
+        //l is s, r is d
+        float l_re = l[n][0];
+        float l_im = l[n][1];
+        float r_re = r[n][0];
+        float r_im = r[n][1];
+        h0 += hs0;
+        h1 += hs1;
+        h2 += hs2;
+        h3 += hs3;
+        l[n][0] = h0 * l_re + h2 * r_re;
+        l[n][1] = h0 * l_im + h2 * r_im;
+        r[n][0] = h1 * l_re + h3 * r_re;
+        r[n][1] = h1 * l_im + h3 * r_im;
+    }
+}
+
+static void ps_stereo_interpolate_ipdopd_c(float (*l)[2], float (*r)[2],
+                                           float h[2][4], float h_step[2][4],
+                                           int len)
+{
+    float h00  = h[0][0],      h10  = h[1][0];
+    float h01  = h[0][1],      h11  = h[1][1];
+    float h02  = h[0][2],      h12  = h[1][2];
+    float h03  = h[0][3],      h13  = h[1][3];
+    float hs00 = h_step[0][0], hs10 = h_step[1][0];
+    float hs01 = h_step[0][1], hs11 = h_step[1][1];
+    float hs02 = h_step[0][2], hs12 = h_step[1][2];
+    float hs03 = h_step[0][3], hs13 = h_step[1][3];
+    int n;
+
+    for (n = 0; n < len; n++) {
+        //l is s, r is d
+        float l_re = l[n][0];
+        float l_im = l[n][1];
+        float r_re = r[n][0];
+        float r_im = r[n][1];
+        h00 += hs00;
+        h01 += hs01;
+        h02 += hs02;
+        h03 += hs03;
+        h10 += hs10;
+        h11 += hs11;
+        h12 += hs12;
+        h13 += hs13;
+
+        l[n][0] = h00 * l_re + h02 * r_re - h10 * l_im - h12 * r_im;
+        l[n][1] = h00 * l_im + h02 * r_im + h10 * l_re + h12 * r_re;
+        r[n][0] = h01 * l_re + h03 * r_re - h11 * l_im - h13 * r_im;
+        r[n][1] = h01 * l_im + h03 * r_im + h11 * l_re + h13 * r_re;
+    }
+}
+
+av_cold void ff_psdsp_init(PSDSPContext *s)
+{
+    s->add_squares            = ps_add_squares_c;
+    s->mul_pair_single        = ps_mul_pair_single_c;
+    s->hybrid_analysis        = ps_hybrid_analysis_c;
+    s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
+    s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
+    s->decorrelate            = ps_decorrelate_c;
+    s->stereo_interpolate[0]  = ps_stereo_interpolate_c;
+    s->stereo_interpolate[1]  = ps_stereo_interpolate_ipdopd_c;
+
+    if (ARCH_ARM)
+        ff_psdsp_init_arm(s);
+    if (ARCH_MIPS)
+        ff_psdsp_init_mips(s);
+}
--- a/libavcodec/aacpsdsp.h
+++ b/libavcodec/aacpsdsp.h
@@ -21,37 +21,34 @@
 #ifndef LIBAVCODEC_AACPSDSP_H
 #define LIBAVCODEC_AACPSDSP_H

-#include "aac_defines.h"
-
 #define PS_QMF_TIME_SLOTS 32
 #define PS_AP_LINKS 3
 #define PS_MAX_AP_DELAY 5

 typedef struct PSDSPContext {
-    void (*add_squares)(INTFLOAT *dst, const INTFLOAT (*src)[2], int n);
-    void (*mul_pair_single)(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
+    void (*add_squares)(float *dst, const float (*src)[2], int n);
+    void (*mul_pair_single)(float (*dst)[2], float (*src0)[2], float *src1,
                            int n);
-    void (*hybrid_analysis)(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
-                            const INTFLOAT (*filter)[8][2],
+    void (*hybrid_analysis)(float (*out)[2], float (*in)[2],
+                            const float (*filter)[8][2],
                            int stride, int n);
-    void (*hybrid_analysis_ileave)(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
+    void (*hybrid_analysis_ileave)(float (*out)[32][2], float L[2][38][64],
                                   int i, int len);
-    void (*hybrid_synthesis_deint)(INTFLOAT out[2][38][64], INTFLOAT (*in)[32][2],
+    void (*hybrid_synthesis_deint)(float out[2][38][64], float (*in)[32][2],
                                   int i, int len);
-    void (*decorrelate)(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
-                        INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
-                        const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
-                        const INTFLOAT *transient_gain,
-                        INTFLOAT g_decay_slope,
+    void (*decorrelate)(float (*out)[2], float (*delay)[2],
+                        float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
+                        const float phi_fract[2], const float (*Q_fract)[2],
+                        const float *transient_gain,
+                        float g_decay_slope,
                        int len);
-    void (*stereo_interpolate[2])(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
-                                  INTFLOAT h[2][4], INTFLOAT h_step[2][4],
+    void (*stereo_interpolate[2])(float (*l)[2], float (*r)[2],
+                                  float h[2][4], float h_step[2][4],
                                  int len);
 } PSDSPContext;

-void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s);
+void ff_psdsp_init(PSDSPContext *s);
 void ff_psdsp_init_arm(PSDSPContext *s);
 void ff_psdsp_init_mips(PSDSPContext *s);
-void ff_psdsp_init_x86(PSDSPContext *s);

 #endif /* LIBAVCODEC_AACPSDSP_H */
--- a/libavcodec/aacpsdsp_fixed.c
+++ b/libavcodec/aacpsdsp_fixed.c
@@ -1,23 +0,0 @@
-/*
- * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#define USE_FIXED 1
-
-#include "aacpsdsp_template.c"
--- a/Show More
+++ b/Show More
@@ -1 +1 @@
 .8.2
 .7.2