Add all seven test bitstreams of Coding Technologies "aacPlus Decoder
Check Package". The streams cover different ways to signal SBR and PS
in different formats.
The qatar implementation makes no sense.
a muxer without timestamps is constant fps thus needs vsync.
the crc/mp5 are special cases that have timestamps yet allow any
nonsensical timestamps.
raw (yuv/rgb) video is constant fps thus needs vsync too.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libavutil: add utility functions to simplify allocation of audio buffers.
libavutil: add planar sample formats and av_sample_fmt_is_planar()
avconv: fix segfault at EOF with delayed pictures
pcmdec: remove unneeded resetting of samples pointer
avconv: remove a now unused parameter from output_packet().
avconv: formatting fixes in output_packet()
avconv: declare some variables in blocks where they are used
avconv: use the same behavior when decoding audio/video/subs
bethsoftvideo: return proper consumed size for palette packets.
cdg: skip packets that don't contain a cdg command.
crcenc: add flags
avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats.
tiffenc: add a private option for selecting compression algorithm
md5enc: add flags
ARM: remove needless .text/.align directives
Conflicts:
doc/APIchanges
libavcodec/tiffenc.c
libavutil/avutil.h
libavutil/samplefmt.c
libavutil/samplefmt.h
tests/ref/fate/bethsoft-vid
tests/ref/fate/cdgraphics
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/nuv
tests/ref/fate/tiertex-seq
tests/ref/fate/tscc-32bit
tests/ref/fate/vmnc-32bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framecrc, as it prints dts.
Many FATE changes, because avconv is no longer duplicating frames in
those tests.
Also added -vsync 0 for some tests to prevent avconv from dropping
frames until it can be fixed more properly.
AVFMT_NOTIMESTAMPS for md5, as it ignores the timestamps.
AVFMT_VARIABLE_FPS for framemd5, as it prints dts.
-vsync 0 for the vp8 test is needed because with vsync 2 the timestamp
guessing code gets confused by an altref frame that is never displayed
and drops a frame later.
* qatar/master: (22 commits)
aacdec: Fix PS in ADTS.
avconv: Consistently use PIX_FMT_NONE.
dsputil: use cpuflags in x86 emu_edge_core
dsputil: use movups instead of movdqu in ff_emu_edge_core_sse()
wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
mov: Remove some redundant and obsolete comments.
Add libavutil/mathematics.h #includes for INFINITY
doxy: structure libavformat groups
doxy: introduce an empty structure in libavcodec
doxy: provide a start page and document libavutil
doxy: cleanup pixfmt.h
regtest: split video encode/decode tests into individual targets
ARM: add explicit .arch and .fpu directives to asm.S
pthread: do not touch has_b_frames
avconv: cleanup the transcoding loop in output_packet().
avconv: split subtitle transcoding out of output_packet().
avconv: split video transcoding out of output_packet().
avconv: split audio transcoding out of output_packet().
avconv: reindent.
avconv: move streamcopy-only code out of decoding loop.
...
Conflicts:
avconv.c
libavcodec/aaccoder.c
libavcodec/pthread.c
libavcodec/version.h
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/mem.h
tests/ref/vsynth1/dv
tests/ref/vsynth1/mpeg2thread
tests/ref/vsynth2/dv
tests/ref/vsynth2/mpeg2thread
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
libopencore-amr: check output buffer size before decoding
libopencore-amr: remove unneeded buf_size==0 check.
libopencore-amr: remove unneeded frame_count field.
aac_latm: remove unneeded check for zero-size packet.
pcmdec: fix output buffer size check by calculating the actual output size prior to decoding.
pcmdec: move codec-specific variable declarations to the corresponding codec blocks.
pcmdec: return buf_size instead of src-buf.
avcodec: remove the Zork PCM encoder.
pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8.
pcmenc: remove unneeded sample_fmt check.
pcmdec: move number of channels check to pcm_decode_init()
pcmdec: remove unnecessary check for sample_fmt change
pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets the sample size.
pcmdec: do not needlessly set *data_size to 0
alacdec: remove unneeded NULL or zero-size packet checks.
alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO()
alacdec: ask for a sample for unsupported sample depths.
alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels
alacdec: move some declarations to the top of the function
alacdec: always use get_sbits_long() for uncompressed samples
...
Conflicts:
libavcodec/pcm.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore
the encoder based on the decoder is also incorrect. There is no good reason to
keep the encoder.
* qatar/master:
id3v2: fix doxy comment - 'machine byte order' makes no sense on char arrays
VC1: restore mistakenly removed code
twinvq: check output buffer size before decoding
twinvq: return an error when the packet size is too small
lavf: export some forgotten symbols with non-av prefixes.
swscale: update altivec yuv2planeX asm to new per-plane API.
swscale: make yuv2yuvX_10_sse2/avx 8/9/16-bits aware.
yuv2planeX10 SIMD
swscale: decide whether to use yuv2plane1/X on a per-plane basis.
swscale: reintroduce full precision in 16-bit output.
Split up yuv2yuvX functions
Split out yuv2yuv1 luma and chroma in order to make them generic DSP functions
lavc: replace references to deprecated AVCodecContext.error_recognition to use AVCodecContext.err_recognition
lavc: translate non-flag-based er options into flag-based ef options at codec open
add -err_filter AVOptions to access flag-based error recognition
h264_weight: initialize "height" function argument properly.
presets: spelling error in libvpx 1080p50_60
avplay: fix fullscreen behaviour with SDL 1.2.14 on Mac OS X
Conflicts:
ffplay.c
libavformat/libavformat.v
libswscale/swscale.c
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
flvdec: Do not call parse_keyframes_index with a NULL stream
libspeexdec: include system headers before local headers
libspeexdec: return meaningful error codes
libspeexdec: cosmetics: reindent
libspeexdec: decode one frame at a time.
swscale: fix signed shift overflows in ff_yuv2rgb_c_init_tables()
Move timefilter code from lavf to lavd.
mov: add support for hdvd and pgapmetadata atoms
mov: rename function _stik, some indentation cosmetics
mov: rename function _int8 to remove ambiguity, some indentation cosmetics
mov: parse the gnre atom
mp3on4: check for allocation failures in decode_init_mp3on4()
mp3on4: create a separate flush function for MP3onMP4.
mp3on4: ensure that the frame channel count does not exceed the codec channel count.
mp3on4: set channel layout
mp3on4: fix the output channel order
mp3on4: allocate temp buffer with av_malloc() instead of on the stack.
mp3on4: copy MPADSPContext from first context to all contexts.
fmtconvert: port float_to_int16_interleave() 2-channel x86 inline asm to yasm
fmtconvert: port int32_to_float_fmul_scalar() x86 inline asm to yasm
...
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_ps.c
libavcodec/h264dsp_template.c
libavcodec/h264idct_template.c
libavcodec/h264pred.c
libavcodec/h264pred_template.c
libavcodec/x86/h264dsp_mmx.c
libavdevice/Makefile
libavdevice/jack_audio.c
libavformat/Makefile
libavformat/flvdec.c
libavformat/flvenc.c
libavutil/pixfmt.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (47 commits)
lavc: hide private symbols.
lavc: deprecate img_get_alpha_info().
lavc: use avpriv_ prefix for ff_toupper4.
lavc: use avpriv_ prefix for ff_copy_bits and align_put_bits.
lavc: use avpriv_ prefix for ff_ac3_parse_header.
lavc: use avpriv_ prefix for ff_frame_rate_tab.
lavc: rename ff_find_start_code to avpriv_mpv_find_start_code
lavc: use avpriv_ prefix for ff_split_xiph_headers.
lavc: use avpriv_ prefix for ff_dirac_parse_sequence_header.
lavc: use avpriv_ prefix for some dv symbols used in lavf.
lavc: use avpriv_ prefix for some flac symbols used in lavf.
lavc: use avpriv_ prefix for some mpeg4audio symbols used in lavf.
lavc: use avpriv_ prefix for some mpegaudio symbols used in lavf.
lavc: use avpriv_ prefix for ff_aac_parse_header().
lavf: hide private symbols.
lavf: use avpriv_ prefix for some dv functions.
lavf: use avpriv_ prefix for ff_new_chapter().
avcodec: add CODEC_CAP_DELAY note to avcodec_decode_audio3() documentation
avcodec: clarify the CODEC_CAP_DELAY note in avcodec_decode_video2()
avcodec: clarify documentation of CODEC_CAP_DELAY
...
Conflicts:
configure
doc/general.texi
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/dv.c
libavcodec/dvdata.c
libavcodec/dvdata.h
libavcodec/libspeexenc.c
libavcodec/mpegvideo.c
libavcodec/version.h
libavformat/avidec.c
libavformat/dv.c
libavformat/dv.h
libavformat/flvenc.c
libavformat/mov.c
libavformat/mp3enc.c
libavformat/oggparsespeex.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
enable CODEC_CAP_DELAY to flush any remaining frames in the buffer.
Stop decoding when the FN_QUIT command is found so that a trailing seek table
isn't decoded as a normal frame.
decode all channels in the same call to avcodec_decode_audio3() so that
decoding will not stop after the first channel of the last frame.
Updated FATE reference. More valid audio is now decoded.
* qatar/master:
sunrast: Check for out of bounds reads
lavc: rename AV_ER_* options to AV_EF_* and rename AGGRESSIVE to BUFFER
lavc: replace API-bump-triggered AVCodecContext field change with shorter, non-conflicting name
Add libvpx presets.
doc/avtools: add forgotten part to stream specifiers description
swscale: prevent overflow during initialization
g722: Add a fate test for the encoder
fate: Add a target for creating a 16000 Hz mono synthetic audio file
macosx: use the default surface on newer sdl
Conflicts:
ffplay.c
libavcodec/avcodec.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
prores: get correct size for coded V plane if alpha is present
prores: do not set pixel format on codec init
pthread: prevent updating AVCodecContext from itself in frame_thread_free
pthread: copy coded frame dimensions in update_context_from_thread
vp8: prevent read from uninitialized memory in decode_mvs
vp8: force reallocation in update_thread_context after frame size change
vp8: fix return value if update_dimensions fails
matroskadec: fix out of bounds write
adpcmdec: calculate actual number of output samples for each decoder.
adpcmdec: check remaining buffer size before decoding next block in the ADPCM IMA WAV decoder.
adpcmdec: do not terminate early in ADPCM IMA Duck DK3 decoder.
adpcmdec: remove unneeded buf_size==0 check.
adpcmdec: remove unneeded zeroing of *data_size
dnxhdenc: fixed signed multiplication overflow
Conflicts:
tests/ref/fate/prores-alpha
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The pixel format is not known until the frame header is parsed.
Guessing it here only causes trouble for the caller if the guess
turns out to be wrong (and actually causes very wrong output by
avconv/avplay).
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
proresdsp: fix function prototypes.
prores-idct: fix overflow in c code.
fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
prores: add missing feature warning for alpha
mov: 10l: Terminate string with 0 not '0'
mov: Prevent illegal writes when chapter titles are very short.
prores: add appropriate -fix_fmt parameter to FATE command
riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER
Conflicts:
libavcodec/avcodec.h
libavformat/mov.c
tests/fate/prores.mak
tests/ref/acodec/g726
tests/ref/fate/prores-alpha
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* qatar/master: (22 commits)
prores: add FATE tests
id3v2: reduce the scope of some non-globally-used symbols/structures
id3v2: cosmetics: move some declarations before the places they are used
shorten: remove the flush function.
shn: do not allow seeking in the raw shn demuxer.
avformat: add AVInputFormat flag AVFMT_NO_BYTE_SEEK.
avformat: update AVInputFormat allowed flags
avformat: don't unconditionally call ff_read_frame_flush() when trying to seek.
truespeech: use sizeof() instead of hardcoded sizes
truespeech: remove unneeded variable, 'consumed'
truespeech: simplify truespeech_read_frame() by using get_bits()
truespeech: decode directly to output buffer instead of a temp buffer
truespeech: check to make sure channels == 1
truespeech: check for large enough output buffer rather than truncating output
truespeech: remove unneeded zero-size packet check.
mlpdec: return meaningful error codes instead of -1
mlpdec: remove unnecessary wrapper function
mlpdec: only calculate output size once
mlpdec: validate that the reported channel count matches the actual output channel count
pcm: reduce pointer type casting
...
Conflicts:
libavformat/avformat.h
libavformat/id3v2.c
libavformat/id3v2.h
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpeg4dec: use unsigned type for startcode in ff_mpeg4_decode_picture_header
mpeg124: use sign_extend() function
ac3dec: use get_sbits() instead of manually sign-extending
4xm: fix signed overflow
wmavoice: fix a signed overflow
mpegvideo_enc: fix a signed overflow
crc: fix signed overflow
fate: run avconv with -nostats flag
avtools: add -v as alias for -loglevel
avconv: always print stats with AV_LOG_INFO
doc/avconv: add forgotten output/per-stream info to -filter
avconv: add -stats option to enable/disable printing encoding progress
avconv: in output_packet() don't set decoded_data_size for video.
avconv: remove pointless always true condition
avconv: factorize common code in transcode_init()
zmbv: remove memcpy() of decoded frame
mpeg12enc: use sign_extend() function
h264pred: use unsigned types for pixel values, fix signed overflows
h264: fix signed overflows in x*0x01010101 expressions
h264pred: remove unused variables
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (24 commits)
mpegps: Use av_get_packet() instead of poorly emulating it.
motionpixels: decode only the 111 complete frames for fate
mpc8: Check out of bound bands limit
xan: Prevent NULL dereference with missing palette
xan: Check for out of bound reads in xan_huffman_decode()
xan: Fixed out of bound accesses in xan_unpack()
motionpixels: Prevent calling init_vlc() with invalid parameters
shorten: Fix out of bound writes in fix_bitshift()
dsicinav: Check for out of bounds writes
tiertexseqv: Check for out of bound reads
quickdraw: Check for out of bound reads
dsicinav: Check for out of bounds reads
motionpixels: Fix the size of workspace buffers
motionpixels: Clear FF_INPUT_BUFFER_PADDING_SIZE bytes at the end of the temporary buffer
wmavoice: Check for corrupted extra data
wmavoice: Check for out of bound writes
xan: Prevent NULL dereferences with missing reference frame
bink: Prevent NULL dereferences with missing reference frame
wavpack: Reset internal state on corrupted blocks
wmapro: Validate the number of audio channels before using it
...
Conflicts:
libavcodec/h264.c
libavcodec/xan.c
tests/ref/fate/motionpixels
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
avconv: Reformat s16 volume adjustment.
ARM: NEON optimised vector_fmac_scalar()
dca: use vector_fmac_scalar from dsputil
dsputil: add vector_fmac_scalar()
latmenc: Fix private options
vf_unsharp: store hsub/vsub in the filter context
vf_unsharp: adopt a more natural order of params in apply_unsharp()
vf_unsharp: rename method "unsharpen" to "apply_unsharp"
vf_scale: apply the same transform to the aspect during init that is applied per frame
vf_pad: fix "vsub" variable value computation
vf_scale: add a "sar" variable
lavfi: fix realloc size computation in avfilter_add_format()
vsrc_color: use internal timebase
lavfi: fix signature for avfilter_graph_parse() and avfilter_graph_config()
graphparser: prefer void * over AVClass * for log contexts
avfiltergraph: use meaningful error codes
avconv: Initialize return value for codec copy path.
fate: use 'run' helper for seek-test
fate: remove seek-mpeg2reuse test
Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
...
Conflicts:
doc/filters.texi
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/vf_scale.c
libavfilter/vsrc_color.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Makes the code less obfuscated and fixes encoding one video stream to
several outputs.
Also use avcodec_alloc_frame() instead of allocating AVFrame on stack.
Breaks me_threshold in avconv, as motion vectors aren't passed through
lavfi. They could be copied manually, but I don't think this misfeature
is useful enough to justify ugly hacks.
* qatar/master: (21 commits)
fate: allow testing with libavfilter disabled
x86: XOP/FMA4 CPU detection support
ws_snd: misc cosmetic clean-ups
ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead.
ws_snd: use memcpy() and memset() instead of loops
ws_snd: use samples pointer for loop termination instead of a separate iterator variable.
ws_snd: make sure number of channels is 1
ws_snd: add some checks to prevent buffer overread or overwrite.
ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
flacdec: fix buffer size checking in get_metadata_size()
rtp: Simplify ff_rtp_get_payload_type
rtpenc: Add a payload type private option
rtp: Correct ff_rtp_get_payload_type documentation
avconv: replace all fprintf() by av_log().
avconv: change av_log verbosity from ERROR to FATAL for fatal errors.
cmdutils: replace fprintf() by av_log()
avtools: parse loglevel before all the other options.
oggdec: add support for Xiph's CELT codec
sol: return error if av_get_packet() fails.
cosmetics: reindent and pretty-print
...
Conflicts:
avconv.c
cmdutils.c
libavcodec/avcodec.h
libavcodec/version.h
libavformat/oggparsecelt.c
libavformat/utils.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtp: factorize dynamic payload type fallback
flvdec: Ignore the index if it's from a creator known to be different
cmdutils: move grow_array out of #if CONFIG_AVFILTER
avconv: actually set InputFile.rate_emu
ratecontrol: update last_qscale_for sooner
Fix unnecessary shift with 9/10bit vertical scaling
prores: mark prores as intra-only in libavformat/utils.c:is_intra_only()
prores: return more meaningful error values
prores: improve error message wording
prores: cosmetics: prettyprinting, drop useless parentheses
prores: lowercase AVCodec name entry
Conflicts:
cmdutils.c
libavcodec/proresdec_lgpl.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Add LATM demuxer
avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
8svx/iff: fix decoding of compressed stereo 8svx files.
8svx: log an error message if output buffer is too small
8svx: check packet size before reading the initial sample value.
8svx: output 8-bit samples instead of 16-bit.
8svx: split delta decoding into a separate function.
mp4: Don't read an empty Decoder Config Descriptor
fate.sh: Ignore errors from rm command during cleanup.
fate.sh: Run git-pull in quiet mode to avoid console spam.
Apple ProRes decoder
rtmp: Make the input FLV parser handle data cut at any point
rv34: Check for invalid slices offsets
eval: test isnan(sqrt(-1)) instead of just sqrt(-1)
Conflicts:
Changelog
libavcodec/8svx.c
libavcodec/proresdec.c
libavcodec/version.h
libavformat/iff.c
libavformat/version.h
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The install directory being deleted might not be present if the build failed.
This can lead to annoying error output if the FATE client is run as a cronjob.
Similar to libswscale this does resampling and format convertion, just for audio
instead of video.
changing sampling rate, sample formats, channel layouts and sample packing all
in one with a very simple public interface.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavfi: add select filter
oggdec: fix out of bound write in the ogg demuxer
movenc: create an alternate group for each media type
lavd: add libcdio-paranoia input device for audio CD grabbing
rawdec: refactor private option for raw video demuxers
pcmdec: use unique classes for all pcm demuxers.
rawdec: g722 is always 1 channel/16kHz
Conflicts:
Changelog
configure
doc/filters.texi
libavdevice/avdevice.h
libavfilter/avfilter.h
libavfilter/vf_select.c
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This uses the RIFF header stored size to figure out the expected AVI file size, instead
of the actual file. To work fully it requires handling failed avio_seek() instead
of assuming they always succeed.
Some fate file has been cut off and contains half a frame at the end which previously
was not output during demuxing. This frame is now output to encoder, thus fate
diff update.
Merge video_qscale (set by -qscale) and audio_qscale (set by -aq) into
one 'qscale' field in the options context. Add a shortcut -q for -qscale
and make -aq an alias for -q:a.
* qatar/master:
ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
wavpack: fix wrong return value in wavpack_decode_block()
avconv: fix parsing metadata specifiers.
fate: use +frame+slice named constants instead of '3'
mpeg12: propagate more real return values through chunk decode error return and fix some indentation
wavpack: use context reset in appropriate places
avconv: move mux_preload and mux_max_delay to options context
avconv: move bitstream filters to options context.
avconv: move rate_emu to options context.
avconv: move max_frames to options context.
avconv: move metadata to options context.
avconv: move ts scale to options context.
avconv: move chapter maps to options context.
avconv: move metadata maps to options context.
avconv: move codec_names to options context.
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavc: fix type for thread_type option
avconv: move format to options context
avconv: move limit_filesize to options context
avconv: move start_time, recording_time and input_ts_offset to options context
avconv: add a context for options.
cmdutils: allow storing per-stream/chapter/.... options in a generic way
cmdutils: split per-option code out of parse_options().
cmdutils: add support for caller-provided option context.
cmdutils: declare only one pointer type in OptionDef
cmdutils: move grow_array() from avconv to cmdutils.
cmdutils: move exit_program() declaration to cmdutils from avconv
http: Consider the stream as seekable if the reply contains Accept-Ranges: bytes
nutenc: add namespace to the api facing functions
Conflicts:
avconv.c
cmdutils.c
cmdutils.h
ffmpeg.c
ffplay.c
ffprobe.c
ffserver.c
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fifo: add FIFO API test program, and fate test
fifo: add av_fifo_peek2(), and deprecate av_fifo_peek()
postprocess.c: filter name needs to be double 0 terminated
doxygen: fix wrong comment syntax, //< vs. ///<
doxygen: drop pointless star from pointer variable names
Replace deprecated av_find_stream_info() by avformat_find_stream_info().
xmv: eliminate superfluous zeroing of zero data
configure: fix typo in avconv dependency list
Conflicts:
configure
doc/APIchanges
libavutil/Makefile
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libx264: only use ABR mode when the user explicitly set bitrate.
libx264: use medium preset by default.
mp2 encoder: make 128k the default bitrate.
movenc: use libx264 by default when possible for mov, mp4 and psp
avienc: saner default audio codec.
matroskaenc: saner default codecs.
avplay: add examples of how to specify size/pixel format through private options
lavc: add A|E|D flags to "ac" and "ar" options
Conflicts:
doc/ffplay.texi
libavcodec/libx264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avconv: Fix spelling errors.
aac: Only set sample rate and object type from ADTS if output hasn't been configured.
aac: Set SBR and PS to unsignalled during headerless and ADTS initialization.
aac: Only output configure if audio was found.
avconv: save two levels of indentation in flush_encoders()
avconv: factor flushing encoders out of output_packet().
avconv: factor out initializing input streams.
avconv: remove -intra option.
avconv: reset streamid_map between output files.
avconv: make timer_start a local var in transcode().
avconv: cosmetics, move OutputStream.
avconv: remove two unused macros.
Conflicts:
avconv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
h264: hide reference frame errors unless requested
swscale: split hScale() function pointer into h[cy]Scale().
Move clipd macros to x86util.asm.
avconv: reindent.
avconv: rescue poor abused start_time global.
avconv: rescue poor abused recording_time global.
avconv: merge two loops in output_packet().
avconv: fix broken indentation.
avconv: get rid of the arbitrary MAX_FILES limit.
avconv: get rid of the output_streams_for_file vs. ost_table schizophrenia
avconv: add a wrapper for output AVFormatContexts and merge output_opts into it
avconv: make itsscale syntax consistent with other options.
avconv: factor out adding input streams.
avconv: Factorize combining auto vsync with format.
avconv: Factorize video resampling.
avconv: Don't unnecessarily convert ipts to a double.
ffmpeg: remove unsed variable nopts
RV3/4 parser: remove unused variable 'off'
add XMV demuxer
rmdec: parse FPS in RealMedia properly
...
Conflicts:
avconv.c
libavformat/version.h
libswscale/swscale.c
tests/ref/fate/lmlm4-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
First, container stores only DTS and not PTS as it was believed.
Second, multiple frames in a packet store timestamp instead of position
after the frame length.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Its only reason for its existence was ffmpeg's inability to properly assign
AVOptions to streams. Now this is not a problem anymore, so 'ab' should
go.
* qatar/master:
swscale: add dithering to yuv2yuvX_altivec_real
rv34: free+allocate buffer instead of reallocating it to preserve alignment
h264: add missing brackets.
swscale: use 15-bit intermediates for 9/10-bit scaling.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b2c087871dafc7d030b2d48457ddff597dfd4925':
Move x86util.asm from libavcodec/ to libavutil/.
Move x86inc.asm to libavutil/.
APIchanges: note error_recognition in lavf
lavf: add support for error_recognition, use it in avidec, and bump minor API version
avconv: change semantics of -map
avconv: get rid of new* options.
cmdutils: allow precisely specifying a stream for AVOptions.
configure: add missing CFLAGS to fix building on the HURD
libx264: Include hint for possible values for configuring libx264
cmdutils: allow ':'-separated modifiers in option names.
avconv: make -map_metadata work consistently with the other options
avconv: remove deprecated options.
avconv: make -map_chapters accept only the input file index.
Make a copy of ffmpeg under a new name -- avconv.
ffmpeg: add a warning stating that the program is deprecated.
Add weighted motion compensation for RV40 B-frames
RV3/4: calculate B-frame motion weights once per frame
Move RV3/4-specific DSP functions into their own context
mjpeg: propagate decode errors from ff_mjpeg_decode_sos and ff_mjpeg_decode_dqt
h264: notice memory allocation failure
Conflicts:
.gitignore
Makefile
cmdutils.c
configure
doc/ffplay.texi
doc/ffprobe.texi
doc/ffserver.texi
libavcodec/libx264.c
libavformat/avformat.h
libavformat/avidec.c
libavformat/version.h
tests/lavf-regression.sh
tests/lavfi-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Revert "swscale: use 15-bit intermediates for 9/10-bit scaling."
swscale: use 15-bit intermediates for 9/10-bit scaling.
dct32: Add SSE2 ASM optimizations
Correct chroma vector calculation for RealVideo 3.
lavf: Add an option to discard corrupted frames
mpegts: Mark wrongly-sized packets as corrupted
mpegts: Move scan test to handle_packets
mpegts: Mark corrupted packets
mpegts: Reset continuity counter on seek
mpegts: Fix for continuity counter
mpegts: Silence "can't seek" warning on unseekable
apichange: add an entry for AV_PKT_FLAG_CORRUPT
avpacket: signal possibly corrupted packets
mpeg4videodec: remove dead code that would have detected erroneous encoding
aac: Remove some suspicious illegal memcpy()s from LTP.
bink: Eliminate unnecessary shadow declaration.
Conflicts:
doc/APIchanges
libavcodec/version.h
libavformat/avformat.h
libavformat/options.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Old version divided it wrong, which resulted in chroma drift (visible on FATE
sample too as dirty trails left by clouds).
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
dnxhddec: optimise dnxhd_decode_dct_block()
rtp: remove disabled code
eac3enc: use different numbers of blocks per frame to allow higher bitrates
dnxhd: add regression test for 10-bit
dnxhd: 10-bit support
dsputil: update per-arch init funcs for non-h264 high bit depth
dsputil: template get_pixels() for different bit depths
dsputil: create 16/32-bit dctcoef versions of some functions
jfdctint: add 10-bit version
mov: add clcp type track as Subtitle stream.
mpeg4: add Mpeg4 Profiles names.
mpeg4: decode Level Profile for MPEG4 Part 2.
ffprobe: display bitstream level.
imgconvert: remove unused glue and xglue macros
Conflicts:
libavcodec/dsputil_template.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
simple_idct: simplify some ifdeffery
simple_idct: remove code for DCTELEM != int16
Remove VLAs in ff_amrwb_lsp2lpc()
fate: make vsynth tests depend on only the relevant vref
rtsp: remove disabled code
dsputil: restore mistakenly removed hunk of disabled code
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ffmpeg: fix some indentation
ffmpeg: fix operation with --disable-avfilter
simple_idct: remove disabled code
motion_est: remove disabled code
vc1: remove disabled code
fate: separate lavf-mxf_d10 test from lavf-mxf
cabac: Move code only used in the cabac test program to cabac.c.
ffplay: warn that -pix_fmt is no longer working, suggest alternative
ffplay: warn that -s is no longer working, suggest alternative
lavf: rename enc variable in utils.c:has_codec_parameters()
lavf: use designated initialisers for all (de)muxers.
wav: remove a use of deprecated AV_METADATA_ macro
rmdec: remove useless ap parameter from rm_read_header_old()
dct-test: remove write-only variable
des: fix #if conditional around P_shuffle
Use LOCAL_ALIGNED in ff_check_alignment()
Conflicts:
ffmpeg.c
libavformat/avidec.c
libavformat/matroskaenc.c
libavformat/mp3enc.c
libavformat/oggenc.c
libavformat/utils.c
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b5849f77095439e994b11c25e6063d443b36c228': (21 commits)
ac3enc: merge AC3MDCTContext with AC3EncodeContext.
ac3enc: prefer passing AC3EncodeContext rather than AVCodecContext
ac3enc: fix memleak
mpeg1video: add CODEC_CAP_SLICE_THREADS.
lavf: fix segfault in av_open_input_stream()
mpegtsenc: set Random Access indicator on keyframe start packets
lavf: Cleanup try_decode_frame() logic.
Replace some gotos that lead to single return statements by direct return.
build: move tests/seek_test.c to libavformat and reuse generic build rules
mxfenc: include needed header for ff_iso8601_to_unix_time() prototype
Add a check for strptime().
lavf: factor out conversion of ISO8601 string to unix time
wav: parse 'bext' metadata
wav: keep parsing until EOF if the input is seekable and we know the size of the data tag
wav: Refactor the tag checking into a switch statement
wav: make sure neither data_size nor sample_count is negative.
wav: refactor the 'fmt ' tag search and parsing.
wav: add an option for writing BEXT chunk
ffmpeg: get rid of a pointless limit on number of streams.
ffmpeg: remove an unused define.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This variable is set to the same value for all directories.
Adding the -L flags directly to LDFLAGS is simpler and achieves
the same thing.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* commit '142e76f1055de5dde44696e71a5f63f2cb11dedf':
swscale: fix crash with dithering due incorrect offset calculation.
matroskadec: fix stupid typo (!= -> ==)
build: remove duplicates from order-only directory prerequisite list
build: rework rules for things in the tools dir
configure: fix --cpu=host with gcc 4.6
ARM: use const macro to define constant data in asm
bitdepth: simplify FUNC/FUNCC macros
dsputil: remove ff_emulated_edge_mc macro used in one place
9/10-bit: simplify clipping macros
matroskadec: reindent
matroskadec: defer parsing of cues element until we seek.
lavc: add support for codec-specific defaults.
lavc: make avcodec_alloc_context3 officially public.
lavc: remove a half-working attempt at different defaults for audio/video codecs.
ac3dec: add a drc_scale private option
lavf: add avformat_find_stream_info()
lavc: introduce avcodec_open2() as a replacement for avcodec_open().
Conflicts:
Makefile
libavcodec/utils.c
libavformat/avformat.h
libswscale/swscale_internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* sws_32bit_integration:
regtests/sws: update checksums for recent changes
sws: dont mess with XInc when the code needing it isnt used
sws: Fix chroma init for 32bit buffers.
swscale: error dithering for 16/9/10-bit to 8-bit.
swscale: fix overflow in 16-bit vertical scaling.
swscale: fix crash in 8-bpc bilinear output without alpha.
swscale: fix 16-bit scaling when output is 8-bits.
sws: fix non native endian 9-15 bit input with 16bit out
sws: disable scale16 when int32 is used
sws: fix rgb -> 16bit
sws: fix uv overwrite in 32bt
sws: fix gray16_1
sws:ix yuv2rgb48_1_c_template()
sws: fix 16/32 bug from merge
swscale: for >8bit scaling, read in native bit-depth.
swscale: fix another yuv range conversion overflow in 16bit scaling. (cherry picked from commit 81cc7d0bd1)
swscale: fix yuv range correction when using 16-bit scaling. (cherry picked from commit e0b8fff6c7)
swscale: implement >8bit scaling support.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Declaring tools associated with each library in their respective
makefiles allows these tools to easily depend on the correct
prerequisites and link against the libs they need.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (28 commits)
mp3enc: write a xing frame containing number of frames in the file
lavf: update AVStream.nb_frames when muxing.
ffmpeg: remove unused variables from InputStream.
doc: update ffmpeg -ar and -ac documentation to reflect reality.
ffmpeg: remove pointless if (nb_input_files)
ffmpeg: merge input_files_ts_offset into input_files.
ffmpeg: merge input_codecs into input_streams.
ffmpeg: drop AV prefixes from struct names.
ffmpeg: deprecate loop_input and loop_output options
gif: add loop private option.
img2: add loop private option.
AVOptions: in av_opt_find() don't return named constants unless unit is specified.
x11grab: replace undocumented nomouse hackery with a private option.
dict: extend documentation.
lls: whitespace cosmetics
docs: Use proper markup for a literal command line option
docs: Remove a remark that isn't relevant any longer
docs: Explain how to regenerate import libraries with MSVC tools
docs: Mention that libraries for MSVC can be built with a cross compiler
docs: Remove old docs that mention setting up a build environment with lib.exe
...
Conflicts:
doc/ffmpeg.texi
doc/general.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/dnxhddata.c
libavformat/mp3enc.c
libavformat/utils.c
libavutil/Makefile
tests/copycooker.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We operated on 31-bits, but with e.g. lanczos scaling, values can
add up to beyond 0x80000000, thus leading to output of zeroes. Drop
one bit of precision fixes this.
* qatar/master:
ffserver: remove unused variable.
Remove unused and outdated TODO file.
gitignore: Drop individual .d ignore; it is already covered by a wildcard.
lavf: deprecate AVStream.quality.
bink: pass Bink version to audio decoder through extradata instead of codec_tag.
libpostproc: Remove disabled code.
flashsv: improve some comments and fix some wrong ones
flashsv: Eliminate redundant variable indirection.
flashsv: set reference frame type to full frame
flashsv: replace bitstream description by a link to the specification
flashsv: convert a debug av_log into av_dlog
flashsv: simplify condition
flashsv: return more meaningful error values
flashsv: cosmetics: break some overly long lines
flashsv: cosmetics: drop some unnecessary parentheses
swscale: amend documentation to mention use of native depth for scaling.
eval: add missing comma to tests.
eval: fix memleak.
H.264: make loopfilter bS const where applicable
Conflicts:
libavcodec/binkaudio.c
libavformat/bink.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (36 commits)
ARM: allow unaligned buffer in fixed-point NEON FFT4
fate: test more FFT etc sizes
dca: set AVCodecContext frame_size for DTS audio
YASM: Shut up unused variable compiler warning with --disable-yasm.
x86_32: Fix build on x86_32 with --disable-yasm.
iirfilter: add fate test
doxygen: Add qmul docs.
ogg: propagate return values and return more meaningful error values
H.264: fix overreads of qscale_table
Remove unused static tables and static inline functions.
eval: clear Parser instances before using
dct-test: remove 'ref' function pointer from tables
build: Remove deleted 'check' target from .PHONY list.
oggdec: Abort Ogg header parsing when encountering a data packet.
Add LGPL license boilerplate to files lacking it.
mxfenc: small typo fix
doxygen: Fix documentation for some VP8 functions.
sha: use AV_RB32() instead of assuming buffer can be cast to uint32_t*
des: allow unaligned input and output buffers
aes: allow unaligned input and output buffers
...
Conflicts:
libavcodec/dct-test.c
libavcodec/libvpxenc.c
libavcodec/x86/dsputil_mmx.c
libavcodec/x86/h264_qpel_mmx.c
libavfilter/x86/gradfun.c
libavformat/oggdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Setting CMP=null uses stdout from the test as the diff record in
the report without further inspection. This can be useful when
the exit status of the test itself determines success while the
output is informative.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master: (40 commits)
H.264: template left MB handling
H.264: faster fill_decode_caches
H.264: faster write_back_*
H.264: faster fill_filter_caches
H.264: make filter_mb_fast support the case of unavailable top mb
Do not include log.h in avutil.h
Do not include pixfmt.h in avutil.h
Do not include rational.h in avutil.h
Do not include mathematics.h in avutil.h
Do not include intfloat_readwrite.h in avutil.h
Remove return statements following infinite loops without break
RTSP: Doxygen comment cleanup
doxygen: Escape '\' in Doxygen documentation.
md5: cosmetics
md5: use AV_WL32 to write result
md5: add fate test
md5: include correct headers
md5: fix test program
doxygen: Drop array size declarations from Doxygen parameter names.
doxygen: Fix parameter names to match the function prototypes.
...
Conflicts:
libavcodec/x86/dsputil_mmx.c
libavformat/flvenc.c
libavformat/oggenc.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
get_bits: remove x86 inline asm in A32 bitstream reader
doc: Remove outdated information about our issue tracker
avidec: Factor out the sync fucntionality.
fate-aac: Expand coverage.
ac3dsp: add x86-optimized versions of ac3dsp.extract_exponents().
ac3dsp: simplify extract_exponents() now that it does not need to do clipping.
ac3enc: clip coefficients after MDCT.
ac3enc: add int32_t array clipping function to DSPUtil, including x86 versions.
swscale: for >8bit scaling, read in native bit-depth.
matroskadec: matroska_read_seek after after EBML_STOP leads to failure.
doxygen: fix usage of @file directive in libavutil/{dict,file}.h
doxygen: Help doxygen parser to understand the DECLARE_ALIGNED and offsetof macros
Conflicts:
doc/issue_tracker.txt
libavformat/avidec.c
libavutil/dict.h
libswscale/swscale.c
libswscale/utils.c
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
cosmetics: fix some then/than typos
doxygen: Include libavcodec and libavformat examples into the documentation
avutil: elaborate documentation for av_get_random_seed
Add support for aac streams in mp4/mov without extradata.
aes: whitespace cosmetics
adler32: whitespace cosmetics
swscale: fix another yuv range conversion overflow in 16bit scaling.
Fix cpu flags test program
opt-test: Add missing braces to silence compiler warnings.
build: Eliminate obsolete test targets.
udp: Fix a compilation warning
swscale: Unbreak build with --enable-small
base64: add fate test
aes: improve test program and add fate test
adler32: make test program more useful and add fate test
swscale: fix yuv range correction when using 16-bit scaling.
aacenc: Make chan_map const correct
Conflicts:
Makefile
doc/examples/muxing-example.c
libavformat/udp.c
libavutil/random_seed.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rational-test: Add proper main() declaration to fix gcc warnings.
configure: Add vdpau and dxva2 to configure results output.
Remove unused, never built libavutil/pca.[ch]
matroskadec: forward parsing errors to caller.
av_find_stream_info: simplify EAGAIN handling.
aacenc: Fix determination of Mid/Side Mode.
psymodel: Remove the single channel analysis function
aacenc: Implement dummy channel group analysis that just calls the single channel analysis for each channel.
psymodel: Add channels and channel groups to the psymodel.
ARM: remove check for PLD instruction
fate: move amr[nw]b test rules into separate files
ogg: fix double free when finding length of small chained oggs.
swscale: implement >8bit scaling support.
build: fix creation of tools dir with make 3.81
build: Mark all-yes Makefile target as phony.
pixfmt: fix YUV422/444 wrong endian comment
build: create output directories as needed
Add new yuv444 pixfmts to avcodec_align_dimensions2
Conflicts:
Makefile
configure
libavutil/pca.c
libavutil/pca.h
libavutil/pixfmt.h
libswscale/swscale.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixdesc
tests/ref/lavfi/pixfmts_copy
tests/ref/lavfi/pixfmts_null
tests/ref/lavfi/pixfmts_scale
tests/ref/lavfi/pixfmts_vflip
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (21 commits)
swscale: Add Doxygen for hyscale_fast/hScale.
fate: enable lavfi-pixmt tests on big endian systems
PPC: swscale: disable altivec functions for unsupported formats
fate: merge identical pixdesc_be/le tests
swscale: Add Doxygen for yuv2planar*/yuv2packed* functions.
build: call texi2pod.pl with full path instead of symlink
build: include sub-makefiles using full path instead of symlinks
swscale: update big endian reference values after dff5a835.
wavpack: skip blocks with no samples
cosmetics: remove outdated comment that is no longer true
build: replace some addprefix/addsuffix with substitution refs
avutil: Remove unused arbitrary precision integer code.
configure: Drop check for availability of ten assembler operands.
aacenc: Save channel configuration for later use.
aacenc: Fix codebook trellising for zeroed bands.
swscale: change prototypes of scaled YUV output functions.
swscale: re-add support for non-native endianness.
swscale: disentangle yuv2rgbX_c_full() into small functions.
swscale: split yuv2packed[12X]_c() remainders into small functions.
swscale: split yuv2packedX_altivec in smaller functions.
...
Conflicts:
Makefile
configure
libavcodec/x86/dsputil_mmx.c
libavfilter/Makefile
libavformat/Makefile
libavutil/integer.c
libavutil/integer.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/x86/swscale_template.c
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doxygen: Consistently use '@' instead of '\' for Doxygen markup.
Use av_printf_format to check the usage of printf style functions
Add av_printf_format, for marking printf style format strings and their parameters
ARM: enable thumb for Cortex-M* CPUs
nsvdec: Propagate error values instead of returning 0 in nsv_read_header().
build: remove SRC_PATH_BARE variable
build: move basic rules and variables to main Makefile
build: move special targets to end of main Makefile
lavdev: improve feedback in case of invalid frame rate/size
vfwcap: prefer "framerate_q" over "fps" in vfw_read_header()
v4l2: prefer "framerate_q" over "fps" in v4l2_set_parameters()
fbdev: prefer "framerate_q" over "fps" in device context
bktr: prefer "framerate" over "fps" for grab_read_header()
ALSA: implement channel layout for playback.
alsa: support unsigned variants of already supported signed formats.
alsa: add support for more formats.
ARM: allow building in Thumb2 mode
Conflicts:
common.mak
doc/APIchanges
libavcodec/vdpau.h
libavdevice/alsa-audio-common.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Setting SRC_PATH to "." when building in-tree removes the need
for a quoted version of the source path since out-of-tree builds
are not possible if the pathname contains spaces.
Signed-off-by: Mans Rullgard <mans@mansr.com>
All tests are run through the fate-run.sh script which already
sets up redirections. Using the outputs set there simplifies
things somewhat.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
ac3enc: use correct alignment and length in channel coupling dsp functions.
ffmpeg: don't abuse a global for passing framerate from input to output
ffmpeg: don't abuse a global for passing channels from input to output
ffmpeg: don't abuse a global for passing samplerate from input to output
ARM: update ff_h264_idct8_add4_neon for 4:4:4 changes
swscale: use SwsContext for av_log when available
swscale: Remove HAVE_MMX from files that are only compiled with MMX enabled.
swscale: Fix compilation with --disable-mmx2.
Conflicts:
ffmpeg.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: Add -U__STRICT_ANSI__ to CPPFLAGS on Cygwin and DOS.
aacdec: fix typo in scalefactor clipping check
fate: fix fate-h264-conformance-frext-pph10i4-panasonic-a crcs.
fate: update 9/10bit refs.
h264: Properly set coded_{width, height} when parsing H.264.
x86 asm: Add SECTION_TEXT to dct32_sse.asm.
Fix 9/10 bit in swscale.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: make executable again
LATM/AAC: Free previously initialized context on reinit.
configure: Do not unconditionally add -Wall to host CFLAGS.
configure: Set OS/2 objformat to a.out.
Add support for a.out object format to assembler macros.
fate: disable threading for encoding
fate: add comment field
fate: allow overriding default build and install dirs
mpegtsenc: Add an AVClass pointer to the private data
mpegaudio: clean up #includes
mpegaudio: move all header parsing to mpegaudiodecheader.[ch]
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This explicitly disables threading for encoding as slices are otherwise
automatically activated. This should be dropped once option resetting
between files is fully implemented.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This adds a comment field to the report header, suitable for
extra information not covered by the automatic fields.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
APIchanges: fill in date and commit for request_sample_fmt
Add floating-point sample format support to the ac3, eac3, dca, aac, and vorbis decoders.
Add support for request_sample_format in ffmpeg and ffplay.
Add APIchanges entry for request_sample_fmt.
Add request_sample_fmt field to AVCodecContext.
Add float_interleave() to FmtConvertContext with x86-optimized versions.
Remove unused make variable SEEK_REFFILE
fate: remove redundant aref and vref references
fate: remove do_ffmpeg_nocheck function
fate: do not collect -benchmark output
mpegaudiodec: remove decode_end() function
fate: run aref and vref as regular tests
mpegaudio: sanitise compute_antialias_* names
mpeg12: add slice-threading checks to slice-threading initializers.
h264: copy pixel_shift between slice threading contexts.
mdec: enable frame-level multithreading.
mdec.c: fix overread.
Conflicts:
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/dca.c
libavcodec/h264.c
libavcodec/mdec.c
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/version.h
libavcodec/vorbisdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This function is essentially an alias for run_ffmpeg and is only
used in one place. This patch removes the function and replaces
the call with the equivalent (simpler) run_ffmpeg call.
Signed-off-by: Mans Rullgard <mans@mansr.com>
The old regtest scripts pass -benchmark and collect the utime values.
As these values are never used, this machinery can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
These tests create reference files used for psnr calculation in
the other codec tests. Treating them as (mostly) regular tests
simplifies the makefile and makes them visible in the fate reports.
The latter makes errors in these runs easier to identify.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet, move stereo interleaving from the iff demuxer to the
decoder, and introduce an 8svx_raw decoder which performs
stereo interleaving.
This is required for handling stereo data correctly, indeed samples
are stored like:
LLLLLL....RRRRRR
that is all left samples are at the beginning of the chunk, all right
samples at the end, so it is necessary to store and process the whole
buffer in order to decode each frame. Thus the decoder needs all the
audio chunk before it can return interleaved data.
Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
Also remove code that overwrites the C versions of functions in
sws_init_swScale_altivec(), so that it uses the C functions of files
if no altivec-optimized version exists.
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (32 commits)
10-bit H.264 x86 chroma v loopfilter asm
Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
Fix crash of interlaced MPEG2 decoding
h264pred: fix one more aliasing violation.
doc/APIchanges: fill in missing hashes and dates.
flacenc: use proper initializers for AVOption default values.
lavc: deprecate named constants for deprecated antialias_algo.
aac: workaround for compilation on cygwin
swscale: extend YUV422p support to 10bits depth
tiff: add support for inverted FillOrder for uncompressed data
Remove unused softfloat implementation.
h264pred: fix aliasing violations.
rotozoom: Eliminate French variable name.
rotozoom: Check return value of fread().
rotozoom: Return an error value instead of calling exit().
rotozoom: Make init_demo() return int and check for errors on invocation.
rotozoom: Drop silly UINT8 typedef.
rotozoom: Drop some unnecessary parentheses.
rotozoom: K&R coding style cosmetics
rtsp: Only do keepalive using GET_PARAMETER if the server supports it
...
Conflicts:
Changelog
cmdutils.c
doc/APIchanges
doc/general.texi
ffmpeg.c
ffplay.c
libavcodec/h264pred_template.c
libavcodec/resample.c
libavutil/pixfmt.h
libavutil/softfloat.c
libavutil/softfloat.h
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix handling of input if not in native endianness, and add support for
9/10-bit output. This allows us to force endianness of YUV420P 9/10bit
in the H264/10bit fate tests, which should fix them on big-endian
systems.
The file does not decode correctly yet the checksums match this wrongly
decoded file. Thus the checksums must be wrong.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
framecrc returns different values when one swiches endianness,
this apparently has been missed by "the fork" who added the 10bit fate
tests. Sorry for missing this during the merge.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes the warnings:
tests/rotozoom.c:252: warning: ignoring return value of ‘fread’, declared with attribute warn_unused_result
tests/rotozoom.c:254: warning: ignoring return value of ‘fread’, declared with attribute warn_unused_result
* qatar/master: (30 commits)
AVOptions: make default_val a union, as proposed in AVOption2.
arm/h264pred: add missing argument type.
h264dsp_mmx: place bracket outside #if/#endif block.
lavf/utils: fix ff_interleave_compare_dts corner case.
fate: add 10-bit H264 tests.
h264: do not print "too many references" warning for intra-only.
Enable decoding of high bit depth h264.
Adds 8-, 9- and 10-bit versions of some of the functions used by the h264 decoder.
Add support for higher QP values in h264.
Add the notion of pixel size in h264 related functions.
Make the h264 loop filter bit depth aware.
Template dsputil_template.c with respect to pixel size, etc.
Template h264idct_template.c with respect to pixel size, etc.
Preparatory patch for high bit depth h264 decoding support.
Move some functions in dsputil.c into a new file dsputil_template.c.
Move the functions in h264idct into a new file h264idct_template.c.
Move the functions in h264pred.c into a new file h264pred_template.c.
Preparatory patch for high bit depth h264 decoding support.
Add pixel formats for 9- and 10-bit yuv420p.
Choose h264 chroma dc dequant function dynamically.
...
Conflicts:
doc/APIchanges
ffmpeg.c
ffplay.c
libavcodec/alpha/dsputil_alpha.c
libavcodec/arm/dsputil_init_arm.c
libavcodec/arm/dsputil_init_armv6.c
libavcodec/arm/dsputil_init_neon.c
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/h264pred_init_arm.c
libavcodec/bfin/dsputil_bfin.c
libavcodec/dsputil.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_loopfilter.c
libavcodec/h264_ps.c
libavcodec/h264_refs.c
libavcodec/h264dsp.c
libavcodec/h264idct.c
libavcodec/h264pred.c
libavcodec/mlib/dsputil_mlib.c
libavcodec/options.c
libavcodec/ppc/dsputil_altivec.c
libavcodec/ppc/dsputil_ppc.c
libavcodec/ppc/h264_altivec.c
libavcodec/ps2/dsputil_mmi.c
libavcodec/sh4/dsputil_align.c
libavcodec/sh4/dsputil_sh4.c
libavcodec/sparc/dsputil_vis.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/options.c
libavformat/utils.c
libavutil/pixfmt.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/swscale_template.c
tests/ref/seek/lavf_avi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Looks better for some cases, worse for others, overall not much difference.
Its more correct though.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This should fix behavior introduced by commit
96573c0d76. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
mpegaudiodec: group #includes more sanely
mpegaudio: remove #if 0 blocks
ffmpeg.c: reset avoptions after each input/output file.
ffmpeg.c: store per-output stream sws flags.
mpegaudio: remove CONFIG_MPEGAUDIO_HP option
mpegtsenc: Clear st->priv_data when freeing it
udp: Fix receiving RTP data over multicast
rtpproto: Remove an unused variable
regtest: fix wma tests
NOT pulled: mpegaudio: remove CONFIG_AUDIO_NONSHORT
regtest: separate flags for encoding and decoding
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds $DEC_OPTS to the wma decode commands, making tests pass
on systems where the bitexact flag is needed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This separates encoding and decoding flags, and passes them together
with the related file argument instead of all at the start of the
command line.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
log: Fix an oob array read.
cosmetics: trim trailing whitespace in postproc
Ban strncpy() it's too easy to misuse.
psymodel: Remove wrapper functions.
aacenc: Replace loop counters in aac_encode_frame() with more descriptive 'ch' and 'w'.
regtest: remove redundant flags in jpg test
regtest: use run_ffmpeg in do_image_formats
regtest: simplify encoding functions
ffmpeg.c: check for interlaced flag in the correct place.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The video encoding options were needlessly split in two parameters
which are merged. The do_audio_encoding function did not use its
second argument, so this can be removed.
Signed-off-by: Mans Rullgard <mans@mansr.com>
In do_lavfi_pixfmts(), provide the filter arguments to showfiltfmts,
since some filter may require non-null or non-empty argument string
for working properly.
* qatar/master: (23 commits)
doc: Check standalone compilation before submitting new components.
Fix standalone compilation of pipe protocol.
Fix standalone compilation of ac3_fixed encoder.
Fix standalone compilation of binkaudio_dct / binkaudio_rdft decoders.
Fix standalone compilation of IMC decoder.
Fix standalone compilation of WTV demuxer.
Fix standalone compilation of MXPEG decoder.
flashsv: K&R cosmetics
matroskaenc: fix memory leak
vc1: make overlap filter for I-frames bit-exact.
vc1dec: use s->start/end_mb_y instead of passing them as function args.
Revert "VC1: merge idct8x8, coeff adjustments and put_pixels."
Replace strncpy() with av_strlcpy().
indeo3: Eliminate use of long.
get_bits: make cache unsigned to eliminate undefined signed overflow.
asfdec: fix assert failure on invalid files
avfilter: check malloc return values.
Not pulled as reason for reindent is not pulled: mpegvideo: reindent.
nutenc: check malloc return values.
Not pulled due to much simpler solution in ffmpeg *: don't av_malloc(0).
...
Conflicts:
doc/developer.texi
libavcodec/Makefile
libavcodec/get_bits.h
libavcodec/mpegvideo.c
libavformat/Makefile
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 78431098f9)
Tested with mplayer based on this report
http://thread.gmane.org/gmane.comp.video.mplayer.user/66043/focus=66063
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/master:
APIChanges: document git revision for CODEC_CAP_SLICE_THREADS addition.
Introduce slice threads flag.
FATE: allow forcing thread-type when doing threaded fate runs.
Use av_log_ask_for_sample() where appropriate.
error: sort, pack, and align error code and string definitions
The stabilization period after version bumps should be one month, not one week.
applehttp: Expose the stream bitrate via metadata
doc: Add some initial docs on the applehttp demuxer
Provide a fallback version of the libm function trunc
libavdevice: Define _XOPEN_SOURCE for usleep
lavc: provide deprecated avcodec_thread_init until next major version
lavc: provide the opt.h header until the next bump
error: change AVERROR_EOF value
error: remove AVERROR_NUMEXPECTED
error: add error code AVERROR_OPTION_NOT_FOUND, and use it in opt.c
Conflicts:
libavcodec/h264.c
libavutil/error.c
libavutil/error.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3dec: fix processing of delta bit allocation information.
vc1: fix fate-vc1 after previous commit.
wmv3dec: fix playback of complex WMV3 files using simple_idct.
make av_dup_packet() more cautious on allocation failures
make containers pass palette change in AVPacket
introduce side information for AVPacket
Politic commits that have not been pulled:
Update regtest checksums after revision 6001dad.
Replace more FFmpeg references by Libav.
Replace references to ffmpeg-devel with libav-devel; fix roundup URL.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (32 commits)
libopencore-amr, libvo-amrwbenc: Allow enabling DTX via private AVOptions
libopencore-amr, libvo-amrwbenc: Only check the bitrate when changed
libopencore-amr, libvo-amrwbenc: Find the closest matching bitrate
libvo-*: Fix up the long codec names
libavcodec: Mark AVCodec->priv_class const
swscale: Factorize FAST_BGR2YV12 definition.
libvo-aacenc: Only produce extradata if the global header flag is set
lavf: postpone removal of public metadata conversion API
lavc: postpone removal of request_channels
lavc: postpone removal of audioconvert and sample_fmt wrappers
lavf: postpone removal of deprecated avio functions
libopencore-amr: Cosmetics: Rewrap and align
libopencore-amr, libvo-amrbwenc: Rename variables and functions
libopencore-amr: Convert commented out debug logging into av_dlog
libopencore-amr: Remove an unused state variable
libvo-amrwbenc: Don't explicitly store bitrate modes in the bitrate table
libopencore-amr: Remove a useless local variable
libopencore-amr, libvo-amrwbenc: Make the bitrate/mode mapping array static const
libopencore-amr, libvo-amrwbenc: Return proper error codes in most places
libopencore-amr: Don't print carriage returns in log messages
...
Conflicts:
doc/developer.texi
libavcodec/avcodec.h
libavcodec/libvo-aacenc.c
libavcodec/libvo-amrwbenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proto: include os_support.h in network.h
matroskaenc: don't write an empty Cues element.
lavc: add a FF_API_REQUEST_CHANNELS deprecation macro
avio: move extern url_interrupt_cb declaration from avio.h to url.h
avio: make av_register_protocol2 internal.
avio: avio_ prefix for url_set_interrupt_cb.
avio: AVIO_ prefixes for URL_ open flags.
proto: introduce listen option in tcp
doc: clarify configure features
proto: factor ff_network_wait_fd and use it on udp
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix parser: mark av_parser_parse() for removal on next major bump
swscale: postpone sws_getContext removal until next major bump.
fate: add AAC LATM test
mmst: get rid of deprecated AVERRORs
lxfdec: use AVERROR(ENOMEM) instead of deprecated AVERROR_NOMEM.
Reemove remaining uses of deprecated AVERROR_NOTSUPP.
REIMPLEMENTED in 2 lines of code: lavf: if id3v2 tag is present and all else fails, guess by file extension
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
ac3enc: move extract_exponents inner loop to ac3dsp
avio: deprecate url_get_filename().
avio: deprecate url_max_packet_size().
avio: make url_get_file_handle() internal.
avio: make url_filesize() internal.
avio: make url_close() internal.
avio: make url_seek() internal.
avio: cosmetics, move AVSEEK_SIZE/FORCE declarations together
avio: make url_write() internal.
avio: make url_read_complete() internal.
avio: make url_read() internal.
avio: make url_open() internal.
avio: make url_connect internal.
avio: make url_alloc internal.
applehttp: Merge two for loops
applehttp: Restructure the demuxer to use a custom AVIOContext
applehttp: Move finished and target_duration to the variant struct
aacenc: reduce the number of loop index variables
avio: deprecate url_open_protocol
avio: deprecate url_poll and URLPollEntry
...
Conflicts:
libavformat/applehttp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: fix partial run when no samples path is specified
ARM: NEON fixed-point forward MDCT
ARM: NEON fixed-point FFT
lavf: bump minor version and add an APIChanges entry for avio changes
avio: simplify url_open_dyn_buf_internal by using avio_alloc_context()
avio: make url_fdopen internal.
avio: make url_open_dyn_packet_buf internal.
avio: avio_ prefix for url_close_dyn_buf
avio: avio_ prefix for url_open_dyn_buf
avio: introduce an AVIOContext.seekable field
ac3enc: use generic fixed-point mdct
lavfi: add fade filter
Change yadif to not use out of picture lines.
lavc: deprecate AVCodecContext.antialias_algo
lavc: mark mb_qmin/mb_qmax for removal on next major bump.
Conflicts:
doc/filters.texi
libavcodec/ac3enc_fixed.h
libavcodec/ac3enc_float.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/vf_fade.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the AC3 encoder use the shared fixed-point MDCT rather
than its own implementation. The checksum changes are due to
different rounding in the MDCT.
Signed-off-by: Mans Rullgard <mans@mansr.com>
As per issue2629, most 23.976fps matroska H.264 files are incorrectly
detected as 24fps, as the matroska timestamps usually have only
millisecond precision.
Fix that by doubling the amount of timestamps inspected for frame rate
for streams that have coarse time base. This also fixes 29.970 detection
in matroska.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ac3enc: ARM optimised ac3_compute_matissa_size
ac3: armv6 optimised bit_alloc_calc_bap
fate: simplify fft test rules
avio: document avio_alloc_context.
lavf: make compute_chapters_end less picky.
sierravmd: fix Indeo3 videos
FFT: simplify fft8()
fate: add fixed-point fft/mdct tests
Fixed-point support in fft-test
ape: check that number of seektable entries is equal to number of frames
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master:
dsputil: allow to skip drawing of top/bottom edges.
Split fate-psx-str-v3 into a video-only and audio-only test.
Conflicts:
libavcodec/dsputil.c
libavcodec/mpegvideo.c
libavcodec/snow.c
libavcodec/x86/dsputil_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master:
avio: make udp_set_remote_url/get_local_port internal.
asfdec: also subtract preroll when reading simple index object
matroskaenc: remove a variable that's unused after bc17bd9.
avio: cosmetics - nicer vertical alignment.
Remove unnecessary icc version checks
Disable 'attribute "foo" ignored' warnings from icc
rtsp: Don't use a locale dependent format string
Add xd55 codec tag for XDCAM HD422 720p25 CBR files.
configure: get libavcodec version from new version.h header
lavc: move the version macros to a new installed header.
matroskaenc: simplify get_aac_sample_rates by using ff_mpeg4audio_get_config
Do not use format string "%0.3f" for RTSP Range field.
Add apply_window_int16() to DSPContext with x86-optimized versions and use it in the ac3_fixed encoder.
Document usage of import libraries created by dlltool
configure: Set the correct lib target for arm/wince dlltool
fate: simplify regression-funcs.sh
fate: add support for multithread testing
Conflicts:
libavformat/rtspdec.c
libavutil/attributes.h
libavutil/internal.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master: (33 commits)
Fix an infinite loop when RoQ encoded generated a frame with a size greater than the maximum valid size.
Add kbdwin.o to AC3 decoder
Detect byte-swapped AC-3 and support decoding it directly.
cosmetics: indentation
Always copy input data for AC3 decoder.
ac3enc: make sym_quant() branch-free
cosmetics: indentation
Add a CPU flag for the Atom processor.
id3v2: skip broken tags with invalid size
id3v2: don't explicitly skip padding
Make sure kbhit() is in conio.h
fate: update wmv8-drm reference
vc1: make P-frame deblock filter bit-exact.
configure: Add the -D parameter to the dlltool command
amr: Set the AVFMT_GENERIC_INDEX flag
amr: Set the pkt->pos field properly to the start of the packet
amr: Set the codec->bit_rate field based on the last packet
rtsp: Specify unicast for TCP interleaved streams, too
Set the correct target for mingw64 dlltool
applehttp: Change the variable for stream position in seconds into int64_t
...
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/ac3dec.c
libavformat/avio.h
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not Pulled:
commit 42cfb3835b
Author: Mans Rullgard <mans@mansr.com>
Date: Mon Feb 28 18:06:58 2011 +0000
Remove Sonic experimental audio codec
commit 2912e87a6c
Author: Mans Rullgard <mans@mansr.com>
Date: Fri Mar 18 17:35:10 2011 +0000
Replace FFmpeg with Libav in licence headers
This updates the seek test reference to match de11ee9. Before this
change, most of the seeks requested positions before the supposed
start of the file (the preroll time), resulting in the first packet
being returned. With the preroll subtracted, some of these seeks
will land within the file and some beyond the end, thus returning
a different set of packets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This is to match the value in every (E-)AC-3 file from commercial sources.
It has a negligible effect on audio quality.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This patch changes the exponent difference threshold in the exponent
strategy decision function of the AC-3 encoder. I tested lowering in
increments of 100. From 1000 down to 500 generally increased in quality
with each step, but 400 was generally much worse.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Instead of saving huge raw files, use the md5: output pseudo-protocol
to calculate the checksum of the file directly. This is especially
useful when testing on remote targets as it avoids transferring 3.6GB
over the network.
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixed-point AC-3 encoder renamed to ac3_fixed.
Regression test acodec-ac3 renamed to acodec-ac3_fixed.
Regression test lavf-rm changed to use ac3_fixed encoder.
Originally committed as revision 26209 to svn://svn.ffmpeg.org/ffmpeg/trunk
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.
Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.
Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor)
Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk
The rm demuxer has timestamp bugs, so this test is sensitive to changes in
timestamp correction. The previous commit did not make output any better or worse
on this test, just different.
See https://roundup.ffmpeg.org/issue2288 for details.
Originally committed as revision 25432 to svn://svn.ffmpeg.org/ffmpeg/trunk
and add a test for regular GSM as fate-gsm.
Fixes a 8kHz sample from issue 113.
Originally committed as revision 25313 to svn://svn.ffmpeg.org/ffmpeg/trunk
Slightly more intuitive and required by a pending changes for making
the filter parametric.
Originally committed as revision 25184 to svn://svn.ffmpeg.org/ffmpeg/trunk
instead of current "make output file of size less than ss".
Also use it to make MP3 tests more readable (using -fs xxx where xxx is
the requested output size, not something slightly lower).
Originally committed as revision 24884 to svn://svn.ffmpeg.org/ffmpeg/trunk
This moves some groups of tests for single codecs to separate files,
and adds shorthands for running all tests in a group.
Originally committed as revision 24697 to svn://svn.ffmpeg.org/ffmpeg/trunk
Increase readability and robustness, as the test result is not going
to differ if the order of the pixfmts codes changes.
Originally committed as revision 24665 to svn://svn.ffmpeg.org/ffmpeg/trunk
The corresponding lavfi-pixfmts BE tests are not yet added, as there
are some bugs in the scaler (scaling rgba, argb, bgra, abgr, yuva420p)
which result in differences with the LE reference, and I cannot
visually check the generated files on BE.
Originally committed as revision 24657 to svn://svn.ffmpeg.org/ffmpeg/trunk
Introduce the function do_lavfi_pixfmts(), and use it for generating a
pixfmts test for each different filter.
Originally committed as revision 24654 to svn://svn.ffmpeg.org/ffmpeg/trunk
The 'tar' variable should be set to a command writing a tar archive
of the named files to stdout, typically "tar c" or "tar cf -"
Originally committed as revision 24649 to svn://svn.ffmpeg.org/ffmpeg/trunk
Add the lavfi-showfiltfmts dependency in the Makefile, and correctly
use the $target_exec and $target_path variables for invoking the
lavfi-showfiltfmts tool.
Originally committed as revision 24645 to svn://svn.ffmpeg.org/ffmpeg/trunk
otherwise the test will be running whatever ffmpeg is installed on the
host system.
Originally committed as revision 24644 to svn://svn.ffmpeg.org/ffmpeg/trunk
test output files, and add a prefix with the name of the test.
Make per-filter grouping of the generated output files easier, which
is more useful than per-pixel-format grouping.
Originally committed as revision 24643 to svn://svn.ffmpeg.org/ffmpeg/trunk
Consistent with the lavfi pixdesc test code, and slightly improve
readability.
Originally committed as revision 24642 to svn://svn.ffmpeg.org/ffmpeg/trunk
The diff may provide useful information even if the command was
unsuccessful. The test is still treated as failed in this case.
Originally committed as revision 24353 to svn://svn.ffmpeg.org/ffmpeg/trunk
This test verifies the pixdesc code by comparing the output with and
without a filter which should have no effect on the image. Since the
available pixel formats depend on the byte order of the machine, a
simple reference checksum is not possible.
The test originally tried to solve this by generating a reference file
on the fly. The problem with this is that the test framework expects
the reference file in the source tree, and writing to the source tree
is not allowed.
To avoid complicating the test framework, we instead provide two
reference files and select which to use based on the byte order.
Originally committed as revision 24330 to svn://svn.ffmpeg.org/ffmpeg/trunk
All tests use the provided helper functions so prepending $target_exec
and using eval is no longer required.
Originally committed as revision 24317 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows CMP=stddev in test rules. The test passes if the reported
stddev is <= the FUZZ value (default 1).
Originally committed as revision 24289 to svn://svn.ffmpeg.org/ffmpeg/trunk
Log:
Add msmpeg4v1 regtest
Added:
trunk/tests/ref/fate/msmpeg4v1
Modified:
trunk/tests/fate2.mak
According to Mans, "make test" tests already msmpeg4v1.
Originally committed as revision 24260 to svn://svn.ffmpeg.org/ffmpeg/trunk
The byte count printed excludes the header, and offsets are applied
after the the headers are skipped.
Reference files updated to reflect new output. Some stddev/psnr values
have changed slightly due to headers no longer being compared.
Originally committed as revision 24143 to svn://svn.ffmpeg.org/ffmpeg/trunk
This adds the 'oneoff' comparison method. It compares two s16le pcm
files with a max difference <= the fuzz factor treated as success.
Default fuzz is 1. On failure, the full tiny_psnr output is printed.
Originally committed as revision 24134 to svn://svn.ffmpeg.org/ffmpeg/trunk
This passes the following additional variables to the test script:
- CMP method to use comparing output with reference, default 'diff'
- REF reference file, default 'tests/ref/fate/${test}'
- FUZZ maximum fuzz factor in comparison, meaning depends on CMP
Originally committed as revision 24133 to svn://svn.ffmpeg.org/ffmpeg/trunk
Regression test reference updates are due to the extra output
from tiny_psnr.
Patch by Vitor Sessak
Originally committed as revision 24132 to svn://svn.ffmpeg.org/ffmpeg/trunk
Author: bcoudurier
Date: Sat Jul 3 03:11:04 2010 +0000
Set graph swscale opts before parsing it, that way opts are available
when auto-adding scalers.
It changed the swscale flags used by the auto-added scalers, and so
the output video.
Originally committed as revision 24065 to svn://svn.ffmpeg.org/ffmpeg/trunk
and bgr4_byte from the lavfi-pix_fmts test.
The formats are now supported by NUT.
Originally committed as revision 23510 to svn://svn.ffmpeg.org/ffmpeg/trunk
Start them on keyframes when reasonable, and delay writing audio packets
to help ensure that there's audio samples available for the first frame in
clusters.
Patch by James Zern <jzern at google>
Originally committed as revision 23473 to svn://svn.ffmpeg.org/ffmpeg/trunk
formats which are not currently supported by NUT and which cannot be
played by ffplay.
Originally committed as revision 23458 to svn://svn.ffmpeg.org/ffmpeg/trunk
target regtest-lavfi_pix_fmts.
The lavfi_pix_fmts test is disabled, this because there are
many tests which are failing, and there are still some output files
which cannot be played by NUT/ffplay.
Originally committed as revision 23297 to svn://svn.ffmpeg.org/ffmpeg/trunk
This isn't exactly semantically equivalent, but the field has already been
long abused to mean this, and writing it helps in determining a decent cfr
time base when transcoding from a mkv where the video codec stores none (VP8).
Originally committed as revision 23284 to svn://svn.ffmpeg.org/ffmpeg/trunk
A patched version of ffmpeg supporting video filters is required for
getting this working; thus make lavfitest is supposed to work only in
the libavfilter repository for now.
Originally committed as revision 22586 to svn://svn.ffmpeg.org/ffmpeg/trunk
This adds a "fate" make target which runs the full FATE test suite.
Individual tests can be run with "make fate-$testname".
The location of the FATE test samples must be specified with the
--samples=PATH option to configure.
The tests/fate-update.sh script regenerates the references files and
test list from the online FATE database. These are checked in since
generating them requires non-standard tools.
Originally committed as revision 22552 to svn://svn.ffmpeg.org/ffmpeg/trunk
This takes into account whether the granule defines the start or end times
of packets, and sets the correct file offset of the associated page.
Originally committed as revision 22462 to svn://svn.ffmpeg.org/ffmpeg/trunk
Otherwise it gets set automatically to a page midstream and prevents seeking
to the first page.
Originally committed as revision 22454 to svn://svn.ffmpeg.org/ffmpeg/trunk
ffserver-regression.sh doesn't need anything else from
regression-funcs.sh, and sourcing the entire file there
breaks things.
Originally committed as revision 22304 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seeking on image sequences doesn't actually work, so this
test isn't very useful until that capability is added.
Originally committed as revision 22286 to svn://svn.ffmpeg.org/ffmpeg/trunk
This test generates many output files, and keeping them separate
is convenient.
Originally committed as revision 22157 to svn://svn.ffmpeg.org/ffmpeg/trunk
This reverts r22151. It prevents the seektest from running at all and
is thus of no use whatsoever.
Originally committed as revision 22152 to svn://svn.ffmpeg.org/ffmpeg/trunk
testing new formats since 2 years namely r11831.
This likely will uncover bugs that gone unnoticed in the last 2 years
due to lack of testing.
Originally committed as revision 22151 to svn://svn.ffmpeg.org/ffmpeg/trunk
This correct the stop point for demuxing with -vcodec copy and -t as well as
packet interleaving. (we already diddrop packets but kept demuxing them
for too long due to opts being wrong)
the change to ffm is due to 2 packets with timestamp 0 being stored
in different order.
Originally committed as revision 21626 to svn://svn.ffmpeg.org/ffmpeg/trunk
This fixes at least ogg encoding with -t where the file was slightly too long.
Originally committed as revision 21598 to svn://svn.ffmpeg.org/ffmpeg/trunk
With this change, the output is checked immediately after each test
has run. This means commands like "make regtest-mpeg2" can now be
used to run a single test and get meaningful results.
By default, make will abort if any test fails. To run all tests
regardless, use make -k.
Originally committed as revision 21254 to svn://svn.ffmpeg.org/ffmpeg/trunk
This disables all console output from the *-regression.sh scripts by
default. If the V environment variable is set to 1, ffmpeg commands
are printed before being run. If V is greater than 1, ffmpeg output
goes to the console, otherwise stderr is redirected to a file.
Originally committed as revision 21253 to svn://svn.ffmpeg.org/ffmpeg/trunk
scheutzow at alcatel-lucent dot com and Niobos, niobos at dest-unreach dot be.
Originally committed as revision 21199 to svn://svn.ffmpeg.org/ffmpeg/trunk
just channels>2. The 16bit case is from MSDN, the 48khz is from less a reliable
www page (sorry i cant find the URL anymore).
Originally committed as revision 20892 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes the output nicely aligned even in for those cases, and might
also avoid some issues with printf implementations that can not handle well
very large float values (e.g. msvcrt without MinGW wrapper IIRC).
Originally committed as revision 20334 to svn://svn.ffmpeg.org/ffmpeg/trunk
definitions of variables defined using the positional arguments of the
script.
Should ease the split of the file.
Originally committed as revision 20322 to svn://svn.ffmpeg.org/ffmpeg/trunk
Hopefully this fixes regression tests on DOS.
It breaks alignment of the output a bit more, but this should be easy to
fix with a second patch.
Originally committed as revision 20282 to svn://svn.ffmpeg.org/ffmpeg/trunk
The au and aiff demuxers would always return an empty packet before
detecting EOF, this is now fixed with av_get_packet etc. returning EOF.
Originally committed as revision 20136 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes issue 975: high db peak levels when encoding mp2
Original patch by Lasse Reinhold, lassemikkelreinhold hotmail
Originally committed as revision 20100 to svn://svn.ffmpeg.org/ffmpeg/trunk
tests, as the use of huffyuv may cause the auto-insertion of a scale
filter, thus masking the mis-behavior of some filter.
Originally committed as revision 20088 to svn://svn.ffmpeg.org/ffmpeg/trunk
The specification does not say which value to use for unused
parts, so fill all unused bytes with 0xff, which is consistent
with what DV usually uses for reserved or unused parts.
Originally committed as revision 20084 to svn://svn.ffmpeg.org/ffmpeg/trunk
The target is disabled.
See the thread: "[PATCH] libavfilter-soc: regression test for libavfilter".
Originally committed as revision 20057 to svn://svn.ffmpeg.org/ffmpeg/trunk
Nothing useful was ever written to this file. It did, however,
often cause spurious failures when the file was left behind by
another user on the system.
Originally committed as revision 19938 to svn://svn.ffmpeg.org/ffmpeg/trunk
complex as originally planned.
Typo fixed (the background moves less linear now) and regression test
results updated.
Originally committed as revision 19478 to svn://svn.ffmpeg.org/ffmpeg/trunk
r19300
Rewrite mp3 parser. New code is much simpler and does not drop
stuff at random.
Originally committed as revision 19306 to svn://svn.ffmpeg.org/ffmpeg/trunk
This requires updating the gxf regression checksum since the gxf muxer puts
the filename in files it creates.
Originally committed as revision 18253 to svn://svn.ffmpeg.org/ffmpeg/trunk
single index chunk) and is always empty anyway. See "[PATCH] rmenc.c: remove
index writing" thread.
Originally committed as revision 18119 to svn://svn.ffmpeg.org/ffmpeg/trunk
expected change because the ogg muxer with FLAC encoder uses the maximum
frame size estimate since it cannot rewrite the real maximum frame size
at end of encoding.
Originally committed as revision 18094 to svn://svn.ffmpeg.org/ffmpeg/trunk
seeking before data_offset and is more consistent with how the generic index
seeking code handles it.
Originally committed as revision 17964 to svn://svn.ffmpeg.org/ffmpeg/trunk
for swf before and after, but it now incorrectly creates additional
streams.
Originally committed as revision 17913 to svn://svn.ffmpeg.org/ffmpeg/trunk
AC-3 decoding regression test fails with gcc 2.95.3 because of missing
SSE support.
Originally committed as revision 15625 to svn://svn.ffmpeg.org/ffmpeg/trunk
Plain C, x86-32 and -64 have been tested and should work, other
archs that had asm optmizations in swscale likely will need some fixes
to either fall back on C if SWS_BITEXACT is set or make the asm match C.
This also disables the PAL8 test as neither swscale nor the old scaler
really support PAL8 output, imgconvert supported a fixed 666 palette
as output and swscale supports fixed 884 and 422.
Originally committed as revision 15305 to svn://svn.ffmpeg.org/ffmpeg/trunk
bits_per_coded_sample but that cannot be done seperately.
Patch by Luca Abeni
Also reset the minor version and fix the forgotton change to libfaad.
Note: The API/ABI should not be considered stable yet, there still may
be a change done here or there if some developer has some cleanup ideas and
patches!
Originally committed as revision 15262 to svn://svn.ffmpeg.org/ffmpeg/trunk
The matroska demuxer now index every streams so seek on stream 1 now works.
Originally committed as revision 15254 to svn://svn.ffmpeg.org/ffmpeg/trunk
residuals, Note this does not change RGB32 as we need to check this
against some decoder that supports it.
Originally committed as revision 15055 to svn://svn.ffmpeg.org/ffmpeg/trunk
