* qatar/release/0.7: (84 commits)
id3v2: fix skipping extended header in id3v2.4
Update RELEASE file for 0.7.5
lcl: use AVERROR_INVALIDDATA instead of AVERROR_UNKNOWN
kgv1dec: Increase offsets array size so it is large enough.
kgv1: use avctx->get/release_buffer().
kvmc: fix invalid reads
nsvdec: Propagate error values instead of returning 0 in nsv_read_header().
mjpegbdec: Fix overflow in SOS.
shorten: Use separate pointers for the allocated memory for decoded samples.
shorten: check for realloc failure (cherry picked from commit 9e5e2c2d010c05c10337e9c1ec9d0d61495e0c9c)
atrac3: Fix crash in tonal component decoding.
ws_snd1: Fix wrong samples count and crash.
ws_snd: add some checks to prevent buffer overread or overwrite. (cherry picked from commit 417364ce1f979031ef6fee661fc15e1869bdb1b4)
ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
dca: include libavutil/mathematics.h for possibly missing M_SQRT1_2
h264: stricter reference limit enforcement.
jvdec: unbreak video decoding
xxan: don't read before start of buffer in av_memcpy_backptr().
dsicinvideo: validate buffer offset before copying pixels.
huffyuv: add padding to classic (v1) huffman tables.
...
Conflicts:
RELEASE
libavcodec/atrac3.c
libavcodec/h264.c
libavcodec/h264_parser.c
libavcodec/kgv1dec.c
libavcodec/shorten.c
libavcodec/svq3.c
libavcodec/ws-snd1.c
libavcodec/xxan.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.
CC:libav-stable@libav.org
(cherry picked from commit 51ddf35c9017018e58c15275ff5b129647a0c94d)
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* qatar/release/0.7: (73 commits)
Update Changelog for 0.7.2 release
Update RELEASE file for 0.7.2
lavf: do not set codec_tag for rawvideo
fate: allow testing with libavfilter disabled
fate: separate lavf-mxf_d10 test from lavf-mxf
Fix memory (re)allocation in matroskadec.c, related to MSVR-11-0080.
movenc: fix NULL reference in mov_write_tkhd_tag
movenc: create an alternate group for each media type
flvdec: Check for overflow before allocating arrays
ppc: fix some pointer to integer casts
ppc: fix 32-bit PIC build
rv34: Check for invalid slice offsets
rv34: Fix potential overreads
rv34: Avoid NULL dereference on corrupted bitstream
rv10: Reject slices that does not have the same type as the first one
lavf: Fix context pointer in av_open_input_stream when avformat_open_input fails
oggdec: fix out of bound write in the ogg demuxer
Fixed size given to init_get_bits().
smacker: fix a few off by 1 errors
Check for invalid VLC value in smacker decoder.
...
Conflicts:
RELEASE
libavcodec/avs.c
libavcodec/ppc/asm.S
libavcodec/rv34.c
libavcodec/xan.c
libavdevice/alsa-audio.h
libavformat/flvdec.c
libavformat/gxf.c
libavformat/utils.c
libswscale/x86/swscale_template.c
tests/ref/lavf/mov
tests/ref/lavf/mxf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
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>
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>
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>
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
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
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
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
