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* fate/vp8-size-change: set bitexact flagMichael Niedermayer2012-03-291-27/+27
| | | | Signed-off-by: Michael Niedermayer <[email protected]>
* Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-292-0/+345
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: asf: only set index_read if the index contained entries. cabac: add overread protection to BRANCHLESS_GET_CABAC(). cabac: increment jump locations by one in callers of BRANCHLESS_GET_CABAC(). cabac: remove unused argument from BRANCHLESS_GET_CABAC_UPDATE(). cabac: use struct+offset instead of memory operand in BRANCHLESS_GET_CABAC(). h264: add overread protection to get_cabac_bypass_sign_x86(). h264: reindent get_cabac_bypass_sign_x86(). h264: use struct offsets in get_cabac_bypass_sign_x86(). h264: fix overreads in cabac reader. wmall: fix seeking. lagarith: fix buffer overreads. dvdec: drop unnecessary dv_tablegen.h #include build: fix doc generation errors in parallel builds Replace memset(0) by zero initializations. faandct: Remove FAAN_POSTSCALE define and related code. dvenc: print allowed profiles if the video doesn't conform to any of them. avcodec_encode_{audio,video}: only reallocate output packet when it has non-zero size. FATE: add a test for vp8 with changing frame size. fate: add kgv1 fate test. oggdec: calculate correct timestamps in Ogg/FLAC Conflicts: libavcodec/4xm.c libavcodec/cook.c libavcodec/dvdata.c libavcodec/dvdsubdec.c libavcodec/lagarith.c libavcodec/lagarithrac.c libavcodec/utils.c tests/fate/video.mak Merged-by: Michael Niedermayer <[email protected]>
| * FATE: add a test for vp8 with changing frame size.Anton Khirnov2012-03-281-0/+31
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| * fate: add kgv1 fate test.Ronald S. Bultje2012-03-271-0/+314
| | | | | | | | Tested to be bit-exact across x86-64, x86-32 and ppc.
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-241-0/+1
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: rv34: error out on size changes with frame threading aacsbr: Add a debug check to sbr_mapping. aac: Reset some state variables when turning SBR off aac: Reset PS parameters on header decode failure. fate: add wmalossless test. aacsbr: handle m_max values smaller than 4. Conflicts: libavcodec/aacsbr.c tests/fate/lossless-audio.mak Merged-by: Michael Niedermayer <[email protected]>
| * fate: add wmalossless test.Ronald S. Bultje2012-03-231-0/+1
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| * FATE: Add ZeroCodec testDerek Buitenhuis2012-03-221-0/+39
| | | | | | | | | | Signed-off-by: Derek Buitenhuis <[email protected]> Signed-off-by: Anton Khirnov <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-217-741/+802
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (27 commits) avconv: free packet in write_frame() when discarding due to frame number limit FATE: use +/- flag option syntax for vp8 emu-edge tests lavf: make av_interleave_packet_per_dts() private. lavf: deprecate av_read_packet(). oggdec: output correct timestamps for Vorbis avconv: pass input stream timestamps to audio encoders lavc: shrink encoded audio packet size after encoding. xa: set correct bit rate xa: do not set bit_rate, block_align, or bits_per_coded_sample xa: fix end-of-file handling xa: fix timestamp calculation bink: fix typo in FFALIGN() argument bink: align plane width to 8 when calculating bundle sizes doc: pass -Idoc texi2html and texi2pod doc: texi2pod: add -I flag movenc: Add a min_frag_duration option rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers libavformat: Set the default for the max_delay option to -1 Generate manpages for AV{Format,Codec}Context AVOptions. doc/avconv: remove entries for AVOptions. ... Conflicts: doc/Makefile doc/ffmpeg.texi doc/muxers.texi ffmpeg.c libavcodec/Makefile libavcodec/options.c libavcodec/vp8.c libavformat/options.c tests/fate/demux.mak tests/ref/fate/truemotion1-15 tests/ref/fate/truemotion1-24 Merged-by: Michael Niedermayer <[email protected]>
| * avconv: pass input stream timestamps to audio encodersJustin Ruggles2012-03-205-740/+740
| | | | | | | | | | 5 FATE test references updated due to using demuxer-generated timestamps that are either not sample-accurate or are slightly off in the input file.
| * xa: fix timestamp calculationJustin Ruggles2012-03-201-30/+30
| | | | | | | | The packet duration is always 28 samples.
| * FATE: change fate-maxis-xa to a normal demuxing testPaul B Mahol2012-03-191-1/+31
| | | | | | | | | | Signed-off-by: Paul B Mahol <[email protected]> Signed-off-by: Justin Ruggles <[email protected]>
| * FATE: add test for adpcm-ea-maxis-xaPaul B Mahol2012-03-191-0/+31
| | | | | | | | | | Signed-off-by: Paul B Mahol <[email protected]> Signed-off-by: Justin Ruggles <[email protected]>
* | fate/zerocodec: fix permissionsMichael Niedermayer2012-03-201-0/+0
| | | | | | | | | | Reported-by: Deamon404 Signed-off-by: Michael Niedermayer <[email protected]>
* | FATE: Add ZeroCodec testDerek Buitenhuis2012-03-201-0/+39
| | | | | | | | | | Signed-off-by: Derek Buitenhuis <[email protected]> Signed-off-by: Michael Niedermayer <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-203-142/+142
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (35 commits) fix space type in Changelog ZeroCodec Decoder RealAudio Lossless decoder rtpenc: Use AVFormatContext.packet_size instead of a private option url: Document the expected behaviour of url_read libavformat: Use AVFormatContext.probesize in init_input docs: Fix a stray reference to tags in the generic doxy on dicts cosmetics: Align some AVInput/OutputFormat declarations zmbv: check decompress result zmbv: correct indentation adpcm: convert adpcm_thp to bytestream2. adpcm: convert adpcm_yamaha to bytestream2. adpcm: convert adpcm_swf to bytestream2. adpcm: convert adpcm_sbpro to bytestream2. adpcm: convert adpcm_ct to bytestream2. adpcm: convert adpcm_ima_amv/smjpeg to bytestream2. adpcm: convert adpcm_ea_xas to bytestream2. adpcm: convert adpcm_ea_r1/2/3 to bytestream2. adpcm: convert ea_maxis_xa to bytestream2. adpcm: convert adpcm_ea to bytestream2. ... Conflicts: Changelog libavcodec/Makefile libavcodec/adpcm.c libavcodec/allcodecs.c libavcodec/avcodec.h libavcodec/version.h libavcodec/zerocodec.c libavcodec/zmbv.c libavformat/riff.c libavformat/url.h tests/ref/fate/truemotion1-15 tests/ref/fate/truemotion1-24 Merged-by: Michael Niedermayer <[email protected]>
| * adpcm: fix nb_samples rounding for adpcm_ima_dk3, and update reference.Ronald S. Bultje2012-03-183-142/+142
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* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-181-1/+0
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: fate: make compare() function compatible with POSIX bc Update Janne's email address. APIchanges: Replace Subversion revision numbers by Git hashes. bytestream: Eliminate one level of pointless macro indirection. xwd: convert to bytestream2. vqavideo: port to bytestream2 API Read preset files with suffix .avpreset prores: allow user to set fixed quantiser lavf: remove some disabled code. lavf: only set average frame rate for video. lavf: remove a pointless check. avcodec: add XBM encoder Conflicts: Changelog cmdutils.c cmdutils.h doc/APIchanges libavcodec/Makefile libavcodec/avcodec.h libavcodec/version.h libavcodec/vqavideo.c libavformat/img2enc.c libavformat/utils.c Merged-by: Michael Niedermayer <[email protected]>
| * FATE: add test for cdxl demuxerPaul B Mahol2012-03-121-0/+21
| | | | | | | | | | Signed-off-by: Paul B Mahol <[email protected]> Signed-off-by: Martin Storsjö <[email protected]>
* | FATE: add test for cdxl demuxerPaul B Mahol2012-03-111-0/+21
| | | | | | | | | | Signed-off-by: Paul B Mahol <[email protected]> Signed-off-by: Michael Niedermayer <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-064-22/+23
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (31 commits) cdxl demux: do not create packets with uninitialized data at EOF. Replace computations of remaining bits with calls to get_bits_left(). amrnb/amrwb: Remove get_bits usage. cosmetics: reindent avformat: do not require a pixel/sample format if there is no decoder avformat: do not fill-in audio packet duration in compute_pkt_fields() lavf: Use av_get_audio_frame_duration() in get_audio_frame_size() dca_parser: parse the sample rate and frame durations libspeexdec: do not set AVCodecContext.frame_size libopencore-amr: do not set AVCodecContext.frame_size alsdec: do not set AVCodecContext.frame_size siff: do not set AVCodecContext.frame_size amr demuxer: do not set AVCodecContext.frame_size. aiffdec: do not set AVCodecContext.frame_size mov: do not set AVCodecContext.frame_size ape: do not set AVCodecContext.frame_size. rdt: remove workaround for infinite loop with aac avformat: do not require frame_size in avformat_find_stream_info() for CELT avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3 avformat: do not require frame_size in avformat_find_stream_info() for AAC ... Conflicts: doc/APIchanges libavcodec/Makefile libavcodec/avcodec.h libavcodec/h264.c libavcodec/h264_ps.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/dsputil_mmx.c libavformat/utils.c Merged-by: Michael Niedermayer <[email protected]>
| * lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()Justin Ruggles2012-03-052-20/+21
| | | | | | | | | | | | | | | | | | | | Also, do not give AVCodecContext.frame_size priority for muxing. Updated 2 FATE references: dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified by -t 2 in the FATE test wmv8-drm-nodec - durations are not needed. previously they were estimated using the packet size and average bit rate.
| * lavf: deobfuscate read_frame_internal().Anton Khirnov2012-03-052-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Split off packet parsing into a separate function. Parse full packets at once and store them in a queue, eliminating the need for tracking parsing state in AVStream. The horrible unreadable loop in read_frame_internal() now isn't weirdly ordered and doesn't contain evil gotos, so it should be much easier to understand. compute_pkt_fields() now invents slightly different timestamps for two raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't be more wrong (or right) than previous ones.
* | lavf: Do not compute the packet duration based on the bitrate if the ↵Michael Niedermayer2012-03-041-6/+6
| | | | | | | | | | | | | | | | | | frame_size can be determined. This fixes issues when the bitrate is variable or inaccurate but the frame size has not been determined yet. Signed-off-by: Michael Niedermayer <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-042-22/+22
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: tiertexseq: set correct block_align for audio tiertexseq: set audio stream start time to 0 voc/avs: Do not change the sample rate mid-stream. segafilm: use the sample rate as the time base for audio streams ea: fix audio pts psx-str: fix audio pts vqf: set packet duration tta demuxer: set packet duration mpegaudio_parser: do not ignore information from the first parsed frame mpegaudio_parser: be less picky about the start position thp: set audio packet durations avcodec: add a Vorbis parser to get packet duration vorbisdec: read the previous window flag for long windows lavc: free the output packet when encoding failed or produced no output. lavc: preserve avpkt->destruct in ff_alloc_packet(). lavc: clarify the meaning of AVCodecContext.frame_number. mpegts: Pad the packet buffer in handle_packet(). mpegts: Do not call read_sl_header() when no bytes remain in the buffer. Conflicts: libavcodec/mpegaudio_parser.c libavcodec/version.h libavformat/mpegts.c tests/ref/fate/pva-demux Merged-by: Michael Niedermayer <[email protected]>
| * tiertexseq: set audio stream start time to 0Justin Ruggles2012-03-031-21/+21
| | | | | | | | | | Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
| * vqf: set packet durationJustin Ruggles2012-03-031-1/+1
| | | | | | | | | | | | Fixes timestamp calculation. The FATE reference is updated because timestamp calculations are now more accurate. Previous timestamps were based on average bit rate.
| * mpegaudio_parser: do not ignore information from the first parsed frameJustin Ruggles2012-03-032-9/+9
| | | | | | | | Update some demuxing and seeking fate tests.
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-032-168/+168
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (29 commits) amrwb: remove duplicate arguments from extrapolate_isf(). amrwb: error out early if mode is invalid. h264: change underread for 10bit QPEL to overread. matroska: check buffer size for RM-style byte reordering. vp8: disable mmx functions with sse/sse2 counterparts on x86-64. vp8: change int stride to ptrdiff_t stride. wma: fix invalid buffer size assumptions causing random overreads. Windows Media Audio Lossless decoder rv10/20: Fix slice overflow with checked bitstream reader. h263dec: Disallow width/height changing with frame threads. rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size. rmdec: Honor .RMF tag size rather than assuming 18. g722: Fix the QMF scaling r3d: don't set codec timebase. electronicarts: set timebase for tgv video. electronicarts: parse the framerate for cmv video. ogg: don't set codec timebase electronicarts: don't set codec timebase avs: don't set codec timebase wavpack: Fix an integer overflow ... Conflicts: libavcodec/arm/vp8dsp_init_arm.c libavcodec/fraps.c libavcodec/h264.c libavcodec/mpeg4videodec.c libavcodec/mpegvideo.c libavcodec/msmpeg4.c libavcodec/pnmdec.c libavcodec/qpeg.c libavcodec/rawenc.c libavcodec/ulti.c libavcodec/vcr1.c libavcodec/version.h libavcodec/wmalosslessdec.c libavformat/electronicarts.c libswscale/ppc/yuv2rgb_altivec.c tests/ref/acodec/g722 tests/ref/fate/ea-cmv Merged-by: Michael Niedermayer <[email protected]>
| * g722: Fix the QMF scalingMartin Storsjö2012-03-022-168/+168
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This fixes clipping if the encoder input used the full 16 bit input range (samples with a magnitude below 16383 worked fine). The filtered subband samples should be 15 bit maximum, while the code earlier produced them scaled to 16 bit. This makes the decoder output have double the magnitude compared to before. The spec reference samples doesn't test the QMF at all, which was why this part slipped past initially. Signed-off-by: Martin Storsjö <[email protected]>
| * electronicarts: set timebase for tgv video.Anton Khirnov2012-03-022-87/+87
| | | | | | | | | | | | | | | | The container has no timestamps and the framerate isn't stored in the data either. The decoder sets codec timebase to experimentally found value 1/15. Do the same for the demuxer too, it should at least be better than the default 1/90000.
| * electronicarts: parse the framerate for cmv video.Anton Khirnov2012-03-021-195/+195
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| * electronicarts: don't set codec timebaseAnton Khirnov2012-03-023-390/+390
| | | | | | | | | | | | Demuxers are not supposed to set it. Set stream timebase and framerates instead (this is a cfr container with no timestamps).
* | lavf: fix update_initial_durations() so it handles missing durations with ↵Michael Niedermayer2012-03-021-3/+3
| | | | | | | | | | | | | | | | | | the initial timestamp being known. This fixes duplicate timestamps on mp2 in ts with non seekable input. It also fixed the fate pva demux timestamps. Signed-off-by: Michael Niedermayer <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-012-19/+11
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (58 commits) amrnbdec: check frame size before decoding. cscd: use negative error values to indicate decode_init() failures. h264: prevent overreads in intra PCM decoding. FATE: do not decode audio in the nuv test. dxa: set audio stream time base using the sample rate psx-str: do not allow seeking by bytes asfdec: Do not set AVCodecContext.frame_size vqf: set packet parameters after av_new_packet() mpegaudiodec: use DSPUtil.butterflies_float(). FATE: add mp3 test for sample that exhibited false overreads fate: add cdxl test for bit line plane arrangement vmnc: return error on decode_init() failure. libvorbis: add/update error messages libvorbis: use AVFifoBuffer for output packet buffer libvorbis: remove unneeded e_o_s check libvorbis: check return values for functions that can return errors libvorbis: use float input instead of s16 libvorbis: do not flush libvorbis analysis if dsp state was not initialized libvorbis: use VBR by default, with default quality of 3 libvorbis: fix use of minrate/maxrate AVOptions ... Conflicts: Changelog doc/APIchanges libavcodec/avcodec.h libavcodec/dpxenc.c libavcodec/libvorbis.c libavcodec/vmnc.c libavformat/asfdec.c libavformat/id3v2enc.c libavformat/internal.h libavformat/mp3enc.c libavformat/utils.c libavformat/version.h libswscale/utils.c tests/fate/video.mak tests/ref/fate/nuv tests/ref/fate/prores-alpha tests/ref/lavf/ffm tests/ref/vsynth1/prores tests/ref/vsynth2/prores Merged-by: Michael Niedermayer <[email protected]>
| * FATE: do not decode audio in the nuv test.Justin Ruggles2012-02-291-19/+0
| | | | | | | | We already have sufficient coverage for 16-bit pcm.
| * fate: add cdxl test for bit line plane arrangementPaul B Mahol2012-02-291-0/+11
| | | | | | | | | | Signed-off-by: Paul B Mahol <[email protected]> Signed-off-by: Justin Ruggles <[email protected]>
| * prores: handle 444 chroma in right orderKostya Shishkov2012-02-291-2/+2
| | | | | | | | | | | | | | ProRes codes chroma blocks in 444 mode in different order than luma blocks, so make both decoder and encoder read/write chroma blocks in right order. Reported by Phil Barrett
| * fate: Overhaul WavPack coverageDerek Buitenhuis2012-02-2727-1/+26
| | | | | | | | | | | | | | | | WavPack has a comprehensive test suite, and a bunch of corner cases. Signed-off-by: Derek Buitenhuis <[email protected]> Signed-off-by: Ronald S. Bultje <[email protected]>
| * lavf: don't guess r_frame_rate from either stream or codec timebase.Anton Khirnov2012-02-261-1/+1
| | | | | | | | | | | | | | Neither of those is guaranteed to be connected to framerate in any way (if it even exists). Fixes bug 56.
| * avconv: saner output video timebase.Anton Khirnov2012-02-2651-3775/+3778
| | | | | | | | | | | | | | | | | | | | r_frame_rate should in theory have something to do with input framerate, but in practice it is often made up from thin air by lavf. So unless we are targeting a constant output framerate, it's better to just use input stream timebase. Brings back dropped frames in nuv and cscd tests introduced in cd1ad18a6539bd7fc2dc4c1740fbcbd498c0c0a2
* | eval: add root() to solve f(x)=0Michael Niedermayer2012-02-271-0/+6
| | | | | | | | Signed-off-by: Michael Niedermayer <[email protected]>
* | eval: Allow specifying the variable id.Michael Niedermayer2012-02-261-2/+2
| | | | | | | | | | Reviewed-by: Nicolas George <[email protected]> Signed-off-by: Michael Niedermayer <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-02-264-0/+78
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (34 commits) mlp_parser: fix the channel mask value used for the top surround channel vorbisenc: check all allocations for failure roqaudioenc: return AVERROR codes instead of -1 roqaudioenc: set correct bit rate roqaudioenc: use AVCodecContext.frame_size correctly. roqaudioenc: remove unneeded sample_fmt check ra144enc: use int16_t* for input samples rather than void* ra144enc: set AVCodecContext.coded_frame ra144enc: remove unneeded sample_fmt check nellymoserenc: set AVCodecContext.coded_frame nellymoserenc: improve error checking in encode_init() nellymoserenc: return AVERROR codes instead of -1 libvorbis: improve error checking in oggvorbis_encode_init() mpegaudioenc: return AVERROR codes instead of -1 libfaac: improve error checking and handling in Faac_encode_init() avutil: add AVERROR_UNKNOWN check for coded_frame allocation failure in several audio encoders audio encoders: do not set coded_frame->key_frame. g722enc: check for trellis data allocation error libspeexenc: export encoder delay through AVCodecContext.delay ... Conflicts: doc/APIchanges libavcodec/avcodec.h libavcodec/fraps.c libavcodec/kgv1dec.c libavcodec/libfaac.c libavcodec/libgsm.c libavcodec/libvorbis.c libavcodec/mlp_parser.c libavcodec/roqaudioenc.c libavcodec/vorbisenc.c libavutil/avutil.h libavutil/error.c libavutil/error.h Merged-by: Michael Niedermayer <[email protected]>
| * fate: add tests for cdxl videoPaul B Mahol2012-02-254-0/+78
| | | | | | | | | | Signed-off-by: Paul B Mahol <[email protected]> Signed-off-by: Justin Ruggles <[email protected]>
| * lavf: don't set AVCodecContext.has_b_frames in compute_pkt_fields().Anton Khirnov2012-02-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | It is not supposed to be done outside lavc. This is basically a revert of 818062f2f346df30f4ec0c0c1f54e8025cc3a80a. It is unclear what issue this was supposed to fix, if it reappears again it will have to be fixed in a more proper place. The wtv-demux test change is because the sample starts with a B-frame.
* | fate: Overhaul WavPack coverageDerek Buitenhuis2012-02-2527-1/+26
| | | | | | | | | | | | | | | | WavPack has a comprehensive test suite, and a bunch of corner cases. Signed-off-by: Derek Buitenhuis <[email protected]> Signed-off-by: Michael Niedermayer <[email protected]>
* | fate: add forgotten random_seed refMichael Niedermayer2012-02-241-0/+1
| | | | | | | | Signed-off-by: Michael Niedermayer <[email protected]>
* | eval: Add taylor series evaluation support.Michael Niedermayer2012-02-221-0/+6
| | | | | | | | Signed-off-by: Michael Niedermayer <[email protected]>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-02-171-71/+72
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: shorten: Use separate pointers for the allocated memory for decoded samples. atrac3: Fix crash in tonal component decoding. ws_snd1: Fix wrong samples counts. movenc: Don't set a default sample duration when creating ismv rtp: Factorize the check for distinguishing RTCP packets from RTP golomb: avoid infinite loop on all-zero input (or end of buffer). bethsoftvid: synchronize video timestamps with audio sample rate bethsoftvid: add audio stream only after getting the first audio packet bethsoftvid: Set video packet duration instead of accumulating pts. bethsoftvid: set packet key frame flag for audio and I-frame video packets. bethsoftvid: fix read_packet() return codes. bethsoftvid: pass palette in side data instead of in a separate packet. sdp: Ignore RTCP packets when autodetecting RTP streams proresenc: initialise 'sign' variable mpegaudio: replace memcpy by SIMD code vc1: prevent using last_frame as a reference for I/P first frame. Conflicts: libavcodec/atrac3.c libavcodec/golomb.h libavcodec/shorten.c libavcodec/ws-snd1.c tests/ref/fate/bethsoft-vid Merged-by: Michael Niedermayer <[email protected]>
| * bethsoftvid: synchronize video timestamps with audio sample rateJustin Ruggles2012-02-161-70/+71
| | | | | | | | | | | | | | | | According to unofficial documentation, the video rate is locked to the audio sample rate. This results in proper synchronization of audio and video timestamps from the demuxer. This only works if the first audio packet occurs before the first video packet or the audio sample rate is the default rate of 11111 Hz, both of which are true for all samples in our archive.
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