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* Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-06
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()Justin Ruggles2012-03-05
| | | | | | | | | | | | | | | | | | | | 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.
| * aiffdec: do not set AVCodecContext.frame_sizeJustin Ruggles2012-03-05
| | | | | | | | | | | | | | | | | | It is unnecessary. Also, for some codecs we're reading more than 1 frame per packet. Instead we use a private context variable to calculate the bit rate, stream duration, and packet durations. Updated FATE seek test, which has slightly different timestamps due to a more accurate bit rate calculation.
| * lavf: deobfuscate read_frame_internal().Anton Khirnov2012-03-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-05
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (27 commits) cmdutils: use new avcodec_is_decoder/encoder() functions. lavc: make codec_is_decoder/encoder() public. lavc: deprecate AVCodecContext.sub_id. libcdio: add a forgotten AVClass to the private context. swscale: remove "cpu flags" from -sws_flags description. proresenc: give user a possibility to alter some encoding parameters vorbisenc: add output buffer overwrite protection libopencore-amrnbenc: fix end-of-stream handling ra144enc: fix end-of-stream handling nellymoserenc: zero any leftover packet bytes nellymoserenc: use proper MDCT overlap delay qpeg: Use bytestream2 functions to prevent buffer overreads. swscale: make %rep unconditional. vp8: convert simple loopfilter x86 assembly to use named arguments. vp8: convert idct x86 assembly to use named arguments. vp8: convert mc x86 assembly to use named arguments. vp8: convert loopfilter x86 assembly to use cpuflags(). vp8: convert idct/mc x86 assembly to use cpuflags(). swscale: remove now unnecessary hack. x86inc: don't "bake" stack_offset in named arguments. ... Conflicts: cmdutils.c doc/APIchanges libavcodec/mpeg12.c libavcodec/options.c libavcodec/qpeg.c libavcodec/utils.c libavcodec/version.h libavdevice/libcdio.c tests/lavf-regression.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * fate: Add sunrast regression testDerek Buitenhuis2012-03-03
| | | | | | | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
| * wmaenc: fix m/s stereo encoding for the first frameJustin Ruggles2012-03-03
| | | | | | | | | | | | | | | | 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
* | lavf: Do not compute the packet duration based on the bitrate if the ↵Michael Niedermayer2012-03-04
| | | | | | | | | | | | | | | | | | 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 <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-04
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * tiertexseq: set audio stream start time to 0Justin Ruggles2012-03-03
| | | | | | | | | | Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
| * voc/avs: Do not change the sample rate mid-stream.Justin Ruggles2012-03-03
| | | | | | | | | | Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.
| * vqf: set packet durationJustin Ruggles2012-03-03
| | | | | | | | | | | | 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 Ruggles2012-03-03
| | | | | | | | Update some demuxing and seeking fate tests.
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-03
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * g722: Fix the QMF scalingMartin Storsjö2012-03-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 <martin@martin.st>
| * electronicarts: set timebase for tgv video.Anton Khirnov2012-03-02
| | | | | | | | | | | | | | | | 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 Khirnov2012-03-02
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| * electronicarts: don't set codec timebaseAnton Khirnov2012-03-02
| | | | | | | | | | | | 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 Niedermayer2012-03-02
| | | | | | | | | | | | | | | | | | 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 <michaelni@gmx.at>
* | fate: Add sunrast regression testDerek Buitenhuis2012-03-02
| | | | | | | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-01
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * FATE: do not decode audio in the nuv test.Justin Ruggles2012-02-29
| | | | | | | | We already have sufficient coverage for 16-bit pcm.
| * fate: add cdxl test for bit line plane arrangementPaul B Mahol2012-02-29
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
| * dpxenc: Don't include the libavcodec ident if bitexact mode is enabledMartin Storsjö2012-02-29
| | | | | | | | | | | | This avoids breaking fate every time the lavc version is bumped. Signed-off-by: Martin Storsjö <martin@martin.st>
| * prores: store and retrieve extended colourspace informationKostya Shishkov2012-02-29
| | | | | | | | Based on the patch by Phil Barrett.
| * prores: handle 444 chroma in right orderKostya Shishkov2012-02-29
| | | | | | | | | | | | | | 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
| * mpegvideo_enc: add chroma/luma_elim_threshold private options.Anton Khirnov2012-02-29
| | | | | | | | Deprecate corresponding AVCodecContext fields.
| * fate: Overhaul WavPack coverageDerek Buitenhuis2012-02-27
| | | | | | | | | | | | | | | | WavPack has a comprehensive test suite, and a bunch of corner cases. Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
| * lavf: don't guess r_frame_rate from either stream or codec timebase.Anton Khirnov2012-02-26
| | | | | | | | | | | | | | 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 Khirnov2012-02-26
| | | | | | | | | | | | | | | | | | | | 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
| * rawdec: set timebase to 1/fps.Anton Khirnov2012-02-26
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* | fate: disable deshake test as it isnt binary identical across archs.Michael Niedermayer2012-02-29
| | | | | | | | | | | | probably due to float rounding Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: switch lavfi-pp targets which need bitexact flags to use them.Michael Niedermayer2012-02-29
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: More libpostproc tests to improve coverage.Michael Niedermayer2012-02-29
| | | | | | | | | | Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | tests: add libpostproc testMichael Niedermayer2012-02-28
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | fate: add test for the life video source.Nicolas George2012-02-27
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* | fate: add test for the unsharp video filter.Nicolas George2012-02-27
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* | fate: add test for the transpose video filter.Nicolas George2012-02-27
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* | fate: add test for the thumbnail video filter.Nicolas George2012-02-27
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* | fate: add test for the select video filter.Nicolas George2012-02-27
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* | fate: add test for the pad video filter.Nicolas George2012-02-27
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* | fate: add test for the overlay video filter.Nicolas George2012-02-27
| | | | | | | | | | The test also covers somewhat the split, scale, pad and fifo filters and the graph parser.
* | fate: add test for the fade video filter.Nicolas George2012-02-27
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* | fate: add test for the drawbox video filter.Nicolas George2012-02-27
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* | fate: add test for the deshake video filter.Nicolas George2012-02-27
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* | fate: add tests for the setsar and setdar video filters.Nicolas George2012-02-27
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* | fate: add tests for testsrc.Nicolas George2012-02-27
| | | | | | | | | | This test uses the lavfi device and is ready for more similar tests.
* | eval: add root() to solve f(x)=0Michael Niedermayer2012-02-27
| | | | | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | eval: Allow specifying the variable id.Michael Niedermayer2012-02-26
| | | | | | | | | | Reviewed-by: Nicolas George <nicolas.george@normalesup.org> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-02-26
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>