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* Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-24
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * fate: add wmalossless test.Ronald S. Bultje2012-03-23
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| * FATE: Add ZeroCodec testDerek Buitenhuis2012-03-22
| | | | | | | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-22
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: (26 commits) adxenc: use AVCodec.encode2() adxenc: Use the AVFrame in ADXContext for coded_frame indeo4: fix out-of-bounds function call. configure: Restructure help output. configure: Internal-only components should not be command-line selectable. vorbisenc: use AVCodec.encode2() libvorbis: use AVCodec.encode2() libopencore-amrnbenc: use AVCodec.encode2() ra144enc: use AVCodec.encode2() nellymoserenc: use AVCodec.encode2() roqaudioenc: use AVCodec.encode2() libspeex: use AVCodec.encode2() libvo_amrwbenc: use AVCodec.encode2() libvo_aacenc: use AVCodec.encode2() wmaenc: use AVCodec.encode2() mpegaudioenc: use AVCodec.encode2() libmp3lame: use AVCodec.encode2() libgsmenc: use AVCodec.encode2() libfaac: use AVCodec.encode2() g726enc: use AVCodec.encode2() ... Conflicts: configure libavcodec/Makefile libavcodec/ac3enc.c libavcodec/adxenc.c libavcodec/libgsm.c libavcodec/libvorbis.c libavcodec/vorbisenc.c libavcodec/wmaenc.c tests/ref/acodec/g722 tests/ref/lavf/asf tests/ref/lavf/ffm tests/ref/lavf/mkv tests/ref/lavf/mpg tests/ref/lavf/rm tests/ref/lavf/ts tests/ref/seek/lavf_asf tests/ref/seek/lavf_ffm tests/ref/seek/lavf_mkv tests/ref/seek/lavf_mpg tests/ref/seek/lavf_rm tests/ref/seek/lavf_ts Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * mpegaudioenc: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | | | | | Update FATE references due to encoder delay.
| * g722enc: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | | | | | | | FATE reference updated due timestamp rounding because of resampling from 44100 Hz to 16000 Hz in avconv.
| * ac3enc: update to AVCodec.encode2()Justin Ruggles2012-03-20
| | | | | | | | Update FATE references due to encoder delay.
* | Fix type and codetype fields in the MMF header for some phones.Vidar Madsen2012-03-21
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* | vf_pad: port to new drawutils API.Nicolas George2012-03-21
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* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-21
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * avconv: pass input stream timestamps to audio encodersJustin Ruggles2012-03-20
| | | | | | | | | | 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 Ruggles2012-03-20
| | | | | | | | The packet duration is always 28 samples.
| * FATE: change fate-maxis-xa to a normal demuxing testPaul B Mahol2012-03-19
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
| * FATE: add test for adpcm-ea-maxis-xaPaul B Mahol2012-03-19
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* | fate/zerocodec: fix permissionsMichael Niedermayer2012-03-20
| | | | | | | | | | Reported-by: Deamon404 Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | FATE: Add ZeroCodec testDerek Buitenhuis2012-03-20
| | | | | | | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-20
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
| * adpcm: fix nb_samples rounding for adpcm_ima_dk3, and update reference.Ronald S. Bultje2012-03-18
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* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-18
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * 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 <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-17
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: resample: allocate a large enough output buffer fate: fix enc_dec_pcm tests with remote target wmaenc: remove bit-exact hack FATE: remove WMA acodec tests FATE: add WMAv1 and WMAv2 encode/decode tests with fuzzy comparison FATE: add AC-3 and E-AC-3 encode/decode tests with fuzzy comparison qtrle: Use bytestream2 functions to prevent buffer overreads. vqavideo: check malloc return values x11grab: fix a memory leak exposed by valgrind threads: fix old frames returned after avcodec_flush_buffers() MPV: always mark dummy frames as reference h264: fix deadlocks on incomplete reference frame decoding. mpeg4: report frame decoding completion at ff_MPV_frame_end(). mimic: don't use self as reference, and report completion at end of decode(). Conflicts: libavcodec/h264.c libavcodec/qtrle.c libavcodec/resample.c libavcodec/vqavideo.c libavdevice/x11grab.c tests/ref/seek/wmav1_asf tests/ref/seek/wmav2_asf Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * FATE: remove WMA acodec testsJustin Ruggles2012-03-17
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* | mpegvideo: don't pretend the first frame is always a key frameWolfram Gloger2012-03-16
| | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Wolfram Gloger <wmglo@dent.med.uni-muenchen.de> Modify the parser initialization so that parsers can set pict_type themselves. Use this in the mpegvideo parser so that initial frames are not unconditionally I frames. I have had this in my tree for several years. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | Merge remote-tracking branch 'qatar/master'Michael Niedermayer2012-03-15
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * qatar/master: h264: stricter reference limit enforcement. h264: increase reference poc list from 16 to 32. xa_adpcm: limit filter to prevent xa_adpcm_table[] array bounds overruns. snow: check reference frame indices. snow: reject unsupported chroma shifts. Add ffvhuff encoding and decoding regression test anm: convert to bytestream2 API bytestream: add more unchecked variants for bytestream2 API jvdec: unbreak video decoding jv demux: set video stream duration fate: add pam image regression test Conflicts: libavcodec/adpcm.c libavcodec/anm.c libavcodec/h264.c libavcodec/mpegvideo.h libavcodec/snowdec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
| * Add ffvhuff encoding and decoding regression testPaul B Mahol2012-03-14
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
| * fate: add pam image regression testPaul B Mahol2012-03-14
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Janne Grunau <janne-libav@jannau.net>
| * FATE: add test for cdxl demuxerPaul B Mahol2012-03-12
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* | FATE: add test for cdxl demuxerPaul B Mahol2012-03-11
| | | | | | | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | lavf: Add system to seperate relative timestamps from absolute ones.Michael Niedermayer2012-03-09
| | | | | | | | | | | | | | | | | | With this we can always know if a timestamp is based on added durations from an unknown origin or if it is based on a correct timestamp (and possibly added durations) This should fix some bugs where this distinction was mixed up. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* | 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.
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