| Commit message (Collapse) | Author | Age | Files | Lines | ||
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| * | mov: Do not read past the end of the ctts_data table. | Alex Converse | 2012-03-22 | 1 | -1/+1 | |
| | | | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | mov: Add missing terminator to mov_ch_layout_map_1ch. | Alex Converse | 2012-03-22 | 1 | -0/+1 | |
| | | | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: Libav-stable@libav.org | |||||
| * | asf: reset side data elements on packet copy. | Ronald S. Bultje | 2012-03-22 | 1 | -0/+2 | |
| | | | | | | | | Prevents crash (double free) when free()ing the original packet. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | wmavoice: fix stack overread. | Ronald S. Bultje | 2012-03-22 | 1 | -2/+3 | |
| | | | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | wmalossless: error out if a subframe is not used by any channel. | Ronald S. Bultje | 2012-03-22 | 1 | -3/+10 | |
| | | | | | | | | Prevents infinite loop because min_channel_len never increments. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | vqa: check palette chunk size before reading data. | Ronald S. Bultje | 2012-03-22 | 1 | -1/+1 | |
| | | | | | | | | Prevents overreads beyond buffer boundaries. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | wmalossless: reset sample pointer for each subframe. | Ronald S. Bultje | 2012-03-22 | 1 | -12/+19 | |
| | | | | | | | | Prevents overwrites when some subframes only encode some channels. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | wmalossless: error out on invalid values for order. | Ronald S. Bultje | 2012-03-22 | 1 | -9/+25 | |
| | | | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org | |||||
| * | FATE: Add ZeroCodec test | Derek Buitenhuis | 2012-03-22 | 2 | -0/+42 | |
| | | | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net> | |||||
| * | oggparseogm: fix order of arguments of avpriv_set_pts_info(). | Michael Niedermayer | 2012-03-22 | 1 | -1/+1 | |
| | | | | | Signed-off-by: Anton Khirnov <anton@khirnov.net> | |||||
| * | pngenc: better upper bound for encoded frame size. | Anton Khirnov | 2012-03-22 | 1 | -13/+17 | |
| | | | | | | | Fixes encoding very large pictures. Thanks to Костя for providing the formula. | |||||
| * | aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-C | Justin Ruggles | 2012-03-22 | 1 | -0/+7 | |
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| * | aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFF | Justin Ruggles | 2012-03-22 | 1 | -13/+8 | |
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| * | aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-C | Justin Ruggles | 2012-03-22 | 1 | -5/+4 | |
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| * | aiffdec: do not set bit rate if block duration is unknown | Justin Ruggles | 2012-03-22 | 1 | -2/+4 | |
| | | | | | CC: libav-stable@libav.org | |||||
| * | wmall: output packet only if we have decoded some samples | Kostya Shishkov | 2012-03-22 | 1 | -2/+2 | |
| | | | | | | Also set CODEC_CAP_DELAY to indicate that decoder may still have some undecoded data left in internal buffer. | |||||
| * | adxenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -16/+20 | |
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| * | adxenc: Use the AVFrame in ADXContext for coded_frame | Justin Ruggles | 2012-03-21 | 1 | -8/+2 | |
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| * | indeo4: fix out-of-bounds function call. | Ronald S. Bultje | 2012-03-21 | 1 | -1/+2 | |
| | | | | | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com> | |||||
| * | configure: Restructure help output. | Diego Biurrun | 2012-03-21 | 1 | -32/+45 | |
| | | | | | | Break some of the longer sections into smaller sensible pieces; make some option descriptions and option ordering more consistent. | |||||
| * | configure: Internal-only components should not be command-line selectable. | Diego Biurrun | 2012-03-21 | 1 | -11/+7 | |
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| * | vorbisenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -12/+31 | |
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| * | libvorbis: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 2 | -21/+51 | |
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| * | libopencore-amrnbenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -14/+37 | |
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| * | ra144enc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 3 | -15/+36 | |
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| * | nellymoserenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 2 | -11/+35 | |
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| * | roqaudioenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -15/+27 | |
| | | | | | | The first frame pts must be saved until we have 8 frames since RoQ audio requires 8 frames in the first packet. | |||||
| * | libspeex: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 2 | -20/+37 | |
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| * | libvo_amrwbenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -8/+29 | |
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| * | libvo_aacenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -25/+96 | |
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| * | wmaenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-21 | 1 | -17/+31 | |
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| * | mpegaudioenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 11 | -110/+131 | |
| | | | | | Update FATE references due to encoder delay. | |||||
| * | libmp3lame: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 2 | -22/+52 | |
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| * | libgsmenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 1 | -10/+24 | |
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| * | libfaac: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 2 | -12/+52 | |
| | | | | | | Encoder output is delayed by several frames, so we keep a queue of input frame timing info to match up with corresponding output packets. | |||||
| * | g726enc: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 1 | -8/+21 | |
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| * | g722enc: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 2 | -13/+35 | |
| | | | | | | FATE reference updated due timestamp rounding because of resampling from 44100 Hz to 16000 Hz in avconv. | |||||
| * | flacenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 1 | -21/+31 | |
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| * | adpcmenc: update to AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 1 | -17/+31 | |
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| * | ac3enc: update to AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 8 | -22/+41 | |
| | | | | | Update FATE references due to encoder delay. | |||||
| * | aacenc: use AVCodec.encode2() | Justin Ruggles | 2012-03-20 | 3 | -18/+51 | |
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| * | avcodec: add code for a frame queue for use by audio encoders with delay | Justin Ruggles | 2012-03-20 | 2 | -0/+252 | |
| | | | | | | This simplifies matching of timestamps between input frames and output packets. | |||||
| * | avconv: free packet in write_frame() when discarding due to frame number limit | Justin Ruggles | 2012-03-20 | 1 | -1/+3 | |
| | | | | | Fixes a memleak when using the -frames option with audio. | |||||
| * | FATE: use +/- flag option syntax for vp8 emu-edge tests | Justin Ruggles | 2012-03-20 | 1 | -1/+1 | |
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| * | lavf: make av_interleave_packet_per_dts() private. | Anton Khirnov | 2012-03-20 | 5 | -16/+39 | |
| | | | | | | There is no reason for it to be public, it's only meant to be used internally. | |||||
| * | lavf: deprecate av_read_packet(). | Anton Khirnov | 2012-03-20 | 6 | -8/+34 | |
| | | | | | | | The caller can achieve the same effect (i.e. getting raw unparsed/mangled packets) with av_read_frame() and AVFMT_FLAG_NOPARSE | AVFMT_FLAG_NOFILLIN | |||||
| * | oggdec: output correct timestamps for Vorbis | Justin Ruggles | 2012-03-20 | 2 | -3/+82 | |
| | | | | | | | | | | Takes encoder delay into account by comparing first the coded page duration with the calculated page duration. Handles last packet duration if needed, also by comparing coded duration with calculated duration. Also does better handling of timestamp generation for packets in the first page for streamed ogg files where the start time is not necessarily zero. | |||||
| * | avconv: pass input stream timestamps to audio encoders | Justin Ruggles | 2012-03-20 | 6 | -740/+741 | |
| | | | | | | 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. | |||||
| * | lavc: shrink encoded audio packet size after encoding. | Justin Ruggles | 2012-03-20 | 1 | -1/+8 | |
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| * | xa: set correct bit rate | Justin Ruggles | 2012-03-20 | 1 | -0/+3 | |
| | | | | | Also fixes stream duration calculation. | |||||
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index : ffmpeg-streaming | |
| Raptor Engineering's fork of FFmpeg with streaming enhancements https://git.ffmpeg.org/ffmpeg.git | Raptor Engineering, LLC |
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