summaryrefslogtreecommitdiff
Commit message (Collapse)AuthorAge
* aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-CJustin Ruggles2012-03-22
|
* aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFFJustin Ruggles2012-03-22
|
* aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-CJustin Ruggles2012-03-22
|
* aiffdec: do not set bit rate if block duration is unknownJustin Ruggles2012-03-22
| | | | CC: libav-stable@libav.org
* wmall: output packet only if we have decoded some samplesKostya Shishkov2012-03-22
| | | | | Also set CODEC_CAP_DELAY to indicate that decoder may still have some undecoded data left in internal buffer.
* adxenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* adxenc: Use the AVFrame in ADXContext for coded_frameJustin Ruggles2012-03-21
|
* indeo4: fix out-of-bounds function call.Ronald S. Bultje2012-03-21
| | | | | | | 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 Biurrun2012-03-21
| | | | | 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 Biurrun2012-03-21
|
* vorbisenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* libvorbis: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* libopencore-amrnbenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* ra144enc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* nellymoserenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* roqaudioenc: use AVCodec.encode2()Justin Ruggles2012-03-21
| | | | | 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 Ruggles2012-03-21
|
* libvo_amrwbenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* libvo_aacenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* wmaenc: use AVCodec.encode2()Justin Ruggles2012-03-21
|
* mpegaudioenc: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | Update FATE references due to encoder delay.
* libmp3lame: use AVCodec.encode2()Justin Ruggles2012-03-20
|
* libgsmenc: use AVCodec.encode2()Justin Ruggles2012-03-20
|
* libfaac: use AVCodec.encode2()Justin Ruggles2012-03-20
| | | | | 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 Ruggles2012-03-20
|
* 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.
* flacenc: use AVCodec.encode2()Justin Ruggles2012-03-20
|
* adpcmenc: update to AVCodec.encode2()Justin Ruggles2012-03-20
|
* ac3enc: update to AVCodec.encode2()Justin Ruggles2012-03-20
| | | | Update FATE references due to encoder delay.
* aacenc: use AVCodec.encode2()Justin Ruggles2012-03-20
|
* avcodec: add code for a frame queue for use by audio encoders with delayJustin Ruggles2012-03-20
| | | | | This simplifies matching of timestamps between input frames and output packets.
* avconv: free packet in write_frame() when discarding due to frame number limitJustin Ruggles2012-03-20
| | | | Fixes a memleak when using the -frames option with audio.
* FATE: use +/- flag option syntax for vp8 emu-edge testsJustin Ruggles2012-03-20
|
* lavf: make av_interleave_packet_per_dts() private.Anton Khirnov2012-03-20
| | | | | There is no reason for it to be public, it's only meant to be used internally.
* lavf: deprecate av_read_packet().Anton Khirnov2012-03-20
| | | | | | 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 VorbisJustin Ruggles2012-03-20
| | | | | | | | | 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 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.
* lavc: shrink encoded audio packet size after encoding.Justin Ruggles2012-03-20
|
* xa: set correct bit rateJustin Ruggles2012-03-20
| | | | Also fixes stream duration calculation.
* xa: do not set bit_rate, block_align, or bits_per_coded_sampleJustin Ruggles2012-03-20
| | | | The values in the header refer to decoded data, not compressed data.
* xa: fix end-of-file handlingJustin Ruggles2012-03-20
| | | | | Do not output an extra packet when out_size is reached. Also return AVERROR_EOF instead of AVERROR(EIO).
* xa: fix timestamp calculationJustin Ruggles2012-03-20
| | | | The packet duration is always 28 samples.
* bink: fix typo in FFALIGN() argumentKostya Shishkov2012-03-20
|
* bink: align plane width to 8 when calculating bundle sizesKostya Shishkov2012-03-20
| | | | This fixes decoding of Bink files with non-multiple-of-16 width.
* doc: pass -Idoc texi2html and texi2podMans Rullgard2012-03-20
| | | | | | This fixes doc generation in build tree separate from source. Signed-off-by: Mans Rullgard <mans@mansr.com>
* doc: texi2pod: add -I flagMans Rullgard2012-03-20
| | | | | | | This allows specifying additional directories to search for @include files. Signed-off-by: Mans Rullgard <mans@mansr.com>
* movenc: Add a min_frag_duration optionMartin Storsjö2012-03-20
| | | | | | | | The other fragmentation options (frag_duration, frag_size and frag_keyframe) are combined with OR, cutting fragments at the first of the conditions being fulfilled. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxersMartin Storsjö2012-03-20
| | | | | | | This enables reordering of UDP packets by default, unless the caller explicitly sets -max_delay 0. Signed-off-by: Martin Storsjö <martin@martin.st>
* libavformat: Set the default for the max_delay option to -1Martin Storsjö2012-03-20
| | | | | | | | | | Make the muxers/demuxers that use the field handle the default -1 in the same way as 0. This allows distinguishing an intentionally set 0 from the default value where the user hasn't set it. Signed-off-by: Martin Storsjö <martin@martin.st>
* Generate manpages for AV{Format,Codec}Context AVOptions.Anton Khirnov2012-03-20
|
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-18 21:10:21 +0000