FFmpeg

Commit Graph

Author	SHA1	Message	Date
Justin Ruggles	c019070fda	riffenc: use av_get_audio_frame_duration() For encoding, frame_size is not a reliable indicator of packet duration. Also, we don't want to have to force the demuxer to find frame_size for stream copy to work.	13 years ago
Justin Ruggles	9524cf79df	avcodec: add av_get_audio_frame_duration() function. This is a utility function for the user to get the frame duration based on the codec id, frame size in bytes, and various AVCodecContext parameters.	13 years ago
Justin Ruggles	6699d07480	avcodec: add av_get_exact_bits_per_sample() function This only returns bits per sample when it is exactly correct. That is, the codec contains only raw samples with no frame headers or padding. This applies to basically all PCM codecs and a small subset of ADPCM codecs.	13 years ago
Anton Khirnov	27c7ca9c12	lavf: deobfuscate read_frame_internal(). 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.	13 years ago
Anton Khirnov	dcee811505	lavf: make read_from_packet_buffer() more flexible. Make packet buffer a parameter, don't hardcode it to be AVFormatContext.packet_buffer. Also move the function higher in the file, since it will be called from read_frame_internal().	13 years ago
Anton Khirnov	52b0943f10	lavf: factorize freeing a packet buffer.	13 years ago
Fabian Greffrath	c9dbac36ad	Fix format string vulnerability detected by -Wformat-security. Signed-off-by: Diego Biurrun <diego@biurrun.de>	13 years ago
Diego Biurrun	0a41f47dc1	dv: Do not redundantly initialize struct members to zero.	13 years ago
Mans Rullgard	356ee8d7de	x86: clean up ff_dsputil_init_mmx() This splits ff_dsputil_init_mmx() into multiple functions, one for each MMX/SSE level, somewhat simplifying the nested conditions. Signed-off-by: Mans Rullgard <mans@mansr.com> Signed-off-by: Diego Biurrun <diego@biurrun.de>	13 years ago
Anton Khirnov	3faa141d15	cmdutils: use new avcodec_is_decoder/encoder() functions. Fixes listing encoders.	13 years ago
Anton Khirnov	44fe77b350	lavc: make codec_is_decoder/encoder() public.	13 years ago
Anton Khirnov	02beb9826b	lavc: deprecate AVCodecContext.sub_id. In most places where it's used, it's as a pointless write-only field. Only rv10 decoder actually reads from it, but it stores some internal version info in it. There is no reason for it to be in a public field.	13 years ago
Anton Khirnov	87392b1fd5	libcdio: add a forgotten AVClass to the private context.	13 years ago
Ronald S. Bultje	1c97b5c4a3	swscale: remove "cpu flags" from -sws_flags description.	13 years ago
Kostya Shishkov	4db4b53dc8	proresenc: give user a possibility to alter some encoding parameters This allows user to select quantisation matrix from different profile, stamp frames with custom vendor string and change target bitrate.	13 years ago
Justin Ruggles	1ba08c94f5	vorbisenc: add output buffer overwrite protection	13 years ago
Justin Ruggles	fe78470a8b	libopencore-amrnbenc: fix end-of-stream handling Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush the encoder at the end of encoding. This is needed in order to have all input samples decoded.	13 years ago
Justin Ruggles	b0350c1c30	ra144enc: fix end-of-stream handling Use CODEC_CAP_DELAY and CODEC_CAP_SMALL_LAST_FRAME to properly pad and flush the encoder at the end of encoding. This is needed in order to have all input samples decoded.	13 years ago
Justin Ruggles	29e2c85310	nellymoserenc: zero any leftover packet bytes fixes writing of uninitialized packet data	13 years ago
Justin Ruggles	6c7a01621c	nellymoserenc: use proper MDCT overlap delay	13 years ago
Aneesh Dogra	3e9cd8b4b0	qpeg: Use bytestream2 functions to prevent buffer overreads. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>	13 years ago
Ronald S. Bultje	dccb2cd3f9	swscale: make %rep unconditional. Fixes pre-processing with latest versions of nasm.	13 years ago
Ronald S. Bultje	b4188f0d46	vp8: convert simple loopfilter x86 assembly to use named arguments.	13 years ago
Ronald S. Bultje	8476ca3b4e	vp8: convert idct x86 assembly to use named arguments.	13 years ago
Ronald S. Bultje	21ffc78fd7	vp8: convert mc x86 assembly to use named arguments.	13 years ago
Ronald S. Bultje	28170f1a39	vp8: convert loopfilter x86 assembly to use cpuflags().	13 years ago
Ronald S. Bultje	e25be47154	vp8: convert idct/mc x86 assembly to use cpuflags().	13 years ago
Ronald S. Bultje	8249a23fc1	swscale: remove now unnecessary hack.	13 years ago
Loren Merritt	0f53d0cf4b	x86inc: don't "bake" stack_offset in named arguments. Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>	13 years ago
Derek Buitenhuis	6aa6e3e814	fate: Add sunrast regression test Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>	13 years ago
Justin Ruggles	51ddf35c90	wmaenc: fix m/s stereo encoding for the first frame 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	13 years ago
Justin Ruggles	8ed7488ea3	wmaenc: return s->block_align instead of recalculating it	13 years ago
Justin Ruggles	5d652e063b	wmaenc: check final frame size against output packet size Currently we have an assert() that prevents the frame from being too large, but it is more user-friendly to give an error message instead of aborting on assert(). This condition is quite unlikely due to the minimum bit rate check in encode_init(), but it is still worth having.	13 years ago
Justin Ruggles	dfc4fdedf8	wmaenc: require a large enough output buffer to prevent overwrites The maximum theoretical frame size is around 17000 bytes. Although in practice it will generally be much smaller, we require a larger buffer just to be safe. CC: libav-stable@libav.org	13 years ago
Justin Ruggles	1ec075cfec	wmaenc: limit allowed sample rate to 48kHz ff_wma_init() allows up to 50kHz, but this generates an exponent band size table that requires 65 bands. The code assumes 25 bands in many places, and using sample rates higher than 48kHz will lead to buffer overwrites. CC:libav-stable@libav.org	13 years ago
Justin Ruggles	c2b8dea182	wmaenc: limit block_align to MAX_CODED_SUPERFRAME_SIZE This is near the theoretical limit for wma frame size and is the most that our decoder can handle. Allowing higher bit rates will just end up padding each frame with empty bytes. Fixes invalid writes for avconv when using very high bit rates. CC:libav-stable@libav.org	13 years ago
Justin Ruggles	b7beabab4b	tiertexseq: set correct block_align for audio	13 years ago
Justin Ruggles	f9cf91d822	tiertexseq: set audio stream start time to 0 Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.	13 years ago
Justin Ruggles	0883109b27	voc/avs: Do not change the sample rate mid-stream. Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.	13 years ago
Justin Ruggles	4da374f8a9	segafilm: use the sample rate as the time base for audio streams	13 years ago
Justin Ruggles	ea289186f0	ea: fix audio pts The time base is 1 / sample_rate, not 90000. Several more codecs encode the sample count in the first 4 bytes of the chunk, so we set the durations accordingly. Also, we can set start_time and packet duration instead of keeping track of the sample count in the demuxer.	13 years ago
Justin Ruggles	01be6fa926	psx-str: fix audio pts Each packet has 18 sectors with 224/channels samples in each sector.	13 years ago
Justin Ruggles	d0ab585074	vqf: set packet duration Fixes timestamp calculation. The FATE reference is updated because timestamp calculations are now more accurate. Previous timestamps were based on average bit rate.	13 years ago
Justin Ruggles	101c369b7c	tta demuxer: set packet duration	13 years ago
Justin Ruggles	0b8b7db01b	mpegaudio_parser: do not ignore information from the first parsed frame Update some demuxing and seeking fate tests.	13 years ago
Michael Niedermayer	6776a8f189	mpegaudio_parser: be less picky about the start position Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>	13 years ago
Justin Ruggles	5a9b952201	thp: set audio packet durations	13 years ago
Justin Ruggles	5602a464c9	avcodec: add a Vorbis parser to get packet duration This also allows for removing some of the Vorbis-related hacks.	13 years ago
Justin Ruggles	737ca4482b	vorbisdec: read the previous window flag for long windows When reading sequentially, we are using the actual flag from the previous frame, but when seeking we do not know what the previous window flag was, so we need to read it from the bitstream.	13 years ago
Anton Khirnov	7fb6c9225c	lavc: free the output packet when encoding failed or produced no output.	13 years ago

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32720 Commits (c019070fda6468d16bb5d0891e203cc3fe87605e) All Branches Search

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