FFmpeg

Commit Graph

Author	SHA1	Message	Date
Martin Storsjö	94f1b11a6f	rtpenc: Fix the AVRational used for av_rescale_q_rnd The current one has a zero denominator - this is what was intended in `14aecc50fa`. Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Martin Storsjö	a887c87c23	udp: Print an error message if bind fails Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Ronald S. Bultje	a93b572ae4	smacker: error out if palette copy-with-offset overruns palette size. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Carl Eugen Hoyos	a294a7a1b3	mov: Allow last chunk to have an arbitrary number of samples. Fixes ticket #673. (cherry picked from commit `8dcd2a41ec`) Signed-off-by: Alex Converse <alex.converse@gmail.com>	13 years ago
Reimar Döffinger	632eb1bbae	cdxl demux: do not create packets with uninitialized data at EOF. Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de> Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>	13 years ago
Justin Ruggles	94cf64b81f	cosmetics: reindent	13 years ago
Justin Ruggles	8c1d6ac66a	avformat: do not require a pixel/sample format if there is no decoder Also, do not keep trying to find and open a decoder in try_decode_frame() if we already tried and failed once. Fixes always searching until max_analyze_duration in avformat_find_stream_info() when demuxing codecs without a decoder.	13 years ago
Justin Ruggles	a7fa75684d	avformat: do not fill-in audio packet duration in compute_pkt_fields() Use the estimated duration only to calculate missing timestamps if needed.	13 years ago
Justin Ruggles	6c65cf58fd	lavf: Use av_get_audio_frame_duration() in get_audio_frame_size() 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.	13 years ago
Justin Ruggles	f1e73100d9	siff: do not set AVCodecContext.frame_size also, properly set AVCodecContext.bits_per_coded_sample, AVStreasm.start_time, and AVPacket.duration.	13 years ago
Justin Ruggles	ec2e767bf3	amr demuxer: do not set AVCodecContext.frame_size. it is not necessary.	13 years ago
Justin Ruggles	8d1a20aa7c	aiffdec: do not set AVCodecContext.frame_size 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.	13 years ago
Justin Ruggles	237a855caf	mov: do not set AVCodecContext.frame_size It is not necessary.	13 years ago
Justin Ruggles	9727264220	ape: do not set AVCodecContext.frame_size. prevents lavf from setting incorrect packet durations.	13 years ago
Justin Ruggles	2dd18d4435	rdt: remove workaround for infinite loop with aac avformat_find_stream_info() no longer hangs while waiting for AAC frame_size	13 years ago
Justin Ruggles	9c365fe8ae	avformat: do not require frame_size in avformat_find_stream_info() for CELT In Ogg/CELT, frame_size is found in the same place as the sample_rate and channels, so we do not need to force the frame_size to be parsed.	13 years ago
Justin Ruggles	fbc8c59679	avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3 It was only needed to avoid a bad time base (and thus non-monotone timestamps) for stream copy to avi.	13 years ago
Justin Ruggles	84b6ae0808	avformat: do not require frame_size in avformat_find_stream_info() for AAC We already will get the needed info because of CODEC_CAP_CHANNEL_CONF	13 years ago
Justin Ruggles	620b88a302	swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_size This way we can do stream copy without having the demuxer wait until frame_size has been set.	13 years ago
Justin Ruggles	14aecc50fa	rtpenc: use av_get_audio_frame_duration() for max_frames_per_packet It is more reliable than AVCodecContext.frame_size for codecs with constant packet duration.	13 years ago
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
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
Diego Biurrun	0a41f47dc1	dv: Do not redundantly initialize struct members to zero.	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	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
Alex Converse	1aa708988a	mpegts: Pad the packet buffer in handle_packet(). This allows it to be used with get_bits without the thread of overreads. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Alex Converse	4df369692e	mpegts: Do not call read_sl_header() when no bytes remain in the buffer. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Ronald S. Bultje	9c239f6026	matroska: check buffer size for RM-style byte reordering. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Alex Converse	1697c29d75	rmdec: Honor .RMF tag size rather than assuming 18.	13 years ago
Anton Khirnov	56bf24ad78	r3d: don't set codec timebase. It's not supposed to be set by demuxers. Set avg_frame_rate and r_frame_rate instead.	13 years ago
Anton Khirnov	efec3bc65a	electronicarts: set timebase for tgv video. 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.	13 years ago
Anton Khirnov	e39400c3a8	electronicarts: parse the framerate for cmv video.	13 years ago
Anton Khirnov	1bb3990b56	ogg: don't set codec timebase Demuxers are not supposed to set it.	13 years ago
Anton Khirnov	1d3144c318	electronicarts: don't set codec timebase Demuxers are not supposed to set it. Set stream timebase and framerates instead (this is a cfr container with no timestamps).	13 years ago
Anton Khirnov	10a6e0c346	avs: don't set codec timebase Demuxers are not supposed to set it. Set r_frame_rate and avg_frame_rate instead.	13 years ago
Ingo Brückl	c05e2be9a2	mp3dec: Fix reading file size and frames in VBRI headers The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes" come first, "Frames" behind. Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Alex Converse <alex.converse@gmail.com>	13 years ago
Diego Biurrun	75c553eb26	rmdec: adjust printf format string specifier to fix warning libavformat/rmdec.c:383: warning: format ‘%d’ expects type ‘int’, but argument 7 has type ‘int64_t’	13 years ago
Martin Storsjö	984b914c55	rtpenc: Use MB info side data for splitting H263 packets for RFC 2190 This makes the packetization spec compliant for cases where one single GOB doesn't fit into an RTP packet. Signed-off-by: Martin Storsjö <martin@martin.st>	13 years ago
Justin Ruggles	929dd8c108	dxa: set audio stream time base using the sample rate	13 years ago
Justin Ruggles	aa831c4093	psx-str: do not allow seeking by bytes	13 years ago

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8346 Commits (83632cbb1111f56cb3987c6c56b63dbd8b3fdce1)