FFmpeg

Commit Graph

Author	SHA1	Message	Date
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
Anton Khirnov	e42e9b0e4d	lavc: preserve avpkt->destruct in ff_alloc_packet(). Also, don't bother with saving/restoring data, av_init_packet doesn't touch it.	13 years ago
Anton Khirnov	c179c9e19d	lavc: clarify the meaning of AVCodecContext.frame_number.	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	9d87374ec0	amrwb: remove duplicate arguments from extrapolate_isf(). Prevents warnings because the dst and src overlap (are the same) in the memcpy() inside the function. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Ronald S. Bultje	154b8bb800	amrwb: error out early if mode is invalid. Prevents using the invalid mode as an index in a static array, which would generate invalid reads. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Ronald S. Bultje	291c9b6285	h264: change underread for 10bit QPEL to overread. This prevents us from reading before the start of the buffer, and thus prevents crashes resulting from this behaviour. Fixes bug 237.	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
Ronald S. Bultje	45549339bc	vp8: disable mmx functions with sse/sse2 counterparts on x86-64. x86-64 is guaranteed to have at least SSE2, therefore the MMX/MMX2 functions will never be used in practice.	13 years ago
Ronald S. Bultje	bd66f073fe	vp8: change int stride to ptrdiff_t stride. On 64bit platforms with 32bit int, this means we won't have to sign- extend the integer anymore.	13 years ago
Ronald S. Bultje	349b7977e4	wma: fix invalid buffer size assumptions causing random overreads. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org	13 years ago
Mashiat Sarker Shakkhar	9d25f1f619	Windows Media Audio Lossless decoder Decodes 16-bit WMA Lossless encoded files. 24-bit is not supported yet. Bitstream parser written by Andreas Öman with contributions from Baptiste Coudurier and Ulion. Includes a number of bug-fixes from Benjamin Larsson, Michael Niedermayer and Konstantin Shishkov, shine and polish by Diego Biurrun. Signed-off-by: Diego Biurrun <diego@biurrun.de>	13 years ago
Alex Converse	9243ec4a50	rv10/20: Fix slice overflow with checked bitstream reader.	13 years ago
Michael Niedermayer	71db86d53b	h263dec: Disallow width/height changing with frame threads. Fixes CVE-2011-3937 Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michaelni@gmx.at>	13 years ago
Alex Converse	2f6528537f	rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size. 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
Martin Storsjö	b087ce2bee	g722: Fix the QMF scaling 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>	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

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32748 Commits (2254b559cbcfc0418135f09add37c0a5866b1981) All Branches Search

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