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* vp8: convert loopfilter x86 assembly to use cpuflags().Ronald S. Bultje2012-03-03
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* vp8: convert idct/mc x86 assembly to use cpuflags().Ronald S. Bultje2012-03-03
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* swscale: remove now unnecessary hack.Ronald S. Bultje2012-03-03
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* x86inc: don't "bake" stack_offset in named arguments.Loren Merritt2012-03-03
| | | | Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* fate: Add sunrast regression testDerek Buitenhuis2012-03-03
| | | | | Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* wmaenc: fix m/s stereo encoding for the first frameJustin Ruggles2012-03-03
| | | | | | | | 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
* wmaenc: return s->block_align instead of recalculating itJustin Ruggles2012-03-03
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* wmaenc: check final frame size against output packet sizeJustin Ruggles2012-03-03
| | | | | | | 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.
* wmaenc: require a large enough output buffer to prevent overwritesJustin Ruggles2012-03-03
| | | | | | | | 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
* wmaenc: limit allowed sample rate to 48kHzJustin Ruggles2012-03-03
| | | | | | | | | 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
* wmaenc: limit block_align to MAX_CODED_SUPERFRAME_SIZEJustin Ruggles2012-03-03
| | | | | | | | | | 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
* tiertexseq: set correct block_align for audioJustin Ruggles2012-03-03
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* tiertexseq: set audio stream start time to 0Justin Ruggles2012-03-03
| | | | | Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
* voc/avs: Do not change the sample rate mid-stream.Justin Ruggles2012-03-03
| | | | | Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.
* segafilm: use the sample rate as the time base for audio streamsJustin Ruggles2012-03-03
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* ea: fix audio ptsJustin Ruggles2012-03-03
| | | | | | | 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.
* psx-str: fix audio ptsJustin Ruggles2012-03-03
| | | | Each packet has 18 sectors with 224/channels samples in each sector.
* vqf: set packet durationJustin Ruggles2012-03-03
| | | | | | Fixes timestamp calculation. The FATE reference is updated because timestamp calculations are now more accurate. Previous timestamps were based on average bit rate.
* tta demuxer: set packet durationJustin Ruggles2012-03-03
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* mpegaudio_parser: do not ignore information from the first parsed frameJustin Ruggles2012-03-03
| | | | Update some demuxing and seeking fate tests.
* mpegaudio_parser: be less picky about the start positionMichael Niedermayer2012-03-03
| | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* thp: set audio packet durationsJustin Ruggles2012-03-03
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* avcodec: add a Vorbis parser to get packet durationJustin Ruggles2012-03-03
| | | | This also allows for removing some of the Vorbis-related hacks.
* vorbisdec: read the previous window flag for long windowsJustin Ruggles2012-03-03
| | | | | | 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.
* lavc: free the output packet when encoding failed or produced no output.Anton Khirnov2012-03-03
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* lavc: preserve avpkt->destruct in ff_alloc_packet().Anton Khirnov2012-03-03
| | | | | Also, don't bother with saving/restoring data, av_init_packet doesn't touch it.
* lavc: clarify the meaning of AVCodecContext.frame_number.Anton Khirnov2012-03-03
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* mpegts: Pad the packet buffer in handle_packet().Alex Converse2012-03-02
| | | | | | | 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
* mpegts: Do not call read_sl_header() when no bytes remain in the buffer.Alex Converse2012-03-02
| | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* amrwb: remove duplicate arguments from extrapolate_isf().Ronald S. Bultje2012-03-02
| | | | | | | | 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
* amrwb: error out early if mode is invalid.Ronald S. Bultje2012-03-02
| | | | | | | | 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
* h264: change underread for 10bit QPEL to overread.Ronald S. Bultje2012-03-02
| | | | | This prevents us from reading before the start of the buffer, and thus prevents crashes resulting from this behaviour. Fixes bug 237.
* matroska: check buffer size for RM-style byte reordering.Ronald S. Bultje2012-03-02
| | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* vp8: disable mmx functions with sse/sse2 counterparts on x86-64.Ronald S. Bultje2012-03-02
| | | | | x86-64 is guaranteed to have at least SSE2, therefore the MMX/MMX2 functions will never be used in practice.
* vp8: change int stride to ptrdiff_t stride.Ronald S. Bultje2012-03-02
| | | | | On 64bit platforms with 32bit int, this means we won't have to sign- extend the integer anymore.
* wma: fix invalid buffer size assumptions causing random overreads.Ronald S. Bultje2012-03-02
| | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* Windows Media Audio Lossless decoderMashiat Sarker Shakkhar2012-03-02
| | | | | | | | | | | | 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>
* rv10/20: Fix slice overflow with checked bitstream reader.Alex Converse2012-03-02
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* h263dec: Disallow width/height changing with frame threads.Michael Niedermayer2012-03-02
| | | | | | | Fixes CVE-2011-3937 Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* rv10/20: Fix a buffer overread caused by losing track of the remaining ↵Alex Converse2012-03-02
| | | | | | | buffer size. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* rmdec: Honor .RMF tag size rather than assuming 18.Alex Converse2012-03-02
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* g722: Fix the QMF scalingMartin Storsjö2012-03-02
| | | | | | | | | | | | | | | 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>
* r3d: don't set codec timebase.Anton Khirnov2012-03-02
| | | | | | It's not supposed to be set by demuxers. Set avg_frame_rate and r_frame_rate instead.
* electronicarts: set timebase for tgv video.Anton Khirnov2012-03-02
| | | | | | | | 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.
* electronicarts: parse the framerate for cmv video.Anton Khirnov2012-03-02
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* ogg: don't set codec timebaseAnton Khirnov2012-03-02
| | | | Demuxers are not supposed to set it.
* electronicarts: don't set codec timebaseAnton Khirnov2012-03-02
| | | | | | Demuxers are not supposed to set it. Set stream timebase and framerates instead (this is a cfr container with no timestamps).
* avs: don't set codec timebaseAnton Khirnov2012-03-02
| | | | | Demuxers are not supposed to set it. Set r_frame_rate and avg_frame_rate instead.
* wavpack: Fix an integer overflowDerek Buitenhuis2012-03-02
| | | | | | | | | | Integer Overflow Checker detected an integer overflow while FATE was running. See: http://fate.libav.org/x86_64-linux-ioc/ Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* swscale: K&R formatting cosmetics for PowerPC code (part II/II)Diego Biurrun2012-03-02
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