| Commit message (Collapse) | Author | Age |
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The "pts for a given stream" was nonsensical.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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This reduces code size when either piece has been disabled.
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Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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It adds unnecessary complication for insignificant usability improvement.
The user really should know if they'll need resampling compensation before
opening the context.
Note that only the documentation has changed. The current functionality will
still work until the next major bump.
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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This makes it easier for receivers to decide what to do if data
is lost.
Refactor calculating the max payload size, to avoid hardcoding the
header size in too many places, reducing the number of lines that
have to be touched if the header is adjusted further.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Introduced in d7d6efe42b0d.
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Since a NAL_DPA can start a new frame it has to be handled before
ff_thread_finish_setup is called.
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Fixes CVE-2012-2783
CC: libav-stable@libav.org
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Returning 0 for failure is misleading.
CC: libav-stable@libav.org
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Fixes CVE-2012-2791.
CC: libav-stable@libav.org
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Fixes build errors with nasm introduced in 6f40e9f070f7 for stack
memory alignment. Noticed by BugMaster.
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This requires to move the avcodec_default_free_buffers() call to
ff_MPV_common_end() since otherwise delayed pictures would get freed
during a size change.
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Direct rendering capable decoders call get_buffer() which will set the
frame parameters.
Prevents frames with wrong parameters when a decoder outputs delayed
frames after a resolution or pixel format change.
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Fixes a crash in the fuzzed sample sample_varPAR.avi_s26638 with
alternating bit depths.
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Note the time base for the option. Add an additional example description.
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Currently it will do padding, but it does not properly handle
start-of-stream trimming as documented.
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The max_comp option is supposed to limit maximum compensation, but
currently the clipped value is not actually used.
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Interlacing is not supported at all and mismanaged down the normal
codepaths causing possible buffer management issues.
CC: libav-stable@libav.org
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Found-by: pawlkt
CC: libav-stable@libav.org
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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The previous comment about the buffer was wildly inaccurate and
misleading.
Signed-off-by: Martin Storsjö <martin@martin.st>
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These stem from back when both the rtp muxer and rtp depacketizing
shared the same struct.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This fixes potential buffer overwrites.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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This decoder is quite simple and none of the MpegEncContext complexity
is actually needed.
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Use this in VP8/H264-8bit loopfilter functions so they can be used if
there is no aligned stack (e.g. MSVC 32bit or ICC 10.x).
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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This allows AudioMix to be treated the same way as other conversion contexts
and removes the requirement to allocate it at the same time as the
AVAudioResampleContext.
The current matrix get/set functions are split between the public interface
and AudioMix private functions.
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