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* vcr1: return a meaningful error code.Anton Khirnov2013-01-14
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* rpza: return a meaningful error code.Anton Khirnov2013-01-14
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* qdrw: cosmetics, reformatAnton Khirnov2013-01-14
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* qdrw: return meaningful error codes.Anton Khirnov2013-01-14
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* qtrle: return a meaningful error code.Anton Khirnov2013-01-14
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* gifdec: return meaningful error codes.Anton Khirnov2013-01-14
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* interplayvideo: remove a static variable.Anton Khirnov2013-01-14
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* interplayvideo: return meaningful error codes.Anton Khirnov2013-01-14
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* lcldec: return meaningful error codes.Anton Khirnov2013-01-14
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* targa: return meaningful error codes.Anton Khirnov2013-01-14
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* qpeg: return a meaningful error code.Anton Khirnov2013-01-14
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* nuv: return meaningful error codes.Anton Khirnov2013-01-14
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* pcx: return meaningful error codes.Anton Khirnov2013-01-14
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* tmv: return meaningful error codes.Anton Khirnov2013-01-14
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* msrle: return meaningful error codes.Anton Khirnov2013-01-14
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* cscd: return meaningful error codes.Anton Khirnov2013-01-14
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* yadif: x86: fix build for compilers without aligned stackDaniel Kang2013-01-14
| | | | | | | Manually load registers to avoid using 8 registers on x86_32 with compilers that do not align the stack (e.g. MSVC). Signed-off-by: Diego Biurrun <diego@biurrun.de>
* lavc: introduce the convenience function init_get_bits8Luca Barbato2013-01-14
| | | | | Accept the buffer size in bytes and check for overflow before passing the value in bits to init_get_bits.
* lavc: check for overflow in init_get_bitsLuca Barbato2013-01-14
| | | | | | | Fix an undefined behaviour and make the function return a proper error in case of overflow. CC: libav-stable@libav.org
* avpacket: free side data in av_free_packet().Anton Khirnov2013-01-13
| | | | | | | | | Freeing it in av_destruct_packet(), as is done currently, would mean that we allow it to be allocated with other means. But that would make av_packet_new_side_data() unsafe. Side data is not expected to be large, so copying it if required shouldn't be a problem.
* v4l2: do not assert on a value received from outside of LibavAnton Khirnov2013-01-13
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* v4l2: set the average framerate instead of codec timebase.Anton Khirnov2013-01-13
| | | | Codec timebase is supposed to be set by decoders only.
* v4l2: avoid pointless indirection.Anton Khirnov2013-01-13
| | | | | v4l2_read_header() does no cleanup, so it can return directly, without any need for goto.
* vdpau: Add VC-1 decoding via hwaccel infrastructureRémi Denis-Courmont2013-01-13
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* vdpau: Add H.264 decoding via hwaccel infrastructureRémi Denis-Courmont2013-01-13
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* vdpau: Add MPEG-4 decoding via hwaccel infrastructureRémi Denis-Courmont2013-01-13
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* vdpau: Add MPEG-1/2 decoding via hwaccel infrastructureRémi Denis-Courmont2013-01-13
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* vdpau: Add context and common helpers for hwaccel supportRémi Denis-Courmont2013-01-13
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* rtpdec: Send a valid "delay since SR" value in the RTCP RR packetsMartin Storsjö2013-01-12
| | | | | | | | | | | Previously, we always signalled a zero time since the last RTCP SR, which is dubious. The code also suggested that this would be the difference in RTP NTP time units (32.32 fixed point), while it actually is in in 1/65536 second units. (RFC 3550 section 6.4.1) Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec: Calculate and report packet reception jitterMartin Storsjö2013-01-12
| | | | | | | | | This brings back some code that was added originally in 4a6cc061 but never was used, and was removed as unused in 4cc843fa. The code is updated to actually work and is tested to return sane values. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec: Fix the calculation of expected number of packetsMartin Storsjö2013-01-12
| | | | | | | | | The base_seq variable is set to first_seq - 1 (in rtp_init_sequence), so no + 1 is needed here. This avoids reporting 1 lost packet from the start. Signed-off-by: Martin Storsjö <martin@martin.st>
* fate: vp3: Fix fate-vp3-coeff-level64 test dependenciesDiego Biurrun2013-01-12
| | | | Also rename the test to reflect that the video track is Theora, not VP3.
* fate: cosmetics: Order some test entriesDiego Biurrun2013-01-12
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* Add version bump and APIchanges entry for Add AV_PIX_FMT_VDPAU.Diego Biurrun2013-01-12
| | | | Also fix a lavu version typo in APIchanges.
* pixfmt: add picture format for VDPAURémi Denis-Courmont2013-01-12
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* ratecontrol: K&R cosmetic formattingLuca Barbato2013-01-12
| | | | Signed-off-by: Diego Biurrun <diego@biurrun.de>
* rtpdec: Remove a useless todo commentMartin Storsjö2013-01-12
| | | | | | | | | The question can be answered: No, we do not know the initial sequence number from the SDP. In certain cases, it can be known from the RTP-Info response header in RTSP though. (In that case, we use it as timestamp origin, but not for rtp receiver statistics.) Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Remove an outdated commentMartin Storsjö2013-01-12
| | | | | | | | It is unclear what the bug exactly was and if it ever was fixed, and we don't even support decoding via faad any longer. The comment has been present since d0deedcb in 2006. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Remove references to weirdly named variables in other filesMartin Storsjö2013-01-12
| | | | | | | | One of them is renamed now, but mentioning it by name serves no purpose here. The other table mentioned ceased to exist under that name in 4934884a1 in 2006. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtp: Rename a static variable to normal naming conventionsMartin Storsjö2013-01-12
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtp: Cosmetic cleanupMartin Storsjö2013-01-12
| | | | | | | Remove leftover debug comments, fix brace placement and add whitespace, remove unnecessary and weirdly placed braces. Signed-off-by: Martin Storsjö <martin@martin.st>
* configure: enable pic for shared libs on AArch64Marcin Juszkiewicz2013-01-11
| | | | | Signed-off-by: Marcin Juszkiewicz <marcin.juszkiewicz@linaro.org> Signed-off-by: Martin Storsjö <martin@martin.st>
* zmbv: Reset the decoder on keyframe errorsLuca Barbato2013-01-11
| | | | Prevent the crash on fuzzed files as reported in bug 63.
* vc1dec: prevent a crash due missing pred_flag parameterVladimir Pantelic2013-01-11
| | | | | | | Handle pred_flag parameter not given to get_mvdata_interlaced() Signed-off-by: Vladimir Pantelic <vladoman@gmail.com> Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* matroska: Fix use after freeDale Curtis2013-01-11
| | | | | Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* vp3: Fix double free in vp3_decode_end()Ronald Bultje2013-01-10
| | | | | Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* fate: Split fate-siff test into demuxer and VB video decoder testDiego Biurrun2013-01-10
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* rtpdec_vp8: Don't trim too much data from broken framesMartin Storsjö2013-01-10
| | | | | | | | | | | | | Previously, for broken frames, we only returned the first partition of the frame (we would append all the received packets to the packet buffer, then set pkt->size to the size of the first partition, since the rest of the frame could have lost data inbetween) - now instead return the full buffered data we have, but don't append anything more to the buffer after the lost packet discontinuity. Decoding the truncated packet should hopefully get better quality than trimming out everything after the first partition. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec_vp8: Simplify code by using an existing helper functionMartin Storsjö2013-01-10
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec: Add a terminating null byte at the end of the SDES/CNAMEMartin Storsjö2013-01-10
| | | | | | | | | | | | | | | | This is required by RFC 3550 (section 6.5): The list of items in each chunk MUST be terminated by one or more null octets, the first of which is interpreted as an item type of zero to denote the end of the list. This was implicitly added as padding before, unless the host name length matched up so no padding was added. This makes wireshark parse the packets properly if other RTCP items are appended to the same packet. Signed-off-by: Martin Storsjö <martin@martin.st>
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