- 13 Jan, 2013 9 commits
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Anton Khirnov authored
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.
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Anton Khirnov authored
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Anton Khirnov authored
Codec timebase is supposed to be set by decoders only.
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Anton Khirnov authored
v4l2_read_header() does no cleanup, so it can return directly, without any need for goto.
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Rémi Denis-Courmont authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Rémi Denis-Courmont authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Rémi Denis-Courmont authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Rémi Denis-Courmont authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Rémi Denis-Courmont authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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- 12 Jan, 2013 8 commits
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Martin Storsjö authored
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>
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Martin Storsjö authored
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>
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Martin Storsjö authored
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>
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Diego Biurrun authored
Also rename the test to reflect that the video track is Theora, not VP3.
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Diego Biurrun authored
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Diego Biurrun authored
Also fix a lavu version typo in APIchanges.
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Rémi Denis-Courmont authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Luca Barbato authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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- 11 Jan, 2013 8 commits
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Martin Storsjö authored
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>
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Martin Storsjö authored
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>
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Martin Storsjö authored
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 4934884a in 2006. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Remove leftover debug comments, fix brace placement and add whitespace, remove unnecessary and weirdly placed braces. Signed-off-by: Martin Storsjö <martin@martin.st>
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Marcin Juszkiewicz authored
Signed-off-by: Marcin Juszkiewicz <marcin.juszkiewicz@linaro.org> Signed-off-by: Martin Storsjö <martin@martin.st>
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Luca Barbato authored
Prevent the crash on fuzzed files as reported in bug 63.
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Vladimir Pantelic authored
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>
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- 10 Jan, 2013 6 commits
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Dale Curtis authored
Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Ronald Bultje authored
Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Diego Biurrun authored
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Martin Storsjö authored
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>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
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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- 09 Jan, 2013 9 commits
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Luca Barbato authored
Use the libavutil replacement.
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Luca Barbato authored
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Justin Ruggles authored
This matches the AVInputFormat.read_packet() API.
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Justin Ruggles authored
Add some additional checks for EOF and print error messages on an incomplete header or packet. FATE reference updated for id-cin-video due to the demuxer no longer returning a partial video packet at EOF.
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Justin Ruggles authored
chunk_size is unsigned 32-bit, but av_get_packet() takes a signed int as the packet size.
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Justin Ruggles authored
Also, do not allow seek-by-byte, as there is no way to find the next packet boundary.
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Justin Ruggles authored
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Justin Ruggles authored
Also, use 1 / sample_rate for audio stream time_base.
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Justin Ruggles authored
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