"analyzeduration" is not used to detect the input duration, but to
specify the max probe data duration. Fix option description and related
doc entry accordingly.
* commit 'abae27ed3acd0a7c54f11760c5be2d2653c4edf8':
rtpdec: Fix the calculation of expected number of packets
fate: vp3: Fix fate-vp3-coeff-level64 test dependencies
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
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>
* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1':
rtsp: Remove an outdated comment
rtsp: Remove references to weirdly named variables in other files
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c44784c9bb9d0ddf5d39d0dfa640816a57b8f457':
rtp: Rename a static variable to normal naming conventions
rtp: Cosmetic cleanup
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
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>
Fixes division by zero
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7b8c5b263bc680eff5710bee5994de39d47fc15e':
vc1dec: prevent a crash due missing pred_flag parameter
matroska: Fix use after free
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec_vp8: Don't trim too much data from broken frames
rtpdec_vp8: Simplify code by using an existing helper function
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ed79093222ceb42f0c3a39095a69af0b32be5450':
rtpdec: Add a terminating null byte at the end of the SDES/CNAME
yuv4mpeg: do not use deprecated functions
oggdec: fix faulty cleanup prototype
idcin: return 0 from idcin_read_packet() on success.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7040e479a1530b2eda4b89a182d5eb50a77bd907':
idcin: allow seeking back to the first packet
idcin: set AV_PKT_FLAG_KEY for video packets with a palette
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ccc0ffb1ba3fc1adb05a9f56dfc26131e61db3fb':
idcin: set start_time and packet duration instead of manually tracking pts.
idcin: set channel_layout
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '12c2530b1d87fa94f81ea97df575b77c825e6f4f':
idcin: fix check for presence of an audio stream
idcin: validate header parameters
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c88d245c9866e48cb8a238b7564964c1fcf3315f':
au: use ff_raw_write_packet()
au: set stream start time and packet durations
Conflicts:
libavformat/au.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'af68a2baae6761044cbed95575e8bcfebf55c6f1':
au: use %u when printing id and channels since they are unsigned
au: validate sample rate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c837b38dd33a11c3810e988a60193a858eb4f58c':
au: move skipping of unused data to before parameter validation
au: do not arbitrarily limit channel count
Conflicts:
libavformat/au.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9a7b56883d1333cdfcdf0fa7584a333841b86114':
au: set bit rate
au: validate bits-per-sample separately from codec tag
rtpdec_vp8: Mark broken packets with AV_PKT_FLAG_CORRUPT
Conflicts:
libavformat/au.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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>
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>
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.
