Commit Graph

36289 Commits

Author SHA1 Message Date
Anton Khirnov
49dc82eef7 v4l2: do not assert on a value received from outside of Libav 2013-01-13 16:31:17 +01:00
Anton Khirnov
838b849e70 v4l2: set the average framerate instead of codec timebase.
Codec timebase is supposed to be set by decoders only.
2013-01-13 16:31:09 +01:00
Anton Khirnov
246da0b135 v4l2: avoid pointless indirection.
v4l2_read_header() does no cleanup, so it can return directly, without
any need for goto.
2013-01-13 16:30:38 +01:00
Rémi Denis-Courmont
7752532789 vdpau: Add VC-1 decoding via hwaccel infrastructure
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-13 14:54:18 +01:00
Rémi Denis-Courmont
51b56a0716 vdpau: Add H.264 decoding via hwaccel infrastructure
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-13 14:54:08 +01:00
Rémi Denis-Courmont
200e8ac920 vdpau: Add MPEG-4 decoding via hwaccel infrastructure
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-13 14:54:02 +01:00
Rémi Denis-Courmont
aaf9d0e8f3 vdpau: Add MPEG-1/2 decoding via hwaccel infrastructure
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-13 14:53:47 +01:00
Rémi Denis-Courmont
44e065d56c vdpau: Add context and common helpers for hwaccel support
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-13 14:41:27 +01:00
Martin Storsjö
22c436c85e rtpdec: Send a valid "delay since SR" value in the RTCP RR packets
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>
2013-01-12 19:55:49 +02:00
Martin Storsjö
e568db4025 rtpdec: Calculate and report packet reception jitter
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>
2013-01-12 19:53:53 +02:00
Martin Storsjö
abae27ed3a rtpdec: Fix the calculation of expected number of packets
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>
2013-01-12 19:48:41 +02:00
Diego Biurrun
0f94c0b1b6 fate: vp3: Fix fate-vp3-coeff-level64 test dependencies
Also rename the test to reflect that the video track is Theora, not VP3.
2013-01-12 17:19:27 +01:00
Diego Biurrun
4d3b144c5e fate: cosmetics: Order some test entries 2013-01-12 17:19:27 +01:00
Diego Biurrun
f89466ad6f Add version bump and APIchanges entry for Add AV_PIX_FMT_VDPAU.
Also fix a lavu version typo in APIchanges.
2013-01-12 12:28:12 +01:00
Rémi Denis-Courmont
169fb94f0f pixfmt: add picture format for VDPAU
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-12 12:00:49 +01:00
Luca Barbato
f61272f0ef ratecontrol: K&R cosmetic formatting
Signed-off-by: Diego Biurrun <diego@biurrun.de>
2013-01-12 12:00:49 +01:00
Martin Storsjö
f6804c3e1b rtpdec: Remove a useless todo comment
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>
2013-01-12 00:02:17 +02:00
Martin Storsjö
54cb096ee4 rtsp: Remove an outdated comment
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>
2013-01-12 00:02:11 +02:00
Martin Storsjö
3900d53fb1 rtsp: Remove references to weirdly named variables in other files
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>
2013-01-12 00:02:04 +02:00
Martin Storsjö
c44784c9bb rtp: Rename a static variable to normal naming conventions
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-12 00:01:51 +02:00
Martin Storsjö
58b5971881 rtp: Cosmetic cleanup
Remove leftover debug comments, fix brace placement and
add whitespace, remove unnecessary and weirdly placed braces.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-12 00:01:28 +02:00
Marcin Juszkiewicz
d11cb13b0e configure: enable pic for shared libs on AArch64
Signed-off-by: Marcin Juszkiewicz <marcin.juszkiewicz@linaro.org>
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-11 11:01:05 +02:00
Luca Barbato
c1d1ef4ecd zmbv: Reset the decoder on keyframe errors
Prevent the crash on fuzzed files as reported in bug 63.
2013-01-11 05:07:42 +01:00
Vladimir Pantelic
7b8c5b263b vc1dec: prevent a crash due missing pred_flag parameter
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>
2013-01-11 04:57:41 +01:00
Dale Curtis
ae3d416369 matroska: Fix use after free
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2013-01-11 00:12:08 +01:00
Ronald Bultje
ec86ba5731 vp3: Fix double free in vp3_decode_end()
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2013-01-10 20:05:42 +01:00
Diego Biurrun
f8936c6f90 fate: Split fate-siff test into demuxer and VB video decoder test 2013-01-10 17:54:47 +01:00
Martin Storsjö
76c40fbef0 rtpdec_vp8: Don't trim too much data from broken frames
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>
2013-01-10 09:43:01 +02:00
Martin Storsjö
3b366c3aa0 rtpdec_vp8: Simplify code by using an existing helper function
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-01-10 09:41:44 +02:00
Martin Storsjö
ed79093222 rtpdec: Add a terminating null byte at the end of the SDES/CNAME
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>
2013-01-10 09:40:49 +02:00
Luca Barbato
a800fd5fc7 yuv4mpeg: do not use deprecated functions
Use the libavutil replacement.
2013-01-09 21:07:49 +01:00
Luca Barbato
fba8e5b608 oggdec: fix faulty cleanup prototype 2013-01-09 21:07:48 +01:00
Justin Ruggles
06deaf8ad3 idcin: return 0 from idcin_read_packet() on success.
This matches the AVInputFormat.read_packet() API.
2013-01-09 14:49:07 -05:00
Justin Ruggles
5d0450461f idcin: better error handling
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.
2013-01-09 14:49:07 -05:00
Justin Ruggles
33f58c3616 idcin: check for integer overflow when calling av_get_packet()
chunk_size is unsigned 32-bit, but av_get_packet() takes a signed int as the
packet size.
2013-01-09 14:49:06 -05:00
Justin Ruggles
7040e479a1 idcin: allow seeking back to the first packet
Also, do not allow seek-by-byte, as there is no way to find the next packet
boundary.
2013-01-09 14:49:06 -05:00
Justin Ruggles
49543373f3 idcin: set AV_PKT_FLAG_KEY for video packets with a palette 2013-01-09 14:49:06 -05:00
Justin Ruggles
ccc0ffb1ba idcin: set start_time and packet duration instead of manually tracking pts.
Also, use 1 / sample_rate for audio stream time_base.
2013-01-09 14:49:06 -05:00
Justin Ruggles
4b840930da idcin: set channel_layout 2013-01-09 14:49:06 -05:00
Justin Ruggles
12c2530b1d idcin: fix check for presence of an audio stream 2013-01-09 14:49:06 -05:00
Justin Ruggles
b0c96e0613 idcin: validate header parameters
Avoids using unsupported parameters and signed integer overflows.
2013-01-09 14:49:06 -05:00
Justin Ruggles
f7bf72a4a1 idcinvideo: correctly set AVFrame defaults 2013-01-09 14:49:06 -05:00
Daniel Kang
899157b308 yadif: Port inline assembly to yasm
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
2013-01-09 18:41:02 +01:00
Justin Ruggles
f7a3c540c5 au: remove unnecessary casts 2013-01-09 11:52:57 -05:00
Justin Ruggles
2f8207b1c6 au: return AVERROR codes instead of -1 2013-01-09 11:52:57 -05:00
Justin Ruggles
fd9147f114 au: cosmetics: pretty-print and remove pointless comments 2013-01-09 11:52:57 -05:00
Justin Ruggles
c88d245c98 au: use ff_raw_write_packet() 2013-01-09 11:52:57 -05:00
Justin Ruggles
bdd00e2d1b au: set stream start time and packet durations 2013-01-09 11:52:57 -05:00
Justin Ruggles
af68a2baae au: use %u when printing id and channels since they are unsigned 2013-01-09 11:52:57 -05:00
Justin Ruggles
47d029a4c1 au: validate sample rate 2013-01-09 11:52:57 -05:00