This makes more sense than mapping to AV_CODEC_ID_SUBRIP. Nothing
indicates that a T.140 track contains subrip sub-titles.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '353b492d0f2a21ae8eb829db1ac01b54b2a4d202':
rtpdec: Change enc_name to a pointer instead of a fixed-size buffer
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When receiving an RTCP packet, the difference between the last RTCP
timestamp and the base timestamp may be negative. As these timestamps
are of the uint32_t type, the result becomes a large integer. Cast
the difference to int32_t to avoid this issue.
The result of this issue is very large start times for RTSP
streams, and difficulty to restart correctly after a pause.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Martin Storsjö <martin@martin.st>
When receiving an RTCP packet, the difference between the last RTCP
timestamp and the base timestamp may be negative. As these timestamps
are of the uint32_t type, the result becomes a large integer. Cast
the difference to int32_t to avoid this issue.
The result of this issue is very large start times for RTSP
streams, and difficulty to restart correctly after a pause.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The packetizer only supports splitting at GOB headers - if
such aren't available frequently enough, it splits at any
random byte offset (not at a macroblock boundary either, which
would be allowed by the spec) and sends a payload header pretend
that it starts with a GOB header.
As long as a receiver doesn't try to handle such cases cleverly
but just drops broken frames, this shouldn't matter too much
in practice.
Signed-off-by: Martin Storsjö <martin@martin.st>
The ones left using av_gettime are NTP timestamps (for RTCP,
which is specified to send the actual current realtime clock
in RTCP SR packets), and the NUT muxer timestamper, which is
documented as using wallclock time.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'c463dfc7e49929a9891884312b23b27d14729c51':
rtpdec_hevc: Drop a duplicated, nonstandard entry
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The RFC spec draft only specifies the "H265" name - there is no
specification saying how to interpret "HEVC" (if such a packet
format is specified it could be an entirely different format).
Since this is a very new standard (still a draft), there is little
need for compatibility with existing, broken implementations. Therefore
remove the extra alias, to avoid the risk of encouraging incorrect
usage.
Intentionally keeping the ff_hevc_dynamic_handler name for the
handler, to use "hevc" consistently as name for the codec instead
of "h265" within the library internals as long as there only is one
single variant in actual use.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '0307cc2253e76772b1c645ac6117d08da87a147c':
rtpdec: pass an AVFormatContext to ff_parse_fmtp()
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'feeafb4adabd5c17de1738ed9962e40892b20edb':
lavf: do not export av_register_{rtp,rdt}_dynamic_payload_handlers from shared objects
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The warnings are false positives, older gcc versions (such as 4.5)
think the variables can be used uninitialized while they in
practice can't, while newer (4.6) gets it right.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'f53490cc0c809975f8238d5a9edbd26f83bd2f84':
rtpdec/srtp: Handle CSRC fields being present
rtpdec: Check the return value from av_new_packet
ac3dec: fix non-optimal dithering of zero bit mantissas
Conflicts:
libavcodec/ac3dec.c
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c6f1dc8e4cd967ae056698eafb891a08003c211c':
rtpdec: Move setting the parsing flags to the actual depacketizers
rtpdec: Split handling of mpeg12 audio/video to a separate depacketizer
Conflicts:
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Untested, due to lack of rtp stream with CSRCs, but the code as
is does not work with multiple CSRCs
Reviewed-by: Luca Abeni <lucabe72@email.it>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
