ff_write_chained essentially duplicated the functionality of
av_packet_rescale_ts. This has been changed.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
There is no reason for these functions to modify the given packets at
all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
PCR generation was based on counting packets for both CBR and VBR streams.
Couting packets might have worked for CBR streams (when muxrate was specified)
but it only took into account the packets of a service (or the packets of the
PCR stream lately), so even that was problematic for multi program streams.
The new code works on actual timestamps for both CBR and VBR streams. For VBR
streams the behaviour of the old code is simulated by selecting a PCR interval
which is the highest multiple of the frame duration but still less than 100 ms.
It should be trivial to add support for setting the PCR interval for VBR
streams as well in a later patch.
The accuracy of PCR packets for CBR streams was greatly improved by preemtively
sending them at PCR intervals even if sending the payload of another stream
is in progress.
This may fix these tickets:
- #5750
- #7524
Signed-off-by: Marton Balint <cus@passwd.hu>
The MPEG-TS muxer had a serious bug related to the use of multiple programs:
in that case, the PCR pid selection was incomplete for all services except one.
This patch solves this problem and selects a stream to become PCR for each
service, preferably the video stream.
This patch also moves pcr calculation attributes to MpegTSWriteStream from
MpegTSService. PCR is a per-stream and not per-service thing, so it was
misleading to refer to it as something that is per-service.
Also remove *service from MpegTSWriteStream because a stream can belong to
multiple services so it was misleading to select one for each stream.
You can check the result with this example command:
./ffmpeg -loglevel verbose -y -f lavfi -i \
"testsrc=s=64x64:d=10,split=2[out0][tmp1];[tmp1]vflip[out1];sine=d=10,asetnsamples=1152[out2]" \
-flags +bitexact -fflags +bitexact -sws_flags +accurate_rnd+bitexact \
-codec:v libx264 -codec:a mp2 -pix_fmt yuv420p \
-map '0✌️0' \
-map '0✌️1' \
-map '0🅰️0' \
-program st=0:st=2 -program st=1:st=2 -program st=2 -program st=0 -f mpegts out.ts
You should now see this:
[mpegts @ 0x37505c0] service 1 using PCR in pid=256
[mpegts @ 0x37505c0] service 2 using PCR in pid=257
[mpegts @ 0x37505c0] service 3 using PCR in pid=258
[mpegts @ 0x37505c0] service 4 using PCR in pid=256
Fixes ticket #8039.
v2: a video is stream is preferred if there are no programs, just like before
the patch.
Signed-off-by: Marton Balint <cus@passwd.hu>
Up until now, when performing a SEEK_END seek, the subfile protocol
ignored the desired position (relative to EOF) and used the current
absolute offset in the input file instead.
And when performing a SEEK_CUR seek, the current position has been
ignored.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
=== PROBLEM ===
I was trying to record h264 + aac streams from an RTSP server to mp4 file. using this command line:
ffmpeg -v verbose -y -i "rtsp://<ip>/my_resources" -codec copy -bsf:a aac_adtstoasc test.mp4
FFmpeg then fail to record audio and output this logs:
[rtsp @ 0xcda1f0] The profile-level-id field size is invalid (40)
[rtsp @ 0xcda1f0] Error parsing AU headers
...
[rtsp @ 0xcda1f0] Could not find codec parameters for stream 1 (Audio: aac, 48000 Hz, 1 channels): unspecified sample format
In SDP provided by my RTSP server I had this fmtp line:
a=fmtp:98 streamType=5; profile-level-id=40; mode=AAC-hbr; config=1188; sizeLength=13; indexLength=3; indexDeltaLength=3;
In FFmpeg code, I found a check introduced by commit 24130234cd. It disallows values greater than 32 for fmtp line parameters.
RFC-4566 (SDP: Session Description Protocol) do not give any limit of size on interger parameters given in an fmtp line.
However, In RFC-6416 (RTP Payload Format for MPEG-4 Audio/Visual Streams) give examples of "profile-level-id" values for AAC, up to 55.
=== FIX ===
As each parameter may have its own min and max values
I propose to introduce a range for each parameter.
For this patch I used RFC-3640 and ISO/IEC 14496-1 as reference for validity ranges.
This patch fix my problem and I now can record my RTSP AAC stream to mp4.
It has passed the full fate tests suite sucessfully.
Signed-off-by: Olivier Maignial <olivier.maignial@smile.fr>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add http_seekable option for HTTP partial requests, when The
EXT-X-BYTERANGE tag indicates that a Media Segment is a sub-range
of the resource identified by its URI, we can use HTTP partial
requests to get the Media Segment.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
The previous calculation code did not account for the fact that the
copy_offset for the start of the frame array is at index 0, yet the
scan line number from the rfc4175 RTP header starts at 1.
This caused 2 issues to appear:
- The first scan line was being copied into the array where the second
scan line should be. This caused the resulting video to have a green
line at the top of it.
- Since the packet containing the last scan line would fail the
calculation, the packet with the RTP marker would not be processed
which caused a log message saying "Missed previous RTP marker" to be
outputted for each frame.
Signed-off-by: Jacob Siddall <kobe@live.com.au>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -1241665686 + -1340629419 cannot be represented in type 'int'
Fixes: 15922/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5692826442006528
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: out of array access
Fixes: 15365/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5716153105645568
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
From https://aomediacodec.github.io/av1-isobmff/#codecsparam, the parameters
sample entry 4CC, profile, level, tier, and bitDepth are all mandatory fields.
All the other fields are optional, mutually inclusive (all or none).
Fixes ticket #8049
Signed-off-by: James Almer <jamrial@gmail.com>
codecpar->extradata is not going to change between packets. New extradata
is instead propagated using packet side data.
Use ff_alloc_extradata() as well.
Signed-off-by: James Almer <jamrial@gmail.com>
avio_read can successfully return even when less than the requested
amount of input was read. wavdec's bext parsing mistakenly assumed a
successful avio_read always read the full amount that was requested.
The result could be dictionary tags populated with partially
uninitialized values.
This change also fixes a broken assertion in wav_parse_bext_string that
was off-by-one, though no known current usage of that method hits that
broken case.
Chromium bug: 987270
Signed-off-by: Matt Wolenetz <wolenetz@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Replace the same code logic with ensure_playlist(), it's will
help reusable blocks of code.
Reviewed-by: Jun Zhao <barryjzhao@tencent.com>
Signed-off-by: vacingfang <vacingfang@tencent.com>
av_packet_unref have reseted the AVPacket, so don't need to call
reset_packet after that.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
A value of zero for sample_count in trun box is not
prohibited by 14496-12 section 8.8.8. 4a9d32baca
disallowed this which led the demuxer to error out
when reading the header of valid files.
Fixes: signed integer overflow: 9151595917793558550 + 297519050751678697 cannot be represented in type 'long'
Fixes: 15496/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5722866475073536
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: Timeout
Fixes: 15567/clusterfuzz-testcase-minimized-ffmpeg_DEMUXER_fuzzer-5758451487080448
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
tomajsjiang
Andreas Rheinhardt
Marton Balint
Nicolas George
Michael Niedermayer
Carl Eugen Hoyos
Olivier Maignial
Jun Zhao
Jacob Siddall
James Almer
Steven Liu
Paul B Mahol
Matt Wolenetz
vacingfang
Baptiste Coudurier
Gyan Doshi
Leo Zhang