This patch fixes Bug #8469
If x264 baseline profile is used with other profiles,
start_pts will be initialized to audio stream's first pts,
while the duration is calculated based on video stream's pts.
In this patch the start_pts is initialized with the correct stream's first pts.
Signed-off-by: Hongcheng Zhong <sj.hc_Zhong@sjtu.edu.cn>
Reviewed-by: Steven Liu <liuqi05@kuaishou.com>
Add support for WavPack DSD files to the existing WavPack decoder using
avcodec/dsd to perform the 8:1 decimation to 32-bit float samples. We must
serialize the dsd2pcm operation (cross-boundary filtering) but would like
to use frame-level multithreading for the CPU-intensive DSD decompression,
and this is accomplished with ff_thread_report/await_progress(). Because
the dsd2pcm operation is independent across channels we use slice-based
multithreading for that part.
Also a few things were removed from the existing WavPack decoder that
weren't being used (primarily the SavedContext stuff) and the WavPack
demuxer was enhanced to correctly determine the sampling rate of DSD
files (and of course to no longer reject them).
Signed-off-by: David Bryant <david@wavpack.com>
If 'write_colr' movflag is set, then movflag 'prefer_icc' can
be used to first look for an AV_PKT_DATA_ICC_PROFILE entry to
encode.
If ICC profile doesn't exist, default behaviour enabled by
'write_colr' occurs.
Signed-off-by: vectronic <hello.vectronic@gmail.com>
Supports connecting to a RabbitMQ broker via AMQP version 0-9-1.
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Dashdec can able to handle MPEG-2 TS streams by default as well,
used MP4Box to create the segmented MPEG-2 TS files for
verification.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
ff_alloc_extradata() already sets the size of the extradata so doing it
again is unnecessary.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Gyan Doshi <ffmpeg@gyani.pro>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The standard does not seem to require the counter to be zero based, but some
checker tools (MyriadBits MXFInspect, Interra Baton) have validations against 0
start...
Fixes ticket #6781.
Signed-off-by: Marton Balint <cus@passwd.hu>
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>
Prevents memleaks when the trailer is never written or when shifting the
data fails when writing the trailer.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Otherwise the user is not able to override the SRT API connect timeout above 5
sec without also setting the timeout option.
Signed-off-by: Marton Balint <cus@passwd.hu>
Return os error code if available, check for both SRT_EASYNCRCV and
SRT_EASYNCSND when transforming them to EAGAIN and only display error if it is
not EAGAIN.
Signed-off-by: Marton Balint <cus@passwd.hu>
After f8990c5f41 we properly set non-blocking
mode which makes the connect() call return always 0 even if no connection can
be established.
Fix this by always doing a poll after calling connect(). Also there was some
leftover copy paste code which checks for various errors which are simply not
possible with SRT.
Signed-off-by: Marton Balint <cus@passwd.hu>
rw_timeout is the generic URLcontext option, not the protocol specific timeout
option, also ?rw_timeout never worked because ?timeout was parsed instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
This ensures it's written at the beginning of a segment in non streaming mode
when segment duration differs from fragment duration.
Signed-off-by: James Almer <jamrial@gmail.com>
Such errors may make sense for specific formats, but general parsing
logic shouldn't be treating these as errors regardless of the error
recognition mode.
Fixes loading of the following wave when using -err_detect explode:
https://cs.chromium.org/chromium/src/third_party/blink/web_tests/external/wpt/webaudio/resources/4ch-440.wav
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If an audio stream is present, the Sega FILM muxer checks for its
compability with the container during init, so that the very same check
needn't be repeated during writing the trailer.
Essentially the same is true for the presence of a video stream: It has
already been checked during init. Furthermore, after the check for the
presence of a video stream succeeded, a pointer is set to point to the
video stream. Yet said pointer (which was NULL before) will be
derefenced anyway regardless of the result of the check. Coverity thus
complained about this in CID 1434155 and removing this pointless check
will also fix this issue.
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>
Commit 36e156bef0 ("avformat/spdifenc: fix handling of large TrueHD
frame") added an obviously incorrect bitshift that caused incorrect
samples-per-frame calculation for TrueHD streams over 48kHz.
Fix that.
The TrueHD IEC 61937 encapsulation code uses a very naive method of
always inserting 24 TrueHD frames evenly in a MAT frame. This does not
work for larger frames as they may exceed the size of 1/24th of a MAT
frame.
To fix that, use the input_timing field in the TrueHD frame to determine
the proper position of the TrueHD frame in the MAT frame. That field is
basically a dts field, telling the time to feed this frame to the
decoder in sample count units.
This can cause a TrueHD frame to be split between two MAT frames, so a
second concatenation hd_buf is added, alternating with the first buffer.
Large frames are preceded by smaller frames that have input_timing
values that cause the frames to be sent out faster than the nominal rate
(i.e. increasing decoder latency, long decoder buffer), allowing the
larger frames to then be sent out slower than the nominal rate as the
decoder has enough data buffered to keep it busy.
This is preparation for adding a second hd_buf in a followup commit.
Also, slightly improve the comments for hd_buf_x members to clarify
which ones are actually used and kept up-to-date depending on which
codec is being muxed.
wtvfile_read_packet did not abide by the requirements of an
AVIOContext's read_packet-function: If it did not read anything,
it returned zero, which currently leads to a warning in read_packet_wrapper
in aviobuf.c. Said warning will be an av_assert2 as soon as
FF_API_OLD_AVIO_EOF_0 is zero (probably the next major version bump).
So instead forward the error code from the underlying protocol.
This error/assert is triggered in the wtv-demux FATE test.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Paul B Mahol
Steven Liu
Hongcheng Zhong
David Bryant
vectronic
Andriy Gelman
Zane van Iperen
phunkyfish
Jun Zhao
Andreas Rheinhardt
Marton Balint
Michael Niedermayer
James Almer
Dale Curtis
Anssi Hannula