Seeing the offset of video and audio streams to each other is often
a useful metric in diagnosing and understanding issues with playback
or transcoding.
This commit adds those offsets to the stream info print.
Signed-off-by: softworkz <softworkz@hotmail.com>
This add support for WebVTT subtitles in HLS streams.
Just like for separate audio streams, it supports all available
WebVTT streams in all renditions.
No new options are added, it just works and provides subtitles streams
like any other demuxer.
The code prevents downloading subtitle segments which are farther
in the future than the main segments, to avoid loading hundreds
of subtitle segments in advance.
Signed-off-by: softworkz <softworkz@hotmail.com>
see: Re: [FFmpeg-devel] [PATCH 2/2] avformat/id3v2: Check that decode_str() did advance
but in short, first our API since 16years says we dont have multiple values per key
(which it supports since 9 years only)
and it causes some problems for ffprobe apparently.
I do believe the original patch is the correct direction but it requires
more changes. So revert this until other things are in place and until
we have a consensus.
This reverts commit 80b77e8e8d.
fix ticket #11531
the rtsp aac did not marked keyframe which cannot easy copy to output.
because f265f9c9d0 commit change the AAC props to match xHE-AAC.
in some formats like MOV, need_parsing is set, so AAC can be still parsed be keyframe
but rtsp did not, so this patch add it
Signed-off-by: Jack Lau <jacklau1222@qq.com>
Reviewed-by: Zhao Zhili <quinkblack@foxmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The current behavior when using restart_with_keyframe is that it will
recover if it also encounters any audio packet, as they are flagged as a
keyframe.
The expectation is that packets are dropped until the next _video_
keyframe.
To fix that, check if exists a video stream, if it exists check the
packet stream codec type, only letting it recover when it is a video
one. If there is no video stream, resume to the original behavior, not
checking the codec type.
Fixes ticket: #11467
Signed-off-by: Arthur Grillo <arthurgrillo@riseup.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes https://trac.ffmpeg.org/ticket/6949
Ordinary text frames in ID3v2 are allowed to have multiple
(null-separated) values. This technically isn't allowed in TXXX,
but it's used in practice by Picard, and supporting it is harmless.
TIPL/IPL (Involved People List) and TMCL (Musician Credits List) work
similarly to TXXX, but alternate key-value-key-value.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixed warning about OBU count being wrong, which can only be
determined when the number of OBUs in the header is non-zero,
not the other way round.
Signed-off-by: Chris Hodges <chris.hodges@axis.com>
Also fixes a -Wdeclaration-after-statement warning.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Not all rtp formats require the video dimensions to be available
up front. H264 and HEVC will send them as stream parameters.
The flag is restrictive and prevents RTP repacketization
without parsing the codec information out of the stream.
This change checks to see if the codec parameters are available
on the rtp formats that need it.
Signed-off-by: Koushik Dutta <koushd@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
fix ticket: 10786
parse the SPS from extradata and
get profile_compatibility, tier, constraints which was been hard code before.
HEVC CODECS Attribute reference to: ISO/IEC14496-15
Signed-off-by: Jack Lau <jacklau1222@qq.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The webvtt extension is sometimes used in HLS playlists.
Signed-off-by: softworkz <softworkz@hotmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The ///< or /**< form of doxygen comments are only to be used
when the documentation follows the member and the comment
block starts on the same line as the member. This commit
fixes wrong uses of them; in particular, this fixes the comment
for mb_height in H.264 SPS's structure which was wrongly added
to mb_width.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The precision should be enough for primaries and luminance.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Also remove the condition of AMF_DATA_TYPE_BOOL when parse color
info. There is no AMF_DATA_TYPE_BOOL type in color info.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
1. Rename metaVideoColor to meta_color_info
2. Allocated FLVMetaVideoColor together with FLVContext
3. Improve the use of meta_color_info_flag. Do a sequence of strcmp
only if meta_color_info_flag is FLV_COLOR_INFO_FLAG_PARSING.
4. Check return value of amf_parse_object().
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
sizeof AVMasteringDisplayMetadata isn't part of ABI.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Reviewed-by: Steven Liu <lingjiujianke@gmail.com>
Reviewed-by: James Almer <jamrial@gmail.com>
When using a 64bit PutBitContext (i.e. on x64), put_bits_no_assert()
can naturally write up to 63 bits. So one can avoid treating the
cases <32bits, 32 bits and <63 bits differently.
As it turns out, no user actually wants to write 64 bit at once
(maybe except testprograms).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Otherwise compilers might emit code that presumes there to be
a GOT which can't be fixed by the linker.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
'avs3' is registered at mp4ra.org. The Avs3ConfigurationBox 'av3c'
inside 'avs3' hasn't been registered yet, but is specified by the
AVS3 spec.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Change the type of logctx from void* to AVFormatContext*. This is in
preparation for the next commit.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(setting to 100 as a reasonable compromise)
The change has caused regressions for many users and consumers.
Playlist reloads only happen when a playlist doesn't indicate that it
has ended (via #EXT-X-ENDLIST), which means that the addition of future
segments is still expected.
It is well possible that an HLS server is temporarily unable to serve
further segments but resumes after some time, either indicating a
discontinuity or even by fully catching up.
With a segment length of 3s, a max_reload value of 1000 corresponds to
a duration of 50 minutes which appears to be a reasonable default.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add RTP packetizer and depacketizer according to (most)
of the official AV1 RTP specification. This enables
streaming via RTSP between ffmpeg and ffmpeg and has
also been tested to work with AV1 RTSP streams via
GStreamer.
It also adds the required SDP attributes for AV1.
AV1 RTP encoding is marked as experimental due to
draft specification status, debug amount reduced
and other changes suggested by Tristan.
Added optional code for searching the sequence
header to determine the first packet for broken
AV1 encoders / parsers.
Stops depacketizing on corruption until next keyframe,
no longer prematurely issues packet on decoding if
temporal unit was not complete yet.
Change-Id: I90f5c5b9d577908a0d713606706b5654fde5f910
Signed-off-by: Chris Hodges <chrishod@axis.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This changes the default behaviour of ffmpeg where content-type headers
on an input gives an absolut probe score (of 75) to instead give a bonus
score (of 30). This gives the probe a better chance to arrive at the
correct format by (hopefully) giving a large enough bonus to push edge
cases in the right direction (MPEG-PS vs MP3, I am looking at you) while
also not adversly punishing clearer cases (raw ADTS marked as
"audio/mpeg" for example).
This patch was regression tested against 20 million recent podcast
submissions (after content-type propagation was added to
original-storage), and 50k Juno vodcasts submissions (dito). No adverse
effects observed (but the bonus may still need tweaking if other edge
cases are detected in production).
When an Info-tag is present, marking initial and trailing samples as
padding, those samples should not be included in the calculation of track
duration.
This solves a surprising user experience where converting a WAV->MP3->WAV,
ffprobe will show the duration of the mp3 as slightly longer than both the
input and the output.
As a result, the estimated duration and imprecise seek-results of some
FATE-tests have been updated.
