The bits_per_raw_sample represents the number of bits of precision per sample.
The field is added at the logical place, not at the end as the code was just
recently added
This fixes the regression about losing the audio sample precision information
The change in the fate test checksum un-does the change from the merge
Previous version reviewed by: wm4 <nfxjfg@googlemail.com>
Previous version reviewed by: Dominik 'Rathann' Mierzejewski <dominik@greysector.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
This reverts commit 8ed82d8174.
SMPTE S377-1-2009c defines in F.4.1 that the Video Line Map should
always be an array with two 32 bit integers as elements. This is
repeated in G.2.12 with actual examples for progressive content,
where the second value would always be 0.
Additionally, the IRT MXF analyser also lists this as the only
error in the MXF output from ffmpeg: https://mxf-analyser-cloud.irt.de
Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Since timecode_frame)start is a private option now, it stays at the default,
and is no longer written to the file.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This is never mentioned in the specifications, and decoders work
just as fine without it. Update the fate references since the compressed
file is smaller.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Also support disabling them as they seem to cause problems to some
Users. They are also not allowed in IRT D-10 thus the default for
mxf_d10 is not to write them
This also decreases the filesize when no user comment are stored
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The CMP variable seems to have been inherited from fate-api-seek which set it to null
the mxf reference needed a change due to c7e14a279f
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Compute individual stream durations in matroska muxer.
Write them as string tags in the same format as mkvmerge tool does.
Signed-off-by: Sasi Inguva <isasi@google.com>
Previous version Reviewed-by: tim nicholson <nichot20@yahoo.com>
Previous version Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously unset, and some software mishandles files if it is absent
Signed-off-by: Tim Nicholson <tim.nicholson@bbc.co.uk>
Reviewed-by: tomas.hardin@codemill.se
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
According to the DPX file format description found at
http://www.fileformat.info/format/dpx/egff.htm the ImageElement part of
the GenericImageHeader also contains an an offset to the real image data
beside the same member that can be found in the GenericFileHeader.
Libav keeps this member empty (=0) while some applications expects it to
be filled properly. FATE test updated accordingly.
Bug-Id: 742
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This is a generic solution that will not reqiore modifications when new options are added.
This also fixes problem with current implementation when qmin or qmax=-1.
Only 8 bits was sent and read back as 255.
Fixes#1275Fixes#1461
Signed-off-by: Lukasz Marek <lukasz.m.luki2@gmail.com>
It is derived from the actual equations of the specs. In
particular, it is closer to the inverse of what the encoder uses.
fate tests accordingly updated.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes assertion failure
Fixes Ticket3822
as a side-effect this makes some mkv files a few bytes smaller
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.
Partially undoes commit 2c4e08d89327595f7f4be57dda4b3775e1198d5e:
riff: always generate a proper WAVEFORMATEX structure in
ff_put_wav_header
A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the
use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs.
This flag is used in the Matroska muxer (the cause of the original
change) and in the ASF muxer, because the specifications for
these formats indicate explicitly that WAVEFORMATEX should be used.
Muxers for other formats will return to the original behavior of writing
PCMWAVEFORMAT when writing a header for raw PCM.
In particular, this causes raw PCM in WAV to generate the canonical
44-byte header expected by some tools.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The rational for this is another issue that plex has exposed. When it is
conducting a transcode of video to HLS for streaming, my father noticed
artifacts when played on his GoogleTV (NSZ-GT1). He sent me a test file
and I reproduced it on my device of the same model. It is important to
note that the artifacts were not present when streaming to VLC or QuickTime
Player. I copied the command-line that plex used, and conducted all of the
following tests using FFmpeg git.
Transcode to HLS: artifacts on playback
Transcode to TS: playback is fine
Cat HLS segments into a single TS: playback is fine
Segment single TS file to segments: artifacts on playback
Segment single TS file to segments using Apple's HLS segmenter: playback is
fine
At this point I carefully examined the differences between Apple's HLS
segmenter output and FFmpeg's. Among the considerable differences, I
noticed that the video PES packets always had a 0 length. So I continued:
Transcode to HLS using FFmpeg with 0 length PES packets: playback is fine.
Segment single TS to segments with 0 length PES packets: playback is fine.
All failures mentioned are only on the GTV since it is the only player on
which I could reproduce artifacts. I only tested the GTV, VLC, and
QuickTime Player though, so my test case is limited. I do not know if
other players exhibit this issue.
Since it was useful last time, I have uploaded the test file as
hls_pes_packet_length.m4v along with its associated txt file which contains
the transcode command-line that was used.
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Michael Niedermayer
Luca Barbato
Anton Khirnov
Marton Balint
Sebastian Dröge
Derek Buitenhuis
Vittorio Giovara
Sasi Inguva
Juanjo
Michael Niedermayer
Mark Reid
Tim Nicholson
Robert Krüger
Zoltan Kovacs
Lukasz Marek
Christophe Gisquet
Peter B
Josh Allmann
Marc-Antoine Arnaud
Andrew Kelley
Carl Eugen Hoyos
Daniel Verkamp
Graham Booker