This commit adds support for storing DFPWM audio in a WAV container.
It uses the WAVEFORMATEXTENSIBLE structure, following these conventions:
https://gist.github.com/MCJack123/90c24b64c8e626c7f130b57e9800962c
The implementation is very simple: it just adds the GUID to the list of
WAV GUIDs, and modifies the WAV muxer to always use WAVEFORMATEXTENSIBLE
format with that GUID.
This creates a standard container format for DFPWM besides raw data.
It will allow users to transfer DFPWM audio in a standard container
format, with the sample rate and channel count contained in the file
as opposed to being an external parameter as in the raw format.
This format is already supported in my AUKit library, which is the CC
analog to libav (albeit much smaller). Support in other applications is TBD.
Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
Prefer to error than to create a broken file. Closes ticket #5829.
Effectively disables remuxing adpcm_swf from flv -> wav.
Signed-off-by: Zane van Iperen <zane@zanevaniperen.com>
In 1ec2b3de5a, the extradata size was affected when the raster was
signaled as flipped due to user-set option rather than via extradata.
This resulted in a wrong header size being written. Fixed.
Some legacy applications such as AVI2MVE expect raw RGB bitmaps
to be stored bottom-up, whereas our RIFF BITMAPINFOHEADER assumes
they are always stored top-down and thus write a negative value
for height. This can prevent reading of these files.
Option flipped_raw_rgb added to AVI and Matroska muxers
which will write positive value for height when enabled.
Note that the user has to flip the bitmaps beforehand using other
means such as the vflip filter.
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 unbreaks muxing-encoding
Example:
ffmpeg -i matrixbench_mpeg2.mpg new.avi
-rw-r----- 1 michael michael 226035354 Jan 1 16:27 new.avi
-rw-r----- 1 michael michael 10016802 Jan 1 16:28 ref.avi
Also av_get_audio_frame_duration() itself uses frame_size
This reverts commit 29e6606e9b, reversing
changes made to 53448461a7.
It will not be set unless the muxing codec context is also the encoding
context, which is discouraged. When the frame size is not known from
av_get_audio_frame_duration(), the fallback should still be good enough.
Allow writing an empty channel mask into the wave format header. Useful
if the input file contains an unknown channel layout.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It will not be set unless the codec context is used as the encoding
context, which is discouraged. For MP2, av_get_audio_frame_duration()
will already set the frame size properly. For MP3, set the frame size
explicitly.
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).
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>
Vittorio Giovara
Jack Bruienne
Andreas Rheinhardt
Zane van Iperen
Gyan Doshi
Aleksandr Slobodeniuk
Carl Eugen Hoyos
Florian Diemer
Mats Peterson
Anton Khirnov
Michael Niedermayer
Tobias Rapp
Juanjo
Michael Niedermayer
Benoit Fouet
nu774
Daniel Verkamp
Clément Bœsch
James Almer
Peter Ross
Diego Biurrun