VP6 alpha in EA format is a second VP6 encoded video stream where only the Y
component is used and is interpreted as the alpha channel of the first VP6
stream. The alpha VP6 stream is muxed separately from the main VP6 stream, has
its own stream headers and packet headers. In theory the two streams might not
even have the same resolution (although most likely that is not something that
is seen or supported in the wild), but the format is capable of doing it.
Merged VP6 alpha (also known as the VP6A codec) means that a packet of the
video stream contains the corresponding packet of both VP6 substreams like
this:
{OffsetOfAlpha, DataPacket, AlphaDataPacket}
So data and alpha data of a frame is merged to a single packet, this is how VP6
video with alpha is muxed in FLV and SWF.
The first approach is more like how the demuxer sees data in the EA format,
unfortunately it is different to what the FLV or SWF format expects, so -
having no better place for it in the framework - I decided to do an optional
format conversion in the EA demuxer.
Signed-off-by: Marton Balint <cus@passwd.hu>
Add PCR at keyframe can be undesirable when -pcr_period is
specified. Add an flag to disable this behavior.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Users can't make anything with its content.
Making it opaque might allow us to avoid one level of indirection.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
According to the HEIF specification (ISO/IEC 23008-12) Section
7.5.3.1, tracks with handler_type 'auxv' must contain a 'auxi' box
in its SampleEntry to notify the nature of the auxiliary track to the
decoder.
The content is the same as the 'auxC' box. So parameterize and re-use
the existing function.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: James Zern <jzern@google.com>
Opus can be decoded to multiple samplerates (namely 48kHz, 24KHz,
16Khz, 12 KHz and 8Khz); libopus as well as our encoder wrapper
support these sample rates. The OpusHead contains a field for
this original samplerate. Yet the pre-skip (and the granule-position
in the Ogg-Opus mapping in general) are always in the 48KHz clock,
irrespective of the original sample rate.
Before commit c3c22bee63, our libopus
encoder was buggy: It did not account for the fact that the pre-skip
field is always according to a 48kHz clock and wrote a too small
value in case one uses the encoder with a sample rate other than 48kHz;
this discrepancy between CodecDelay and OpusHead led to Firefox
rejecting such streams.
In order to account for that, said commit made the muxer always use
48kHz instead of the actual sample rate to convert the initial_padding
(in samples in the stream's sample rate) to ns. This meant that both
fields are now off by the same factor, so Firefox was happy.
Then commit f4bdeddc3c fixed the issue
in libopusenc; so the OpusHead is correct, but the CodecDelay is
still off*. This commit fixes this by effectively reverting
c3c22bee63.
*: Firefox seems to no longer abort when CodecDelay and OpusHead
are off.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This new function makes it possible to use avio_printf() functionality from
a function taking a variable list of arguments.
Signed-off-by: Marton Balint <cus@passwd.hu>
Add an AVIF muxer by re-using the existing the mov/mp4 muxer.
AVIF Specification: https://aomediacodec.github.io/av1-avif
Sample usage for still image:
ffmpeg -i image.png -c:v libaom-av1 -still-picture 1 image.avif
Sample usage for animated AVIF image:
ffmpeg -i video.mp4 animated.avif
We can re-use any of the AV1 encoding options that will make
sense for image encoding (like bitrate, tiles, encoding speed,
etc).
The files generated by this muxer has been verified to be valid
AVIF files by the following:
1) Displays on Chrome (both still and animated images).
2) Displays on Firefox (only still images, firefox does not support
animated AVIF yet).
3) Verified to be valid by Compliance Warden:
https://github.com/gpac/ComplianceWarden
Fixes the encoder/muxer part of Trac Ticket #7621
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
This patch adds support for:
- ffplay ipfs://<cid>
- ffplay ipns://<cid>
IPFS data can be played from so called "ipfs gateways".
A gateway is essentially a webserver that gives access to the
distributed IPFS network.
This protocol support (ipfs and ipns) therefore translates
ipfs:// and ipns:// to a http:// url. This resulting url is
then handled by the http protocol. It could also be https
depending on the gateway provided.
To use this protocol, a gateway must be provided.
If you do nothing it will try to find it in your
$HOME/.ipfs/gateway file. The ways to set it manually are:
1. Define a -gateway <url> to the gateway.
2. Define $IPFS_GATEWAY with the full http link to the gateway.
3. Define $IPFS_PATH and point it to the IPFS data path.
4. Have IPFS running in your local user folder (under $HOME/.ipfs).
Signed-off-by: Mark Gaiser <markg85@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also bump the minor versions of all libraries, to signify the
API change of splitting the version.h headers and adding the
new version_major.h header.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
This patch builds on my previous DFPWM codec patch, adding a raw
audio format to be able to read/write the raw files that are most commonly
used (as no other container format supports it yet).
The muxers are mostly copied from the PCM demuxer and the raw muxers, as
DFPWM is typically stored as raw data.
Please see the previous patch for more information on DFPWM.
Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
following 625ea2d, redirect caching is performed according to the http
response headers, there's no need to have it as an option -
always start from the original uri, and apply any redirects according
to the redirect_cache dictionary.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This is similar to the faststart option of the mov muxer, yet
in contrast to it it works together with reserve_index_space
(the equivalent to reserved_moov_size): If the reserved space
does not suffice, the data is shifted; if not, the Cues are
written at the front without shifting the data.
Several tests that cover (not only) this have been added.
Implements #7017.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is done a second time for 5.0 because master was
merged into 5.0 so that it contains the recent DOVI additions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
added "cache_redirect" option to http.
when enabled, requests issued after seek will use the latest redirected
url.
when disabled, each call to seek will revert to the original url that
was sent to http_open.
currently, the default remains 'enabled', until the next major
libavformat bump, where it will change to 'disabled'.
Because the hls_ts_options will be misunderstand by user,
and then user can use hls_segment_options instead of hls_ts_options.
Signed-off-by: Steven Liu <liuqi05@kuaishou.com>
Otherwise there is no way to detect an error returned by avio_close() because
ff_format_io_close cannot get the return value.
Checking the return value of the close function is important in order to check
if all data was successfully written and the underlying close() operation was
successful.
It can also be useful even for read mode because it can return any pending
AVIOContext error, so the user don't have to manually check AVIOContext->error.
In order to still support if the user overrides io_close, the generic code only
uses io_close2 if io_close is either NULL or the default io_close callback.
Signed-off-by: Marton Balint <cus@passwd.hu>
In 45bfe8b838, short_seek_threshold was removed
from the public AVIO struct. Although this option was private and not intended
to be used by public API users, it was nonetheless, because it provided functionality
that could otherwise not be gained via public API.
This was especially important for networked I/O like HTTP, where the internal
size for lavf could be way to small depending on the specifics of a user's
usecase, such as reading interlavd media files from cloud storage.
Add an AVOption to make this functionality accessible to the HTTP client.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
There is no reason to wrap them in #ifndef guards, they should only be
defined here and nowhere else. The define guards just add the
possibility to accidentally use the same FF_API name in different
libraries.
Such fields can be seen as generally useful in cases where the
API user is not implementing custom AVIO callbacks, but still would
like to know if data is being read or written out, such as in case
data is being read from input but no AVPacket has been received yet.
Originally added as a private entry in commit
3f75e5116b, but its grouping with
the comment noting its private state was missed during merging of
the field from Libav (most likely due to an already existing field
in between).
This information is coded in a standard MP4 KindBox and utilizes the
scheme and values as per the DASH role scheme defined in MPEG-DASH.
Other schemes are technically allowed, but where multiple schemes
define the same concepts, the DASH scheme should be utilized.
Such flagging is additionally utilized by the DASH-IF CMAF ingest
specification, enabling an encoder to inform the following component
of the roles of the incoming media streams.
A test is added for this functionality in a similar manner to the
matroska test.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Up until now, the Matroska muxer did not use the dispositions it is
given as-is; instead it by default overrode the disposition of the first
track of a kind (audio, video, subtitles) if no track of this kind has
the default disposition set. And up until recently, it also enforced
by default that no more than one track of each kind be marked as
default.
The rationale for the former is that there are lots of containers which
lack the concept of default streams, so that it is not uncommon for no
stream to be marked as default at all; the rationale for the latter was
that up until recently, it was dubious whether the Matroska specification
allowed more than one default stream for track type (e.g. mkvmerge
disallowed it). It was this point which led to the implementation of
the above mentioned behaviour inspired by mkvmerge.
Yet the Matroska specifications have changed and now explicitly allow
to set more than one track of each type as default, so that the main
reason of not using the dispositions as-is was rendered moot. Therefore
this commit changes the default to pass the disposition through.
The matroska-mpegts-remux FATE-test has been updated to still use the
old "infer" mode so that it is still covered by FATE; the
matroska-zero-length-block test has also been updated to cover
the infer_no_subs mode. The references for lots of other FATE tests
needed to be updated because of a newly added FlagDefault element with
value zero (whereas a FlagDefault with value 1 needn't be coded at all,
as it coincided with the default value of said element).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
