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>
The lensfun filter, at present, loads its database from a path hardcoded
at build time. This may not be known or available to end users.
Added option db_path allows custom path.
Calculate Spatial Info (SI) and Temporal Info (TI) scores for a video, as defined
in ITU-T P.910: Subjective video quality assessment methods for multimedia
applications.
This comment only applies to the scenario in which one uses
the AVCodecContexts embedded in AVStreams. Yet this code sample
stopped doing so in 9897d9f4e074cdc6c7f2409885ddefe300f18dc7;
and the last major version bump even removed the public
AVCodecContexts in AVStreams. So just remove this comment.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids unnecessary churn and build breakage for users, by
making sure the whole version.h is included like it has been so far,
while keeping the benefit of not needing to rebuild most files in
the ffmpeg tree on minor/micro bumps.
Signed-off-by: Martin Storsjö <martin@martin.st>
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 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>
From the wiki page (https://wiki.vexatos.com/dfpwm):
> DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec
> created by Ben “GreaseMonkey” Russell in 2012, originally to be used
> as a voice codec for asiekierka's pixmess, a C remake of 64pixels.
> It is a 1-bit-per-sample codec which uses a dynamic-strength one-pole
> low-pass filter as a predictor. Due to the fact that a raw DPFWM decoding
> creates a high-pitched whine, it is often followed by some post-processing
> filters to make the stream more listenable.
It has recently gained popularity through the ComputerCraft mod for
Minecraft, which added support for audio through this codec, as well as
the Computronics expansion which preceeded the official support. These
both implement the slightly adjusted 1a version of the codec, which is
the version I have chosen for this patch.
This patch adds a new codec (with encoding and decoding) for DFPWM1a.
The codec sources are pretty simple: they use the reference codec with
a basic wrapper to connect it to the FFmpeg AVCodec system.
To clarify, the codec does not have a specific sample rate - it is
provided by the container (or user), which is typically 48000, but has
also been known to be 32768. The codec does not specify channel info
either, and it's pretty much always used with one mono channel.
However, since it appears that libavcodec expects both sample rate and
channel count to be handled by either the codec or container, I have
made the decision to allow multiple channels interleaved, which as far
as I know has never been used, but it works fine here nevertheless. The
accompanying raw format has a channels option to set this. (I expect
most users of this will not use multiple channels, but it remains an
option just in case.)
This patch will be highly useful to ComputerCraft developers who are
working with audio, as it is the standard format for audio, and there
are few user-friendly encoders out there, and even fewer decoders. It
will streamline the process for importing and listening to audio,
replacing the need to write code or use tools that require very
specific input formats.
You may use the CraftOS-PC program (https://www.craftos-pc.cc) to test
out DFPWM playback. To use it, run the program and type this command:
"attach left speaker" Then run "speaker play <file.dfpwm>" for each file.
The app runs in a sandbox, so files have to be transferred in first;
the easiest way to do this is to simply drag the file on the window.
(Or copy files to the folder at https://www.craftos-pc.cc/docs/saves.)
Sample DFPWM files can be generated with an online tool at
https://music.madefor.cc. This is the current best way to encode DFPWM
files. Simply drag an audio file onto the page, and it will encode it,
giving a download link on the page.
I've made sure to update all of the docs as per Developer§7, and I've
tested it as per section 8. Test files encoded to DFPWM play correctly
in ComputerCraft, and other files that work in CC are correctly decoded.
I have also verified that corrupt files do not crash the decoder - this
should theoretically not be an issue as the result size is constant with
respect to the input size.
Signed-off-by: Jack Bruienne <jackbruienne@gmail.com>
They correspond to the relevant fields from the packet that follows the
one where the expressions are being applied.
Signed-off-by: James Almer <jamrial@gmail.com>
Fix: #7706. After commit 5fdcf85bbf, vaapi encoder's performance
decrease. The reason is that vaRenderPicture() and vaSyncBuffer() are
called at the same time (vaRenderPicture() always followed by a
vaSyncBuffer()). Now I changed them to be called in a asynchronous way,
which will make better use of hardware.
Async_depth is added to increase encoder's performance. The frames that
are sent to hardware are stored in a fifo. Encoder will sync output
after async fifo is full.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Users should switch to the superior AVFifo API.
Unfortunately AVFifoBuffer fields cannot be marked as deprecated because
it would trigger a warning wherever fifo.h is #included, due to
inlined av_fifo_peek2().
Many AVFifoBuffer users operate on fixed-size elements (e.g. pointers),
but the current FIFO API deals exclusively in bytes, requiring extra
complexity in all these callers.
Add a new AVFifo API creating a FIFO with an element size
that may be larger than a byte. All operations on such a FIFO then
operate on complete elements.
This API does not reuse AVFifoBuffer and its API at all, but instead uses
an opaque struct called AVFifo. The AVFifoBuffer API will be deprecated
in a future commit once all of its users have been switched to the new
API.
Not reusing AVFifoBuffer also allowed to use the full range of size_t
from the beginning.
Add intra refresh support to hevc_qsv as well.
Add an new intra refresh type: "horizontal", and an new param
ref_cycle_dist. This param specify the distance between the
beginnings of the intra-refresh cycles in frames.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Add b_strategy option to hevc_qsv. By enabling this option, encoder can
use b frames as reference.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Add transform_skip option to hevc_qsv. By enabling this option,
the transform_skip_enabled_flag in PPS will be set to 1.
This option is supported on the platform equal or newer than ICL.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Add low latency P-pyramid support to qsv. This feature relates to
command line option "-p_strategy". To enable this flag, user also
need to set "-bf" to 0. P-strategy has two modes "1-simple" and
"2-pyramid". The details of the two models refer to
https://github.com/Intel-Media-SDK/MediaSDK/blob/master/doc/mediasdk-man.md#preftype
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Add dblk_idc option to 264_qsv and hevc_qsv. Turining on this opion can
disable deblocking.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Add max_frame_size support to hevc_qsv as well.
Signed-off-by: Wenbin Chen <wenbin.chen@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
The SDK may insert picture timing SEI for hevc and the code to set mfx
parameter has been added in qsvenc, however the corresponding option is
missing in the hevc option array
Reviewed-by: Limin Wang <lance.lmwang@gmail.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.com>
Overlay one video on the top of another.
It takes two inputs and has one output. The first input is the "main" video on
which the second input is overlaid. This filter requires same memory layout for
all the inputs.
An example command to use this filter to overlay overlay.mp4 at the top-left
corner of the main.mp4:
ffmpeg -init_hw_device vaapi=foo:/dev/dri/renderD128 \
-hwaccel vaapi -hwaccel_device foo -hwaccel_output_format vaapi -c:v h264 -i main.mp4 \
-hwaccel vaapi -hwaccel_device foo -hwaccel_output_format vaapi -c:v h264 -i overlay.mp4 \
-filter_complex "[0:v][1:v]overlay_vaapi=0:0:100💯0.5[t1]" \
-map "[t1]" -an -c:v h264_vaapi -y out_vaapi.mp4
Signed-off-by: U. Artie Eoff <ullysses.a.eoff@intel.com>
Signed-off-by: Xinpeng Sun <xinpeng.sun@intel.com>
Signed-off-by: Zachary Zhou <zachary.zhou@intel.com>
Signed-off-by: Fei Wang <fei.w.wang@intel.com>
Signed-off-by: Haihao Xiang <haihao.xiang@intel.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>
In order to be able to extend this struct later (as the Dolby Vision RPU
evolves), all of the 'container' structs are considered extensible, and
the individual constituent fields must instead be accessed via offsets.
The precedent for this style of access is set in
<libavutil/detection_bbox.h>
Signed-off-by: Niklas Haas <git@haasn.dev>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Users may take the description literally which leads to inverted
results.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
Reviewed-by: Jun Zhao <barryjzhao@tencent.com
Very high stts sample deltas may occasionally be intended but usually
they are written in error or used to store a negative value for dts correction
when treated as signed 32-bit integers.
This option lets the user set an upper limit, beyond which the delta is clamped to 1.
Values greater than the limit if negative when cast to int32 are used to adjust onward dts.
Unit is the track time scale. Default is UINT_MAX - 48000*10 which
allows upto a 10 second dts correction for 48 kHz audio streams while
accommodating 99.9% of uint32 range.
Signed-off-by: Gyan Doshi <ffmpeg@gyani.pro>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Most of user data unregistered SEIs are privated data which defined by user/
encoder. currently, the user data unregistered SEIs found in input are forwarded
as side-data to encoders directly, it'll cause the reencoded output including some
useless UDU SEIs.
I prefer to add one option to enable/disable it and default is off after I saw
the patch by Andreas Rheinhardt:
https://patchwork.ffmpeg.org/project/ffmpeg/patch/AM7PR03MB66607C2DB65E1AD49D975CF18F7B9@AM7PR03MB6660.eurprd03.prod.outlook.com/
How to test by cli:
ffmpeg -y -f lavfi -i testsrc -c:v libx264 -frames:v 1 a.ts
ffmpeg -y -i a.ts -c:v libx264 -udu_sei 1 b.ts
ffmpeg -y -i a.ts -c:v libx264 -udu_sei 0 c.ts
# check the user data unregistered SEIs, you'll see two UDU SEIs for b.ts.
# and mediainfo will show with wrong encoding setting info
ffmpeg -i b.ts -vf showinfo -f null -
ffmpeg -i c.ts -vf showinfo -f null -
This fixes tickets #9500 and #9557.
Reviewed-by: "zhilizhao(赵志立)" <quinkblack@foxmail.com>
Signed-off-by: Limin Wang <lance.lmwang@gmail.com>
The dshow avdevice ignores timestamps for video frames provided by the
DirectShow device, instead using wallclock time, apparently because the
implementer of this code had a device that provided unreliable
timestamps. Me (and others) would like to use the device's timestamps.
The new use_video_device_timestamps option for dshow device enables them
to do so. Since the majority of video devices out there probably provide
fine timestamps, this patch sets the default to using the device
timestamps, which means best fidelity timestamps are used by default.
Using the new option, the user can switch this off and revert to the old
behavior, so a fall back remains available in case the device provides
broken timestamps.
add use_video_device_timestamps to docs.
Closes: #8620
Signed-off-by: Diederick Niehorster <dcnieho@gmail.com>
Reviewed-by: Roger Pack <rogerdpack2@gmail.com>
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>
Because the hls_ts_options will be misunderstand by user that only can
be used in mpegts segments option. So add this option for segments.
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>
Treat values returned from av_dict_get() as const, since they are
internal to AVDictionary.
Signed-off-by: Chad Fraleigh <chadf@triularity.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently, the code doing this is spread over several places and may
behave in unexpected ways. E.g. automatic 'default' marking is only done
for streams fed by complex filtergraphs. It is also applied in the order
in which the output streams are initialized, which is effectively
random.
Move processing the dispositions at the end of open_output_file(), when
we already have all the necessary information.
Apply the automatic default marking only if no explicit -disposition
options were supplied by the user, and apply it to the first stream of
each type (excluding attached pics) when there is more than one stream
of that type and no default markings were copied from the input streams.
Explicitly document the new behavior.
Changes the results of some tests, where the output file gets a default
disposition, while it previously did not.
Unfortunately pad_len and pad_dur behaviour was different if 0 was specified,
pad_dur handled 0 duration as infinity, for pad_len, infinity was -1.
Let's make the behaviour consistent by handling 0 duration for pad_dur and
whole_dur as indeed 0 duration. This somewhat changes the behaviour of the
filter if 0 was explicitly specified, but deprecating the old option and adding
a new for the corrected behaviour seemed a bit overkill. So let's document the
change instead.
Signed-off-by: Marton Balint <cus@passwd.hu>
Since texinfo 6.8, there's no longer an INLINE_CONTENTS variable.
makeinfo: warning: set_from_init_file: unknown variable INLINE_CONTENTS
texinfo commit 62a6adfb33b006e187483779974bbd45f0f782b1 replaced
INLINE_CONTENTS with OUTPUT_CONTENTS_LOCATION.
texinfo commit 41f8ed4eb42bf6daa7df7007afd946875597452d replaced
OUTPUT_CONTENTS_LOCATION with CONTENTS_OUTPUT_LOCATION.
With texinfo 6.8 and above, the same as INLINE_CONTENTS=1 could be
achieved by CONTENTS_OUTPUT_LOCATION=inline.
https://www.gnu.org/software/texinfo/manual/texinfo/html_node/HTML-Customization-Variables.html
Since texinfo commit 6a5ceab6a48a4f052baad9f3474d741428409fd7, the
formatting functions, in particular begin_file, program_string and
end_file, are prefixed with format_, i.e. format_begin_file, etc.
This patch fixes building the documentation when texinfo 6.8, or
above, is used:
Unknown formatting type begin_file
at /usr/bin/makeinfo line 415.
Unknown formatting type program_string
at /usr/bin/makeinfo line 415.
Unknown formatting type end_file
at /usr/bin/makeinfo line 415.
Try to make the feature easier to use, especially since the user
have enabled -strict experimental manually. The user shouldn't
be surprised that hls_playlist is enabled for lhls automatically,
so change the log level from warning to info for that.
