Its use for muxing is not documented, in practice it is incremented per
each packet successfully passed to the muxer's write_packet(). Since
there is a lot of indirection between ffmpeg receiving a packet from the
encoder and it actually being written (e.g. bitstream filters, the
interleaving queue), using nb_frames here is incorrect.
Add a new counter for packets received from encoder instead.
This field is currently used by checks
- skipping packets before the first keyframe
- skipping packets before start time
to test whether any packets have been output already. But since
frame_number is incremented after the bitstream filters are applied
(which may involve delay), this use is incorrect. The keyframe check
works around this by adding an extra flag, the start-time check does
not.
Simplify both checks by replacing the seen_kf flag with a flag tracking
whether any packets have been output by do_streamcopy().
Bitstream filters inserted between the input and output were never drained,
resulting in packets being lost if the bsf had any buffered.
Signed-off-by: James Almer <jamrial@gmail.com>
This was almost completely redundant. The only functionality that's no longer
available after this removal is the videotoolbox_pixfmt arg, which has been
obsolete for several years.
send_frame_to_filters() sends a frame to all the filters that
need said frame; for every filter except the last one this involves
creating a reference to the frame, because
av_buffersrc_add_frame_flags() by default takes ownership of
the supplied references. Yet said function has a flag which
changes its behaviour to create a reference itself.
This commit uses this flag and stops creating the references itself;
this allows to remove the spare AVFrame holding the temporary
references; it also avoids unreferencing said frame.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
As well as the custom get_buffer2() implementation which would become a
redundant wrapper for avcodec_default_get_buffer2() after this
Signed-off-by: James Almer <jamrial@gmail.com>
This way the CLI accepts for "filter_threads" the same values as for the
libavcodec specific option "threads".
Fixes FATE with THREADS=auto which was broken in bdc1bdf3f5.
Signed-off-by: James Almer <jamrial@gmail.com>
These were intended to pass options to auto-inserted avresample
resampling filters. Yet FFmpeg uses swresample for this purpose
(with its own AVDictionary swr_opts similar to resample_opts).
Therefore said options were not forwarded any more since commit
911417f0b34e611bf084319c5b5a4e4e630da940; moreover since commit
420cedd497 avresample options are
not even recognized and ignored any more. Yet there are still
remnants of all of this. This commit gets rid of them.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This allows user set hw_device_ctx instead of hw_frames_ctx for QSV
decoders, hence we may remove the ad-hoc libmfx setup code from FFmpeg.
"-hwaccel_output_format format" is applied to QSV decoders after
removing the ad-hoc libmfx code. In order to keep compatibility with old
commandlines, the default format is set to AV_PIX_FMT_QSV, but this
behavior will be removed in the future. Please set "-hwaccel_output_format qsv"
explicitly if AV_PIX_FMT_QSV is expected.
The normal device stuff works for QSV decoders now, user may use
"-init_hw_device args" to initialise device and "-hwaccel_device
devicename" to select a device for QSV decoders.
"-qsv_device device" which was added for workarounding device selection
in the ad-hoc libmfx code still works
For example:
$> ffmpeg -init_hw_device qsv=qsv:hw_any,child_device=/dev/dri/card0
-hwaccel qsv -c:v h264_qsv -i input.h264 -f null -
/dev/dri/renderD128 is actually open for h264_qsv decoder in the above
command without this patch. After applying this patch, /dev/dri/card0
is used.
$> ffmpeg -init_hw_device vaapi=va:/dev/dri/card0 -init_hw_device
qsv=hw@va -hwaccel_device hw -hwaccel qsv -c:v h264_qsv -i input.h264
-f null -
device hw of type qsv is not usable in the above command without this
patch. After applying this patch, this command works as expected.
Reviewed-by: Soft Works <softworkz@hotmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The obstacle to do so was in filter_codec_opts: It uses searches
the AVCodec for options via the AV_OPT_SEARCH_FAKE_OBJ method, which
requires using a void * that points to a pointer to a const AVClass.
When using const AVCodec *, one can not simply use a pointer that points
to the AVCodec's pointer to its AVClass, as said pointer is const, too.
This is fixed by using a temporary pointer to the AVClass.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
At present, progress stats are updated at a hardcoded interval of
half a second. For long processes, this can lead to bloated
logs and progress reports.
Users can now set a custom period using option -stats_period
Default is kept at 0.5 seconds.
This way the old max queue size limit based behavior for streams
where each individual packet is large is kept, while for smaller
streams more packets can be buffered (current default is at 50
megabytes per stream).
For some explanation, by default ffmpeg copies packets from before
the appointed seek point/start time and puts them into the local
muxing queue. Before, it getting utilized was much less likely
since as soon as the filter chain was initialized, the encoder
(and thus output stream) was also initialized.
Now, since we will be pushing the encoder initialization to when the
first AVFrame is decoded and filtered - which only happens after
the exact seek point is hit as packets are ignored until then -
this queue will be seeing much more usage.
In more layman's terms, this attempts to fix cases such as where:
- seek point ends up being 5 seconds before requested time.
- audio is set to copy, and thus immediately begins filling the
muxing queue.
- video is being encoded, and thus all received packets are skipped
until the requested time is hit.
The user has no business modifying the underlying AVCodec.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Currently, ffmpeg inserts scale filter by default in the filter graph
to force the whole decoded stream to scale into the same size with the
first frame. It's not quite make sense in resolution changing cases if
user wants the rawvideo without any scale.
Using autoscale/noautoscale as an output option to indicate whether auto
inserting the scale filter in the filter graph:
-noautoscale or -autoscale 0:
disable the default auto scale filter inserting.
ffmpeg -y -i input.mp4 out1.yuv -noautoscale out2.yuv -autoscale 0 out3.yuv
Update docs.
Suggested-by: Mark Thompson <sw@jkqxz.net>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: U. Artie Eoff <ullysses.a.eoff@intel.com>
Signed-off-by: Linjie Fu <linjie.fu@intel.com>
Each time the sub2video structure is initialized, the sub2video
subpicture is initialized together with the first received heartbeat.
The heartbeat's PTS is utilized as the subpicture start time.
Additionally, add some documentation on the stages.
It's a duplicate of the properly implemented nvdec libavcodec hwaccel
Reviewed-by: Timo Rothenpieler <timo@rothenpieler.org>
Signed-off-by: James Almer <jamrial@gmail.com>
Forced key frames generation functionality was assuming the first PTS
value as zero, but, when 'copyts' is enabled, the first PTS can be any
big number. This was eventually forcing all the frames as key frames.
To resolve this issue, update has been made to use first input pts as
reference pts.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
