This reverts commit 628a73f8f3.
At the time of said commit there was talk of removing the audio bitrate
"ab" option to bring FFmpeg in line with what Libav has done in 2012 in
commit 041cd5a0c5. By having different
option flags for the "ab" and the ordinay bitrate "b" option is is
possible to have different default bitrates for audio and video. In
order to maintain this behaviour and not break user scripts the commit
to be reverted added code to ffmpeg.c that set the bitrate value to the
audio default for audio codecs, but only if AVCodec.defaults didn't
exist (as in this case the default would be codec-default and not
affected by the "ab" removal).
This had the downside of being an API violation, because
AVCodec.defaults is not a public field. Given that the "ab" option
and its audio-specific default value have never been removed,
said API violation can be simply fixed by reverting said commit.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.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 reverts commit b3a0548a98.
This breaks the usage of swscale options, scale_sws_opts should be
passed to auto-inserted scale-filters.
Signed-off-by: Linjie Fu <linjie.justin.fu@gmail.com>
When both -filter_threads and -threads are specified, the latter takes
effect. Since -threads is an encoder option and -filter_threads is a
filter option, it makes sense for the -filter_threads to take
precedence.
Besides being nicer code this also has the advantage of not making
assumptions about the internal implementation: While it is documented
that the AVFilter.inputs and AVFilter.outputs arrays are terminated
by a zeroed sentinel, one is not allowed to infer that one can just
check avfilter_pad_get_name(padarray, i) to see whether one has reached
the sentinel:
It could be that the pointer to the string is contained
in a different structure than AVFilterPad that needs to be accessed
first: return pad->struct->string.
It could be that for small strings an internal buffer in
AVFilterPad is used (to avoid a relocation) whereas for longer strings
an external string is used; this is useful to avoid relocations:
return pad->string_ptr ? pad->string_ptr : pad->interal_string
Or it could be that the name has a default value:
return pad->name ? pad->name : "default"
(This actually made sense for us because the name of most of our
AVFilterPads is just "default"; doing so would save lots of relocations.)
The only thing one is allowed to infer from the existence of the
sentinel is that one is allowed to use avfilter_pad_count() to get
the number of pads. Therefore it is used.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently user may use '-init_hw_device type=name' to initialize a hw
device, however the key parameter is ignored when use '-init_hw_device
type=name,key=value'. After applying this patch, user may set key
parameter if needed.
Reviewed-by: Soft Works <softworkz@hotmail.com>
Signed-off-by: James Almer <jamrial@gmail.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>
677a030b26 introduced more printable
side data types in ffprobe, however the Audio Service Type side data
'type' field that was introduced aliases an existing field of the same
name within the side data array, which can lead to JSON output like:
"side_data_list": [
{
"side_data_type": "Audio Service Type",
"type": 0
},
{
"side_data_type": "Stereo 3D",
"type": "side by side",
"inverted": 1
}
]
This, while technically valid JSON, is considered bad practice, since it
forces all downstream users to manually parse it and check all types;
it makes simple deserialization impossible. Worse, in som loosely
type languages, it can lead to silent bugs if exising code assumed
it was a different type.
As such, rename this second "type" field to "service_type".
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
11d3b03fcb added consideration of default stream disposition for audio
and video when choosing the 'best' stream among all the inputs. This can
lead to video streams with lower resolution or audio streams with fewer
channels being selected.
Stream disposition, however, only sets a priority for a stream
among all other streams in the *same input*. It cannot set a priority
for a stream across all inputs.
This patch sets a middle-way and selects the best stream from each file
with default disposition considered. Then it discards disposition weight
and selects best stream as per the original criteria of highest
resolution for video and most channels for audio.
Having the override before autodetection meant that the overridden
value got overwritten by the autodetected result each time,
effectively disabling the ability to utilize the `-top` option
for override purposes.
Somehow I missed this in fbb44bc51a ,
even though the lines were within the context. Probably the code
originally being after this logic had something to do with it,
but previously it only touched the avformat context's codecpar,
which did not affect the encoder codec context whatsoever.
Fixes#9320Fixes#9339
Read rate enforcement delayed till first decoded frame is obtained, to
speed up init of output streams.
Thanks to Linjie Fu <linjie.justin.fu@gmail.com> for the initial patch.
if input start time is not 0 -t is inaccurate doing stream copy,
will record extra duration according to input start time.
it should base on following cases:
input video start time from 60s, duration is 300s,
1. stream copy:
ffmpeg -ss 40 -t 60 -i in.mp4 -c copy -y out.mp4
open_input_file() will seek to 100 and set ts_offset to -100,
process_input() will offset pkt->pts with ts_offset to make it 0,
so when do_streamcopy() with -t, exits when ist->pts >= recording_time.
2. stream copy with -copyts:
ffmpeg -ss 40 -t 60 -copyts -i in.mp4 -c copy -y out.mp4
open_input_file() will seek to 100 and set ts_offset to 0,
process_input() will keep raw pkt->pts as ts_offset is 0,
so when do_streamcopy() with -t, exits when
ist->pts >= (recording_time+f->start_time+f->ctx->start_time).
3. stream copy with -copyts -start_at_zero:
ffmpeg -ss 40 -t 60 -copyts -start_at_zero -i in.mp4 -c copy -y out.mp4
open_input_file() will seek to 120 and set ts_offset to -60 as start_to_zero option,
process_input() will offset pkt->pts with input file start time,
so when do_streamcopy() with -t, exits when ist->pts >= (recording_time+f->start_time).
0 60 40 60 360
|_______|_____|_______|_______________________|
start -ss -t
This fixes ticket #9141.
Signed-off-by: Shiwang.Xie <shiwang.xie666@outlook.com>
xmllint (silently) replaces the ' with " when fixing and validating the output
of ffprobe in fate-ffprobe_xsd.
Reviewed-by: Tobias Rapp <t.rapp@noa-archive.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Besides being unnecessary it is also safer: If the error for an
unrecognized option were triggered (which seems to be impossible right
now), it might be that the stream whose codecpar is accessed is NULL.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Otherwise the rate emulation logic in `transcode_step` never gets
hit, and the unavailability flag never gets reset, leading to an
eternal loop with some rate emulation use cases.
This change was missed during the rework of ffmpeg.c, in which
encoder initialization was moved further down the time line in
commit 67be1ce0c6 . Previously,
as the encoder initialization had happened earlier, this state was
not possible (flow getting as far as hitting the rate emulation logic,
yet not having the encoder initialized yet).
Fixes#9160
Use AVSTREAM_EVENT_FLAG_NEW_PACKETS instead, which should provide the
same information in this case.
Finishes removing all uses of this field as started by 87f0c8280c.
Signed-off-by: James Almer <jamrial@gmail.com>
