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).
Looking at 3f75e5116b, the field
was supposed to be private, but during merging the field and the
group that had the comment about it got separated.
Thus, move the actual privately utilized state of this variable
into the private FFIOContext. Additionally, name the private field
somewhat better, so that it does not get confused with the amount
of bytes written out.
Before adts_aac_resync would always bail out after probesize amount
of bytes had been progressed from the start of the input.
Now just query the current position when entering resync, and at most
advance probesize amount of data from that start position.
Fixes#9433
Unlike libx264, libx265 does not handle the chroma format check
on its own side, so in order to not write out values which are
supposed to be ignored according to the specification, we limit
the writing out of chroma sample location to 4:2:0 only.
subtitles.mak's fate-sub tests utilize a more strict comparator
("rawdiff"), which causes the tests fail in case of white space
differences, such as CRLF vs LF. This in turn causes these
ffprobe-using TTML-in-MP4 tests to fail on non-LF systems such as
Windows or wine.
Unlike libx264, libx265 does not have a separate "unspecified"/"auto"
default for color range, so we do always have to specify it.
Thus, we are required to handle the RGB case on the libavcodec
side to enable the correct value to be written out in in case
of RGB content with unspecified color range being received.
In other words:
1. If the user has set color range specifically, follow that.
2. If the user has not set color range specifically, set full
range by default in case of RGB and YUVJ pixel formats.
By default the x264 full range flag is set to -1. By not setting
it to something else, we can let libx264 handle the RGB case.
Additionally, change the preference order to user-specified range
first, and then any fall-back logic left for the YUVJ pix_fmts.
Fixes the capture part of #9374
In 1c42fd9323 the ipcm identifier was
added in order to demux additional raw audio from Sony MP4 files.
Unfortunately, it was not noticed that this same list is utilized
for muxing as well, thus causing ipcm to get preferred compared
to the identifier officially specified in QTFF documentation.
This fixes the order of preference for 24bit PCM, where ipcm is
still allowed, but in24 is the first match - thus being preferred.
Fixes fate-acodec-pcm-s24be.
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
We already require X264_BUILD >= 118, which includes an unconditional
definition of X264_CSP_BGR in itself, thus making this check
effectively always true.
This makes the libx264rgb check work when pkg-config is utilized
and x264.h is not part of the standard include path (as is often
with cross-compilation, or when you just have a custom prefix in
general in f.ex. your home directory).
The X264_BUILD >= 118 required by configure since 2011 should have
X264_CSP_BGR defined unconditionally (it was added a few X264_BUILD
updates earlier), but as 134cba728b
added this additional check, I have kept it for now.
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
Libavcodec can now handle the AV1CodecConfigurationRecord structure
as-is when passed as extradata, so the standard behavior of
read-box-into-extradata should suffice, just like with AVC and HEVC.
Apparently for various image sequences libavformat/utils.c can
calculate rather fancy r_frame_rate values, such as `186/1921`,
and since ffmpeg.c utilizes r_frame_rate for the filter chain
time base, this can quite deteriorate the output frame timing - even
though the user has requested the image sequence to be interpreted
at a specific, constant frame rate.
These made sense before 3ebf449766
when the creation of these boxes was specifically requested by the
user, but now they have become unnecessary as they are just like
many other boxes: If the input has the information required, they
are written, otherwise they are not.
They were moved to verbose verbosity level (which happens to be
the last level still relatively usable), and now appear either once
(normal MP4 muxing), or thrice (with the faststart flag set) in
any normal MP4 usage, without giving much useful information. Thus,
remove them in their current form.
This lets us re-utilize the extradata-related checks in the CBS
to add support for passing the AV1CodecConfigurationRecord
as extradata as-is without further filtering.
We now have the possibility of getting AVFrames here, and we should
not touch the muxer's codecpar after writing the header.
Results of FATE tests change as the MXF and Matroska muxers actually
write down the field/frame coding type of a stream in their
respective headers. Before this change, these values in codecpar
would only be set after the muxer was initialized. Now, the
information is also available for encoder and muxer initialization.
Additionally, reap the first rewards by being able to set the
color related encoding values based on the passed AVFrame.
The only tests that seem to have changed their results with this
change seem to be the MXF tests. There, the muxer writes the
limited/full range flag to the output container if the encoder
is not set to "unspecified".
- For video, this means a single initialization point in do_video_out.
- For audio we unfortunately need to do it in two places just
before the buffer sink is utilized (if av_buffersink_get_samples
would still work according to its specification after a call to
avfilter_graph_request_oldest was made, we could at least remove
the one in transcode_step).
Other adjustments to make things work:
- As the AVFrame PTS adjustment to encoder time base needs the encoder
to be initialized, so it is now moved to do_{video,audio}_out,
right after the encoder has been initialized. Due to this,
the additional parameter in do_video_out is removed as it is no
longer necessary.
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.
Fixes vf_scale outputting RGB AVFrames with limited range flagged
in case either input or output specifically sets the range.
This is the reverse of the logic utilized for RGB and PAL8 content
in sws_setColorspaceDetails.
As it was brought up that the current documentation leaves things
as specific to YCbCr only, ICtCp and RGB are now mentioned.
Additionally, the specifications on which these definitions of
narrow and full range are defined are mentioned.
This way, the documentation of AVColorRange should now match how
most people seem to read interpret it at this point, and thus
flagging RGB AVFrames as full range is valid not only according to
common sense, but also the enum definition.
If the average bit rate cannot be calculated, such as in the case
of streamed fragmented mp4, utilize various available parameters
in priority order.
Tests are updated where the esds or btrt or ISML manifest boxes'
output changes.
This is utilized by various media ingests to figure out the bit
rate of the content you are pushing towards it, so write it for
video, audio and subtitle tracks in case at least one nonzero value
is available. It is only mentioned for timed metadata sample
descriptions in QTFF, so limit it only to ISOBMFF (MODE_MP4) mode.
Updates the FATE tests which have their results changed due to the
20 extra bytes being written per track.
Currently the utilized AVBPrint API is internally limited to unsigned
integers, so if we limit the file size as well as the amount to read
to UINT_MAX - 1, we do not require additional limiting to be performed
on the values.
This change is based on the fact that initially the 8*1024 value added
in 96d70694ae was only for the case where
the file size was not known. It was not a maximum file size limit.
In 2912118898 this was reworked to be
a maximum manifest file size limit, while its commit message appears
to only note that it added support for larger manifest file sizes.
This should enable various unfortunately large MPEG-DASH manifests,
such as Youtube's multi-megabyte live stream archives to load up
as well as bring back the original intent of the logic.
Validates the set channel layout as well as verifies that the received
layout to the function matches the reference layout, so that it matches
the implemented re-ordering logic.
Fixes#8845
This is as far as 22.2 follows the same channel order as
WaveFormatExtensible's channel mask (and the AV_CH_* defines).
After LFE2 the side channels would follow, but that offset of
one stops us from utilizing them without further tweaks.
This change was verified by using swresample to downmix to 5.1,
and then feeding that to WASAPI.
While having the possibility of non-NOPTS values that can suddenly
jump in time due to adjustments to match PCR is not nice for DVB
subtitles, apparently the parser for this format bases its behavior on
whether the packets' timestamps are NOPTS or not. Thus while we can
adjust timestamps, we should exclude DVB subtitles from the timestamp
unsetting logic.
Fixes#8844
Only this sub-set of channels actually follows the bit mask order
in the official 22.2 channel mapping. Additionally, the 5.1 channels
are there for backwards compatibility with the previous system.
This enables the utilization of 22.2 content until a proper down/up
matrix is added into swresample.
These bits are utilized by channel layouts such as 22.2. If those
are dropped, the returned channel layout is no longer a match
against the AV_CH_LAYOUT define when returned from this function.
Requires some extraneous top side and bottom front channels to be
defined.
According to STD-B59v2, the defined channel layout is:
- FL
- FR
- FC
- LFE1
- BL
- BR
- FLc
- FRc
- BC
- LFE2
- SiL
- SiR
- TpFL
- TpFR
- TpFC
- TpC
- TpBL
- TpBR
- TpSiL
- TpSiR
- TpBC
- BtFC
- BtFL
- BtFR
Until now, we would have only attempted to utilize already decrypted
data if it was enough to fill the size of buffer requested, that could
very well be up to 32 kilobytes.
With keep-alive connections this would just lead to recv blocking
until rw_timeout had been reached, as the connection would not be
officially closed after each transfer. This would also lead to a
loop, as such timed out I/O request would just be attempted again.
By just returning the available decrypted data, keep-alive based
connectivity such as HLS playback is fixed with schannel.
The dec_buf seems to be properly managed between read calls,
and we have no logic to decrypt before attempting socket I/O.
Thus - until now - such data would not be decrypted in case of
connections such as HTTP keep-alive, as the recv call would
always get executed first, block until rw_timeout, and then get
retried by retry_transfer_wrapper.
Thus - if data is received - decrypt all of it right away. This way
it is available for the following requests in case they can be
satisfied with it.
Utilizes a subpicture sample with one decodable subpicture for the
test.
Based on a failing test case in reported by Michael in
https://ffmpeg.org/pipermail/ffmpeg-devel/2019-February/240398.html
which at the time had no test case for it.
Additionally, this is the first test case for the presentation
graphics format.
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.
This way re-initializations properly update end_pts, enabling
sub2video_heartbeat to call sub2video_update as expected to re-init
the sub2video AVFrame's contents and to feed a frame into the filter
chain.
This then fixes memory usage ballooning due to framesync waiting
for secondary input in case of no actual subtitle samples being present
for a while in source after a re-init occurs.
Fixes some rather embarrassing mistakes that somehow passed my
eyes.
* Now catches if memory allocation has failed during bprint usage
by checking av_bprint_is_complete().
* Now catches if adding an ASS rectangle into an AVSubtitle failed.
* Returns AVERROR_INVALIDDATA if we get an invalid region buffer
length.
This enables us to read the data coding type utilized for
a specific private data stream, of which we currently are
interested in ARIB caption streams.
The component tag limitations are according to ARIB TR-B14,
and the component IDs are defined in ARIB STD-B10.
* Outputs ASS lines with basic coloring and font scaling for each
given region.
* Sets the default style to the resolution of the subtitle plane
(for example, 960x540 / 36pt font for profile A).
* Has options to:
* Disable ruby text (which is coded as regions which have
half-height text in libaribb24).
Enabled by default as without positioning ruby text only
confuses as it is usually coded in the beginning of the decoded
subtitle line.
* Set the working directory, in which libaribb24 will read
configuration as well as into which it may save broadcast extra
symbols as PNG.
Unset by default.
The unconventional library check can be explained by the library's
current master branch being licensed as LGPLv3, but at the time of
writing the latest official release is still licensed under GPLv3.
Thus, one either has to wait for the following release, or enable
GPLv3.
Fixes issues when a subtitle packet is received before PCR for the
program has been received, leading to wildly jumping timestamps
on the lavf client side as well as in the re-ordering logic.
This usually happens in case of multiplexes where the PCR of a
program is not taken into account with subtitle tracks' DTS/PTS.
Internally in ISOBMFF the FLAC-in-ISOBMFF draft uses "fLaC"
as the identifier for FLACSampleEntry, and there seems to be no
MPEG-DASH specification for the in-manifest identifier for FLAC.
After testing the browsers' implementations, it seems like all of
the major browser vendors have decided to utilize the MIME type for
FLAC ("audio/flac") as the identifier. This change set leads to
that string being utilized for FLAC streams instead of the sample
entry identifier ("fLaC"), which is the default behavior.
Verified by auri_ on IRC to play with the major browsers.
ISMV lacks any sort of edit list support, as well as tfxd is
effectively the PTS of the fragment for most intents and purposes.
Thus, if b-frames are requested without negative CTS offsets you
end up with N frames' worth of delay (tfxd PTS plus the CTS offset
of the first sample). Negative CTS offsets enable the first sample
to have CTS=DTS, and thus a/v desync due to b-frame reorder delay
is avoided.
These files depend on libavformat, and the vf_srcnn filter
currently is the only thing utilizing these dnn_* files and
already happens to have a dependency on libavformat.
This fixes compilation in cases where libavformat is not a
dependency for libavfilter.
Reported by Kam_ on IRC.
Yet another case of forgotten 0 =! EOF translation.
While the documentation for this specific synchronous read
function does not mention it, the documentation for
`sftp_async_read` documents it, as well as looking at the
implementation of this function leads one to find
`if (handle->eof) { return 0; }`.
Reported by stnutt on IRC.
It exists, so why not use it? Helps one get rid of additional
search path related flags in addition to PKG_CONFIG_{PATH,LIBDIR}
when utilizing a cross-prefix separate from the sysroot.
Enables one to test possibly nonstandard formats such as Opus or
FLAC in ISOBMFF, among other things.
This becomes much more useful if output segment format becomes an
option, or if the WebM segment feature gets removed.
It has not ever been working and has not been validated,
Additionally, mention that the segment file names should be changed
to end with webm instead of m4s, which is utilized for ISOBMFF
fragments.
With certain types of input and the filter chain getting re-initialized
or re-configured, multiple nullptr AVSubtitles can get pushed into
sub2video_update() in a row from sub2video_heartbeat.
This causes end_pts, and on the next round pts, to become INT64_MAX,
latter of which signals EOF in framesync, leading to complete loss of
subtitles from that point on.
Thus, check that the sub2video.end_pts is smaller than INT64_MAX
in a similar fashion to sub2video_flush before sending out the
nullptr AVSubtitle. This keeps premature EOFs from happening in
framesync and the subtitle overlay is kept past the filter chain
re-initializations/configurations.
Enables having proper bit rate values being written into the master
playlist in case of hard-constrained VBR where the maximum bit
rate utilized is known before hand.
Does the same thing as movenc.c, for example.
Signed-off-by: Jan Ekström <jan.ekstrom@aminocom.com>
Makes the uninit function re-entrable, which can be a common case
when an API user first tries to initialize its context, fails, and
then finally unrefs the AVHWDevice.
Fixes a crash reported by sm2345 on IRC.
This FFmpeg-specific "fuzzer fix" was never perfect, but now it
stopped encoding of actual content with a big enough DTS shift.
This returns the function to its original state of results
before negative CTS offsets were added.
I remember dealing with this function before, but somehow had
forgotten about it during VDD. The test cases not tripping this
over also didn't help.
This enables having multiple tracks of the same type which would
be treated as different things by the media server (as opposed to
different bit rate versions of the same track). According to the
smooth streaming specification, just setting the systemLanguage
tag is not enough to note that a track with the same attributes
differs from another one.
Reviewed-by: Martin
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Streaming servers appear to ignore all other language metadata.
Signed-off-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Josh de Kock <josh@itanimul.li>
This way, in case of bit rate not being set, max_bitrate will be
used instead. This enables, for example, re-using max_bitrate
information from the input or doing transcoding with a rate
control mode that is not bit rate based.
Signed-off-by: Jan Ekström <jeebjp@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Widen the values from limited to full range and use BT.709 where it
should be used according to the video resolution:
SD is BT.601, HD is BT.709
Default to BT.709 due to most observed HDMV content being HD.
BT.709 coefficients were gathered from the first two parts of BT.709
to BT.2020 conversion guide in ARIB STD-B62 (Pt. 1, Chapter 6.2.2).
They were additionally confirmed by manually calculating values.
Functionality used before didn't widen the values from limited to
full range. Additionally, now the decoder uses BT.709 where it
should be used according to the video resolution.
Default for not yet set colorimetry is BT.709 due to most observed
HDMV content being HD.
BT.709 coefficients were gathered from the first two parts of BT.709
to BT.2020 conversion guide in ARIB STD-B62 (Pt. 1, Chapter 6.2.2).
They were additionally confirmed by manually calculating values.
Fixes#4637
With cli usage the decoder might have not set the colorspace during
encoder init, manual colorspace override might be needed in such
cases.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This applies commit 5de64bb3 (the source of the above commit message)
to libutvideoenc as well.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The official Ut Video decoder only threads with slices, thus until
now any files encoded by the libavcodec encoder have only been
decodable with a single thread. The default slice count is now
set to subsampled_height / 120.
Also sets slices to 1 for the Ut Video encoder tests to keep them
green.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The official Ut Video decoder only threads with slices, thus until
now any files encoded by the libavcodec encoder have only been
decodable with a single thread. The default slice count is now
set to subsampled_height / 120.
Also sets slices to 1 for the Ut Video encoder tests to keep them
green.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
With cli usage the decoder might have not set the colorspace during
encoder init, manual colorspace override might be needed in such
cases.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Makes it easier to recreate an AVCodecContext for ATRAC3+ decoding,
which is needed in multimedia frameworks, as well as in general cases
where demuxing and decoding are separate entities.
Jan EkströmandNiklas Haas
Jan Ekström