Remove now-obsolete code setting packet durations pre-muxing for CFR
encoded video.
Changes output in the following FATE tests:
* numerous adpcm tests
* ffmpeg-filter_complex_audio
* lavf-asf
* lavf-mkv
* lavf-mkv_attachment
* matroska-encoding-delay
All of these change due to the fact that the output duration is now
the actual input data duration and does not include padding added by
the encoder.
* apng-osample: less wrong packet durations are now passed to the muxer.
They are not entirely correct, because the first frame duration should
be 3 rather than 2. This is caused by the vsync code and should be
addressed later, but this change is a step in the right direction.
* tscc2-mov: last output frame has a duration of 11 rather than 1 - this
corresponds to the duration actually returned by the demuxer.
* film-cvid: video frame durations are now 2 rather than 1 - this
corresponds to durations actually returned by the demuxer and matches
the timestamps.
* mpeg2-ticket6677: durations of some video frames are now 2 rather than
1 - this matches the timestamps.
That field was added to store timestamp conversion state for audio
decoding code. Later it started being used by streamcopy as well, but
that use is wrong because, for a given input stream, both decoding and
an arbitrary number of streamcopies may be performed simultaneously.
They would then all overwrite the same state variable.
Store this state in MuxStream instead.
This is the last use of InputStream in of_streamcopy(), so the ist
parameter can now be removed.
It stores codec parameters of the stream submitted to the muxer, which
may be different from the codec parameters in AVStream due to bitstream
filtering.
This avoids the confusing back and forth synchronisation between the
encoder, bitstream filters, and the muxer, now information flows only in
one direction. It also reduces the need for non-muxing code to access
AVStream.
Reduces access to a deeply nested muxer property
OutputStream.st->codecpar->codec_type for this fundamental and immutable
stream property.
Besides making the code shorter, this will allow making the AVStream
(OutputStream.st) private to the muxer in the future.
Set InputStream.decoding_needed/discard/etc. only from
ist_{filter,output},add() functions. Reduces the knowledge of
InputStream internals in muxing/filtering code.
init_input_stream() can print log messages directly, there is no need to
ship them to the caller.
Also, log errors to the InputStream and avoid duplicate information in
the message.
Changing AVCodecContext.sample_aspect_ratio after the encoder was opened
is by itself questionable, but if anywhere it belongs in encoding rather
than filtering code.
Creating a new output stream of a given type is currently done by
calling new_<type>_stream(), which all start by calling
new_output_stream() to allocate the stream and do common init, followed
by type-specific init.
Reverse this structure - the caller now calls the common function
ost_add() with the type as a parameter, which then calls the
type-specific function internally. This will allow adding common code
that runs after type-specific code in future commits.
In most cases this should only occur once or so per stream in an
input, and in case the logic ends up in an eternal loop, it should
be visible to the end user instead of being completely invisible.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
When no packet dts values are available from the container, video
decoding code will currently use its own guessed values, which will then
be propagated to pkt_dts on decoded frames and used as pts in certain
cases. This is inaccurate, fragile, and unnecessarily convoluted.
Simply removing this allows the extrapolation code introduced in the
previous commit to do a better job.
Changes the results of numerous h264 and hevc FATE tests, where no
spurious timestamp gaps are generated anymore. Several tests no longer
need -vsync passthrough.
When no timestamps are available from the container, the video decoding
code will currently use fake dts values - generated in
process_input_packet() based on a combination of information from the
decoder and the parser (obtained via the demuxer) - to generate
timestamps during decoder flushing. This is fragile, hard to follow, and
unnecessarily convoluted, since more reliable information can be
obtained directly from post-decoding values.
The new code keeps track of the last decoded frame pts and estimates its
duration based on a number of heuristics. Timestamps generated when both
pts and pkt_dts are missing are then simple pts+duration of the last frame.
The heuristics are somewhat complicated by the fact that lavf insists on
making up packet timestamps based on its highly incomplete information.
That should be removed in the future, allowing to further simplify this
code.
The results of the following tests change:
* h264-3386 now requires -fps_mode passthrough to avoid dropping frames
at the end; this is a pathology of the interaction of the new and old
code, and the fact that the sample switches from field to frame coding
in the last packet, and will be fixed in following commits
* hevc-conformance-DELTAQP_A_BRCM_4 stops inventing an arbitrary
timestamp gap at the end
* hevc-small422chroma - the single frame output by this test now has a
timestamp of 0, rather than an arbitrary 7
This field contains different values depending on whether the stream is
being decoded or not. When it is, InputStream.pts is set to the
timestamp of the last decoded frame. Otherwise, it is made equal to
InputStream.dts.
Since a given InputStream can be at the same time decoded and
streamcopied to any number of output streams, this use is incorrect, as
decoded frame timestamps can be delayed with respect to input packets by
an arbitrary amount (e.g. depending on the thread count when frame
threading is used).
Replace all uses of InputStream.pts for streamcopy with InputStream.dts,
which is its value when decoding is not performed. Stop setting
InputStream.pts for pure streamcopy.
Also, pass InputStream.dts as a parameter to do_streamcopy(), which
will allow that function to be decoupled from InputStream completely in
the future.
Which is subtitle encoding. Also, check for AVSubtitle.pts rather than
InputStream.pts, since that is the more authoritative value and is
guaranteed to be valid.
That function only contains two checks now - whether the muxer returned
EOF and whether the packet timestamp is before requested output start
time.
The first check is unnecessary, since the packet will just be rejected
by the muxer. The second check is better combined with a related check
directly in do_streamcopy().
Currently, output streams where an input stream is sent directly (i.e.
not through lavfi) are determined by iterating over ALL the output
streams and skipping the irrelevant ones. This is awkward and
inefficient.
The channel layout is set before opening the encoder, in enc_open().
Messing with it in configure_output_audio_filter() cannot accomplish
anything meaningful.
This option adds a long string of numbers to the progress line, where
i-th number contains the base-2 logarithm of the number of times a frame
with this QP value was seen by print_report().
There are multiple problems with this feature:
* despite this existing since 2005, web search shows no indication
that it was ever useful for any meaningful purpose;
* the format of what is printed is entirely undocumented, one has to
find it out from the source code;
* QP values above 31 are silently ignored;
* it only works with one video stream;
* as it relies on global state, it is in conflict with ongoing
architectural changes.
It then seems that the nontrivial cost of maintaining this option is not
worth its negligible (or possibly negative - since it pollutes the
already large option space) value.
Users who really need similar functionality can also implement it
themselves using -vstats.
Current code in print_final_stats(), printing the final summary such as
video:8851kB audio:548kB subtitle:0kB other streams:0kB global headers:20kB muxing overhead: 0.559521%
was written with a single output file in mind and makes very little
sense otherwise.
Print this information in mux_final_stats() instead, one line per output
file. Use the correct filesize, if available.
This is currently done in two places:
* at the end of print_final_stats(), which merely prints a warning if
the total size of all written packets is zero
* at the end of transcode() (under a misleading historical 'close each
encoder' comment), which instead checks the packet count to implement
-abort_on empty_output[_stream]
Consolidate both of these blocks into a single function called from
of_write_trailer(), which is a more appropriate place for this. Also,
return an error code rather than exit immediately, which ensures all
output files are properly closed.
Properly pass muxing return codes through the call stack instead.
Slightly changes behavior in case of errors:
* the output IO stream is closed even if writing the trailer returns an
error, which should be more correct
* all files get properly closed with -xerror, even if one of them fails
It is video encoding-only and does not need to be visible outside of
ffmpeg_enc.c
Also, rename the variable to frames_prev_hist to be consistent with
the naming in do_video_out().
Drop unneeded ctype.h and math.h.
Group all system headers together.
Sort unconditional includes alphabetically.
Group local includes by the library, sort alphabetically.
Several places in the code currently call init_output_stream_wrapper(),
which in turn calls init_output_stream(), which then calls either
enc_open() or init_output_stream_streamcopy(), followed by
of_stream_init(), which tells the muxer the stream is ready for muxing.
All except one of these callers are in the encoding code, which will be
moved to ffmpeg_enc.c. Keeping this structure would then necessitate
ffmpeg_enc.c calling back into the common code in ffmpeg.c, which would
then just call ffmpeg_mux, thus making the already convoluted call chain
even more so.
Simplify the situation by using separate paths for filter-fed output
streams (audio and video encoders) and others (subtitles, streamcopy,
data, attachments).
Encoder initialization is currently split rather arbitrarily between
init_output_stream_encode() and init_output_stream(). Move all of it to
init_output_stream_encode().
The code currently uses lavfi for this, which creates a sort of
configuration dependency loop - the encoder should be ideally
initialized with information from the first audio frame, but to get this
frame one needs to first open the encoder to know the frame size. This
necessitates an awkward workaround, which causes audio handling to be
different from video.
With this change, audio encoder initialization is congruent with video.
For audio AVFrames, nb_samples is typically more trustworthy than
duration. Since sync queues look at durations, make sure they match the
sample count.
The last audio frame in the fate-shortest test is now gone. This is more
correct, since it outlasts the last video frame.
This is more correct, but was not possible before the recently-added
filtergraph parsing API.
Also, only pass hw devices to filters that are flagged as capable of
using them.
Tested-by: Niklas Haas
These fields are ad-hoc and will be deprecated. Use the recently-added
AV_CODEC_FLAG_COPY_OPAQUE to pass arbitrary user data from packets to
frames.
Changes the result of the flcl1905 test, which uses ffprobe to decode
wmav2 with multiple frames per packet. Such packets are handled
internally by calling the decoder's decode callback multiple times,
offsetting the internal packet's data pointer and decreasing its size
after each call. The output pkt_size value before this commit is then
the remaining internal packet size at the time of each internal decode
call.
After this commit, output pkt_size is simply the size of the full packet
submitted by the caller to the decoder. This is more correct, since
internal packets are never seen by the caller and should have no
observable outside effects.
Their usefulness is questionable, very few decoders set them, and their type
should have been int64_t. A replacement field can be added later if a valid use
case is found.
Signed-off-by: Marton Balint <cus@passwd.hu>
Frame counters can overflow relatively easily (INT_MAX number of frames is
slightly more than 1 year for 60 fps content), so make sure we use 64 bit
values for them.
Also deprecate the old 32 bit frame_number attribute.
Signed-off-by: Marton Balint <cus@passwd.hu>
Many filters accept user-provided data that is cumbersome to provide as
text strings - e.g. binary files or very long text. For that reason such
filters typically provide a option whose value is the path from which
the filter loads the actual data.
However, filters doing their own IO internally is a layering violation
that the callers may not expect, and is thus best avoided. With the
recently introduced graph segment parsing API, loading option values
from files can now be handled by the caller.
This commit makes use of the new API in ffmpeg CLI. Any option name in
the filtergraph syntax can now be prefixed with a slash '/'. This will
cause ffmpeg to interpret the value as the path to load the actual value
from.
Analogous to -enc_stats*, but happens right before muxing. Useful
because bitstream filters and the sync queue can modify packets after
encoding and before muxing. Also has access to the muxing timebase.
Since at least 4.4.3, -ab/-b:a help text was in the video section
of ffmpeg -h, but these are audio options.
Signed-off-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Splits the currently handled subtitle at random access point
packets that can be configured to follow a specific output stream.
Currently only subtitle streams which are directly mapped into the
same output in which the heartbeat stream resides are affected.
This way the subtitle - which is known to be shown at this time
can be split and passed to muxer before its full duration is
yet known. This is also a drawback, as this essentially outputs
multiple subtitles from a single input subtitle that continues
over multiple random access points. Thus this feature should not
be utilized in cases where subtitle output latency does not matter.
Co-authored-by: Andrzej Nadachowski <andrzej.nadachowski@24i.com>
Co-authored-by: Bernard Boulay <bernard.boulay@24i.com>
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
This way we can call process_subtitles without causing the decoded
frame counter to get bumped.
Additionally, this now takes into mention all of the decoded
subtitle frames without fix_sub_duration latency/buffering, or filtering
out decoded reset/end subtitles without any rendered rectangles, which
matches the original intent in 4754345027
.
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
This enables us to later call this when generating additional
subtitles for splitting purposes.
Co-authored-by: Andrzej Nadachowski <andrzej.nadachowski@24i.com>
Signed-off-by: Jan Ekström <jan.ekstrom@24i.com>
Current code may, depending on the muxer, decide to use VSYNC_VFR tagged
with the specified framerate, without actually performing framerate
conversion. This is clearly wrong and against the documentation, which
states unambiguously that -r should produce CFR output for video
encoding.
FATE test changes:
* nuv-rtjpeg: replace -r with '-enc_time_base -1', which keeps the
original timebase. Output frames are now produced with proper
durations.
* filter-mpdecimate: just drop the -r option, it is unnecessary
* filter-fps-r: remove, this test makes no sense and actually
produces broken VFR output (with incorrect frame durations).
Instead of manually assembling the string, use av_dict_get_string
which handles things like proper escaping too (even though it is
not yet needed here).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Rather than the encoder timebase. Since the times are parsed as
microseconds, this will not reduce precision, except possibly when
chapter times are used and the chapter timebase happens to be better
aligned with the encoder timebase, which is unlikely.
This will allow parsing the keyframe times earlier (before encoder
timebase is known) in future commits.
There are 8 of them and they are typically used together. Allows to pass
just this struct to forced_kf_apply(), which makes it clear that the
rest of the OutputStream is not accessed there.
Do it in set_dispositions() rather than during stream creation.
Since at this point all other stream information is known, this allows
setting disposition based on metadata, which implements #10015. This
also avoids an extra allocated string in OutputStream that was unused
after of_open().
Replace it with an array of streams in each InputFile. This is a more
accurate reflection of the actual relationship between InputStream and
InputFile.
Analogous to what was previously done to output streams in
7ef7a22251.
Encoding init code will currently fall back to a 25fps default when no
framerate is known or specified, but only if there is a known source
input stream. There is no good reason for this condition, so drop it.
Frame limiting is now handled using sync queues. This code prevents the
sync queue from triggering EOF, resulting in unnecessarily many frames
being decoded, filtered, and then discarded.
Found-by: U. Artie Eoff <ullysses.a.eoff@intel.com>
Specificaly, the of_add_attachments() call (which can add attachment
streams to the output) and the check whether the output file contains
any streams. They both logically belong in create_streams().
Some formats like FLV can dynamically add streams during packet reading.
FFprobe does check for this and reallocates the global stream info, but does
not reallocate InputFrame's streams and decoders when this happens, which,
as a result, could have caused flushing to occur on an out of bounds stream
index, since the flush loop iterates over fmt_ctx's nb_streams, and not
ifile's, despite using ifile's streams.
This fixes an out of bounds read and segfult.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
The current adjustment of input start times just adjusts the tsoffset.
And it does so, by resetting the tsoffset to nullify the new start time.
This leads to breakage of -copyts, ignoring of input_ts_offset, breaking
of -isync as well as breaking wrap correction.
Fixed by taking cognizance of these parameters, and by correcting start times
just before sync offsets are applied.
The current code will override the *_disable fields (set by -vn/-an
options) when creating output streams for unlabeled complex filtergraph
outputs, in order to disable automatic mapping for the corresponding
media type.
However, this will apply not only to automatic mappings, but to manual
ones as well, which should not happen. Avoid this by adding local
variables that are used only for automatic mappings.
Specifically recording_time and stop_time - use local variables instead.
OptionsContext should be input-only to this code. Will allow making it
const in future commits.
This is similar to what was done before for output files and will allow
introducing demuxer-private state in future commits
Unlike for muxing, the code is moved to existing ffmpeg_demux.c rather
than to a new file. The reason is just file size - the demuxing code is
much smaller than muxing.
Now that we have proper options for defining display matrix
overrides, this should no longer be required.
fftools does not have its own versioning, so for now the define is
just set to 1 and disables the functionality if set to zero.
This enables overriding the rotation as well as horizontal/vertical
flip state of a specific video stream on the input side.
Additionally, switch the singular test that was utilizing the rotation
metadata to instead override the input display rotation, thus leading
to the same result.
This is now possible since OutputStream is a child of OutputFile and the
code allocating it can access MuxStream. Avoids the overhead and extra
complexity of allocating two objects instead of one.
Similar to what was previously done for OutputFile/Muxer.
Replace it with an array of streams in each OutputFile. This is a more
accurate reflection of the actual relationship between OutputStream and
OutputFile. This is easier to handle and will allow further
simplifications in future commits.
This is now possible since the code allocating OutputFile can see
sizeof(Muxer). Avoids the overhead and extra complexity of allocating
two objects instead of one.
Similar to what is done e.g. for AVStream/FFStream in lavf.
ffmpeg_opt.c currently contains code for
- parsing the options provided on the command line
- opening and initializing input files based on these options
- opening and initializing output files based on these options
The code dealing with each of these is for the most part disjoint, so it
makes sense to move them to separate files. Beyond reducing the quite
considerable size of ffmpeg_opt.c, this will also allow exposing muxer
internals (currently private to ffmpeg_mux.c) to the initialization
code, thus removing the awkward separation currently in place.
This simplifies the code as there is no other place the error buffer
is needed, so the av_err2str helper macro can be used.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
av_err2str which is a wrapper for av_strerror already calls
strerror_r if available and if not has a fallback for the other
error codes that would be handled by that, so manually calling
strerror again if it fails is not necessary.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
av_err2str which is a wrapper for av_strerror already calls
strerror_r if available and if not has a fallback for the other
error codes that would be handled by that, so manually calling
strerror again if it fails is not necessary.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Currently it would essentially change the find_stream_info setting for
the file it was specified for and all following files, which is unusual
and somewhat unexpected behaviour for a per-file option and not even
documented to behave like this.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
It has been deprecated in favor of the aresample filter for almost 10
years.
Another thing this option can do is drop audio timestamps and have them
generated by the encoding code or the muxer, but
- for encoding, this can already be done with the setpts filter
- for muxing this should almost never be done as timestamp generation by
the muxer is deprecated, but people who really want to do this can use
the setts bitstream filter
av_display_rotation_get will return NAN when the display matrix is invalid,
which would end up printing NAN as an integer in the rotation field. This
is poor for multiple reasons:
* Users of ffprobe have no way of discerning "valid but ugly rotation from
display matrix" from "invalid display matrix".
* It can have unintended consequences on some platforms, such as Linux x86_64,
where NAN is equal to INT64_MIN, which, for example, when printed as JSON,
which uses floating point for all numbers, can end up as invalid JSON or wit
a number that cannot be reserialized as an integer at all.
Since NAN is av_display_rotation_get's error case, just print 0 (no rotation)
when that happens.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
There are two issues here. Firstly, the floating-point comparison
is always true. Seconly, the code depends on the default value of
min_hard_comp implicitly, which can be dangerous.
Partially fixes ticket 9859.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
