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.
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
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.
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.
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().
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
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.
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>
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.
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.
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.
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.
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
update_video_stats() currently uses OutputStream.data_size to print the
total size of the encoded stream so far and the average bitrate.
However, that field is updated in the muxer thread, right before the
packet is sent to the muxer. Not only is this racy, but the numbers may
not match even if muxing was in the main thread due to bitstream
filters, filesize limiting, etc.
Introduce a new counter, data_size_enc, for total size of the packets
received from the encoder and use that in update_video_stats(). Rename
data_size to data_size_mux to indicate its semantics more clearly.
No synchronization is needed for data_size_mux, because it is only read
in the main thread in print_final_stats(), which runs after the muxer
threads are terminated.
It is either equal to OutputStream.enc_ctx->codec, or NULL when enc_ctx
is NULL. Replace the use of enc with enc_ctx->codec, or the equivalent
enc_ctx->codec_* fields where more convenient.
It races with the demuxing thread. Instead, send the information along
with the demuxed packets.
Ideally, the code should stop using the stream-internal parsing
completely, but that requires considerably more effort.
Fixes races, e.g. in:
- fate-h264-brokensps-2580
- fate-h264-extradata-reload
- fate-iv8-demux
- fate-m4v-cfr
- fate-m4v
Use it instead of AVStream.codecpar in the main thread. While
AVStream.codecpar is documented to only be updated when the stream is
added or avformat_find_stream_info(), it is actually updated during
demuxing. Accessing it from a different thread then constitutes a race.
Ideally, some mechanism should eventually be provided for signalling
parameter updates to the user. Then the demuxing thread could pick up
the changes and propagate them to the decoder.
This will allow to move normal offset handling to demuxer thread, since
discontinuities currently have to be processed in the main thread, as
the code uses some decoder-produced values.
InputFile.ts_offset can change during transcoding, due to discontinuity
correction. This should not affect the streamcopy starting timestamp.
Cf. bf2590aed3
-stream_loop is currently handled by destroying the demuxer thread,
seeking, then recreating it anew. This is very messy and conflicts with
the future goal of moving each major ffmpeg component into its own
thread.
Handle -stream_loop directly in the demuxer thread. Looping requires the
demuxer to know the duration of the file, which takes into account the
duration of the last decoded audio frame (if any). Use a thread message
queue to communicate this information from the main thread to the
demuxer thread.
There are currently three possible modes for an output stream:
1) The stream is produced by encoding output from some filtergraph. This
is true when ost->enc_ctx != NULL, or equivalently when
ost->encoding_needed != 0.
2) The stream is produced by copying some input stream's packets. This
is true when ost->enc_ctx == NULL && ost->source_index >= 0.
3) The stream is produced by attaching some file directly. This is true
when ost->enc_ctx == NULL && ost->source_index < 0.
OutputStream.stream_copy is currently used to identify case 2), and
sometimes to confusingly (or even incorrectly) identify case 1). Remove
it, replacing its usage with checking enc_ctx/source_index values.
The -shortest option (which finishes the output file at the time the
shortest stream ends) is currently implemented by faking the -t option
when an output stream ends. This approach is fragile, since it depends
on the frames/packets being processed in a specific order. E.g. there
are currently some situations in which the output file length will
depend unpredictably on unrelated factors like encoder delay. More
importantly, the present work aiming at splitting various ffmpeg
components into different threads will make this approach completely
unworkable, since the frames/packets will arrive in effectively random
order.
This commit introduces a "sync queue", which is essentially a collection
of FIFOs, one per stream. Frames/packets are submitted to these FIFOs
and are then released for further processing (encoding or muxing) when
it is ensured that the frame in question will not cause its stream to
get ahead of the other streams (the logic is similar to libavformat's
interleaving queue).
These sync queues are then used for encoding and/or muxing when the
-shortest option is specified.
A new option – -shortest_buf_duration – controls the maximum number of
queued packets, to avoid runaway memory usage.
This commit changes the results of the following tests:
- copy-shortest[12]: the last audio frame is now gone. This is
correct, since it actually outlasts the last video frame.
- shortest-sub: the video packets following the last subtitle packet are
now gone. This is also correct.
The following commits will add a new buffering stage after bitstream
filters, which should not be taken into account for choosing next
output.
OutputStream.last_mux_dts is also used by the muxing code to make up
missing DTS values - that field is now moved to the muxer-private
MuxStream object.
It is currently called from two places:
- output_packet() in ffmpeg.c, which submits the newly available output
packet to the muxer
- from of_check_init() in ffmpeg_mux.c after the header has been
written, to flush the muxing queue
Some packets will thus be processed by this function twice, so it
requires an extra parameter to indicate the place it is called from and
avoid modifying some state twice.
This is fragile and hard to follow, so split this function into two.
Also rename of_write_packet() to of_submit_packet() to better reflect
its new purpose.
The muxing queue currently lives in OutputStream, which is a very large
struct storing the state for both encoding and muxing. The muxing queue
is only used by the code in ffmpeg_mux, so it makes sense to restrict it
to that file.
This makes the first step towards reducing the scope of OutputStream.
Figure out earlier whether the output stream/file should be bitexact and
store this information in a flag in OutputFile/OutputStream.
Stop accessing the muxer in set_encoder_id(), which will become
forbidden in future commits.
Move the file size checking code to ffmpeg_mux. Use the recently
introduced of_filesize(), making this code consistent with the size
shown by print_report().
Move header_written into it, which is not (and should not be) used by
any code outside of ffmpeg_mux.
In the future this context will contain more muxer-private state that
should not be visible to other code.
This is a per-file input option that adjusts an input's timestamps
with reference to another input, so that emitted packet timestamps
account for the difference between the start times of the two inputs.
Typical use case is to sync two or more live inputs such as from capture
devices. Both the target and reference input source timestamps should be
based on the same clock source.
If either input lacks starting timestamps, then no sync adjustment is made.
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 are always
using 64 bit values for them.
A live stream can easily run for more than a year and the framedup logic breaks
on an overflow.
Signed-off-by: Marton Balint <cus@passwd.hu>
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().
Anton Khirnov
Jan Ekström
Gyan Doshi
Jan Ekström
Andreas Rheinhardt
Marvin Scholz
Marton Balint
James Almer