This was added in 380db56928 as a
temporary crutch that is not needed anymore. The only case where this
code can be triggered is the very first frame, for which InputStream.pts
is always equal to 0.
Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.
New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.
Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.
If input packets have timestamps, they will be propagated to output
frames by the decoder, so at best this block does not do anything.
There can also be an arbitrary amount of delay between packets sent to
the decoder and decoded frames (e.g. due to decoder's intrinsic delay or
frame threading), so deriving any timestamps from packet properties is
wrong.
ts_discontinuity_detect() is applied right after demuxing, while
InputStream.pts is a post-decoding timestamp, which may be delayed with
respect to demuxing by an arbitrary amount (e.g. depending on the thread
count when frame threading is used).
When an encoder exports sum-of-squared-differences information in
encoded packets, print_report() will print PSNR information in the
status line. However,
* the code computing PSNR assumes 8bit 420 video and prints incorrect
values otherwise; there are no issues on trac about this
* only a few encoders (namely aom, vpx, mpegvideo, snow) export this
information; other often-used encoders such as libx26[45] do not
export this, even though they could
This suggests that this feature is not useful and it is better to remove
it rather than spend effort on fixing it.
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.
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.
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.
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.
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
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.
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>
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>
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>
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.
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>
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.
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.
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.
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.
Discontinuity detection/correction is left in the main thread, as it is
entangled with InputStream.next_dts and related variables, which may be
set by decoding code.
Fixes races e.g. in fate-ffmpeg-streamloop after
aae9de0cb2.
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.
This avoids a potential race with the demuxer adding new streams. It is
also more efficient, since we no longer do inter-thread transfers of
packets that will be just discarded.
This undocumented feature runtime-enables dumping input packets. I can
think of no reasonable real-world use case that cannot also be
accomplished in a different way. Keeping this functionality would
interfere with the following commit moving it to the input thread (then
setting the variable would require locking or atomics, which would be
unnecessarily complicated for a feature that probably nobody uses).
