| Commit message (Collapse) | Author | Age | Files | Lines |
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Broke image previews on Linux and Windows (mingw).
This reverts commit b7645e7842703d52c44feaa85cfb3c490e09c8dc.
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For consistency with 36c415e. In practice the include this encapsulates is already included elsewhere, so no underlying change takes place; <pthread.h> is already included.
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For consistency with test/module.defs.
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Also, mark it as static as it is not used outside of this file.
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Update to mingw-w64 5.0-rc2 and pthreads-win32 2.9.1.
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defined.
As may be the case with newer mingw with pthreads precompiled.
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Also drop the scan pass if there is only one subtitle track and "force"
is not specified. When scanning for foreign audio subtitles without
force, you need at least 2 tracks in order for the 10% selection logic
to apply.
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A suitable subtitle is one that matches the users preferred language.
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When doing point-to-point encoding, subtitles can cause a long
delay in finishing the job when the stop point is reached. This
is due to the sparse nature of subtitles. We may not even see
any additional subtitle till we reach the end of the file.
So when all audio and video streams have reached the end point,
force the termination of all subtitle streams by pushing an
end-of-stream buffer into each subtitles input fifo.
This will cause each subtitle sync worker to wake and return
HB_WORK_DONE.
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A full input queue could cause the search to stall
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Only the names have been changed, to protect the innocent.
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After finding the start position, some data prior to the start from
other streams could leak through causing duplicate timestamps in the
output.
Also, improves alignment of stop times of all streams when a stop time
is set.
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Will change VP9 presets' audio to Opus if/when that lands.
Closes #159.
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presets are
veryfast - deadline=good:cpu-used=5
faster - deadline=good:cpu-used=4
fast - deadline=good:cpu-used=3
medium - deadline=good:cpu-used=2
slow - deadline=good:cpu-used=1
slower - deadline=good:cpu-used=0
veryslow - deadline=best:cpu-used=0
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... when applying presets
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cpu-used=0 is very slow, and the various recommendations are to use
cpu-used=2
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HarfBuzz is now enabled when building libass.
Resolves #162.
Additional libass notes:
- Add yasm dependency for better performance.
- Remove no longer valid configure params.
- Disable new coretext and directwrite font selection backends pending additional testing (coretext did not build properly).
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It was dropping subtitles because the "end of CC" marker buffer can have
the same time as the next valid CC which triggered the subtitle overlap
dropping code.
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Use hb_chapter_enqueue/dequeue
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* sync: correct timestamp discontinuities in sync instead of reader
This patch passes discontinuity information through the pipeline till it
reaches sync.c. The timestamps are passed through the pipeline as read
and unmodified to sync.c (instead of attempting to correct
discontinuities in reader). In sync, when we see a discontinuity,
we know where the next timestamp should be based on the timestamp
and duration of the previous buffer (before the discontinuity). So
we calculate an "SCR" offset based on the timestamp after the
discontinuity and what we calculate it should be.
The old discontinuity handling code was broken due to the following.
The MPEG STD timing model relies heavily on the decoder having an STC
that is phase lock looped to the PCRs in the stream. When decoding a
broadcast stream, the decoder can count on the time measure between PCRs
using the STC to match to a high degree of accuracy.
I.e. STC - lastSTC == PCR - lastPCR. When a discontinuity occurs, the
decoder calculates a new PCR offset = PCR - STC. I.e. the offset is the
new PCR value minus what it would have been if there had been no
discontinuity.
The above does not work without a reliable STC, which we do not have.
We have been attempting to approximate one by avereraging the duration
of received packets and extrapolating an "STC" based on the last PTS and
the average packet duration. But this is highly variable and
unreliable.
* decavcodec: fix data type of next_pts
It needs to be double so that partial ticks are not lost
* deccc608sub: clarify comment
* sync: allow queueing more audio
Audio is small, and there is often a significant amount of audio in the
stream before the first video frame.
* sync: improve handling of damaged streams
When data is missing, the audio decoder was extrapolating timestamps
from the last pts before the error caused by the missing data which
caused sync issues.
Also, missing data can cause the video decoder to output a frame out of
order with the wrong scr sequence. Drop such frames.
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brainfart!
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essentially an off-by-one error. OutputBuffer had to wait for one more
buffer before any output was performed after the queue should have
already been filled to it's minimum levels.
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