Some files have the first audio much later into the video (for whatever
reasons). Instead of appending large amounts of silence to the audio
buffer (and refusing to sync if the audio to append is "too large"),
just wait until enough video has played.
It probably happens relatively often that the first packet (or even the
first N packets) of a stream will fail to decode, but decoding will
eventually succeed at a later point. Before commit 261506e3, this was
handled by an explicit retry loop (although this was also for other
purposes), but with then was changed to abort on the first error. This
makes it impossible to decode some audio streams.
Change this so that errors are ignored for the first 50 packets, which
should make it equivalent to the old code.
If you for example use --audio-file, disable the external track, seek,
and enable the external track again, the playback position of the
external file was off, and you would get major A/V desync. This was
actually supposed to work, but broke at some time ago (probably commit
2b87415f). It didn't work, because it attempted to seek the stream if it
was already selected, which was always true due to
reselect_demux_streams() being called before that.
Fix by putting the initial selection and the seek together.
This commit makes audio decoding non-blocking. If e.g. the network is
too slow the playloop will just go to sleep, instead of blocking until
enough data is available.
For video, this was already done with commit 7083f88c. For audio, it's
unfortunately much more complicated, because the audio decoder was used
in a blocking manner. Large changes are required to get around this.
The whole playback restart mechanism must be turned into a statemachine,
especially since it has close interactions with video restart. Lots of
video code is thus also changed.
(For the record, I don't think switching this code to threads would
make this conceptually easier: the code would still have to deal with
external input while blocked, so these in-between states do get visible
[and thus need to be handled] anyway. On the other hand, it certainly
should be possible to modularize this code a bit better.)
This will probably cause a bunch of regressions.
Don't return an EOF code if there's still buffered data.
Also, don't call demux_stream_eof() in the playloop. There's probably
nothing wrong with it, but it's cleaner not to use it.
Also give AD_EOF its own value, so that a decoding error doesn't drain
audio by causing an EOF condition.
There was confusion about what should go into audio pts calculation and
what not (mainly due to the audio push thread). This has been fixed by
using the playing - not written - audio pts (which properly takes into
account the ao's buffer), and incrementing the samples count only by the
amount of samples actually taken from the buffer (unfortunately this
now forces us to keep the lock too long for my taste).
It's unlikely that files with multiple audio tracks and with replaygain
actually happen, but this change might help avoid minor corner cases
with later changes.
Basically, this allows gapless playback with similar files (including
the ordered chapter case), while still being robust in general.
The implementation is quite simplistic on purpose, in order to avoid
all the weird corner cases that can occur when creating the filter
chain. The consequence is that it might do not-gapless playback in
more cases when needed, but if that bothers you, you still can use
the normal gapless mode.
Just using "--gapless-audio" or "--gapless-audio=yes" selects the old
mode.
This code handles buggy AOs (even if all AOs are bug-free, it's good for
robustness). Move handling of it to the AO feed thread. Now this check
doesn't require magic numbers and does exactly what's it supposed to do.
This played the file at a wrong sample rate if the rate was out of
certain bounds.
A comment says this was for the sake of libaf/af_resample.c. This
resampler has been long removed. Our current resampler
(libav/swresample) checks supported sample rates on reconfiguration, and
will error out if a sample rate is not supported. And I think that is
the correct behavior.
This obviously doesn't work. It wasn't much of a problem in the past
because most passthrough formats use 2 channels, which is also the
default for downmix.
This is probably "safer". Without it, we will play 1 sample, because the
logic was written in a way to decode 1 sample if audio is paused. 1
sample usually will initialize the audio PTS, but not play any real
audio. Also see previous commit.
In ancient times, this actually used 1 byte (instead of 1 sample), so
clearly no sample was written, unless the audio was 8-bit mono.
Remove the ao_buffer_playable_samples field. This contained the number
of samples that fill_audio_out_buffers() wanted to write to the AO (i.e.
this data was supposed to be played at some point), but ao_play()
rejected it due to partial fill.
This could happen with many AOs, notably those which align all written
data to an internal period size (often called "outburst" in the AO
code), and the accepted number of samples is rounded down to period
boundaries. The left-over samples at the end were still kept in
mpctx->ao_buffer, and had to be played later.
The reason ao_buffer_playable_samples had to exist was to make sure that
at EOF, the correct number of left-over samples was played (and not
possibly other data in the buffer that had to be sliced off due to
endpts in fill_audio_out_buffers()). (You'd think you could just slice
the entire buffer, but I suspect this wasn't done because the end time
could actually change due to A/V sync changes. Maybe that was the reason
it's so complicated.)
Some commits ago, ao.c gained internal buffering, and ao_play() will
never return partial writes - as long as you don't try to write more
samples than ao_get_space() reports. This is always the case. The only
exception is filling the audio buffers while paused. In this case, we
decode and play only 1 sample in order to initialize decoding (e.g. on
seeking). Actually playing this 1 sample is in fact a bug, but even of
the AO doesn't have period size alignment, you won't notice it. In
summary, this means we can safely remove the code.
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
This field will be moved out of the ao struct. The encoding code was
basically using an invalid way of accessing this field.
Since the AO will be moved into its own thread too and will do its own
buffering, the AO and the playback core might not even agree which
sample a PTS timestamp belongs to. Add some extrapolation code to handle
this case.
This sometimes happened when changing playback speed (= reinitializing
audio) after seeking of playback start. The assertion in audio.c:441 was
triggered, because buffer_playable_samples wasn't reset correctly when
the audio buffer was cleared or shortened. The assertion is correct and
should hold up any time.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
wm4
Rudolf Polzer
Alessandro Ghedini
Martin Herkt