AOs can now call ao_underrun_event() (in any context) if an underrun has
happened. It will print a message.
This will be used in the following commits. But for now, audio.c only
clears the underrun bit, so that subsequent underruns still print the
warning message.
Since the underrun flag will be used in fragile ways by the playback
state machine, there is the "reports_underruns" field that signals
strong support for underrun reporting. (Otherwise, underrun events will
not be used by it.)
This commit tries to prepare for better underrun reporting. The goal is
to report underruns relatively immediately. Until now, this happened
only when play() was called. Change this, and abuse that get_delay() is
called "relatively often" - this reports the underrun immediately in
practice.
Background:
In commit 81e51a15f7 (and also e38b0b245e), we were quite confused
about ALSA underrun handling. The commit message showed uncertainty how
case 3 happened, but it's blindingly obvious and simple.
Actually reading the code shows that ALSA does not have a concept of a
"final chunk" (or we don't use it). It's obvious we never pass the
AOPLAY_FINAL_CHUNK flag along to the ALSA API in any way. The only thing
we do is simply writing a partial fragment. Of course this will cause an
underrun. Doing a partial write saves us the trouble to pad the last
frame with silence, or so.
The main reason why the underrun message was avoided was that play() was
never called with a non-0 sample count again (except if reset() was
called before that). That was OK, at least the goal of avoiding the
unwanted message was reached. (And the original "bogus" message at end
of playback was perfectly correct, as far as ALSA goes.)
If network stalls, play() will called again only once new data is
available. Obviously, this could take a long time, thus it's too late.
It turns out that case 2) mentioned in the previous commit happened
quite often when playback ended normally.
There is probably a legitimate underrun with normal buffer sizes (100
ms, 4 fragments, gapless audio in "weak" mode). This is a result of the
player waiting for video to end, and/or the time needed to kill the
video window. The former case means that it depends on your test case
whether it happens (a file where video ends slightly before audio is
less likely to trigger it).
This in turn is due to how gapless playback works. Achieving not having
a "gap" requires queuing the audio of the next file without playing a
partial chunk (as AOPLAY_FINAL_CHUNK would do). The partial chunk is
then played as part of the first chunk played from the next file. But if
it detects "later" that there is no next file, it still needs to get rid
of the last fragment with AOPLAY_FINAL_CHUNK. At this point it's too
late, and an underrun may have actually happened. The way the player
uninits and reinits the entire playback engine for the next file in a
"serial" manner means it cannot know in advance whether this works.
This is the reason why the idiot who added the underrun exception for
the last chunk in play() was wrong (I wrote that btw., before you accuse
me of being rude). Yes, it's a real underrun, and you could probably
hear it.
This XRUN (aka underrun) message was printed in the following
situations:
1) legitimate underrun during playback
2) legitimate underrun when playing final chunk
3) bogus underrun when playing final chunk
The old underrun case (in play()) happens in cases 1) and 2) as well,
but 3) did not happen. It appears 3) is indeed something that happens,
although it's not known for sure. It's still pretty annoying, so remove
the new XRUN message.
When testing, care should be taken to play with buffer sizes, video
versus no video, and gapless enabled/disabled. Also, suspending the
player with Ctrl+Z in the terminal (SIGSTOP) and then resuming is a good
way to trigger a "normal" underrun.
This wasn't used at all in my tests, because it simply passed the
frame directly to libswsresample. (And, by the way, will always do
that, because s64 is so obscure literally NOTHING uses it except
a sample specifically created to test this code. Screw FFmpeg.)
This can be due to unsupported sample formats (see previous commits),
minor allocation failures, and similar things. For identifying the exact
cause it's buried too deep in abstractions. But most time it doesn't
happen anyway, since it's extremely rare that new audio formats are
added.
What an idiotic format. It makes no sense, and should have been
converted to S32 in the demuxer, rather than plague everyone with
another extremely obscure nonsense format. Why doesn't ffmpeg add S24
instead? That's an actually useful format.
May cause compilation failure with old FFmpeg or Libav libs, but I don't
care.
ioctl(..., SNDCTL_DSP_CHANNELS, &nchannels) for not supported
nchannels does not return an error and instead set nchannels to
the default value.
Instead of failing with no audio, fallback to stereo.
This flag makes mpv continue using the PulseAudio driver even if the
sink is suspended.
This can be useful if JACK is running with PulseAudio in bridge mode and
the sink-input assigned to mpv is the one JACK controls, thus being
suspended.
By forcing mpv to still use PulseAudio in this case, the user can now
adjust the sink to an unsuspended one.
This filter wasn't referenced anywhere and thus was dead code. It should
have been in the audio filter list in user_filters.c. This was intended
as compatibility wrapper (to avoid breaking old command lines and config
files), and has no real use. Apparently I forgot to add it to the filter
list (did I even test this shit?), and so it was rotting around for 1.5
years doing nothing (just like myself).
Note that users can just use the libavfilter provided filter to force
resampling, just that it has a different name and different options.
There's also af_format to force inserting auto conversion through the
internal f_swsresample filter.
This may call memmove() with size==0 and a NULL data pointer. In
addition to this being UB with memmove(), I think it's UB to do
arithmetic on a NULL pointer too. Of course, this doesn't matter in
practice at all, and is just stupidity to torture programmers.
Fixes stupid messages with a opus/mkv test file that had an absurdly
huge codec delay.
This file fully skips several frames at the start. ad_lavc.c trimmed
these frames to 0 samples and returned them. The next layer
(f_decoder_wrapper.c) saw discontinuous PTS values, because the PTS
values increased by a frame, but amounted to 0 audio samples. This was
harmless, but logged PTS discontinuity errors.
See manpage additions. This is a huge hack. You can bet there are shit
tons of bugs. It's literally forcing square pegs into round holes.
Hopefully, the manpage wall of text makes it clear enough that the whole
shit can easily crash and burn. (Although it shouldn't literally crash.
That would be a bug. It possibly _could_ start a fire by entering some
sort of endless loop, not a literal one, just something where it tries
to do work without making progress.)
(Some obvious bugs I simply ignored for this initial version, but
there's a number of potential bugs I can't even imagine. Normal playback
should remain completely unaffected, though.)
How this works is also described in the manpage. Basically, we demux in
reverse, then we decode in reverse, then we render in reverse.
The decoding part is the simplest: just reorder the decoder output. This
weirdly integrates with the timeline/ordered chapter code, which also
has special requirements on feeding the packets to the decoder in a
non-straightforward way (it doesn't conflict, although a bugmessmass
breaks correct slicing of segments, so EDL/ordered chapter playback is
broken in backward direction).
Backward demuxing is pretty involved. In theory, it could be much
easier: simply iterating the usual demuxer output backward. But this
just doesn't fit into our code, so there's a cthulhu nightmare of shit.
To be specific, each stream (audio, video) is reversed separately. At
least this means we can do backward playback within cached content (for
example, you could play backwards in a live stream; on that note, it
disables prefetching, which would lead to losing new live video, but
this could be avoided).
The fuckmess also meant that I didn't bother trying to support
subtitles. Subtitles are a problem because they're "sparse" streams.
They need to be "passively" demuxed: you don't try to read a subtitle
packet, you demux audio and video, and then look whether there was a
subtitle packet. This means to get subtitles for a time range, you need
to know that you demuxed video and audio over this range, which becomes
pretty messy when you demux audio and video backwards separately.
Backward display is the most weird (and potentially buggy) part. To
avoid that we need to touch a LOT of timing code, we negate all
timestamps. The basic idea is that due to the navigation, all
comparisons and subtractions of timestamps keep working, and you don't
need to touch every single of them to "reverse" them.
E.g.:
bool before = pts_a < pts_b;
would need to be:
bool before = forward
? pts_a < pts_b
: pts_a > pts_b;
or:
bool before = pts_a * dir < pts_b * dir;
or if you, as it's implemented now, just do this after decoding:
pts_a *= dir;
pts_b *= dir;
and then in the normal timing/renderer code:
bool before = pts_a < pts_b;
Consequently, we don't need many changes in the latter code. But some
assumptions inhererently true for forward playback may have been broken
anyway. What is mainly needed is fixing places where values are passed
between positive and negative "domains". For example, seeking and
timestamp user display always uses positive timestamps. The main mess is
that it's not obvious which domain a given variable should or does use.
Well, in my tests with a single file, it suddenly started to work when I
did this. I'm honestly surprised that it did, and that I didn't have to
change a single line in the timing code past decoder (just something
minor to make external/cached text subtitles display). I committed it
immediately while avoiding thinking about it. But there really likely
are subtle problems of all sorts.
As far as I'm aware, gstreamer also supports backward playback. When I
looked at this years ago, I couldn't find a way to actually try this,
and I didn't revisit it now. Back then I also read talk slides from the
person who implemented it, and I'm not sure if and which ideas I might
have taken from it. It's possible that the timestamp reversal is
inspired by it, but I didn't check. (I think it claimed that it could
avoid large changes by changing a sign?)
VapourSynth has some sort of reverse function, which provides a backward
view on a video. The function itself is trivial to implement, as
VapourSynth aims to provide random access to video by frame numbers (so
you just request decreasing frame numbers). From what I remember, it
wasn't exactly fluid, but it worked. It's implemented by creating an
index, and seeking to the target on demand, and a bunch of caching. mpv
could use it, but it would either require using VapourSynth as demuxer
and decoder for everything, or replacing the current file every time
something is supposed to be played backwards.
FFmpeg's libavfilter has reversal filters for audio and video. These
require buffering the entire media data of the file, and don't really
fit into mpv's architecture. It could be used by playing a libavfilter
graph that also demuxes, but that's like VapourSynth but worse.
Apparently, for bit streaming DTS-HD MA is specified to be handled as an
eight channel (7.1) bit stream, while DTS-HD HRA is specified to be
handled as a stereo bit stream.
Define a variable for this, and utilize it to set the correct values
for both the DTS-HD bit streaming rate, as well as the channel count
for the SPDIF encoder.
Fixes#6148
According to ALSA doxy, EPIPE is a synonym to SND_PCM_STATE_XRUN,
and that is a state that we should attempt to automatically recover
from. In case recovery fails, log an error and return zero.
A warning message will still be output for each XRUN since those
are not something we should generally be receiving.
This has been way too long coming, and for me to notice that a
whole lot of ao_alsa functions do an early return if the AO is
paused.
For the STATE_SETUP case, I had this reproduced once, and never
since. Still, seems like we can start calling this function before
the ALSA device has been fully initialized so we might as well
early exit in that case.
ao->device_buffer will only affect the enqueue size if the latter
is not specified. In other word, its intended purpose will solely
be setting/guarding the soft buffer size.
This guarantees that the soft buffer size will be consistent no
matter a specific enqueue size is set or not. (In the past it
would drop to the default of the generic audio-buffer option.)
opensles-frames-per-buffer has been renamed to opensles-frames-per
-enqueue, as it was never purposed to set the soft buffer size. It
will only make sure the size is never smaller than itself, just as
before.
opensles-buffer-size-in-ms is introduced to allow easy tuning of
the relative (i.e. in time) soft buffer size (and enqueue size,
unless the aforementioned option is set). As "device buffer" never
really made sense in this AO, this option OVERRIDES audio-buffer
whenever its value (including the default) is larger than 0.
Setting opensl-buffer-size-in-ms to 1 allows you to equate the soft
buffer size to the absolute enqueue size set with opensl-frames-per
-enqueue conveniently (unless it is less than 1ms).
When both are set to 0, audio-buffer will be the ultimate fallback.
If audio-buffer is also 0, the AO errors out.
Fixes a bug with alsa dmix on Fedora 29. After several minutes,
audio suddenly becomes bad and muted.
Actually, I don't know what causes this. Probably this is a bug in alsa.
In any case, as snd_pcm_status() returns not only 'avail', but also other
fields such as tstamp, htstamp, etc, this could be considered a good
simplification, as only avail is required for this function.
This was always a legacy thing. Remove it by applying an orgy of
mp_get_config_group() calls, and sometimes m_config_cache_alloc() or
mp_read_option_raw().
win32 changes untested.
Until recently, ao_lavc and vo_lavc started encoding whenever the core
happened to send them data. Since audio and video are not initialized at
the same time, and the muxer was not necessarily opened when the first
encoder started to produce data, the resulting packets were put into a
queue. As soon as the muxer was opened, the queue was flushed.
Change this to make the core wait with sending data until all encoders
are initialized. This has the advantage that we don't need to queue up
the packets.
The main change is that we wait with opening the muxer ("writing
headers") until we have data from all streams. This fixes race
conditions at init due to broken assumptions in the old code.
This also changes a lot of other stuff. I found and fixed a few API
violations (often things for which better mechanisms were invented, and
the old ones are not valid anymore). I try to get away from the public
mutex and shared fields in encode_lavc_context. For now it's still
needed for some timestamp-related fields, but most are gone. It also
removes some bad code duplication between audio and video paths.
Print them as a warning.
Note that there may be some cases where it underruns, without being a
bad condition. This could possibly happen e.g. if the last chunk is
written, and then it resumes playback some time after that. Eventually I
want to add more code to avoid such spurious warnings.
There is a dedicated thread for feeding audio to the ALSA API from a
buffer with a larger size. There is little reason to have such a large
device buffer.
