This seems more robust than relying on the audio status to actually be
playing. For files where there is no audio or the audio start is
delayed, this guards against that but it allows the subtraction to
always occur otherwise on normal files with audio.
Ended up being a bad idea. As a property, this inherently has more
functionality and the tradeoff of being able to do --ao-volume wasn't
worth it.
This reverts commit 58ed620c06.
This came up in #13571. playing_audio_pts does not include mpctx->delay
contray to what that implies. The function is meant to only offset the
written audio pts by whatever the internal AO buffer may be.
mpctx->delay is a combination from both the audio and video code, so it
should not be used here. What is wanted is purely the audio pts.
b74c09efbf, a very controversial commit to
say the least, was what introduced this comment, so removing is probably
OK.
When calculating the audio pts, mpv multiplies the ao delay by the
current audio speed and subtracts it from the written audio pts. This
doesn't really make sense though. mpctx->video_pts is never affected by
the playback speed, and this leads to weird behavior like the audio-pts
property changing values while paused merely because the playback speed
changes. Remove the multiplication and simply subtract the delay by a
factor of 1 instead. When updating the av_diff in player/video, this
does actually need to take in account the audio speed so we do the
calculation there.
This adds volume-gain, volume-gain-max, volume-gain-min options, which
can be used to control audio volume and target dynamic range in decibels.
The gain is applied on top of the existing volume setting.
dac977193c changed adding delay to only
when the video is playing but this isn't correct. The video frames
adjust themselves based on the audio, so if we still have audio
processing while the video itself happens to be in some non-playing
state (such as STATUS_READY), the delay still needs to be taken into
account. The correct thing to check is to make sure that it is not
STATUS_EOF. STATUS_EOF covers the case where we have still image, and of
course no video at all is STATUS_EOF. So having a massive bogus delay
value is still avoided.
The point of the mpctx->delay field is for calculating a/v sync and
adjusting the frame timings appropriately. However, the frame_time was
always subtracted from mpctx->delay regardless of the audio status. This
meant that for a video with no audio, every single frame had a
subtraction with nothing ever added to it meaning that you get massive
negative delay times. For weird videos where the audio starts way later,
the massive delay leads to the VO sleeping for basically about as long
as the video was previously playing without audio. This results in
nothing being rendered during that brief period of time and just overall
badness. When using display-sync, it happens to work since the video
doesn't adjust itself based on audio and it renders anyway.
The fix is to simply not touch mpctx->delay in player/video.c unless
there's actually audio playing. This is what the rest of the code
already does aside from setting it to 0 on resets or EOFs. Move the
calculation into adjust_sync after the audio status check. It works
exactly the same as before except that we don't constantly subtract
bogus values when there's no audio playing. The reverse situation in
player/audio.c also has the same issue. For something that is only
audio, mpctx->delay is always added to but nothing will ever subtract
from it. It's not really clear if this particular version could ever
cause a real bug, but logically it needs to be guarded in the same way.
The field here should only be updated if the video is actually playing.
Fixes#12025.
There's an edge cause with gapless audio and pausing. Since, gapless
audio works by sending an EOF immediately, it's possible to pause on the
next file before audio actually finishes playing and thus the sound gets
cut off. The fix is to simply just always do an ao_drain if the ao is
about to set a pause on EOF and we still have audio playing.
Fixes#8898.
So far there was no way to sync video to display and have audio sync to
video without changes in pitch.
With this option the audio does not get resampled (pitch change) and
instead the corrected audio speed is applied to audio filters.
We have previously had a problem where pull AOs (such as pipewire)
would not reinitialize on a format change when going between two
audio-only files. In such a situation, playback would stop after the
first file.
We initially attempted to fix this by explicitly draining on EOF, which
solves that problem but introduces a blocking step where we don't
actually want one, breaking gapless audio, and causing dropped frames
at the end of playback for files with video in them too.
So, let's undo these changes and do something better in the next
commit.
52aed495cb ("audio: drain ao on EOF") introduced logic to drain an AO
when EOF of the input has been reached.
When no AO however is present this leads to a NULL-dereference.
Fixes#10556
This gives pull-based AOs the chance to play all queued audio.
Also it will make sure that the audio has finished playing so we can
reinitialize the AO if format changes are necessary.
Fixes#10018Fixes#9835Fixes#8904
The video sync logic for mpv lies completely within its core at
essentially the highest layer of abstraction. The problem with this is
that it is impossible for VOs to know what video sync mode mpv is
currently using since it has no access to the opts. Because different
video sync modes completely changes how mpv's render loop operates, it's
reasonable that a VO may want to change how it renders based on the
current mode (see the next commit for an example).
Let's just move the video sync option to mp_vo_opts. MPContext, of
course, can still access the value of the option so it only requires
minor changes in player/video.c. Additionally, move the VS_IS_DISP
define from to player/core.h to common/common.h. All VOs already have
access to common/common.h, and there's no need for them to gain access
to everything that's in player/core.h.
reinit_audio_filters_and_output function will free mpctx->ao_chain
when there is a failure during audio initialization. So modify it
to return -1 in case of init failure. The return value is checked
to avoid use after free. Reported by Address Sanitizer when manually
specifying --ao which triggers "Failed to initialize audio driver" error.
Switching to a new file while keeping the AO didn't update the volume.
While there's an explicit audio_update_volume() call in
reinit_audio_chain_src(), it doesn't work, because at that point
ao_chain->ao is still NULL, and it does nothing. That's pretty weird and
might cause other problems (what happens if you try to mute while the AO
is "floating"?). Regarding gapless, trying to use the AO gain for
replaygain is also gross nonsense, because the new replaygain computed
gain would affect audio from the previous file. It looks like replaygain
should be in an af_volume filter maybe. On the other hand, I enjoy
setting ridiculous replaygain-preamp values and compensating with a low
volume setting, which would not work well if both gains were applied to
the audio independently.
For now, just add the missing call. This is orthogonal to fixing
replaygain "properly".
This reverts commit 3239e41277.
I'm fairly sure this is wrong, and my next commit should fix it
properly. I'm not really sure, though. Normally, the AO is set again
by reinit_audio_filters_and_output() after the new audio chain has
decoded a frame and knows the new format. The reason replaygain (and
apparently the thing the reverted commit tried to fix) didn't work is
because they work asynchronously to the audio played by the AO (i.e.
buggy and hard to fix).
Seems to be a slight corner case with the audio API rewrite. When
switching from one file to another one, the volume of the ao would never
be set because the audio chain's ao wasn't set. This caused a bug with
the reset-set-on-file option. The volume/property would be correctly set
internally, but the gain was not actually set when the file switched.
Fixes#8287.
There was a simple oversight that meant audio outputs were
uninitialized during an encoding, which is not allowed, the encoding
would stop with numerous errors.
I added a single line to prevent the call of uninit_audio_out in
reinit_audio_chain if the encoder was active and this appears to have
fixed the problem without breaking anything else.
Fixes#8568
I don't see the point of this. Not doing it may defer an error to later.
That's OK? For now, it seems better to reduce the encoding internal API.
If someone can demonstrate that this is needed, I might reimplement it
in a different way.
Since this is a messy and fragile mechanism, I want it logged (even if
it's somewhat in conflict with the verbose logging policy). On the other
hand, it's unconditionally logged on every playloop iteration. So add
some nonsense to log it only on progress.
Just for the redundant message. The function which is called here,
ao_drain(), does not care in which state it is called, and already
handled this gracefully.
This replaces the two buffers (ao_chain.ao_buffer in the core, and
buffer_state.buffers in the AO) with a single queue. Instead of having a
byte based buffer, the queue is simply a list of audio frames, as output
by the decoder. This should make dataflow simpler and reduce copying.
It also attempts to simplify fill_audio_out_buffers(), the function I
always hated most, because it's full of subtle and buggy logic.
Unfortunately, I got assaulted by corner cases, dumb features (attempt
at seamless looping, really?), and other crap, so it got pretty
complicated again. fill_audio_out_buffers() is still full of subtle and
buggy logic. Maybe it got worse. On the other hand, maybe there really
is some progress. Who knows.
Originally, the data flow parts was meant to be in f_output_chain, but
due to tricky interactions with the playloop code, it's now in the dummy
filter in audio.c.
At least this improves the way the audio PTS is passed to the encoder in
encoding mode. Now it attempts to pass frames directly, along with the
pts, which should minimize timestamp problems. But to be honest, encoder
mode is one big kludge that shouldn't exist in this way.
This commit should be considered pre-alpha code. There are lots of bugs
still hiding.
This affects "pull" AOs only: ao_alsa, ao_pulse, ao_openal, ao_pcm,
ao_lavc. There are changes to the other AOs too, but that's only about
renaming ao_driver.resume to ao_driver.start.
ao_openal is broken because I didn't manage to fix it, so it exits with
an error message. If you want it, why don't _you_ put effort into it? I
see no reason to waste my own precious lifetime over this (I realize the
irony).
ao_alsa loses the poll() mechanism, but it was mostly broken and didn't
really do what it was supposed to. There doesn't seem to be anything in
the ALSA API to watch the playback status without polling (unless you
want to use raw UNIX signals).
No idea if ao_pulse is correct, or whether it's subtly broken now. There
is no documentation, so I can't tell what is correct, without reverse
engineering the whole project. I recommend using ALSA.
This was supposed to be just a simple fix, but somehow it expanded scope
like a train wreck. Very high chance of regressions, but probably only
for the AOs listed above. The rest you can figure out from reading the
diff.
This mode drops or repeats audio data to adapt to video speed, instead
of resampling it or such. It was added to deal with SPDIF. The
implementation was part of fill_audio_out_buffers() - the entire
function is something whose complexity exploded in my face, and which I
want to clean up, and this is hopefully a first step.
Put it in a filter, and mess with the shitty glue code. It's all sort of
roundabout and illogical, but that can be rectified later. The important
part is that it works much like the resample or scaletempo filters.
For PCM audio, this does not work on samples anymore. This makes it much
worse. But for PCM you can use saner mechanisms that sound better. Also,
something about PTS tracking is wrong. But not wasting more time on
this.
I may (optionally) move decoding to a separate thread in a future
change. It's a bit attractive to move the entire decoder wrapper to
there, so if the demuxer has a new packet, it doesn't have to wake up
the main thread, and can directly wake up the decoder. (Although that's
bullshit, since there's a queue in between, and libavcodec's
multi-threaded decoding plays cross-threads ping pong with packets
anyway. On the other hand, the main thread would still have to shuffle
the packets around, so whatever, just seems like better design.)
As preparation, there shouldn't be any mutable state exposed by the
wrapper. But there's still a large number of corner-caseish crap, so
just use setters/getters for them. This recorder thing will inherently
not work, so it'll have to be disabled if threads are used.
This is a bit painful, but probably still the right thing. Like
speculatively pulling teeth.
Try to deal with various corner cases. But when I fix one thing, another
thing breaks. (And it's 50/50 whether I find the breakage immediately or
a few months later.) So results may vary.
The default for--hr-seek is changed to "default" (not creative enough to
find a better name). In this mode, audio seeking is exact if there is no
video, or if the video has only a single frame. This change is actually
pretty dumb, since audio frames are usually small enough that exact
seeking does not really add much. But it gets rid of some weird special
cases.
Internally, the most important change is that is_coverart and is_sparse
handling is merged. is_sparse was originally just a special case for
weird .ts streams that have the corresponding low-level flag set. The
idea is that they're pretty similar anyway, so this would reduce the
number of corner cases. But I'm not sure if this doesn't break the
original intended use case for it (I don't have a sample anyway).
This changes last-frame handling, and respects the duration of the last
frame only if audio is disabled. This is mostly "coincidental" due to
the need to make seeking past EOF trigger player exit, and is caused by
setting STATUS_EOF early. On the other hand, this might have been this
way before (see removed chunk close to it).
A previous commit moved the underrun reporting to report_underruns(),
and called it from get_space(). One reason was that I worried about
printing a log message from a "realtime" callback, so I tried to move it
out of the way. (Though there's little justification other than a bad
feeling. While an older version of the pull code tried to avoid any
mutexes at all in the callback to accommodate "requirements" from APIs
like jackaudio, we gave up on that. Nobody has complained yet.)
Simplify this and move underrun reporting back to the callback. But
instead of printing the message from there, move the message into the
playloop. Change the message slightly, because ao->log is inaccessible,
and without the log prefix (e.g. "[ao/alsa]"), some context is missing.
This is preparation to get rid of the option-to-property bridge
(mp_on_set_option). This is a pretty insane thing that redirects
accesses to options to properties. It was needed in the ever ongoing
transition from something to... something else.
A good example for the need of this bridge is applying profiles at
runtime. This obviously goes through the config parser, but should also
make all changes effective, for which traditionally the property layer
is used.
There isn't much left that needs this bridge. This commit changes a
bunch of options (which also have a property implementation) to use
option change notifications instead. Many of the properties are still
left, but perform unrelated functions like OSD formatting.
This should be mostly compatible. There may be some subtle behavior
changes. For example, "hwdec" and "record-file" do not check for changes
anymore before applying them, so writing the current value to them
suddenly does something, while it was ignored before.
DVB changes untested, but should work.
The --cache-pause feature (enabled by default) will pause playback for a
while if network runs out of data. If this is not done, then playback
will go on frame-wise (as packets are slowly read from the network and
then instantly decoded and displayed). This feature is actually useless,
as you won't get nice playback no matter what if network is too slow,
but I guess I still prefer this behavior for some reason.
This commit changes this behavior from using the demuxer cache state
only, to trying to use underrun information from the AO/VO. This means
if you have a very large audio buffer, then cache-pausing will trigger
once that buffer is depleted, which will be some time _after_ the
demuxer cache has run out.
This requires explicit support from the AO. Otherwise, the behavior
should be mostly the same as before this commit.
This does not care about the AO buffer. In theory, the AO may underrun,
then the player will write some data to the AO buffer, then the AO will
recover and play this bit of data, then the player will probably trigger
the cache-pause behavior. The probability of this happening should be
pretty low, so I will hold off fixing this until the next refactor of
the AO chain (if ever).
The VO underflow detection was devised and tested in 5 minutes, and may
not be correct. At least I'm fairly sure that the combination of all the
factors should make incorrect behavior relatively unlikely, but problems
are possible.
Also, the demux_reader_state.underrun field may be inaccurate. It's only
the present state at the time demux_get_reader_state() was called, and
may exclude past underruns. In theory, this could cause "close" cases to
be missed. Then you might get an audio underrun without cache-pausing
acting on it. If the stars align, this could happen multiple times in
the row, effectively making this feature not work.
The most user-visible consequence of this change is that the user
will now see an AO underrun warning every time the cache runs out.
Maybe this cache-pause feature should just be removed...
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.)
In this case, gapless will most likely not work. It will result in (very
slight) desync, or (more commonly with small buffer sizes), in an
underflow.
I think it would be legitimate to disable gapless at end of playback
completely if video is enabled at all. But this would need an exception
for cover art mode, so I guess the current solution is OK as well.
Dudemanguy
Leonardo Boss
nanahi
Thomas Weißschuh
der richter
sfan5
Christoph Heinrich
Philip Langdale
Aman Karmani
Shreesh Adiga
Your Name
Tom Wilson
wm4