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Commit Graph

648 Commits

Author SHA1 Message Date
wm4
d26a0ae111 ao_oss: fix audio device leak on error
Close the audio device if it was already opened, but the rest of
initialization failed.
2014-09-11 02:05:12 +02:00
wm4
5f80e3f91a ao_oss: use poll(), drop --disable-audio-select support
Replace select() usage with poll() (and reduce code duplication).

Also, while we're at it, drop --disable-audio-select, since it has the
wrong name anyway. And I have doubts that this is needed anywhere. If
it is, it should probably fallback to doing the right thing by default,
instead of requiring the user to do it manually. Since nobody has done
that yet, and since this configure option has been part of MPlayer ever
since ao_oss was added, it's probably safe to say it's not needed.

The '#ifdef SNDCTL_DSP_GETOSPACE' was pointless, since it's already used
unconditionally in another place.
2014-09-11 02:03:15 +02:00
wm4
f744aadb77 ao_pulse: dump library version etc.
Might help with debugging.

Unfortunately, there doesn't seem to be a way to get the actual
pulseaudio server version.
2014-09-10 23:14:06 +02:00
wm4
b578abe81b ao_pulse: fix typo in error message
Closes #1076.
2014-09-08 17:19:53 +02:00
wm4
94113e632f audio/out: fix active waiting during pause again
This was fixed in commit 8432eaefa, and commit 39609fc1 of course broke
it again. This was pretty stupid.
2014-09-06 16:25:27 +02:00
wm4
39609fc19a audio/out/push: redo audio waiting
Improve the logic how the audio thread decides how to wait until the AO
is ready for new data. The previous commit makes some of this easier,
although it turned out that it wasn't required, and we still can handle
AOs with bad get_space implementation (although the new code prints an
error message, and it might fail in obscure situations).

The new code is pretty similar to the old one, and the main thing that
changes is that complicated conditions are tweaked. AO waiting is now
used better (mainly instead of max>0, r>0 is used). Whether to wakeup
is reevaluated every time, instead of somehow doing the wrong thing
and compensating for it with a flag.

This fixes the specific situation when the device buffer is full, and
we don't want to buffer more data. In the old code, this wasn't handled
correctly: the AO went to sleep forever, because it prevented proper
wakeup by the AO driver, and as consequence never asked the core for new
data. Commit 4fa3ffeb was a hack-fix against this, and now that we have
a proper solution, this hack is removed as well.

Also make the refill threshold consistent and always use 1/4 of the
buffer. (The threshold is used for situations when an AO doesn't
support proper waiting or chunked processing.)

This commit will probably cause a bunch of regressions again.
2014-09-06 12:59:04 +02:00
wm4
769ac6fb7b audio/out: always round get_space on period size
Round get_space() results in the same way play() rounds the input size.
Some audio APIs do this for various reasons.

This affects only "push" based AOs. Some of these need no change,
because they either do it already right (like ao_openal), or they seem
not to have any such requirements (like ao_pulse).

Needed for the following commit.
2014-09-06 12:59:00 +02:00
wm4
d9941e01cc ao_sndio: fix a comment
Whether this code was written with the correct assumptions in mind, I
don't know.
2014-09-06 12:58:57 +02:00
wm4
6c9ce5bee2 ao_pcm: minor simplification 2014-09-06 12:58:54 +02:00
wm4
4962a1ece3 ao_oss: minor simplification
Equivalent code.
2014-09-06 12:58:48 +02:00
wm4
439a05d8c3 audio/out: remove old things
Remove the unnecessary indirection through ao fields.

Also fix the inverted result of AOCONTROL_HAS_TEMP_VOLUME. Hopefully the
change is equivalent. But actually, it looks like the old code did it
wrong.
2014-09-06 02:30:57 +02:00
wm4
bdf49d137e audio/out: make EOF handling properly event-based
With --gapless-audio=no, changing from one file to the next apparently
made it hang, until the player was woken up by unrelated events like
input. The reason was that the AO doesn't notify the player of EOF
properly. the played was querying ao_eof_reached(), and then just went
to sleep, without anything waking it up.

Make it event-based: the AO wakes up the playloop if the EOF state
changes.

We could have fixed this in a simpler way by synchronously draining the
AO in these cases. But I think proper event handling is preferable.

Fixes: #1069
CC: @mpv-player/stable (perhaps)
2014-09-05 23:45:54 +02:00
wm4
ce246296b3 af_hrtf: request required samplerate, instead of erroring out
It seems hrtf works in 48khz only - and if that wasn't the input, the
filter just exited with an error. Make it request the 48khz instead. The
player will insert a resampling filter.

Not sure why it wasn't done like this in the first place.
2014-09-05 20:49:35 +02:00
wm4
5f29073abf af_hrtf: cosmetics: reindent misaligned code block 2014-09-05 20:47:54 +02:00
wm4
a7d737a698 audio: make buffer size configurable
Really only for testing.
2014-09-05 01:53:10 +02:00
wm4
8432eaefa0 audio/out: prevent burning CPU when seeking while paused
The audio/video sync code in player/audio.c calls ao_reset() each time
audio decoding is entered, but the player is paused, and there would be
more than 1 sample to skip to make audio start match with video start.
This caused a wakeup feedback loop with push.c.

CC: @mpv-player/stable
2014-08-31 14:48:58 +02:00
wm4
68ff8a0484 Move compat/ and bstr/ directory contents somewhere else
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.

The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
2014-08-29 12:31:52 +02:00
Stefano Pigozzi
f4ccf22e16 coreaudio_device: fix overwriting of user input
Fixes #1030
2014-08-25 10:08:54 +02:00
wm4
26500425f6 ao_dsound: raise default buffer size to 200ms, make it configurable 2014-08-22 16:12:47 +02:00
wm4
218ace2b02 audio: limit on low (and not high) buffer size
The original intention was probably to avoid unnecessarily high numbers
of wakeups. Change it to wait at most 25% of buffer time instead of 75%
until refilling. Might help with the dsound problems in issue #1024, but
I don't know if success is guaranteed.
2014-08-21 22:45:58 +02:00
wm4
61b0163d58 af_lavrresample: minor cosmetics 2014-08-17 03:29:09 +02:00
wm4
defa0a20e0 af_lavcac3enc: lower minimum channel number to 3
It seems only stereo PCM should be passed through.
2014-08-12 23:45:41 +02:00
wm4
be792c085a af_lavcac3enc: change default bitrate to 640
No reason to use less.

Since the name "default" is misleading now, replace it with "auto"
(still recognize the old name).
2014-08-12 23:34:28 +02:00
wm4
b4f72b46e5 ao_dsound: reduce default buffer size
Reduce from 1000ms to 100ms. Since there is an audio thread updating AOs
quickly enough now, requesting such a large buffer size makes no sense
anymore.
2014-08-08 01:56:23 +02:00
wm4
d68a759fa4 Improve setting AVOptions
Use OPT_KEYVALUELIST() for all places where AVOptions are directly set
from mpv command line options. This allows escaping values, better
diagnostics (also no more "pal"), and somehow reduces code size.

Remove the old crappy option parser (av_opts.c).
2014-08-02 03:12:33 +02:00
wm4
6afa1a2afc ao_alsa: disable use of non-interleaved formats by default
Some ALSA plugins take non-interleaved audio, but treat it as
interleaved, which results in various funny bugs. Users keep hitting
this issue, and it just doesn't seem worth the trouble.

CC: @mpv-player/stable
2014-07-30 23:28:44 +02:00
wm4
63d1d53d2f audio: ignore (some) decoding errors on initialization
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.
2014-07-29 18:05:55 +02:00
wm4
261506e36e audio: change playback restart and resyncing
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.
2014-07-28 21:20:37 +02:00
wm4
bc6359313f ao_pulse: allow disabling timing bug workarounds
Add an option that enables using native PulseAudio auto-updated timing
information, instead of the manual calculations added in mplayer2 times.
You can use --ao=pulse:no-latency-hacks to enable the new code. The code
is almost the same as the code that was removed with commit de435ed5,
but I didn't readd some bits I didn't understand. Likewise, the option
will disable the code added with that commit.

In my tests this seemed to work well, though the A/V sync display looks
funny when seeking.

The default is still the old behavior.

See issue #959.
2014-07-26 23:20:09 +02:00
wm4
77d9e4b8a9 ao_pulse: remove hacks for ancient PulseAudio versions
This was needed by very old (0.9) versions only. Get rid of it.

Unfortunately, I can't cross-check with the original bug report, since
the bug URL leads to this:

Internal Server Error

TracError: IOError: [Errno 2] No such file or directory: '/home/lennart/svn/trac/pulseaudio/VERSION'
2014-07-26 23:19:48 +02:00
wm4
7077526ffb ao_null: never fail at initialization
ao_null is used to stop autoprobing (if all AOs before fail to init).
After it come things like ao_pcm, which should never be automatically
selected.

Remove a certain theoretically possible failure case, and force "some"
fallback.
2014-07-26 20:26:57 +02:00
wm4
ac62244983 audio/out: fix initialization failure with win32
mp_make_wakeup_pipe() always fails on win32. If this call fails on Linux
(and e.g. ao_alsa is used), this will probably burn CPU since poll()
won't work on the invalid file descriptor, but whatever, the failure
case is obscure enough.
2014-07-26 20:26:27 +02:00
wm4
ef600041ba audio, client API: check mp_make_wakeup_pipe() return value
Could fail e.g. due to FD exhaustion.
2014-07-25 14:32:45 +02:00
wm4
69eb056333 audio: fix timestamps
Accidentally broken in b6af44d3. For ad_lavc (and in general), the PTS
was not updated correctly when filtering only parts of audio frames,
and for ad_mpg123 and ad_spdif the PTS was additionally offset by the
frame size.

This could lead to incorrect time display, and possibly broken A/V sync.
2014-07-24 15:27:31 +02:00
wm4
fc28e4af4d audio: adjust format change code
Execute the format change based on whether we logically detected EOF
(after filters), instead of when the decode buffer was drained. It's
slightly cleaner. (The requirement of len>0 existed before.)
2014-07-24 15:26:43 +02:00
wm4
986099d323 audio: fix race condition in EOF code
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.
2014-07-24 15:26:07 +02:00
wm4
b77dab0f6e audio: cosmetics
Move a function call, which does not change semantics.

Write the extra buffer sample count in a more straight-forward way; the
old code was not meaningful in any way (anymore).
2014-07-24 15:25:48 +02:00
wm4
6455bcc1da audio: remove unnecessary code
It's true that the decoder can successfully decode, but return no data
(for various reasons). We don't need to handle this specially, though.
We just let the decoder decode some more data. This doesn't increase the
danger of an endless loop either, because audio_decode() already calls
this function until enough is decoded.
2014-07-24 15:25:36 +02:00
Rudolf Polzer
c19ec6f6f6 encode: deal even more with codec->time_base deprecation.
I assume this works too with Libav 10 and FFmpeg d3e51b41.
2014-07-23 16:09:44 +02:00
wm4
80d36a0aa2 ao_pulse: fix potential compilation problem
It seems at least on some platforms (OSX 10.9), the POSIX wait()
function becomes visible, and conflicts with this unrelated function.
Just rename it.
2014-07-22 19:26:53 +02:00
wm4
b6af44d31e audio: move initial decode to generic code
This commit mainly moves the initial decoding of data (done to probe the
audio format) to generic code. This will make it easier to make audio
decoding non-blocking in a later commit.

This commit also changes how decoders return data: instead of having
them write the data into a prepared buffer, they return a reference to
an internal buffer (by setting dec_audio.decoded). This makes it
significantly easier to handle audio format changes, since the decoders
don't really need to care anymore.
2014-07-21 19:29:58 +02:00
wm4
1f9e0a15a1 ad_lavc: drop questionable fallback code
If the decoder didn't set a samplerate, it was initialized from the
container samplerate.

This probably didn't make much sense, because it's passed to the
decoder on initialization (so it could definitely use it). It's an
artifact from commit 66a9eb57 (which removed some Matroska-specific non-
sense), and I've never seen it actually happen since it was made into a
warning. Just get rid of it.
2014-07-21 19:29:58 +02:00
wm4
967add9f0f audio: remove unused metadata field
This was used for replaygain at some point, until replaygain info was
passed through explicitly.
2014-07-21 19:29:58 +02:00
wm4
9736f3309a audio: use symbolic constants instead of magic integers
Similar to commit 26468743.
2014-07-20 20:42:03 +02:00
Rudolf Polzer
073b2becfe ao_lavc: Fix design of audio pts handling.
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).
2014-07-16 16:18:34 +02:00
Rudolf Polzer
e257cbfdbb ao_lavc: Add a missing newline for the log. 2014-07-16 16:18:34 +02:00
Rudolf Polzer
2a985716cd ao_lavc: Fix advancing of audio pts. 2014-07-16 16:18:34 +02:00
wm4
417ffa8b40 Remove some mp_msg calls with no trailing \n
The final goal is all mp_msg calls produce complete lines. We want this
because otherwise, race conditions could corrupt the terminal output,
and it's inconvenient for the client API too. This commit works towards
this goal. There's still code that has this not fixed yet, though.
2014-07-13 20:12:13 +02:00
wm4
fb54a1436a audio: don't wait for draining if paused
Logic for this was missing from pull.c. For push.c it was missing if the
driver didn't support it. But even if the driver supported it (such as
with ao_alsa), strange behavior was observed by users. See issue #933.

Always check explicitly whether the AO is in paused mode, and if so,
don't drain.

Possibly fixes #933.

CC: @mpv-player/stable
2014-07-13 20:06:33 +02:00
wm4
f8c2dd1b78 build: include <strings.h> for strcasecmp()
It happens to work without strings.h on glibc or with _GNU_SOURCE, but
the POSIX standard requires including <strings.h>.

Hopefully fixes OSX build.
2014-07-10 08:29:32 +02:00