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Commit Graph

1007 Commits

Author SHA1 Message Date
Martin Herkt
a17ea73636 af_rubberband: actually fix deadlock
371e5d0 missed this one
2015-02-12 10:15:12 +01:00
wm4
371e5d0665 af_rubberband: fix filter error deadlock
rubberband_available() can return a negative value, which we assigned to
a size_t variable, leading to the frame allocation to fail. This could
spam "Error filtering frame.". (That it spams this instead of exiting
should probably also be considered a bug.)

At least in the realtime mode and in our case, a negative return value
should not have any different meaning from a 0 return value, in
particular because we call rubberband_get_samples_required() or set the
"final" parameter for rubberband_process() to continue/stop processing.
2015-02-12 09:47:01 +01:00
Martin Herkt
2dc49ea866 af_rubberband: change defaults
After some testing, I am fairly convinced that these defaults sound
better than the previous settings. This also eliminates some issue
with random crackling and noise.

Also remove the `stretch` option since it has no effect in
realtime mode.
2015-02-12 00:58:40 +01:00
wm4
6299da2047 af_rubberband: fix breakage
The previous commit on this filter accidentally removed the
RubberBandOptionProcessRealTime option. Without it, the lib prints a
warning and passes the audio through.

Also add the RubberBandOptionSmoothingOn option back. Though for some
reason the output sounds still very wrong.
2015-02-11 21:32:01 +01:00
wm4
df5548a754 af_rubberband: make all librubberband options configurable
librubberband exports a big load of options. Normally, the default
settings (whether they're librubberband defaults or our defaults) should
be sufficient, but since I'm not so sure about this, making it
configurable allows others to figure it out for me.
2015-02-11 17:11:05 +01:00
wm4
6f24a61d84 af_rubberband: attempt to fix audio position calculation
The problem here is that librubberband can buffer an arbitrary amount
of data, but at the same time doesn't provide a way to query how much
data is buffered. So we keep track of this manually, assuming that
librubberband tries to reach the requested time ratio for input and
output (which is probably true).

The disadvantage is that rounding errors could accumulate over time.
(Maybe it should try to round towards keeping the time ratio.)
2015-02-11 16:32:40 +01:00
wm4
76501f4f57 af_rubberband: always calculate and set delay
Basically, add an if and reindent the block instead of exiting early.
2015-02-11 16:32:40 +01:00
wm4
d85aa35ffb af: account for queued frames in audio position calculation
af_rubberband exposed this issue.
2015-02-11 16:32:40 +01:00
wm4
8c055f873f af_rubberband: improve EOF handling
In theory it could happen that draining on EOF happens incrementally,
and then the unconditional reset could have dropped the remaining
buffered audio.
2015-02-11 16:31:35 +01:00
wm4
67aeccc254 audio: fix pool allocation
It reallocated the pool on every request, making the pool completely
useless. Oops.
2015-02-11 11:36:07 +01:00
wm4
b6ab34fc98 af_rubberband: pitch correction with librubberband
If "--af=rubberband" is used, librubberband will be used to speed up or
slow down audio with pitch correction.

This still has some problems: the audio delay is not calculated
correctly, so the audio position jitters around by a few milliseconds.
This will probably ruin video timing.
2015-02-11 00:29:12 +01:00
wm4
81d8c5d519 af_scaletempo: allow changing speed at runtime without reinit
Staring at the code a bit, it turns out that changing speed without
losing state is quite easy. The initialization code is big and
complicated, but most of it is specific only to the configured audio
format, not the speed.

Refactor the code so that changing speed at runtime could work. (It's
not actually used yet - the player code still does a complete reinit.
This will be fixed in the next commit.)

The "if (s->speed_tempo == s->speed_pitch)" looks a bit strange, but
does the same thing as the code did before: speed can be changed only if
exactly one flag is set. If both are set or none, speed can't be
changed.
2015-02-10 22:34:07 +01:00
wm4
2a3d19a9df af_scaletempo: drop detaching or skipping init on speed=1
This code skipped initialization if no speed/pitch change was to be
applied.

It also didn't force conversion of the audio to a supported format,
which is probably the most important case in context of compatibility.
With this change applied, af_scaletempo will always force format
conversion.

To make the change less disruptive, make the filter detach if
unconvertable formats are used. Some users use spdif and also have
"af=scaletempo" in their config, so better not completely break this.

In the case the filter was added with the "speed=both" suboption, the
filter also detached itself in this case; but it's an obscure case, so I
don't care about that.
2015-02-10 22:14:26 +01:00
Stefano Pigozzi
5de7f1c5ac ao_coreaudio: fix small memory leak 2015-02-03 00:40:02 +01:00
Stefano Pigozzi
de4f997752 ao_coreaudio: use device UID instead of ID for selection
Previously we let the user use the audio device ID, but this is not persistent
and can change when plugging in new devices. That of course made it quite
worthless for storing it as a user setting for GUIs, or for user scripts.

In theory getting the kAudioDevicePropertyDeviceUID can fail but it doesn't
on any of my devices, so I'm leaving the error reporting quite high and see if
someone complains.
2015-02-03 00:40:02 +01:00
Stefano Pigozzi
a3be14683a command: add property returning detected audio device
This can be useful to adjust some other audio related properties
at runtime depending on the audio device being used.
2015-02-03 00:40:02 +01:00
wm4
12d822ce44 ao_null: add emulation for certain broken behavior
I'm not sure how common this behavior possibly is; well whatever. This
option will allow reproducing such behavior, and help debugging it.
2015-01-30 21:30:54 +01:00
Ben Boeckel
b1d47786d8 ao_pulse: plug a memory leak 2015-01-25 01:26:11 +01:00
James Ross-Gowan
3c10ed540b ao_wasapi: fix try_format logic in shared mode
The MSDN documentation for IsFormatSupported says a return code of
AUDCLNT_E_UNSUPPORTED_FORMAT means the function "succeeded but the
specified format is not supported in exclusive mode." This seems to
imply that the format is supported in shared mode, and that's what the
old code assumed, however try_format would incorrectly return success
with some drivers.

The remarks section of the documentation contradicts that assumption. It
says that in shared mode, if the audio engine does not support the
caller-specified format or any similar format, ppClosestMatch is set to
NULL and the function returns AUDCLNT_E_UNSUPPORTED_FORMAT. This is the
same as in exclusive mode, so treat AUDCLNT_E_UNSUPPORTED_FORMAT the
same regardless of opt_exclusive. In shared mode, the format selection
code will fall back to the mix format, which should always be supported.
2015-01-23 22:02:15 +11:00
wm4
c0077ac936 ao_alsa: reinitialize if device got broken
Apparently, physically disconnecting the audio device (consider USB
audio) breaks the ALSA device handle forever. It will signal ENODEV.
Fortunately, it's easy for us to handle this, and we can just use
existing mechanisms that will make the playback core close and reopen
the AO. Whether the immediate reopening will actually succeeds really is
ALSA's problem, though.
2015-01-21 19:38:18 +01:00
wm4
1e6b4d31aa ao_coreaudio: reset possibly random errno value
In general, you need to check errno when using strtol(), but as far as I
know, strtol() won't reset errno on success. This has to be done
manually. The code could have failed sporadically if strtol() succeeded,
and errno was already set to one of the checked values.

(This strtol() still isn't fully error checked, but I don't know if it's
intentional, e.g. for parsing a numeric prefix only.)
2015-01-20 14:32:01 +01:00
wm4
d44b4ccba1 ao: never autoselect ao_null
Before this commit, ao_null was used as last fallback. This doesn't make
too much sense. Why would you decode audio just to discard it? Let audio
initialization fail instead. This also handles the weird but possible
corner-case that ao_null might fail initializing, in which case e.g.
ao_pcm could be autoselected. (This happened once, and had to be fixed
manually.)
2015-01-20 14:28:34 +01:00
wm4
3c2ca0cecc ao: refactor --audio-device selection code
This removes the slightly duplicated code for picking the required AO
driver if --audio-device forces one. Now --audio-device reuses the same
code as --ao for this.

As a consequence, ao_alloc_pb() and ao_create() can be merged into
ao_init(). Although the ao_init() argument list, which is already pretty
big, grows by one, it's better than having all these similar sounding
functions around.

Actually, I just wanted to do the change the following commit will do,
but I found this code was more of a mess than it had to be.
2015-01-20 14:25:47 +01:00
wm4
ae641d200a af: remove old filter compatibility hack 2015-01-15 20:13:15 +01:00
wm4
388cf6dc96 audio/filter: switch remaining filters to refcounting
All of these filters are very similar in frame management, and copy data
to a new frame during filtering.
2015-01-15 20:13:14 +01:00
wm4
87fe7d8788 audio/filter: switch remaining in-place filters to refcounting
Adds about 7 lines of boilerplate per filter. This could be avoided by
providing a different entrypoint (something like af->filter_inplace),
which would basically mirror the old interface exactly for this kind of
filter. But I feel like it would just be a hack to support all those
old, useless filters better. (The ideal solution would be using a
language that can do closures to provide a compat. wrapper, but
whatever.)

af_bs2b has terribly repetitious code for setting up filter functions
for each format (most of them useless, in addition to bs2b being
useless), so I did something terrible with macros.

af_sinesuppress had commented code for float filtering (maybe it was
broken; it has been commented every since it was added in 2006). Remove
this code.
2015-01-15 20:13:12 +01:00
wm4
ba0e8b754c af: verify filter input formats
Just to make sure all filters get the correct format. Together wih the
check in af_add_output_frame(), this asserts that

    af->prev->fmt_out == af->fmt_in

This also requires setting the "in" pseudo-filter (s->first) formats
correctly. Before this commit, the fmt_in/fmt_out fields weren't used
for this filter.
2015-01-15 20:10:46 +01:00
wm4
c757a06845 ao_alsa: fix a small memory leak 2015-01-14 22:16:36 +01:00
wm4
e865d255d0 af_lavcac3enc: use refcounted frames 2015-01-14 22:16:30 +01:00
wm4
5d972491bb af_lavfi: use refcounted frames 2015-01-14 22:15:56 +01:00
wm4
9c974b2a1b audio/filter: actually set fmt_in/fmt_out fields 2015-01-14 22:15:51 +01:00
wm4
f6a0a1554c af_scaletempo: use refcounted frames 2015-01-14 22:15:39 +01:00
wm4
218c749a16 af_lavrresample: use refcounted frames 2015-01-14 22:15:31 +01:00
wm4
7b8862760d audio: add missing declaration 2015-01-14 22:15:00 +01:00
wm4
c8ecb66269 ao_pcm: add append mode
Pretty useful for debugging, although a bit useless or possibly
misleading too (see comments in the manpage).
2015-01-14 22:14:56 +01:00
wm4
4cabd08e8a audio: fix initial audio PTS
Commit 5e25a3d2 broke handling of the initial frame (the one decoded
with initial_audio_decode()). It didn't update the pts_offset field,
leading to a shift in timestamps by one audio frame.

Fix by calling the actual decode function in a single place. This
requires slightly more changes than what would be necessary to fix the
bug, but it also somewhat simplifies the data flow.
2015-01-14 22:14:46 +01:00
wm4
3cb2add636 audio: fix assertion failure on audio decoding
There are several cases in which a decoder may need several packets to
produce some output audio. Commit 5e25a3d2 broke this.

Fixes #1471.
2015-01-14 07:58:01 +01:00
wm4
ecca64e182 af_convert24: use refcounted frames
This requires allocating a fully new frame. 32->24 could be in-place,
but this is not possible for 24->32.
2015-01-13 20:17:08 +01:00
wm4
983f5efa3c audio/filters: use refcounted frames for some in-place filters
These are also quite simple, but require requesting write access to the
frames. The error handling (for OOM) is a bit annoying.
2015-01-13 20:17:03 +01:00
wm4
1fde40732e audio/filters: use refcounted frames for some simple filters
These are read-only, and very trivial to convert.
2015-01-13 20:16:59 +01:00
wm4
772c42a95c af_volume: use refcounted frames 2015-01-13 20:15:53 +01:00
wm4
5e25a3d216 audio: use refcounted frames in the filter chain
The goal is switching the whole audio chain to using refcounted frames.
This brings the architecture closer to FFmpeg, enables better
integration with libavfilter, will reduce useless copying somewhat, and
will probably allow better timestamp tracking.

For now, every filter goes through a semi-awful wrapper in
af_do_filter(), though. This will be fixed step by step, and the wrapper
should eventually be removed. Another thing that will have to be done is
improving the timestamp handling and avoiding extra copies for the AO.

Some of the new code is rather similar to the video filter code (the
core filter code basically just has types replaced). Such code
duplication is normally very unwanted, but in this case there's probably
no other choice. On the other hand, this code is pretty simple (even if
somewhat tricky). Maybe there will be unified filter code in the future,
but this is still far away.
2015-01-13 20:15:43 +01:00
wm4
97becbc31b audio: add some utility functions for refcounted frames
Used in the following commits.
2015-01-13 20:14:25 +01:00
wm4
0bbd65b09c audio/filter: remove unused af_calc_filter_multiplier()
The purpose of this function was to filter only as much audio input as
needed to produce a certain amount of audio output. This could (in
theory) avoid excessive buffering when e.g. changing playback speed with
resampling.

Use of this was already removed in commit 5fd8a1e0. No problems were
experienced, so let's assume this feature is practically worthless.
(Though it's possible that it was quite useful over a decade ago, or in
some cornercases with evil files.)
2015-01-13 20:14:02 +01:00
wm4
2c9180f47b ao_pulse: exit AO if stream fails
This can for example reproduced by killing the pulseaudio server. If
this happens, just try to reload the AO, instead of breaking everything
forever.
2015-01-11 04:19:40 +01:00
wm4
7f2b78846b ao_alsa: fix dtshd passthrough
We must not try to remap channels with this. Whethever ALSA gives us,
and whatever we do with it, the result will probably be nonsense.

Untested, as I don't have the required hardware.
2015-01-09 03:58:47 +01:00
wm4
5a7719594e ao: remove coreaudio_exclusive from autoprobing list
Apparently this was a mistake.
2015-01-07 22:31:34 +01:00
wm4
dc2d0539c7 ao_pulse: disable latency calculation hacks by default
This used to be required to workaround PulseAudio bugs. Even later, when
the bugs were (partially?) fixed in PulseAudio, I had the feeling the
hacks gave better behavior. On the other hand, I couldn't actually
reproduce any bad behavior without the hacks lately. On top of this, it
seems our hacks sometimes perform much worse than PulseAudio's native
implementation (see #1430).

So disable the hacks by default, but still leave the code and the option
in case it still helps somewhere. Also, being able to blame PulseAudio's
code by using its native API is much easier than trying to debug our own
(mplayer2-derived) hacks.
2015-01-07 22:23:38 +01:00
wm4
f61b8b312d win32: request UTF-16 API variants, Vista+ APIs, and COM C macros
Put the Vista+ (_WIN32_WINNT) and the COM C (COBJMACROS) defines into
the build system, instead of defining them over and over in the code.
2015-01-07 21:42:44 +01:00
wm4
0f4bf347c5 player: print used number of threads in verbose mode
Also, don't use av_log() for mpv output.
2015-01-05 12:17:55 +01:00
wm4
fda44ecc92 af_volume: dump applied replaygain in verbose mode 2015-01-04 01:35:48 +01:00
Kevin Mitchell
6a6620a554 ao/wasapi: style/code formatting tweaks 2015-01-02 14:50:59 -08:00
Kevin Mitchell
155c8e20ef ao/wasapi: improve exclusive mode format search
fixes #1376
2015-01-02 14:08:47 -08:00
Kevin Mitchell
81948634ca ao/wasapi: revamp set_waveformatex
* bits instead of bytes
* add valid_bits argument
* just pass in the mp_chmap and get the number and wavext channel map from that
* indicate valid bits in waveformat_to_str
* make appropriate accomodations in try_format
2015-01-02 14:08:47 -08:00
Kevin Mitchell
121352cd95 ao/wasapi: add CO_E_NOTINITIALIZED to explain_err
someone on irc reported seeing this error
2015-01-02 14:08:47 -08:00
wm4
4075518011 ao_portaudio: remove this audio output
It's just completely useless. We have good native support for all 3
desktop platforms, and ao_sdl or ao_openal as fallbacks.
2014-12-29 18:53:12 +01:00
wm4
adeada149b ao_alsa: print channel map if setting it fails
This message is printed when the audio device advertised a channel map,
but couldn't set it - which is probably a dmix bug (we'll never know,
ALSA doesn't take bug reports).

Print the requested map, so that the user (maybe) can make a connection
when seeing the message and the actually used channel map, which might
be less confusing. Or at least less useless.
2014-12-29 18:49:11 +01:00
Stefano Pigozzi
21d93690cb ao: add debug log with the detected channel maps
This could be helpful with bug reports.
2014-12-29 17:56:53 +01:00
Stefano Pigozzi
54aea7d5de chmap_sel: add multichannel fallback heuristic
Instead of just failing during channel map selection, try to select a close
layout that makes most sense and upmix/downmix to that instead of failing AO
initialization. The heuristic is rather simple, and uses the following steps:

1) If mono is required always prefer stereo to a multichannel upmix.
2) Search for an upmix that is an exact superset of the required channel map.
3) Search for a downmix that is the exact subset of the required channel map.
4) Search for either an upmix or downmix that is the closest (minimum difference
   of channels) to the required channel map.
2014-12-29 17:56:53 +01:00
Stefano Pigozzi
461ba50ed6 chmap: add a 7.1(rear) layout name
This is common on Apple systems so it's handy to have a label for it.
2014-12-29 17:56:53 +01:00
Stefano Pigozzi
894b172a76 ao_coreaudio: remove useless guard
useless after 069016fd6c
2014-12-27 12:33:44 +01:00
Stefano Pigozzi
15e30e58b2 ao_coreaudio: fix some naming conventions 2014-12-27 12:33:44 +01:00
Stefano Pigozzi
069016fd6c ao_coreaudio: fix channel mapping
There where 3 major errors in the previous code:

1) The kAudioDevicePropertyPreferredChannelLayout selector returns a single
   layout not an array.
2) The check for AudioChannelLayout allocation size was wrong (didn't account
   for variable sized struct).
3) Didn't query the kAudioDevicePropertyPreferredChannelsForStereo selector
   since I didn't know about it's existence.

All of these are fixed.

Might help with #1367
2014-12-27 12:04:58 +01:00
Stefano Pigozzi
9aa7df3446 ao_coreaudio: fix typo 2014-12-27 00:29:21 +01:00
Stefano Pigozzi
4d99315730 ao_coreaudio: move some code to make output readable 2014-12-27 00:27:50 +01:00
Stefano Pigozzi
1391e765a2 ao_coreaudio: add more layout debug outputs
Should help remote debugging #1367 with --msg-level=ao=debug
2014-12-27 00:16:48 +01:00
wm4
3fdb6be316 win32: add mmap() emulation
Makes all of overlay_add work on windows/mingw.

Since we now don't explicitly check for mmap() anymore (it's always
present), this also requires us to make af_export.c compile, but I
haven't tested it.
2014-12-26 17:30:10 +01:00
Stefano Pigozzi
9317071bc3 ao_coreaudio: fix AudioChannelLayout allocations
AudioChannelLayout uses a trailing variable sized array so we need to
query CoreAudio for the size of the struct it is going to need (or the
conversion of that particular layout would fail).

Fixes #1366
2014-12-26 15:04:36 +01:00
wm4
759656d0ba ao_alsa: fix unpause path atfer previous commit
The resume code was accidentally fully removed from this code path.
2014-12-23 13:20:32 +01:00
wm4
d7b5484f51 ao_alsa: fix resuming from suspend mode
snd_pcm_prepare() was not always called, which could result in an
infinite loop.

Whether snd_pcm_prepare() was actually called depended on whether the
device was a hw device (or other characteristics; depending on
snd_pcm_hw_params_can_pause()), and required real suspend (annoying for
testing), so it was somewhat tricky to reproduce without knowing these
things.
2014-12-23 03:59:14 +01:00
wm4
a69f168dff ao_alsa: fix setting mono channel map
When setting the ALSA channel map, we never actually set the map we got
from ALSA directly, but convert it to mpv's, and then back to ALSA's.
mpv and ALSA use different conventions for mono, and there is already an
exception for ALSA->mpv, but not mpv->ALSA.
2014-12-20 17:18:50 +01:00
wm4
0dc455eb16 ao_alsa: remove some dead code
This was only added recently (c1e97161) as an attempt to minimize the
bad impact of channel layout device aliases. But use of these was
removed in commit 49df0132. Now this code does pretty much nothing, and
shouldn't be needed anymore. It does something when using spdif, but
this fallback won't work anyway.
2014-12-20 16:54:00 +01:00
wm4
5b32f30aa1 audio: fix previous commit
This would have always forced mono first (if supported by the AO),
instead of stereo.
2014-12-20 16:48:30 +01:00
wm4
d07c6566cd audio: fix fallback if audio API does not support mono
This makes it fallback to stereo properly.
2014-12-20 16:21:52 +01:00
Stefano Pigozzi
4b65bd5086 ao_coreaudio: fix mono/stereo channel mapping
Needed after af3bbb800d since now we use channel mapping all the time.

Fixes #1357
2014-12-16 13:04:29 +01:00
Stefano Pigozzi
a7e48eca66 ao_coreaudio: add missing goto for error path 2014-12-16 13:04:28 +01:00
Kevin Mitchell
1e5f9d2673 ao/wasapi: use IsEqualGUID and IsEqualPropertyKey
before we were reinventing this wheel
2014-12-16 03:29:51 -08:00
wm4
49df01323e ao_alsa: remove old multichannel method
The "old" method (before the ALSA channel map API) used device aliases
like "surround51" to set the channel layout. The "interesting" part was
that these devices usually redirect to a hardware device. This means
playing stereo would lead you to the "default" device (dmix), while e.g.
5.1 to "surround51", which automatically takes care of the fact that
dmix can't do 5.1.

This is pretty much nonsense, though. It shouldn't depend on the damn
input media file whether the player is going to use shared access (dmix)
or exclusive access (direct hw device).

As a consequence, by default ao_alsa will do only what dmix can do. If
the user actually wants multichannel, he has to select a suitable hw
device with --audio-device. From there on, the correct speaker mapping
will be ensured via the channel mapping API.

The change is preparation for making multichannel output the default (as
far as supported by the audio output API). Of the common APIs, only ALSA
messes up beyond repair, so I feel like this change is needed.

On ancient alsa-lib versions, only stereo and mono can be played with
this branch.
2014-12-15 16:58:03 +01:00
wm4
ae5fd4a809 ao_alsa: add ridiculous hack to deal with braindead ALSA behavior
dmix reports channel layouts it doesn't support. The rest of the
technical part of the story is in the code comment.

This seems to be the only reasonable way to fallback from trying to
initialize certain devices (like dmix) with multichannel audio. We could
probably add support for such padding channels to our audio chain or to
ao_alsa itself, but this would probably be much more work than this
commit.

What dmix does is probably a bug. I've tried to report it to ALSA. Thay
have a link on their website to a bug tracker, but it's a dead link, and
has been for years. I've posted to alsa-devel, but received no reply.
I'm thus assuming this absolutely retarded behavior is by design, and
nothing will happen to improve upon it.

I'm considering sending Lennart Poettering a "thank you" email, because
with PulseAudio, multichannel audio just works (although some other
things just don't work).
2014-12-15 16:40:23 +01:00
Kevin Mitchell
4966a67f71 ao/wasapi: set the ao with the waveformat channelmap
hopefully this fixes #1350
2014-12-15 05:01:38 -08:00
reimar
13b4fb9d28 af_hrtf: Fix out-of-range read.
Based on patch by Yuriy Kaminskiy [yumkam gmail].

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37330 b3059339-0415-0410-9bf9-f77b7e298cf2
Signed-off-by: wm4 <wm4@nowhere>
2014-12-06 17:09:57 +01:00
wm4
020897b5d3 ao_alsa: minor simplification
Whether we print it as warning or error doesn't really matter; we
continue anyway. (I don't actually know what the implications of running
in non-blocking mode are; for what's it worth, when I tested with
explicitly changing to non-blocking, it seemed to work fine anyway, so
don't change that part.)
2014-12-05 16:04:05 +01:00
wm4
c6deee3801 ao_alsa: hackfix mono playback
ALSA returns "FL" as channel layout when trying to play mono. mpv and
libavresample don't like this; in particular, using libavresample to
convert stereo to "FL" fails.
2014-12-05 16:04:05 +01:00
Stefano Pigozzi
254c60e608 coreaudio: don't output too many channel descriptions
for #1279 and #1249
2014-12-05 12:35:34 +01:00
Stefano Pigozzi
f5ac80ea88 coreaudio: add missing \n in log line 2014-12-05 09:57:40 +01:00
Stefano Pigozzi
8e6f3bef36 coreaudio: don't print layout a second time
For #1279
2014-12-05 09:57:06 +01:00
wm4
d6606bcfff ao_alsa: simplify, remove no-block suboption
If no-block was given, the device would be opened with SND_PCM_NOBLOCK.
Also, after opening, blocking mode was unconditionally enabled anyway
with snd_pcm_nonblock(). Further, if opening with SND_PCM_NOBLOCK
failed, opening was retried without this flag.

This doesn't make any sense to me, and I've never heard of someone using
this suboption. I suspect it has to do with ancient ALSA bugs or API
caveats. Remove it and simplify the code.
2014-12-05 01:23:09 +01:00
wm4
c1e97161f4 ao_alsa: try to fallback to "default" device if device is busy
ALSA is crap. It's impossible to make multichannel playback just do the
right thing. dmix (the default on most distros) can do stereo only, and
will refuse to play multichannel. On the other hand, if you try like mpv
(and mplayer) to open a multichannel device (like "surround51" etc.),
this will actually open a hardware device, which will either fail if
dmix is active, or block out dmix if opening succeeds.

This commit falls back to "default" (i.e. dmix) if opening a
multichannel device fails, which is a tiny step towards the right
behavior. (Although fixing it fully is impossible.)
2014-12-04 22:42:07 +01:00
Stefano Pigozzi
9faf482d89 coreaudio: reject descriptions with too many channels
This is a fix attempt for #1279 and #1249.
2014-12-04 21:51:06 +01:00
Stefano Pigozzi
c070d16093 coreaudio: fix more layout prints 2014-12-04 21:51:03 +01:00
Stefano Pigozzi
4db97d3303 coreaudio: fix prints of uint32_t in log_layout 2014-12-04 21:33:38 +01:00
wm4
4be7bdcc0f audio: fix one of the previous commits
Fixes commit 52c51149. It broke multichannel (or possibly everything)
for ao_alsa, ao_oss, ao_sndio.
2014-12-01 18:28:00 +01:00
Stefano Pigozzi
1c0920a8dd ao_coreaudio: initialize fetched properties to zeros
Should hopefully fix #1249 and #1279
2014-12-01 16:51:19 +01:00
wm4
b0ed93d87d audio: allow more than 20 channel map entries
This could trigger an assertion when using ao_alsa or ao_coreaudio. The
code was simply assuming the number of channel maps was bounded
statically (which was true at first in both AOs).

Fix by using dynamic memory allocation. It needs to be explicitly
enabled by the AOs by setting a temp context, because otherwise the
memory couldn't be freed. (Or at least this seems to be the most elegant
solution.)

Fixes #1306.
2014-12-01 15:28:06 +01:00
Kevin Mitchell
67c4117476 ao/wasapi: make set_ao_format EX/EXTENSIBLE agnostic
There is no guarantee that closestMatch returned by IsFormatSupported
is actually a WAVEFORMATEXTENSIBLE.

http://msdn.microsoft.com/en-us/library/windows/desktop/dd370876%28v=vs.85%29.aspx

We should therefore not blindly treat it as such.
2014-12-01 03:40:24 -08:00
Kevin Mitchell
146561cc91 ao/wasapi: fix set_ao_format
Before it used whatever was in ao->format and changed the bits even
though this might have nothing to do with the actual WAVEFORMAT
negotiated with WASAPI.

For example, if the initial ao->format was a float and we had set the
WAVEFORMAT to s24, this would create a non-existent float24 format.
Worse, it might put an u16 into ao->format when WAVEFORMAT described s16.
WASAPI doesn't support unsigned at all as far as I can tell.
2014-12-01 03:40:24 -08:00
Kevin Mitchell
524cdfc3f1 ao/wasapi: show actual waveformat tried
also remove bogus ao_format
2014-12-01 03:40:23 -08:00
Kevin Mitchell
bd33fa7052 ao/wasapi: don't assume 32-bits == float
This was based on old WAVEFORMATEX restrictions
http://msdn.microsoft.com/en-us/library/windows/hardware/ff538799%28v=vs.85%29.aspx

With the new WAVEFORMATEXTENSIBLE, this is no longer a problem. and we
can have s32 or float32 so we need to actually check / set these correctly.

fixes #1287
2014-12-01 03:40:23 -08:00
Kevin Mitchell
2006069ca2 ao/format: add af_fmt_is_float 2014-12-01 03:40:23 -08:00
Kevin Mitchell
96fa3ebd1a ao/wasapi: make sure that < 16-bit pcm never happens
it just sucks. noone should have to listen to that.
2014-12-01 03:40:23 -08:00
Kevin Mitchell
9a0b97d214 ao/wasapi: get rid of WAVEFMT union
It only confused the issue. Replace it's functionality with
waveformat_copy function where needed.
2014-12-01 03:40:23 -08:00
Kevin Mitchell
77f675a151 ao/wasapi: handle VistaBlob failure more gracefully 2014-11-28 10:52:48 -08:00
Kevin Mitchell
b83e447e2b ao/wasapi: remove unnecessary check of audio thread input
it would have caused a deadlock if it fired anyway.
2014-11-28 10:52:48 -08:00
Kevin Mitchell
b0854bc42c ao/wasapi: more consistent/reliable method of computing extra WAVEFORMATEXTENSIBLE size 2014-11-28 10:52:48 -08:00
Kevin Mitchell
8908b80b77 ao/wasapi: more missed cleanup on failure 2014-11-28 10:52:43 -08:00
Kevin Mitchell
7b5baf3b53 ao/wasapi: check return values
Only issue a warning for failure of wasapi_enumerate_devices and
wasapi_fill_VistaBlob.
2014-11-28 10:52:12 -08:00
Kevin Mitchell
14f9719a65 ao/wasapi: make functions return bool that were acting like it
this involved inverting the logic of find_formats, enumerate_devies
and wasapi_fill_VistaBlob. The latter two were trivial as their return
values were not actually checked (to be fixed in a later
commit).
2014-11-28 10:48:36 -08:00
Kevin Mitchell
4f208e6f68 ao/wasapi: check full GUID of KSDATAFORMAT to determine float 2014-11-28 10:48:36 -08:00
Kevin Mitchell
239c880fe2 ao/wasapi: expose GUID and PKEY convenience functions
Give them the prefix mp_ and make them nonstatic.
2014-11-28 10:48:36 -08:00
Kevin Mitchell
e2bc1c5f17 ao/wasapi: remove unused variable 2014-11-28 10:48:36 -08:00
Kevin Mitchell
fb5d88a287 ao/wasapi: safely define PKEY constants
Before these definitions were incorrectly guarded by and #ifdef
but since they aren't macros, this would never be true so that
if they were ever added to mingw headers we would have problems.

rename KSDATAFORMAT constants with the same mp prefix for consistency.
also use DEFINE_GUID rather than defining the bare structure
2014-11-28 10:48:36 -08:00
Kevin Mitchell
81464780e9 ao/wasapi: avoid redundant passing of ao and wasapi_state as arguments
also drop some useless const declaraitons
2014-11-28 10:48:36 -08:00
Kevin Mitchell
3b0b89c0af ao/wasapi: just return 0 unconditionally from the thread
We weren't actually checking this value anyway. We only really
cared about init failure, which was checked another way.
2014-11-28 10:48:36 -08:00
Kevin Mitchell
d40c0d4f90 ao/wasapi: fix unmatched CoUninitialize() on init failure 2014-11-28 10:48:36 -08:00
Kevin Mitchell
d0c37f0731 ao/wasapi: more debugging messages 2014-11-28 10:48:07 -08:00
Kevin Mitchell
ecb491fd95 ao/wasapi: simplify the init retry 2014-11-28 08:02:16 -08:00
Kevin Mitchell
e4aaaf69df ao/wasapi: make get_device_delay return an error code 2014-11-28 08:02:16 -08:00
wm4
4a83f3df11 ao_wasapi: don't treat SetDisplayName() failure as fatal
Same for SetIconPath().
2014-11-27 20:05:29 +01:00
wm4
a98f88a12f mixer: don't show softvol neutral marker on OSD if not using softvol
Also fix the comment on the softvol field.
2014-11-27 10:11:55 +01:00
wm4
cc54377463 Do not call strerror()
...because everything is terrible.

strerror() is not documented as having to be thread-safe by POSIX and
C11. (Which is pretty much bullshit, because both mandate threads and
some form of thread-local storage - so there's no excuse why
implementation couldn't implement this in a thread-safe way. Especially
with C11 this is ridiculous, because there is no way to use threads and
convert error numbers to strings at the same time!)

Since we heavily use threads now, we should avoid unsafe functions like
strerror().

strerror_r() is in POSIX, but GNU/glibc deliberately fucks it up and
gives the function different semantics than the POSIX one. It's a bit of
work to convince this piece of shit to expose the POSIX standard
function, and not the messed up GNU one.

strerror_l() is also in POSIX, but only since the 2008 standard, and
thus is not widespread.

The solution is using avlibc (libavutil, by its official name), which
handles the unportable details for us, mostly. We avoid some pain.
2014-11-26 21:21:56 +01:00
wm4
5b69b76609 ao_alsa: fix channel map in pre-channel map API case
Forgotten in commit 5d5f5b09.
2014-11-25 18:34:24 +01:00
wm4
e1ae936e6b ao_alsa: always enable "plug" plugin for non-default device
This seems safer: otherwise, opening the AO could randomly fail if the
audio formats happens to be not float.

Unfortunately, this only works if the user does not select a device.
Since ALSA devices are arbitrary strings, including plugins with complex
parameters, it's not trivial or maybe even impossible to edit the string
in a way the "plug" plugin is added.

With --audio-device, it would be safe for users to select either
"default" or one of the "plughw" devices. Everything else seems
questionable.
2014-11-25 18:15:45 +01:00
wm4
5d5f5b094b ao_alsa: select and set channel maps via channel map API
Use the ALSA channel map API for querying and selecting supported
channel maps.

Since we (probably?) want to be compatible with ALSA versions before the
change, we still try to select the device name by channel map, and open
that device. There's no way to negotiate a channel map before opening,
so we're stuck with this approach. Fortunately, it seems these devices
allow selecting and setting any other supported channel layout, so maybe
this is not an issue at all. In particular, this avoids selecting the
default (dmix) device, which can only do stereo.

Most code is based on Martin Herkt <lachs0r@srsfckn.biz>'s alsa_ng
branch, with heavy modifications.
2014-11-25 18:09:36 +01:00
wm4
5fb54fa756 ao_alsa: minor fixes
Don't crash if no fallback channel layout could be found (caller can't
handle NULL return from select_chmap()). Apparently this could never
actually happen, though.

Don't treat snd_pcm_hw_params_set_periods_near() failure as fatal error.
Same deal as with snd_pcm_hw_params_set_buffer_time_near().

Actually free channel maps returned by snd_pcm_get_chmap().

Adjust some messages.
2014-11-25 17:27:19 +01:00
wm4
7d6e58471f audio: make mp_audio_config_to_str return a stack-allocated string
Simpler overall.
2014-11-25 11:11:31 +01:00
wm4
8a7b686597 ao_alsa: cleanups
No functional changes.

ALSA_PCM_NEW_HW_PARAMS_API was a pre-ALSA 1.0.0 thing and does nothing
with modern ALSA. It stopped being necessary about 10 years ago.

3 functions are moved to avoid forward references.
2014-11-25 11:10:44 +01:00
wm4
28b6ce39d3 audio: make mp_chmap_to_str() return a stack-allocated string
Simplifies memory management.
2014-11-24 19:56:01 +01:00
wm4
2228d47373 ao_alsa: try to use the channel map reported by ALSA
If ALSA reports a channel map, and it looks like it makes sense (i.e.
could be converted to mpv channel map, and the channel count matches),
then use that instead of the channel map we are assuming.

This is based on code written by lachs0r (alsa_ng branch).
2014-11-24 19:44:26 +01:00
wm4
df43e2d22a ao_pcm: simplify
Also shuts up Coverity.
2014-11-21 10:09:38 +01:00
wm4
9d2aef048d ao_oss: check whether setting samplerate succeeds
Independent from whether the samplerate was accepted or adjusted, errors
returned by the ioctl are fatal errors.

Found by Coverity.
2014-11-21 10:09:26 +01:00
wm4
c6c46f5aa7 ao_lavc: fix setting up AVFrame pointers
The caller set up the "start" pointer array using the number of planes,
the encode() function used the number of channels. This copied
uninitialized values for packed formats, which makes Coverity warn.
2014-11-21 10:09:25 +01:00
wm4
c01a62efbc af_scaletempo: use float division for rate
From what I understand the division is to align the dimension of the
value from seconds to milliseconds. Hard to tell whether the "rounding"
was intentional or not; I'm tipping on "not".

Found by Coverity.
2014-11-21 10:09:15 +01:00
wm4
e082c2c3df Remove some unneeded NULL checks
Found by Coverity; also see commit 85fb2af3.
2014-11-21 09:58:09 +01:00
wm4
524aa99401 audio/out/push: fix off-by-one error
Needs 1 additional free entry.

Found by Coverity.
2014-11-21 03:50:57 +01:00
wm4
459f3aa4f9 ao_lavc: fix dangling pointers
Found by Coverity.
2014-11-21 03:50:52 +01:00
Kevin Mitchell
f2dda72dbc ao/wasapi: only retry resizing the buffer once
like the MSDN example:

http://msdn.microsoft.com/en-us/library/windows/desktop/dd370875%28v=vs.85%29.aspx
2014-11-18 07:50:51 -08:00
Kevin Mitchell
da03334a73 ao/wasapi: keep bufferPeriod in sync on retry
Without this, the retry will fail if they are not equal or
bufferPeriod is zero.
2014-11-18 06:59:26 -08:00
Kevin Mitchell
94ea4435a9 ao/wasapi: refix printf warning for both cygwin and msys
a cast to (unsigned) should do "the right thing".
2014-11-18 05:03:33 -08:00
Kevin Mitchell
19e9c9d1be ao/wasapi: periodicity in shared mode must be zero
IAudioClient::Initialize hnsPeriodicity argument is nonzero only for exclusive mode

http://msdn.microsoft.com/en-us/library/windows/desktop/dd370805%28v=vs.85%29.aspx
2014-11-18 05:03:33 -08:00
Kevin Mitchell
c545c406fa ao/wasapi: increase buffer size to 50 ms
Before it was the default device period, which was too small
causing glitches on on entering/exiting fullscreen.
2014-11-18 05:03:33 -08:00
wm4
d96bd0eaa8 audio/out: always log retrieved audio device size 2014-11-18 12:51:43 +01:00
Jonathan Yong
7697d300d2 ao/wasapi: fix leaked marshaled interface streams
Signed-off-by: Kevin Mitchell <kevmitch@gmail.com>
2014-11-18 02:12:28 -08:00
Kevin Mitchell
22bf0a78df ao/wasapi: Don't free stuff the thread may still be using on timeout
In the unlikely event of a timeout waiting for the audio thread to return,
don't free stuff that it may still be using.
2014-11-17 23:46:38 -08:00
Kevin Mitchell
20d42b3475 ao/wasapi: also free the threadLoop handle on uninit
http://msdn.microsoft.com/en-us/library/windows/desktop/ms682453%28v=vs.85%29.aspx
2014-11-17 23:43:51 -08:00
Kevin Mitchell
23f52fd41b ao/wasapi: fix leaked event handles 2014-11-17 23:32:28 -08:00
Kevin Mitchell
ebd161b256 ao/wasapi: fix race condition in uninit on failure.
When the audio thread fails to properly init, it signals failure
to the main thread, AND THEN starts to clean up. For this to work,
ao_init callback must not return until the thread's cleanup is finished.
This is correctly handled in the ao_uninit callback by waiting for
the thread to exit, so just call that to clean up the main thread.
I have no idea why I didn't do this in the first place.
2014-11-17 23:32:13 -08:00
James Ross-Gowan
d9bac96a9d ao/wasapi: silence format string warnings 2014-11-18 12:19:36 +11:00
wm4
fb86750a67 ao_alsa: check for EAGAIN too
Simply retry on EAGAIN.

I've seen this in several other projects; it might be just cargo-culting
though.
2014-11-17 20:07:59 +01:00
wm4
8b2798cb3e audio/out: switch back to wasapi as default on win32
dsound was set as default, because there were some hard to fix problems
with wasapi. These problems were probably fixed now, so let's try with
wasapi as default again.
2014-11-17 14:07:11 +01:00
Kevin Mitchell
4c8b841fc4 ao/wasapi: request ao reload on thread_feed failures
Even with change notifications, there are still (rare) cases when the
feed thread gets AUDCLIENT_DEVICE_INVALIDATED. So handle failures in
thread_feed by requesting ao_reload.
2014-11-17 04:31:22 -08:00