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* ao_alsa: minor fixeswm42014-11-251-4/+6
| | | | | | | | | | | | | 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.
* audio: make mp_audio_config_to_str return a stack-allocated stringwm42014-11-253-20/+10
| | | | Simpler overall.
* ao_alsa: cleanupswm42014-11-251-97/+57
| | | | | | | | | 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.
* audio: make mp_chmap_to_str() return a stack-allocated stringwm42014-11-245-26/+24
| | | | Simplifies memory management.
* ao_alsa: try to use the channel map reported by ALSAwm42014-11-242-1/+66
| | | | | | | | 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).
* ao_pcm: simplifywm42014-11-211-17/+13
| | | | Also shuts up Coverity.
* ao_oss: check whether setting samplerate succeedswm42014-11-211-2/+4
| | | | | | | Independent from whether the samplerate was accepted or adjusted, errors returned by the ioctl are fatal errors. Found by Coverity.
* ao_lavc: fix setting up AVFrame pointerswm42014-11-211-3/+4
| | | | | | 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.
* af_scaletempo: use float division for ratewm42014-11-211-1/+1
| | | | | | | | 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.
* Remove some unneeded NULL checkswm42014-11-211-5/+6
| | | | Found by Coverity; also see commit 85fb2af3.
* audio/out/push: fix off-by-one errorwm42014-11-211-1/+1
| | | | | | Needs 1 additional free entry. Found by Coverity.
* ao_lavc: fix dangling pointerswm42014-11-211-1/+1
| | | | Found by Coverity.
* ao/wasapi: only retry resizing the buffer onceKevin Mitchell2014-11-181-8/+10
| | | | | | like the MSDN example: http://msdn.microsoft.com/en-us/library/windows/desktop/dd370875%28v=vs.85%29.aspx
* ao/wasapi: keep bufferPeriod in sync on retryKevin Mitchell2014-11-181-1/+4
| | | | | Without this, the retry will fail if they are not equal or bufferPeriod is zero.
* ao/wasapi: refix printf warning for both cygwin and msysKevin Mitchell2014-11-181-2/+2
| | | | a cast to (unsigned) should do "the right thing".
* ao/wasapi: periodicity in shared mode must be zeroKevin Mitchell2014-11-181-3/+6
| | | | | | 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
* ao/wasapi: increase buffer size to 50 msKevin Mitchell2014-11-183-16/+26
| | | | | Before it was the default device period, which was too small causing glitches on on entering/exiting fullscreen.
* audio/out: always log retrieved audio device sizewm42014-11-181-2/+2
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* ao/wasapi: fix leaked marshaled interface streamsJonathan Yong2014-11-181-0/+9
| | | | Signed-off-by: Kevin Mitchell <kevmitch@gmail.com>
* ao/wasapi: Don't free stuff the thread may still be using on timeoutKevin Mitchell2014-11-171-1/+3
| | | | | In the unlikely event of a timeout waiting for the audio thread to return, don't free stuff that it may still be using.
* ao/wasapi: also free the threadLoop handle on uninitKevin Mitchell2014-11-171-0/+1
| | | | http://msdn.microsoft.com/en-us/library/windows/desktop/ms682453%28v=vs.85%29.aspx
* ao/wasapi: fix leaked event handlesKevin Mitchell2014-11-171-6/+5
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* ao/wasapi: fix race condition in uninit on failure.Kevin Mitchell2014-11-171-2/+1
| | | | | | | | | 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.
* ao/wasapi: silence format string warningsJames Ross-Gowan2014-11-182-2/+2
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* ao_alsa: check for EAGAIN toowm42014-11-171-1/+1
| | | | | | | Simply retry on EAGAIN. I've seen this in several other projects; it might be just cargo-culting though.
* audio/out: switch back to wasapi as default on win32wm42014-11-171-3/+3
| | | | | | 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.
* ao/wasapi: request ao reload on thread_feed failuresKevin Mitchell2014-11-171-0/+2
| | | | | | 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.
* ao/wasapi: add retry loop on AUDCLNT_E_DEVICE_IN_USEKevin Mitchell2014-11-171-0/+12
| | | | this works around reinitializing too fast on device property changes
* ao/wasapi: request reset on appropriate eventsKevin Mitchell2014-11-174-56/+117
| | | | | | | | on changes to PKEY_AudioEngine_DeviceFormat, device status, and default device. call ao_reload directly in the change_notify "methods". this requires keeping a device enumerator around for the duration of execution, rather than just for initially querying devices
* ao/wasapi: add convenience functions for change notifiyKevin Mitchell2014-11-171-0/+49
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* ao/wasapi: new wasapi device monitoring interfaceJonathan Yong2014-11-172-0/+179
| | | | | | | | | | | Implement skeleton IMMNotificationClient to watch for changes in the sound device. This will make recovery possible from changes shared mode sample rate, bit depth, "enhancements"/effects and even graceful device removal. http://msdn.microsoft.com/en-us/library/windows/desktop/dd371417%28v=vs.85%29.aspx Signed-off-by: Kevin Mitchell <kevmitch@gmail.com>
* ao/wasapi: look for "multimedia" default device instead of "console"Kevin Mitchell2014-11-171-2/+2
| | | | | | console is more for system notifications / voice command, mpv is most certainly multimedia http://msdn.microsoft.com/en-us/library/windows/desktop/dd370842%28v=vs.85%29.aspx
* ao/wasapi: put loading of default device in it's own functionKevin Mitchell2014-11-171-17/+30
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* ao/wasapi: fix possible null dereference of pDeviceKevin Mitchell2014-11-171-0/+1
| | | | | | IMMDeviceEnumerator::GetDefaultAudioEndpoint may set pDevice to null on failure. http://msdn.microsoft.com/en-us/library/windows/desktop/dd371401%28v=vs.85%29.aspx
* ao/wasapi: tidy up better on failureKevin Mitchell2014-11-173-14/+26
| | | | | | | Before, failures, particularly in the thread loop init, could lead to a bad state for the duration of mpvs execution. Make sure that everything that was initialized gets properly and safely uninitialized.
* ao/wasapi: improve error messages and add more debug statementsKevin Mitchell2014-11-174-98/+140
| | | | | | also enforce more consistency in the exit codes and error handling thanks to Jonathan Yong <10walls@gmail.com>
* ao/wasapi: make calling of thread_init consistent with thread_uninitKevin Mitchell2014-11-173-3/+5
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* ao/wasapi: reenable the reset functionKevin Mitchell2014-11-171-1/+1
| | | | | | the race condition that necessitated disabling this was fixed in e4403523131a69a92a8418bb3714090a408680c7
* ao/wasapi: fix leaked deviceIDJonathan Yong2014-11-171-0/+1
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* af: remove redundant functionwm42014-11-121-9/+2
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* af: check audio params for validitywm42014-11-121-0/+5
| | | | Normally, these should be valid anyway, so this is just being cautious.
* ao_lavc, vo_lavc: Fix crashes in case of multiple init attempts.Rudolf Polzer2014-11-121-0/+8
| | | | | | | | | | | | When initialization failed, vo_lavc may cause an irrecoverable state in the ffmpeg-related structs. Therefore, we reject additional initialization attempts at least until we know a better way to clean up the mess. ao_lavc currently cannot be initialized more than once, yet it's good to do consistent changes there as well. Also, clean up uninit-after-failure handling to be less spammy.
* audio: make sure AVFrame is actually refcountedwm42014-11-111-0/+12
| | | | | | | | | | | The mp_audio_from_avframe() function requires the AVFrame to be refcounted, and merely increases its refcount while referencing the same data. For non-refcounted frames, it simply did nothing and potentially would make the caller pass around a frame with dangling pointers. (libavcodec should always return refcounted frames, but it's not clear what other code does; and also the function should simply work, instead of having weird requirements on its arguments.)
* audio: refuse to allocate frames in invalid formatwm42014-11-111-1/+1
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* audio: make decoders output refcounted frameswm42014-11-108-204/+145
| | | | | | | | | | | | | | This rewrites the audio decode loop to some degree. Audio filters don't do refcounted frames yet, so af.c contains a hacky "emulation". Remove some of the weird heuristic-heavy code in dec_audio.c. Instead of estimating how much audio we need to filter, we always filter full frames. Maybe this should be adjusted later: in case filtering increases the volume of the audio data, we should try not to buffer too much filter output by reducing the input that is fed at once. For ad_spdif.c and ad_mpg123.c, we don't avoid extra copying yet - it doesn't seem worth the trouble.
* audio: add mp_audio_make_writeable()wm42014-11-102-0/+28
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* audio: clear buffer array too with mp_audio_set_null_data()wm42014-11-101-1/+3
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* audio: change how filters are inserted on playback speed changeswm42014-11-104-1/+72
| | | | | | | | | | Use a pseudo-filter when changing speed with resampling, instead of somehow changing a samplerate somewhere. This uses the same underlying mechanism, but is a bit more structured and cleaner. It also makes some of the following changes easier. Since we now always use filters to change audio speed, move most of the work set_playback_speed() does to recreate_audio_filters().
* af_format: remove redundant message prefixeswm42014-11-101-2/+2
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* audio: add function to convert AVFrame to mp_audio referenceswm42014-11-102-0/+51
| | | | | This is somewhat duplicated from ad_lavc.c and af_lavfi.c, but will eventually be used by both.
* audio: add mp_audio_poolwm42014-11-102-4/+66
| | | | | | A helper to allocate refcounted audio frames from a pool. This will replace the static buffer many audio filters use (af->data), because such static buffers are incompatible with refcounting.
* audio: use AVBufferRef to allocate audio frameswm42014-11-102-20/+10
| | | | | | | A first step towards refcounted audio frames. Amazingly, the API just does what we want, and the code becomes simpler. We will need to NIH allocation from a pool, though.
* audio/out/pull: avoid deadlock if audio callback stopswm42014-11-091-26/+40
| | | | | | | | | | | | | | | | | | | | | | | | If the audio callback suddenly stops, and the AO provides no "reset" callback, then reset() could deadlock by waiting on the audio callback forever. The waiting was needed to enter a consistent state, where the audio callback guarantees it won't access the ringbuffer. This in turn is needed because mp_ring_reset() is not concurrency-safe. This active waiting is unavoidable. But the way it was implemented, the audio callback had to call ao_read_data() at least once when reset() is called. Fix this by making ao_read_data() set a flag upon entering and leaving, which basically turns p->state into some sort of spinlock. The audio callback actually never needs to spin, because there are only 2 states: playing audio, or playing silence. This might be a bit surprising, because usually atomic_compare_exchange_strong() requires a retry-loop idiom for correct operation. This commit is needed because ao_wasapi can (or will in the future) randomly stop the audio callback in certain corner cases. Then the player would hang forever in reset().
* audio/out: consistently use double return type for get_delaywm42014-11-0912-27/+25
| | | | | ao_get_delay() returns double, but the get_delay callback still returned float.
* audio/out: make ao_request_reload() idempotentwm42014-11-093-9/+25
| | | | | | | | | | This is what you would expect. Before this commit, each ao_request_reload() call would just queue a reload command, and then recreate the AO for the number of times the function was called. Instead of sending a command, introduce some sort of event retrieval mechanism. At least for the reload case, use atomics, because we're too lazy to setup an extra mutex.
* audio: add --audio-client-name optionwm42014-11-073-5/+7
| | | | | | The main need I see for this is with libmpv - it would be confusing if some application showed up as "mpv" on whateverthehell PulseAudio uses it for (generally it does show up on various PA GUI tools).
* ao_oss: wait for events with poll()wm42014-11-061-0/+13
| | | | | | | | | | The intention is to avoid using the timeout-based fallback. There's some minor hope that this will help with OpenBSD (see #1239), although it probably won't. Some chance that this will cause trouble with obscure OSS implementations or emulations.
* audio/out/push: when using audio wait fallback, recheck conditionwm42014-11-061-1/+2
| | | | | | | | | | | If calling ao->driver->wait() fails, we need to fallback to timeout- based waiting. But it could be that at this point, the mutex was already released (and then re-acquired). So we need to recheck the condition in order to avoid missed wakeups. This probably wasn't an actually occurring problem, but still could cause a small race-condition window if the dynamic fallback is actually used.
* ad_lavc: allow skip samples amount to be larger than 1 packetwm42014-11-031-2/+6
| | | | | Apparently we actually need this. At least the following commit would break without this.
* ao_alsa: don't make snd_pcm_hw_params_set_buffer_time_near() error fatalwm42014-10-311-1/+7
| | | | | | | | | | | | | Apparently this can "sometimes" return an error. In my opinion, this should never return an error: neither the semantics of the function, nor the ALSA documentation or ALSA sample code seem to indicate that a failure is to be expected. I'm not perfectly sure about this though (I blame ALSA being a weird, big, underdocumented API). Since it causes problems for some users, and since there is really no reason why we should abort on such an error, turn it into a warning. Fixes #1231.
* options: accept --audio-channels=autowm42014-10-301-0/+1
| | | | This sounds much more intuitive, while "empty" was a bit of a WTF.
* coreaudio: only list output devicesStefano Pigozzi2014-10-281-0/+12
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* audio: add command/function to reload audio outputwm42014-10-272-0/+10
| | | | | Anticipated use: simple solution for dealing with audio APIs which request configuration changes via events.
* ao_alsa: move parameter append code to a functionwm42014-10-231-16/+27
| | | | | Why not. (I thought I needed this, but my other experiments failed. So this is merely a minor cleanup.)
* rename ao_coreaudio_device.c -> ao_coreaudio_exclusive.cStefano Pigozzi2014-10-231-0/+0
| | | | This is so that the source file name matches the AO name
* coreaudio: redirect IEC61937 to coreaudio_exclusiveStefano Pigozzi2014-10-232-1/+7
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* audio/out: add redirection-on-init mechanismwm42014-10-222-14/+47
| | | | | Looks like this will help us with making --audio-device and spdif work as expected on OSX. To be used ina following commit.
* audio/out: missing error checkwm42014-10-221-0/+2
| | | | Oops.
* audio/out: don't add special devices to --audio-device listwm42014-10-221-2/+2
| | | | | | | | | | | | | | | | | | Since the list associated with --audio-device is supposed to enable simple user-selection, it doesn't make much sense to include overly special things like ao_pcm or ao_null in the list. Specifically, ao_pcm is harmful, because it will just dump all audio to a file named audiodump.wav in the current working directory. The user can't choose the filename (it can be customized, but not through this option), and the working directory might be essentially random, especially if this is used from a GUI. Exclude "strange" entries. We reuse the fact that there's already a simple list ordered by auto-probe priority in order to avoid having to add an additional flag. This is also why coreaudio_exclusive was moved above ao_null: ao_null ends auto-probing and marks the start of "special" outputs, which don't show up on the device, but we want coreaudio_exclusive to be selectable (I think).
* audio/out: include coreaudio_exclusive in auto-probingwm42014-10-221-3/+3
| | | | | | | Move it above ao_null, so that it can be selected during auto-probing (even if it's only last). I see no reason why it should not be included, and it makes the following commit slightly more elegant. (See explanations there.)
* Set thread name for debuggingwm42014-10-191-0/+1
| | | | | | | | | | Especially with other components (libavcodec, OSX stuff), the thread list can get quite populated. Setting the thread name helps when debugging. Since this is not portable, we check the OS variants in waf configure. old-configure just g