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* ao_pulse: exit AO if stream failswm42015-01-251-1/+5
| | | | | | This can for example reproduced by killing the pulseaudio server. If this happens, just try to reload the AO, instead of breaking everything forever.
* ao: remove coreaudio_exclusive from autoprobing listwm42015-01-251-2/+2
| | | | Apparently this was a mistake.
* ao_alsa: fix unpause path atfer previous commitwm42015-01-141-0/+2
| | | | The resume code was accidentally fully removed from this code path.
* ao_alsa: fix resuming from suspend modewm42015-01-141-4/+12
| | | | | | | | | | | 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.
* ao_alsa: fix setting mono channel mapwm42015-01-141-0/+5
| | | | | | | 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.
* audio: fix previous commitwm42015-01-141-4/+4
| | | | | This would have always forced mono first (if supported by the AO), instead of stereo.
* audio: fix fallback if audio API does not support monowm42015-01-141-6/+5
| | | | This makes it fallback to stereo properly.
* ao_coreaudio: add missing goto for error pathStefano Pigozzi2014-12-171-0/+1
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* af_hrtf: Fix out-of-range read.reimar2014-12-081-2/+7
| | | | | | | 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>
* ao_alsa: hackfix mono playbackwm42014-12-081-0/+3
| | | | | | 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.
* coreaudio: don't output too many channel descriptionsStefano Pigozzi2014-12-071-0/+6
| | | | for #1279 and #1249
* coreaudio: add missing \n in log lineStefano Pigozzi2014-12-071-1/+1
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* coreaudio: don't print layout a second timeStefano Pigozzi2014-12-071-1/+1
| | | | For #1279
* ao_alsa: try to fallback to "default" device if device is busywm42014-12-071-1/+6
| | | | | | | | | | | | | 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.)
* coreaudio: reject descriptions with too many channelsStefano Pigozzi2014-12-071-0/+7
| | | | This is a fix attempt for #1279 and #1249.
* coreaudio: fix more layout printsStefano Pigozzi2014-12-071-2/+2
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* coreaudio: fix prints of uint32_t in log_layoutStefano Pigozzi2014-12-071-10/+10
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* ao_coreaudio: initialize fetched properties to zerosStefano Pigozzi2014-12-011-2/+2
| | | | Should hopefully fix #1249 and #1279
* mixer: don't show softvol neutral marker on OSD if not using softvolwm42014-12-011-2/+2
| | | | Also fix the comment on the softvol field.
* audio: fix one of the previous commitswm42014-12-011-1/+1
| | | | | Fixes commit 52c51149. It broke multichannel (or possibly everything) for ao_alsa, ao_oss, ao_sndio.
* audio: allow more than 20 channel map entrieswm42014-12-014-8/+19
| | | | | | | | | | | | | 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.
* ao_alsa: fix channel map in pre-channel map API casewm42014-11-251-0/+1
| | | | Forgotten in commit 5d5f5b09.
* ao_alsa: always enable "plug" plugin for non-default devicewm42014-11-251-3/+2
| | | | | | | | | | | | | | 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.
* ao_alsa: select and set channel maps via channel map APIwm42014-11-251-28/+125
| | | | | | | | | | | | | | | | 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.
* 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/