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* audio: react to --ao and --audio-buffer runtime changeswm42019-12-271-3/+3
| | | | | | Before this commit, runtime changes were only applied if something else caused audio to be reinitialized. Now setting them reinitializes audio explicitly.
* audio: add ao_audiotrack for androidAman Gupta2019-11-192-0/+721
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* audio: fix minor whitespace issue in out/internal.hAman Gupta2019-11-191-1/+1
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* Replace uses of FFMIN/MAX with MPMIN/MAXwm42019-10-311-2/+2
| | | | And remove libavutil includes where possible.
* input: add gamepad support through SDL2Stefano Pigozzi2019-10-231-1/+1
| | | | | | | | | | | | | | | The code is very basic: - only handles gamepads, could be extended for generic joysticks in the future. - only has button mappings for controllers natively supported by SDL2. I heard more can be added through env vars, there's also ways to load mappings from text files, but I'd rather not go there yet. Common ones like Dualshock are supported natively. - analog buttons (TRIGGER and AXIS) are mapped to discrete buttons using an activation threshold. - only supports one gamepad at a time. the feature is intented to use gamepads as evolved remote controls, not play multiplayer games in mpv :)
* audio/out: rip out old unused app/softvolume reportingwm42019-10-117-21/+0
| | | | | | | | | | | This was all dead code. Commit 995c47da9a (over 3 years ago) removed all uses of the controls. It would be nice if AOs could apply a linear gain volume, that only affects the AO's audio stream for low-latency volume adjust and muting. AOCONTROL_HAS_SOFT_VOLUME was supposed to signal this, but to use it, we'd have to thoroughly check whether it really uses the expected semantics, so there's really nothing useful left in this old code.
* audio/out/pull, ao_sdl: implement new underrun reportingwm42019-10-112-2/+8
| | | | | | | | | | | | | | | | See previous commits. ao_sdl is worthless, but it might be a good test for pull-based AOs. This stops using the old underrun reporting if the new one is enabled. Also, since the AO's behavior can in theory not be according to expectations, this needs to be enabled for every single pull AO separately. For some reason, in certain cases I get multiple underrun warnings while cache-pausing is active. It fills the cache, restarts the AO, immediately underruns again, and then fills the cache again. I'm not sure why this happens; maybe ao_sdl tries to catch up when it shouldn't. Who knows.
* audio/out/pull: fix underflow reportingwm42019-10-111-2/+2
| | | | | | | | I think this was _always_ wrong. Due to the line above the first changed line, buffered_bytes==bytes always. I can only hope I broke this in a less under-tested edit when I originally wrote this. Fixes: c5a82f729bd097
* ao_alsa: use AO underrun reportingwm42019-10-111-1/+3
| | | | This enables the change introduced in the previous commit for ao_alsa.
* ao: add API for underrun reportingwm42019-10-113-1/+23
| | | | | | | | | | | | | | AOs can now call ao_underrun_event() (in any context) if an underrun has happened. It will print a message. This will be used in the following commits. But for now, audio.c only clears the underrun bit, so that subsequent underruns still print the warning message. Since the underrun flag will be used in fragile ways by the playback state machine, there is the "reports_underruns" field that signals strong support for underrun reporting. (Otherwise, underrun events will not be used by it.)
* ao_alsa: handle underruns in get_space() toowm42019-10-111-0/+2
| | | | | This is essentially optional. But it will give the higher level code a better guarantee that underruns were tested.
* ao_alsa: mess with underrun handling againwm42019-10-111-6/+27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This commit tries to prepare for better underrun reporting. The goal is to report underruns relatively immediately. Until now, this happened only when play() was called. Change this, and abuse that get_delay() is called "relatively often" - this reports the underrun immediately in practice. Background: In commit 81e51a15f7e1 (and also e38b0b245ed4), we were quite confused about ALSA underrun handling. The commit message showed uncertainty how case 3 happened, but it's blindingly obvious and simple. Actually reading the code shows that ALSA does not have a concept of a "final chunk" (or we don't use it). It's obvious we never pass the AOPLAY_FINAL_CHUNK flag along to the ALSA API in any way. The only thing we do is simply writing a partial fragment. Of course this will cause an underrun. Doing a partial write saves us the trouble to pad the last frame with silence, or so. The main reason why the underrun message was avoided was that play() was never called with a non-0 sample count again (except if reset() was called before that). That was OK, at least the goal of avoiding the unwanted message was reached. (And the original "bogus" message at end of playback was perfectly correct, as far as ALSA goes.) If network stalls, play() will called again only once new data is available. Obviously, this could take a long time, thus it's too late.
* ao_alsa: don't silence legitimate underrun if final chunk underrunswm42019-10-061-4/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | It turns out that case 2) mentioned in the previous commit happened quite often when playback ended normally. There is probably a legitimate underrun with normal buffer sizes (100 ms, 4 fragments, gapless audio in "weak" mode). This is a result of the player waiting for video to end, and/or the time needed to kill the video window. The former case means that it depends on your test case whether it happens (a file where video ends slightly before audio is less likely to trigger it). This in turn is due to how gapless playback works. Achieving not having a "gap" requires queuing the audio of the next file without playing a partial chunk (as AOPLAY_FINAL_CHUNK would do). The partial chunk is then played as part of the first chunk played from the next file. But if it detects "later" that there is no next file, it still needs to get rid of the last fragment with AOPLAY_FINAL_CHUNK. At this point it's too late, and an underrun may have actually happened. The way the player uninits and reinits the entire playback engine for the next file in a "serial" manner means it cannot know in advance whether this works. This is the reason why the idiot who added the underrun exception for the last chunk in play() was wrong (I wrote that btw., before you accuse me of being rude). Yes, it's a real underrun, and you could probably hear it.
* ao_alsa: remove sometimes bogus XRUN messagewm42019-10-061-9/+2
| | | | | | | | | | | | | | | | | | | This XRUN (aka underrun) message was printed in the following situations: 1) legitimate underrun during playback 2) legitimate underrun when playing final chunk 3) bogus underrun when playing final chunk The old underrun case (in play()) happens in cases 1) and 2) as well, but 3) did not happen. It appears 3) is indeed something that happens, although it's not known for sure. It's still pretty annoying, so remove the new XRUN message. When testing, care should be taken to play with buffer sizes, video versus no video, and gapless enabled/disabled. Also, suspending the player with Ctrl+Z in the terminal (SIGSTOP) and then resuming is a good way to trigger a "normal" underrun.
* options: add M_OPT_FILE to some more options that take filesPhilip Sequeira2019-09-271-1/+1
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* ao_pulse: add the newly added mappings for TrueHD/DTS-HD formatsJan Ekström2019-09-271-6/+11
| | | | | Originally DTS-HD was mapped to PA_ENCODING_DTS_IEC61937 which I'm actually not sure if it ever worked.
* ao_oss: Fallback to stereo when the device does not support >2 channelsLeonardo Taccari2019-09-211-6/+10
| | | | | | | | ioctl(..., SNDCTL_DSP_CHANNELS, &nchannels) for not supported nchannels does not return an error and instead set nchannels to the default value. Instead of failing with no audio, fallback to stereo.
* ao_pulse: add --pulse-allow-suspendedTérence Clastres2019-09-211-1/+3
| | | | | | | | | | This flag makes mpv continue using the PulseAudio driver even if the sink is suspended. This can be useful if JACK is running with PulseAudio in bridge mode and the sink-input assigned to mpv is the one JACK controls, thus being suspended. By forcing mpv to still use PulseAudio in this case, the user can now adjust the sink to an unsuspended one.
* ao_opensles: fix delayed audiosfan52019-09-021-1/+1
| | | | | This was forgotten in commit 5a8c48fde2a26fe00c3552e3ccf83a965b6d3576 when the number of buffers was reduced to 1.
* ao/audiounit: include AVAudioSession buffer in latency calcAman Gupta2019-04-051-1/+1
| | | | Signed-off-by: Aman Gupta <aman@tmm1.net>
* ao/audiounit: improve a/v syncAman Gupta2019-04-053-4/+18
| | | | | | | This more closely mimics ao_coreaudio, on which this driver was originally based. Signed-off-by: Aman Gupta <aman@tmm1.net>
* Merge commit '559a400ac36e75a8d73ba263fd7fa6736df1c2da' into ↵Anton Kindestam2018-12-052-3/+29
|\ | | | | | | | | | | wm4-commits--merge-edition This bumps libmpv version to 1.103
| * ao: use a local option structwm42018-05-242-3/+29
| | | | | | | | Instead of accessing MPOpts.
* | ao_audiounit: rename pause function to resetJosh Lehman2018-09-301-1/+1
| | | | | | | | | | AudioUnit output driver uses the pull based api so it should have a reset function instead of a pause function.
* | ao_alsa: log the ALSA state if we get a non-XRUN errorJan Ekström2018-09-291-2/+4
| | | | | | | | | | The ALSA state generally can tell us more information in case we get an unexpected error.
* | ao_alsa: handle XRUNs separately from other errorsJan Ekström2018-09-291-2/+7
| | | | | | | | | | | | | | | | | | According to ALSA doxy, EPIPE is a synonym to SND_PCM_STATE_XRUN, and that is a state that we should attempt to automatically recover from. In case recovery fails, log an error and return zero. A warning message will still be output for each XRUN since those are not something we should generally be receiving.
* | ao_alsa: early exit get_space if paused or ALSA is not readyJan Ekström2018-09-291-0/+5
| | | | | | | | | | | | | | | | | | | | | | This has been way too long coming, and for me to notice that a whole lot of ao_alsa functions do an early return if the AO is paused. For the STATE_SETUP case, I had this reproduced once, and never since. Still, seems like we can start calling this function before the ALSA device has been fully initialized so we might as well early exit in that case.
* | ao_jack: only auto-connect to audio portsNiklas Haas2018-09-261-1/+2
| | | | | | | | | | This prevents ao_jack from auto-connecting to MIDI ports (or other, hypothetical future port types).
* | ao_pulse: fix tlength calculationTom Yan2018-09-011-3/+3
| | | | | | | | also remove the now unused non-sensical af_fmt_seconds_to_bytes.
* | Revert "ao_openal: enable building on OSX"Michael Hoang2018-08-261-14/+0
| | | | | | | | | | | | This reverts commit af6126adbe61fb2b6cc780025246d33df93072e6. Apple's OpenAL support is ridiculously out of date, revert back to just using OpenAL Soft on macOS (fixes #4645).
* | ao_opensles: set numBuffers to 8Tom Yan2018-08-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Apparently some Android builds/forks require this for Bluetooth audio to work as they unexpectedly accept fast flag for it. Shouldn't cause any side-effect (e.g. buffer requirement increased when on wired audio). It's a hardcoded default in the upstream AAudio implementation anyway. Ref.: https://android.googlesource.com/platform/frameworks/av/+/android-8.0.0_r1/media/libaaudio/src/legacy/AudioStreamTrack.cpp#109 https://android.googlesource.com/platform/frameworks/wilhelm/+/android-8.0.0_r1/src/android/AudioPlayer_to_android.cpp#1680 https://android.googlesource.com/platform/frameworks/av/+/android-8.0.0_r1/media/libaudioclient/AudioTrack.cpp#488
* | ao_opensles: rework the heuristic of buffer/enqueue size settingTom Yan2018-08-051-18/+36
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ao->device_buffer will only affect the enqueue size if the latter is not specified. In other word, its intended purpose will solely be setting/guarding the soft buffer size. This guarantees that the soft buffer size will be consistent no matter a specific enqueue size is set or not. (In the past it would drop to the default of the generic audio-buffer option.) opensles-frames-per-buffer has been renamed to opensles-frames-per -enqueue, as it was never purposed to set the soft buffer size. It will only make sure the size is never smaller than itself, just as before. opensles-buffer-size-in-ms is introduced to allow easy tuning of the relative (i.e. in time) soft buffer size (and enqueue size, unless the aforementioned option is set). As "device buffer" never really made sense in this AO, this option OVERRIDES audio-buffer whenever its value (including the default) is larger than 0. Setting opensl-buffer-size-in-ms to 1 allows you to equate the soft buffer size to the absolute enqueue size set with opensl-frames-per -enqueue conveniently (unless it is less than 1ms). When both are set to 0, audio-buffer will be the ultimate fallback. If audio-buffer is also 0, the AO errors out.
* | ao_opensles: allow s32 and float outputTom Yan2018-08-051-27/+15
| | | | | | | | | | OpenSLES (and its AudioTrack backend) in Android can take 32-bit fixed and floating point input since Android L (API 21).
* | ao_alsa: simplify get_space()Jan Ekström2018-06-041-6/+10
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* | ao_alsa: replace snd_pcm_status() with snd_pcm_avail() in get_space()Muhammad Faiz2018-06-041-5/+4
|/ | | | | | | | | | Fixes a bug with alsa dmix on Fedora 29. After several minutes, audio suddenly becomes bad and muted. Actually, I don't know what causes this. Probably this is a bug in alsa. In any case, as snd_pcm_status() returns not only 'avail', but also other fields such as tstamp, htstamp, etc, this could be considered a good simplification, as only avail is required for this function.
* build: make encoding mode non-optionalwm42018-05-031-2/+0
| | | | Makes it easier to not break the build by confusing the ifdeffery.
* encode: get rid of the output packet queuewm42018-05-035-3/+33
| | | | | | | | | | | | Until recently, ao_lavc and vo_lavc started encoding whenever the core happened to send them data. Since audio and video are not initialized at the same time, and the muxer was not necessarily opened when the first encoder started to produce data, the resulting packets were put into a queue. As soon as the muxer was opened, the queue was flushed. Change this to make the core wait with sending data until all encoders are initialized. This has the advantage that we don't need to queue up the packets.
* encode: remove old timestamp handlingwm42018-05-031-46/+6
| | | | | This effectively makes --ocopyts the default. The --ocopyts option itself is also removed, because it's redundant.
* encode: rewrite half of itwm42018-04-291-185/+55
| | | | | | | | | | | | | The main change is that we wait with opening the muxer ("writing headers") until we have data from all streams. This fixes race conditions at init due to broken assumptions in the old code. This also changes a lot of other stuff. I found and fixed a few API violations (often things for which better mechanisms were invented, and the old ones are not valid anymore). I try to get away from the public mutex and shared fields in encode_lavc_context. For now it's still needed for some timestamp-related fields, but most are gone. It also removes some bad code duplication between audio and video paths.
* encode: cosmeticswm42018-04-201-25/+29
| | | | Mostly whitespace changes; some semantic preserving transformations.
* ao_alsa: actually report underruns to userwm42018-04-151-5/+5
| | | | | | | | | Print them as a warning. Note that there may be some cases where it underruns, without being a bad condition. This could possibly happen e.g. if the last chunk is written, and then it resumes playback some time after that. Eventually I want to add more code to avoid such spurious warnings.
* ao_pulse: reduce requested device buffer sizewm42018-04-151-1/+1
| | | | | | Same deal as with the previous commit for ALSA. Untested.
* ao_alsa: reduce requested buffer sizewm42018-04-151-2/+2
| | | | | | There is a dedicated thread for feeding audio to the ALSA API from a buffer with a larger size. There is little reason to have such a large device buffer.
* ao_alsa: add options for controlling period/buffer sizewm42018-04-151-8/+16
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* ao_openal: document the muted↔gain conversionJan Ekström2018-04-151-0/+3
| | | | This struck me as odd for a moment, so adding a comment.
* ao/openal: Add option to set buffering characteristicsLAGonauta2018-04-151-23/+62
| | | | | | | | | One can now set the number of buffers and the buffer size. This can reduce the CPU usage and the total latency stays mostly the same. As there are sync mechanisms the A/V sync continue intact and working. It also modifies 6.1 channel order, as per OpenAL spec and add AOPLAY_FINAL_CHUNK support
* ao/openal: Add better sample format and channel layout selectionLAGonauta2018-04-151-139/+73
| | | | Also re-added floating-point support.
* ao/openal: Add OpenAL Soft extension to get the correct latencyLAGonauta2018-04-151-1/+16
| | | | | | | | OpenAL Soft's AL_SOFT_source_latency extension allows one to correctly get the device output latency, facilitating the syncronization with video. Also added a simpler generic fallback that does not take into account latency of the device.
* ao/openal: Add support for direct channels outputLAGonauta2018-04-151-0/+10
| | | | | | | Uses OpenAL Soft's AL_DIRECT_CHANNELS_SOFT extension and can be controlled through a new CLI option, --openal-direct-channels. This allows one to send the audio data direrctly to the desired channel without effects applied.
* ao/openal: Add hardware mute supportLAGonauta2018-04-151-0/+12
| | | | | While the volume is set on the listener, mute is set on the sound source. Seemed easier that way.
* ao/openal: Use only one source for audio outputLAGonauta2018-04-151-52/+153
| | | | Floating point audio not supported on this commit.
* ao_opensles: let cfg_frames_per_buffer accept buffer size up to 0.5s at 192kHzTom Yan2018-04-051-1/+1
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* ao_opensles: remove useless cfg_sample_rateTom Yan2018-04-051-5/+0
| | | | We should always use the ao-neutral --audio-samplerate option.
* ao_opensles: bump device buffer size to 250msTom Yan2018-04-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Although half (non-fast track on sink rate) or one-third (non-fast track not on sink rate) of the buffer size of the created AudioTrack instance as the SL Enqueue buffer size is basically enough for dropout-free playback, only using the full size can avoid stutter upon (re)start of playback. Here are the various buffer sizes on different track/sink rate when on Bluetooth audio on Android O: aptX @ 48kHz: Sink rate: 48000 Hz 44100 Hz: 10632 frames (241.09 ms) 48000 Hz: 11544 frames (240.50 ms) 88200 Hz: 21216 frames (240.54 ms) 96000 Hz: 23088 frames (240.50 ms) 176400 Hz: 42384 frames (240.27 ms) 192000 Hz: 46128 frames (240.25 ms) SBC/AAC/aptX @ 44.1kHz: Sink rate: 44100 Hz 44100 Hz: 10776 frames (244.35 ms) 48000 Hz: 11748 frames (244.75 ms) 88200 Hz: 21552 frames (244.35 ms) 96000 Hz: 23448 frames (244.25 ms) 176400 Hz: 43056 frames (244.08 ms) 192000 Hz: 46848 frames (244.00 ms) The above results were produced with the following code: import android.media.AudioAttributes; import android.media.AudioFormat; import android.media.AudioTrack; class AudioInfo { public static void main(String[] args) { int nosr = AudioTrack.getNativeOutputSampleRate(3); System.out.printf("Sink rate: %d Hz\n", nosr); int[] rates = {44100,48000,88200,96000,176400,192000}; for (int rate: rates) { AudioAttributes aa = new AudioAttributes.Builder().setFlags(256).build(); AudioFormat af = new AudioFormat.Builder().setSampleRate(rate).build(); AudioTrack at = new AudioTrack(aa, af, 4, 1, 0); int sr = at.getSampleRate(); int bs = at.getBufferSizeInFrames(); float ms = bs * (float) 1000 / sr; at.release(); System.out.printf("%d Hz: %d frames (%.2f ms)\