summaryrefslogtreecommitdiffstats
path: root/audio/out/ao_opensles.c
Commit message (Collapse)AuthorAgeFilesLines
* ao_opensles: add guards for sample rate to useTom Yan2021-11-191-0/+2
| | | | | | Upstream "Wilhelm" (the Android OpenSLES implementation) supports only 8000 <= rate <= 192000. Make sure mpv resamples the audio when necessary.
* audio: redo internal AO APIwm42020-06-011-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | This affects "pull" AOs only: ao_alsa, ao_pulse, ao_openal, ao_pcm, ao_lavc. There are changes to the other AOs too, but that's only about renaming ao_driver.resume to ao_driver.start. ao_openal is broken because I didn't manage to fix it, so it exits with an error message. If you want it, why don't _you_ put effort into it? I see no reason to waste my own precious lifetime over this (I realize the irony). ao_alsa loses the poll() mechanism, but it was mostly broken and didn't really do what it was supposed to. There doesn't seem to be anything in the ALSA API to watch the playback status without polling (unless you want to use raw UNIX signals). No idea if ao_pulse is correct, or whether it's subtly broken now. There is no documentation, so I can't tell what is correct, without reverse engineering the whole project. I recommend using ALSA. This was supposed to be just a simple fix, but somehow it expanded scope like a train wreck. Very high chance of regressions, but probably only for the AOs listed above. The rest you can figure out from reading the diff.
* options: change option macros and all option declarationswm42020-03-181-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Change all OPT_* macros such that they don't define the entire m_option initializer, and instead expand only to a part of it, which sets certain fields. This requires changing almost every option declaration, because they all use these macros. A declaration now always starts with {"name", ... followed by designated initializers only (possibly wrapped in macros). The OPT_* macros now initialize the .offset and .type fields only, sometimes also .priv and others. I think this change makes the option macros less tricky. The old code had to stuff everything into macro arguments (and attempted to allow setting arbitrary fields by letting the user pass designated initializers in the vararg parts). Some of this was made messy due to C99 and C11 not allowing 0-sized varargs with ',' removal. It's also possible that this change is pointless, other than cosmetic preferences. Not too happy about some things. For example, the OPT_CHOICE() indentation I applied looks a bit ugly. Much of this change was done with regex search&replace, but some places required manual editing. In particular, code in "obscure" areas (which I didn't include in compilation) might be broken now. In wayland_common.c the author of some option declarations confused the flags parameter with the default value (though the default value was also properly set below). I fixed this with this change.
* 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_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_opensles: let cfg_frames_per_buffer accept buffer size up to 0.5s at 192kHzTom Yan2018-04-051-1/+1
|
* 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)\n", sr, bs, ms); } } } Therefore bumping the device buffer size to 250ms.
* ao_opensles: do one buffer onlyTom Yan2018-04-051-15/+8
| | | | Doing two buffers causes stutters upon (re)start of playback on Android O for all kinds of sinks.
* ao_opensles: re-flow interface/configuration retrievalJan Ekström2018-03-241-9/+18
| | | | | This manages to make the code more readable. Thanks to MakeGho@IRCnet for the snippet on which this was based.
* ao_opensles: fix audio sync using device latency extensionAman Gupta2018-03-231-3/+20
|
* ao_opensles: bump device buffer size to 200mstomty892018-03-071-1/+1
| | | While the soft buffer size is already by default 200ms, it is not enough to guarantee dropout-free playback on Bluetooth audio. Bumping the device buffer size to the same value seems to suffice.
* ao_opensles: remove set_play_state()tomty892018-03-071-10/+1
| | | Set play state to playing in init() instead. We no longer touch the play state afterwards.
* ao_opensles: clear buffer queue in reset()tomty892018-03-071-1/+2
| | | Avoid resume() from causing SL_RESULT_BUFFER_INSUFFICIENT ("Failed to Enqueue: 7" when seek or resume from pause).
* audio: fix annyoing af_get_best_sample_formats() definitionwm42018-01-251-1/+1
| | | | | | | | | | | | | | | | The af_get_best_sample_formats() function had an argument of int[AF_FORMAT_COUNT], which is slightly incorrect, because it's 0 terminated and should in theory have AF_FORMAT_COUNT+1 entries. It won't actually write this many formats (since some formats are fundamentally incompatible), but it still feels annoying and incorrect. So fix it, and require that callers pass an AF_FORMAT_COUNT+1 array. Note that the array size has no meaning in C function arguments (just another issue with C static arrays being weird and stupid), so get rid of it completely. Not changing the af_lavcac3enc use, since that is rewritten in another branch anyway.
* options: remove deprecated sub-option handling for --vo and --aowm42016-11-251-1/+1
| | | | | | | | Long planned. Leads to some sanity. There still are some rather gross things. Especially g_groups is ugly, and a hack that can hopefully be removed. (There is a plan for it, but whether it's implemented depends on how much energy is left.)
* options: deprecate suboptions for the remaining AO/VOswm42016-09-051-0/+1
|
* ao_opensles: remove 32bit audioJosh de Kock2016-05-221-1/+0
| | | | It's unsupported by android, and can cause problems when trying to play 32bit audio. Removing 32bit fixes it by forcing 16 bit or 8 bit audio.
* ao: initial OpenSL ES supportIlya Zhuravlev2016-02-271-0/+250
OpenSL ES is used on Android. At the moment only stereo output is supported. Two options are supported: 'frames-per-buffer' and 'sample-rate'. To get better latency the user of libmpv should pass values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER) and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).