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* ao_audiounit: rename pause function to resetGravatar Josh Lehman2018-09-30
| | | | | 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 errorGravatar Jan Ekström2018-09-29
| | | | | The ALSA state generally can tell us more information in case we get an unexpected error.
* ao_alsa: handle XRUNs separately from other errorsGravatar Jan Ekström2018-09-29
| | | | | | | | | 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 readyGravatar Jan Ekström2018-09-29
| | | | | | | | | | | 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 portsGravatar Niklas Haas2018-09-26
| | | | | This prevents ao_jack from auto-connecting to MIDI ports (or other, hypothetical future port types).
* ao_pulse: fix tlength calculationGravatar Tom Yan2018-09-01
| | | | also remove the now unused non-sensical af_fmt_seconds_to_bytes.
* Revert "ao_openal: enable building on OSX"Gravatar Michael Hoang2018-08-26
| | | | | | 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 Apple's OpenAL support is ridiculously out of date, revert back to just using OpenAL Soft on macOS (fixes #4645).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 8Gravatar Tom Yan2018-08-13
| | | | | | | | | | | | | | 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 settingGravatar Tom Yan2018-08-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 outputGravatar Tom Yan2018-08-05
| | | | | 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()Gravatar Jan Ekström2018-06-04
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* ao_alsa: replace snd_pcm_status() with snd_pcm_avail() in get_space()Gravatar Muhammad Faiz2018-06-04
| | | | | | | | | | 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-optionalGravatar wm42018-05-03
| | | | Makes it easier to not break the build by confusing the ifdeffery.
* encode: get rid of the output packet queueGravatar wm42018-05-03
| | | | | | | | | | | | 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 handlingGravatar wm42018-05-03
| | | | | This effectively makes --ocopyts the default. The --ocopyts option itself is also removed, because it's redundant.
* encode: rewrite half of itGravatar wm42018-04-29
| | | | | | | | | | | | | 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: cosmeticsGravatar wm42018-04-20
| | | | Mostly whitespace changes; some semantic preserving transformations.
* ao_alsa: actually report underruns to userGravatar wm42018-04-15
| | | | | | | | | 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 sizeGravatar wm42018-04-15
| | | | | | Same deal as with the previous commit for ALSA. Untested.
* ao_alsa: reduce requested buffer sizeGravatar wm42018-04-15
| | | | | | 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 sizeGravatar wm42018-04-15
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* ao_openal: document the muted↔gain conversionGravatar Jan Ekström2018-04-15
| | | | This struck me as odd for a moment, so adding a comment.
* ao/openal: Add option to set buffering characteristicsGravatar LAGonauta2018-04-15
| | | | | | | | | 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 selectionGravatar LAGonauta2018-04-15
| | | | Also re-added floating-point support.
* ao/openal: Add OpenAL Soft extension to get the correct latencyGravatar LAGonauta2018-04-15
| | | | | | | | 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 outputGravatar LAGonauta2018-04-15
| | | | | | | 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 supportGravatar LAGonauta2018-04-15
| | | | | 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 outputGravatar LAGonauta2018-04-15
| | | | Floating point audio not supported on this commit.
* ao_opensles: let cfg_frames_per_buffer accept buffer size up to 0.5s at 192kHzGravatar Tom Yan2018-04-05
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* ao_opensles: remove useless cfg_sample_rateGravatar Tom Yan2018-04-05
| | | | We should always use the ao-neutral --audio-samplerate option.
* ao_opensles: bump device buffer size to 250msGravatar Tom Yan2018-04-05
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 onlyGravatar Tom Yan2018-04-05
| | | | Doing two buffers causes stutters upon (re)start of playback on Android O for all kinds of sinks.
* ao_opensles: re-flow interface/configuration retrievalGravatar Jan Ekström2018-03-24
| | | | | 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 extensionGravatar Aman Gupta2018-03-23
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* ao_sdl: fix default buffer sizeGravatar wm42018-03-08
| | | | | | | | If you set desired.samples to 0, SDL will return a default buffer size on obtained.samples. This was broken, because ceil_power_of_two(0) returns 1. Since 0 is usually not considered a power of two, this is probably correct, but we still want to set desired.samples to 0 in this case.
* ao: do not allow actual buffer size of 0Gravatar wm42018-03-08
| | | | | | | | | You can use --audio-buffer=0 to minimize the audio buffer size. But if the AO reports no device buffer size (like e.g. ao_jack does), then the buffer size is actually 0, and playback can never work properly. Make it fallback to a size of 1, which is unlikely to work properly, but you get what you asked for, instead of a freeze.
* ao_opensles: bump device buffer size to 200msGravatar tomty892018-03-07
| | | 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()Gravatar tomty892018-03-07
| | | Set play state to playing in init() instead. We no longer touch the play state afterwards.
* ao_opensles: clear buffer queue in reset()Gravatar tomty892018-03-07
| | | Avoid resume() from causing SL_RESULT_BUFFER_INSUFFICIENT ("Failed to Enqueue: 7" when seek or resume from pause).
* build: drop support for SDL1Gravatar wm42018-02-13
| | | | | For some reason it was supported for ao_sdl because we've only used SDL1 API.
* ao_null: add --ao-null-format option for debuggingGravatar wm42018-01-30
| | | | Helpful especially to test spdif fallback and so on.
* ao_alsa: reduce verbosity at -vGravatar wm42018-01-25
| | | | | | | Always make the hw params dump function use MSGL_DEBUG, and remove the MSGL_V use. That means you need -v -v to see them. The detailed information is usually not very interesting, so this reduces the log noise.
* audio: fix annyoing af_get_best_sample_formats() definitionGravatar wm42018-01-25
| | | | | | | | | | | | | | | | 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.
* Fix undefined preprocessor behaviorGravatar wm42018-01-18
| | | | | | | | | | | This commit eliminates the following clang warning: warning: macro expansion producing 'defined' has undefined behavior [-Wexpansion-to-defined] Going by the clang commit message, this seems to be explicitly specified as UB by the standard, and they added this warning because MSVC apparently results in different behavior. Whatever, we can just avoid the warning with some small changes.
* Fix various typos in log messagesGravatar Nicolas F2017-12-03
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* ao: minor simplification to gain processing codeGravatar wm42017-11-30
| | | | | Cosmetic move of a variable, and consider an adjustment below 1/256 or so not worth applying (even in the float case).
* ao: simplify hack for float atomicsGravatar wm42017-11-30
| | | | | | | stdatomic.h defines no atomic_float typedef. We can't just use _Atomic unconditionally, because we support compilers without C11 atomics. So just create a custom atomic_float typedef in the wrapper, which uses _Atomic in the C11 code path.
* audio: add audio softvol processing to AOGravatar wm42017-11-29
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This does what af_volume used to do. Since we couldn't relicense it, just rewrite it. Since we don't have a new filter mechanism yet, and the libavfilter is too inconvenient, do applying the volume gain in ao.c directly. This is done before handling the audio data to the driver. Since push.c runs a separate thread, and pull.c is called asynchronously from the audio driver's thread, the volume value needs to be synchronized. There's no existing central mutex, so do some shit with atomics. Since there's no atomic_float type predefined (which is at least needed when using the legacy wrapper), do some nonsense about reinterpret casting the float value to an int for the purpose of atomic access. Not sure if using memcpy() is undefined behavior, but for now I don't care. The advantage of not using a filter is lower complexity (no filter auto insertion), and lower latency (gain processing is done after our internal audio buffer of at least 200ms). Disavdantages include inability to use native volume control _before_ other filters with custom filter chains, and the need to add new processing for each new sample type. Since this doesn't reuse any of the old GPL code, nor does indirectly rely on it, volume and replaygain handling now works in LGPL mode. How to process the gain is inspired by libavfilter's af_volume (LGPL). In particular, we use exactly the same rounding, and we quantize processing for integer sample types by 256 steps. Some of libavfilter's copyright may or may not apply, but I think not, and it's the same license anyway.
* ao_alsa: change license to LGPLGravatar wm42017-11-23
| | | | | | | | | | | | | | | | | | | | | | | | Looks like this is covered by LGPL relicensing agreements now. Notes about contributors who could not be reached or who didn't agree: Commit 7fccb6486e has tons of mp_msg changes look like they are not copyrightable (even if they were, all mp_msg calls were rewritten in mpv times again). The additional play() change looks suspicious, but the function was rewritten several times anyway (first time after that commit in 4f40ec312). Commit 89ed1748ae was rewritten in commit 325311af3 and then again several times after that. Basically all this code is unnecessary in modern mpv and has been removed. No code survived from the following commits: 4d31c3c53, 61ecf838f2, d38968bd, 4deb67c3f. At least two cosmetic typo fixes are not considered as well. Commit 22bb046ad is reverted (this wasn't a valid warning anyway, just a C++-ism icc applied to C). Using the constants is nicer, but at least I don't have to decide whether that change was copyrightable.
* ao_alsa: don't convert twice on retryGravatar wm42017-11-23
| | | | Obscure corner case.
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