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miniaudio
Commit d46e19fb authored by David Reid's avatar David Reid
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PulseAudio: Attempt to fix a deadlock. 

Public issues
  https://github.com/mackron/miniaudio/issues/877
  https://github.com/mackron/miniaudio/issues/881
parent dc423daa
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Showing with 73 additions and 29 deletions
CHANGES.md
View file @ d46e19fb
......@@ -12,6 +12,7 @@ v0.11.22 - TBD
* AAudio: The default minimum SDK version has been increased from 26 to 27 when enabling AAudio. If you need to support version 26, you can use `#define MA_AAUDIO_MIN_ANDROID_SDK_VERSION 26`.
* AAudio: Fix ma_device_get_info() implementation
* PulseAudio: Allow setting the channel map requested from PulseAudio in device configs
* PulseAudio: Attempt to fix a deadlock. This involves converting the PulseAudio main loop from blocking to non-blocking. To restore the old blocking behaviour, you can do so via the device config: `deviceConfig.pulse.blockingMainLoop = MA_TRUE`.
v0.11.21 - 2023-11-15
......
miniaudio.h
View file @ d46e19fb
......@@ -7108,6 +7108,7 @@ struct ma_device_config
const char* pStreamNamePlayback;
const char* pStreamNameCapture;
int channelMap;
ma_bool32 blockingMainLoop;
} pulse;
struct
{
......@@ -7809,7 +7810,7 @@ struct ma_device
union
{
#ifdef MA_SUPPORT_WASAPI
#ifdef MA_SUPPORT_WASAPI
struct
{
/*IAudioClient**/ ma_ptr pAudioClientPlayback;
......@@ -7849,8 +7850,8 @@ struct ma_device
void* hAvrtHandle;
ma_mutex rerouteLock;
} wasapi;
#endif
#ifdef MA_SUPPORT_DSOUND
#endif
#ifdef MA_SUPPORT_DSOUND
struct
{
/*LPDIRECTSOUND*/ ma_ptr pPlayback;
......@@ -7859,8 +7860,8 @@ struct ma_device
/*LPDIRECTSOUNDCAPTURE*/ ma_ptr pCapture;
/*LPDIRECTSOUNDCAPTUREBUFFER*/ ma_ptr pCaptureBuffer;
} dsound;
#endif
#ifdef MA_SUPPORT_WINMM
#endif
#ifdef MA_SUPPORT_WINMM
struct
{
/*HWAVEOUT*/ ma_handle hDevicePlayback;
......@@ -7878,8 +7879,8 @@ struct ma_device
ma_uint8* pIntermediaryBufferCapture;
ma_uint8* _pHeapData; /* Used internally and is used for the heap allocated data for the intermediary buffer and the WAVEHDR structures. */
} winmm;
#endif
#ifdef MA_SUPPORT_ALSA
#endif
#ifdef MA_SUPPORT_ALSA
struct
{
/*snd_pcm_t**/ ma_ptr pPCMPlayback;
......@@ -7893,17 +7894,18 @@ struct ma_device
ma_bool8 isUsingMMapPlayback;
ma_bool8 isUsingMMapCapture;
} alsa;
#endif
#ifdef MA_SUPPORT_PULSEAUDIO
#endif
#ifdef MA_SUPPORT_PULSEAUDIO
struct
{
/*pa_mainloop**/ ma_ptr pMainLoop;
/*pa_context**/ ma_ptr pPulseContext;
/*pa_stream**/ ma_ptr pStreamPlayback;
/*pa_stream**/ ma_ptr pStreamCapture;
ma_bool32 blockingMainLoop;
} pulse;
#endif
#ifdef MA_SUPPORT_JACK
#endif
#ifdef MA_SUPPORT_JACK
struct
{
/*jack_client_t**/ ma_ptr pClient;
......@@ -7912,8 +7914,8 @@ struct ma_device
float* pIntermediaryBufferPlayback; /* Typed as a float because JACK is always floating point. */
float* pIntermediaryBufferCapture;
} jack;
#endif
#ifdef MA_SUPPORT_COREAUDIO
#endif
#ifdef MA_SUPPORT_COREAUDIO
struct
{
ma_uint32 deviceObjectIDPlayback;
......@@ -7933,8 +7935,8 @@ struct ma_device
ma_bool32 isSwitchingCaptureDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
void* pNotificationHandler; /* Only used on mobile platforms. Obj-C object for handling route changes. */
} coreaudio;
#endif
#ifdef MA_SUPPORT_SNDIO
#endif
#ifdef MA_SUPPORT_SNDIO
struct
{
ma_ptr handlePlayback;
......@@ -7942,22 +7944,22 @@ struct ma_device
ma_bool32 isStartedPlayback;
ma_bool32 isStartedCapture;
} sndio;
#endif
#ifdef MA_SUPPORT_AUDIO4
#endif
#ifdef MA_SUPPORT_AUDIO4
struct
{
int fdPlayback;
int fdCapture;
} audio4;
#endif
#ifdef MA_SUPPORT_OSS
#endif
#ifdef MA_SUPPORT_OSS
struct
{
int fdPlayback;
int fdCapture;
} oss;
#endif
#ifdef MA_SUPPORT_AAUDIO
#endif
#ifdef MA_SUPPORT_AAUDIO
struct
{
/*AAudioStream**/ ma_ptr pStreamPlayback;
......@@ -7968,8 +7970,8 @@ struct ma_device
ma_aaudio_allowed_capture_policy allowedCapturePolicy;
ma_bool32 noAutoStartAfterReroute;
} aaudio;
#endif
#ifdef MA_SUPPORT_OPENSL
#endif
#ifdef MA_SUPPORT_OPENSL
struct
{
/*SLObjectItf*/ ma_ptr pOutputMixObj;
......@@ -7987,8 +7989,8 @@ struct ma_device
ma_uint8* pBufferPlayback; /* This is malloc()'d and is used for storing audio data. Typed as ma_uint8 for easy offsetting. */
ma_uint8* pBufferCapture;
} opensl;
#endif
#ifdef MA_SUPPORT_WEBAUDIO
#endif
#ifdef MA_SUPPORT_WEBAUDIO
struct
{
/* AudioWorklets path. */
......@@ -7999,8 +8001,8 @@ struct ma_device
ma_result initResult; /* Set to MA_BUSY while initialization is in progress. */
int deviceIndex; /* We store the device in a list on the JavaScript side. This is used to map our C object to the JS object. */
} webaudio;
#endif
#ifdef MA_SUPPORT_NULL
#endif
#ifdef MA_SUPPORT_NULL
struct
{
ma_thread deviceThread;
......@@ -8017,7 +8019,7 @@ struct ma_device
ma_uint64 lastProcessedFrameCapture;
ma_atomic_bool32 isStarted; /* Read and written by multiple threads. Must be used atomically, and must be 32-bit for compiler compatibility. */
} null_device;
#endif
#endif
};
};
#if defined(_MSC_VER) && !defined(__clang__)
......@@ -30471,6 +30473,7 @@ static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_confi
sampleRate = pDescriptorCapture->sampleRate;
}
pDevice->pulse.blockingMainLoop = pConfig->pulse.blockingMainLoop;
result = ma_init_pa_mainloop_and_pa_context__pulse(pDevice->pContext, pDevice->pContext->pulse.pApplicationName, pDevice->pContext->pulse.pServerName, MA_FALSE, &pDevice->pulse.pMainLoop, &pDevice->pulse.pPulseContext);
......@@ -30956,10 +30959,50 @@ static ma_result ma_device_data_loop__pulse(ma_device* pDevice)
the callbacks deal with it.
*/
while (ma_device_get_state(pDevice) == ma_device_state_started) {
resultPA = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, 1, NULL);
int block = (pDevice->pulse.blockingMainLoop) ? 1 : 0;
resultPA = ((ma_pa_mainloop_iterate_proc)pDevice->pContext->pulse.pa_mainloop_iterate)((ma_pa_mainloop*)pDevice->pulse.pMainLoop, block, NULL);
if (resultPA < 0) {
break;
}
/* If we're not blocking we need to sleep for a bit to prevent the CPU core being pinned at 100% usage. */
if (!block) {
ma_uint32 sleepTimeInMilliseconds;
/*
My original idea was to sleep for an amount of time proportionate to the configured period size, but I
wasn't able to figure out how to make this work without glitching. Instead I'm just going to hardcode
it to 1ms and move on.
*/
#if 0
{
if (((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)((ma_pa_stream*)pDevice->pulse.pStreamPlayback) == 0) {
/* The amount of time we spend sleeping should be proportionate to the size of a period. */
if (pDevice->type == ma_device_type_playback) {
sleepTimeInMilliseconds = (pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalSampleRate) / 1000;
} else {
sleepTimeInMilliseconds = (pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.internalSampleRate) / 1000;
}
/*
At this point the sleep time is equal to the period size in milliseconds. I'm going to divide this by 2
in an attempt to reduce latency as a result of sleeping.
*/
sleepTimeInMilliseconds /= 2;
/* Clamp the sleep time to within reasonable values just in case. */
sleepTimeInMilliseconds = ma_clamp(sleepTimeInMilliseconds, 1, 10);
}
}
#else
{
sleepTimeInMilliseconds = 1;
}
#endif
ma_sleep(sleepTimeInMilliseconds);
}
}
/* NOTE: Don't stop the device here. It'll be done at a higher level. */
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