Merge pull request #7425 from lonesurvivor/master

Fix for the huge audio latency (>200 ms) for the …

drivers/alsa/audio_driver_alsa.cpp

@@ -45,7 +45,7 @@ Error AudioDriverALSA::init() {
 	samples_in = NULL;
 	samples_out = NULL;
 
-	mix_rate = 44100;
+	mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
 	output_format = OUTPUT_STEREO;
 	channels = 2;
 
@@ -70,67 +70,62 @@ Error AudioDriverALSA::init() {
 	ERR_FAIL_COND_V( status<0, ERR_CANT_OPEN );
 
 	snd_pcm_hw_params_alloca(&hwparams);
-	status = snd_pcm_hw_params_any(pcm_handle, hwparams);
 
+	status = snd_pcm_hw_params_any(pcm_handle, hwparams);
 	CHECK_FAIL( status<0 );
 
 	status = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
-
 	CHECK_FAIL( status<0 );
 
 	//not interested in anything else
 	status = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE);
-
 	CHECK_FAIL( status<0 );
 
 	//todo: support 4 and 6
 	status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
-
 	CHECK_FAIL( status<0 );
 
 	status = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &mix_rate, NULL);
-
-
 	CHECK_FAIL( status<0 );
 
 	int latency = GLOBAL_DEF("audio/output_latency",25);
 	buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
 
-	status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &buffer_size, NULL);
-
+	// set buffer size from project settings
+	status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
+	CHECK_FAIL( status<0 );
 
+	// make period size 1/8 
+	period_size = buffer_size >> 3;
+	status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, NULL);
 	CHECK_FAIL( status<0 );
 
 	unsigned int periods=2;
 	status = snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &periods, NULL);
-
 	CHECK_FAIL( status<0 );
 
 	status = snd_pcm_hw_params(pcm_handle,hwparams);
-
 	CHECK_FAIL( status<0 );
 
 	//snd_pcm_hw_params_free(&hwparams);
 
 
 	snd_pcm_sw_params_alloca(&swparams);
+
 	status = snd_pcm_sw_params_current(pcm_handle, swparams);
 	CHECK_FAIL( status<0 );
 
-	status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, buffer_size);
-
+	status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, period_size);
 	CHECK_FAIL( status<0 );
 
 	status = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
-
 	CHECK_FAIL( status<0 );
 
 	status = snd_pcm_sw_params(pcm_handle, swparams);
-
 	CHECK_FAIL( status<0 );
 
-	samples_in = memnew_arr(int32_t, buffer_size*channels);
-	samples_out = memnew_arr(int16_t, buffer_size*channels);
+	samples_in = memnew_arr(int32_t, period_size*channels);
+	samples_out = memnew_arr(int16_t, period_size*channels);
 
 	snd_pcm_nonblock(pcm_handle, 0);
 
@@ -144,36 +139,28 @@ void AudioDriverALSA::thread_func(void* p_udata) {
 
 	AudioDriverALSA* ad = (AudioDriverALSA*)p_udata;
 
-
 	while (!ad->exit_thread) {
-
-
 		if (!ad->active) {
-
-			for (unsigned int i=0; i < ad->buffer_size*ad->channels; i++) {
-
+			for (unsigned int i=0; i < ad->period_size*ad->channels; i++) {
 				ad->samples_out[i] = 0;
 			};
 		} else {
-
 			ad->lock();
 
-			ad->audio_server_process(ad->buffer_size, ad->samples_in);
+			ad->audio_server_process(ad->period_size, ad->samples_in);
 
 			ad->unlock();
 
-			for(unsigned int i=0;i<ad->buffer_size*ad->channels;i++) {
-
+			for(unsigned int i=0;i<ad->period_size*ad->channels;i++) {
 				ad->samples_out[i]=ad->samples_in[i]>>16;
 			}
 		};
 
 
-		int todo = ad->buffer_size; // * ad->channels * 2;
+		int todo = ad->period_size;
 		int total = 0;
 
 		while (todo) {
-
 			if (ad->exit_thread)
 				break;
 			uint8_t* src = (uint8_t*)ad->samples_out;
@@ -184,7 +171,8 @@ void AudioDriverALSA::thread_func(void* p_udata) {
 					break;
 
 				if ( wrote == -EAGAIN ) {
-					usleep(1000); //can't write yet (though this is blocking..)
+					//can't write yet (though this is blocking..)
+					usleep(1000); 
 					continue;
 				}
 				wrote = snd_pcm_recover(ad->pcm_handle, wrote, 0);
@@ -197,9 +185,9 @@ void AudioDriverALSA::thread_func(void* p_udata) {
 				}
 				continue;
 			};
+
 			total += wrote;
 			todo -= wrote;
-
 		};
 	};
 

drivers/alsa/audio_driver_alsa.h

@@ -51,6 +51,7 @@ class AudioDriverALSA : public AudioDriverSW {
 	OutputFormat output_format;
 
 	snd_pcm_uframes_t buffer_size;
+	snd_pcm_uframes_t period_size;
 	int channels;
 
 	bool active;

drivers/pulseaudio/audio_driver_pulseaudio.cpp

@@ -36,48 +36,56 @@
 
 Error AudioDriverPulseAudio::init() {
 
-    active = false;
-    thread_exited = false;
-    exit_thread = false;
+	active = false;
+	thread_exited = false;
+	exit_thread = false;
 	pcm_open = false;
 	samples_in = NULL;
 	samples_out = NULL;
 
-	mix_rate = 44100;
+	mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
 	output_format = OUTPUT_STEREO;
 	channels = 2;
 
-    pa_sample_spec spec;
-    spec.format = PA_SAMPLE_S16LE;
-    spec.channels = channels;
-    spec.rate = mix_rate;
-
-    int error_code;
-    pulse = pa_simple_new(NULL,         // default server
-                          "Godot",      // application name
-                          PA_STREAM_PLAYBACK,
-                          NULL,         // default device
-                          "Sound",      // stream description
-                          &spec,
-                          NULL,         // use default channel map
-                          NULL,         // use default buffering attributes
-                          &error_code
-                          );
-
-    if (pulse == NULL) {
-
-        fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
-        ERR_FAIL_COND_V(pulse == NULL, ERR_CANT_OPEN);
-    }
+	pa_sample_spec spec;
+	spec.format = PA_SAMPLE_S16LE;
+	spec.channels = channels;
+	spec.rate = mix_rate;
+
+	int latency = GLOBAL_DEF("audio/output_latency", 25);
+	buffer_size = nearest_power_of_2(latency * mix_rate / 1000);
+
+	pa_buffer_attr attr;
+	// set to appropriate buffer size from global settings
+	attr.tlength = buffer_size;
+	// set them to be automatically chosen
+	attr.prebuf = (uint32_t)-1;
+	attr.maxlength = (uint32_t)-1;
+	attr.minreq = (uint32_t)-1;
+
+	int error_code;
+	pulse = pa_simple_new(	NULL,         // default server
+                          	"Godot",      // application name
+                          	PA_STREAM_PLAYBACK,
+                          	NULL,         // default device
+                          	"Sound",      // stream description
+                          	&spec,
+                          	NULL,         // use default channel map
+                          	&attr,         // use buffering attributes from above
+                          	&error_code
+                          	);
+
+	if (pulse == NULL) {
+		fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
+		ERR_FAIL_COND_V(pulse == NULL, ERR_CANT_OPEN);
+	}
 
-    int latency = GLOBAL_DEF("audio/output_latency", 25);
-    buffer_size = nearest_power_of_2(latency * mix_rate / 1000);
 
-    samples_in = memnew_arr(int32_t, buffer_size * channels);
-    samples_out = memnew_arr(int16_t, buffer_size * channels);
+	samples_in = memnew_arr(int32_t, buffer_size * channels);
+	samples_out = memnew_arr(int16_t, buffer_size * channels);
 
-    mutex = Mutex::create();
-    thread = Thread::create(AudioDriverPulseAudio::thread_func, this);
+	mutex = Mutex::create();
+	thread = Thread::create(AudioDriverPulseAudio::thread_func, this);
 
 	return OK;
 }
@@ -95,47 +103,40 @@ float AudioDriverPulseAudio::get_latency() {
 
 void AudioDriverPulseAudio::thread_func(void* p_udata) {
 
-    AudioDriverPulseAudio* ad = (AudioDriverPulseAudio*)p_udata;
+	AudioDriverPulseAudio* ad = (AudioDriverPulseAudio*)p_udata;
 
 	while (!ad->exit_thread) {
-
 		if (!ad->active) {
-
-            for (unsigned int i=0; i < ad->buffer_size * ad->channels; i++) {
-
+			for (unsigned int i=0; i < ad->buffer_size * ad->channels; i++) {
 				ad->samples_out[i] = 0;
-            }
+			}
 
 		} else {
-
 			ad->lock();
 
 			ad->audio_server_process(ad->buffer_size, ad->samples_in);
 
 			ad->unlock();
 
-            for (unsigned int i=0; i < ad->buffer_size * ad->channels;i ++) {
-
-                ad->samples_out[i] = ad->samples_in[i] >> 16;
+			for (unsigned int i=0; i < ad->buffer_size * ad->channels;i ++) {
+				ad->samples_out[i] = ad->samples_in[i] >> 16;
 			}
-        }
-
-        // pa_simple_write always consumes the entire buffer
-
-        int error_code;
-        int byte_size = ad->buffer_size * sizeof(int16_t) * ad->channels;
-        if (pa_simple_write(ad->pulse, ad->samples_out, byte_size, &error_code) < 0) {
+        	}
 
-            // can't recover here
-            fprintf(stderr, "PulseAudio failed and can't recover: %s\n", pa_strerror(error_code));
-            ad->active = false;
-            ad->exit_thread = true;
-            break;
-        }
+        	// pa_simple_write always consumes the entire buffer
 
-    }
-
-    ad->thread_exited = true;
+		int error_code;
+		int byte_size = ad->buffer_size * sizeof(int16_t) * ad->channels;
+		if (pa_simple_write(ad->pulse, ad->samples_out, byte_size, &error_code) < 0) {
+			// can't recover here
+			fprintf(stderr, "PulseAudio failed and can't recover: %s\n", pa_strerror(error_code));
+			ad->active = false;
+			ad->exit_thread = true;
+			break;
+		}
+    	}
+
+	ad->thread_exited = true;
 }
 
 void AudioDriverPulseAudio::start() {
@@ -184,10 +185,10 @@ void AudioDriverPulseAudio::finish() {
 	};
 
 	memdelete(thread);
-    if (mutex) {
+	if (mutex) {
 		memdelete(mutex);
-        mutex = NULL;
-    }
+		mutex = NULL;
+	}
 
 	thread = NULL;
 }
@@ -195,9 +196,9 @@ void AudioDriverPulseAudio::finish() {
 AudioDriverPulseAudio::AudioDriverPulseAudio() {
 
 	mutex = NULL;
-    thread = NULL;
-    pulse = NULL;
-    latency=0;
+	thread = NULL;
+	pulse = NULL;
+	latency=0;
 }
 
 AudioDriverPulseAudio::~AudioDriverPulseAudio() {

drivers/rtaudio/audio_driver_rtaudio.cpp

@@ -47,9 +47,7 @@ const char* AudioDriverRtAudio::get_name() const {
 
 }
 
-// Two-channel sawtooth wave generator.
-int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
-	double streamTime, RtAudioStreamStatus status, void *userData ) {
+int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status, void *userData ) {
 
 	if (status) {
 		if (status & RTAUDIO_INPUT_OVERFLOW) {
@@ -64,8 +62,6 @@ int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigne
 	AudioDriverRtAudio *self = (AudioDriverRtAudio*)userData;
 
 	if (self->mutex->try_lock()!=OK) {
-
-
 		// what should i do..
 		for(unsigned int i=0;i<nBufferFrames;i++)
 			buffer[i]=0;
@@ -100,61 +96,89 @@ Error AudioDriverRtAudio::init() {
 	else
 		output_format=OUTPUT_STEREO;
 
-
 	RtAudio::StreamParameters parameters;
 	parameters.deviceId = dac->getDefaultOutputDevice();
 	RtAudio::StreamOptions options;
+
+	// set the desired numberOfBuffers
+	unsigned int target_number_of_buffers = 4;
+	options.numberOfBuffers = target_number_of_buffers;
+
 //	options.
 //	RtAudioStreamFlags flags;      /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). *///
 //	unsigned int numberOfBuffers;  /*!< Number of stream buffers. */
 //	std::string streamName;        /*!< A stream name (currently used only in Jack). */
 //	int priority;                  /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
 
-
 	parameters.firstChannel = 0;
 	mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
 
 	int latency = GLOBAL_DEF("audio/output_latency",25);
-	unsigned int buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
+	// calculate desired buffer_size, taking the desired numberOfBuffers into account (latency depends on numberOfBuffers*buffer_size)
+	unsigned int buffer_size = nearest_power_of_2( latency * mix_rate / 1000 / target_number_of_buffers);
+
 	if (OS::get_singleton()->is_stdout_verbose()) {
 		print_line("audio buffer size: "+itos(buffer_size));
 	}
 
-//	bool success=false;
-
-	while( true) {
-
-		switch(output_format) {
-
-			case OUTPUT_MONO: parameters.nChannels = 1; break;
-			case OUTPUT_STEREO: parameters.nChannels = 2; break;
-			case OUTPUT_QUAD: parameters.nChannels = 4; break;
-			case OUTPUT_5_1: parameters.nChannels = 6; break;
-		};
+	short int tries = 2;
 
+	while(true) {
+		while( true) {
+			switch(output_format) {
+				case OUTPUT_MONO: parameters.nChannels = 1; break;
+				case OUTPUT_STEREO: parameters.nChannels = 2; break;
+				case OUTPUT_QUAD: parameters.nChannels = 4; break;
+				case OUTPUT_5_1: parameters.nChannels = 6; break;
+			};
 
-		try {
-			dac->openStream( &parameters, NULL, RTAUDIO_SINT32,
-			    mix_rate, &buffer_size, &callback, this,&options );
-			mutex = Mutex::create(true);
-			active=true;
+			try {
+				dac->openStream( &parameters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_size, &callback, this,&options );
+				mutex = Mutex::create(true);
+				active=true;
+
+				break;
+			} catch ( RtAudioError& e ) {
+				// try with less channels
+				ERR_PRINT("Unable to open audio, retrying with fewer channels..");
+
+				switch(output_format) {
+					case OUTPUT_MONO: ERR_EXPLAIN("Unable to open audio."); ERR_FAIL_V( ERR_UNAVAILABLE ); break;
+					case OUTPUT_STEREO: output_format=OUTPUT_MONO; break;
+					case OUTPUT_QUAD: output_format=OUTPUT_STEREO; break;
+					case OUTPUT_5_1: output_format=OUTPUT_QUAD; break;
+				};
+			}
+		}
 
+		// compare actual numberOfBuffers with the desired one. If not equal, close and reopen the stream with adjusted buffer size, so the desired output_latency is still correct
+		if(target_number_of_buffers != options.numberOfBuffers) {
+			if(tries <= 0) {
+				ERR_EXPLAIN("RtAudio: Unable to set correct number of buffers.");
+				ERR_FAIL_V( ERR_UNAVAILABLE ); 
+				break;
+			}
+			
+			try {
+				dac->closeStream();
+			} catch ( RtAudioError& e ) {
+				ERR_PRINT(e.what());
+				ERR_FAIL_V( ERR_UNAVAILABLE );
+				break;
+			}
+			if (OS::get_singleton()->is_stdout_verbose()) 
+				print_line("RtAudio: Desired number of buffers (" + itos(target_number_of_buffers) + ") not available. Using " + itos(options.numberOfBuffers) + " instead. Reopening stream with adjusted buffer_size.");
+
+			// new buffer size dependent on the ratio between set and actual numberOfBuffers
+			buffer_size = buffer_size / (options.numberOfBuffers / target_number_of_buffers);
+			target_number_of_buffers = options.numberOfBuffers;
+			tries--;
+		} else {
 			break;
-        } catch ( RtAudioError& e ) {
-			// try with less channels
-
-			ERR_PRINT("Unable to open audio, retrying with fewer channels..");
-
-			switch(output_format) {
-
-				case OUTPUT_MONO: ERR_EXPLAIN("Unable to open audio."); ERR_FAIL_V( ERR_UNAVAILABLE ); break;
-				case OUTPUT_STEREO: output_format=OUTPUT_MONO; break;
-				case OUTPUT_QUAD: output_format=OUTPUT_STEREO; break;
-				case OUTPUT_5_1: output_format=OUTPUT_QUAD; break;
-			};
 		}
-	}
+			
 
+	}
 
 	return OK;
 }
@@ -190,7 +214,6 @@ void AudioDriverRtAudio::unlock() {
 
 void AudioDriverRtAudio::finish() {
 
-
 	 if ( active && dac->isStreamOpen() )
 		 dac->closeStream();
 	 if (mutex)
@@ -203,6 +226,7 @@ void AudioDriverRtAudio::finish() {
 
 AudioDriverRtAudio::AudioDriverRtAudio()
 {
+
 	mutex=NULL;
 	mix_rate=44100;
 	output_format=OUTPUT_STEREO;