Fix for the huge audio latency of the SamplePlayer (>200 ms)
- fixes PulseAudio, ALSA and RtAudio driver - cleans up the driver files for better readability (mostly whitespace-related stuff) - makes ALSA and Pulseaudio actually use the global setting "audio/mix_rate" for the sample rate instead of a fixed value (RtAudio did this already)
This commit is contained in:
lonesurvivor
parent
495d059a74
commit
da6b6c2dd7
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@ -45,7 +45,7 @@ Error AudioDriverALSA::init() {
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samples_in = NULL;
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samples_out = NULL;
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mix_rate = 44100;
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mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
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output_format = OUTPUT_STEREO;
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channels = 2;
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@ -70,67 +70,62 @@ Error AudioDriverALSA::init() {
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ERR_FAIL_COND_V( status<0, ERR_CANT_OPEN );
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snd_pcm_hw_params_alloca(&hwparams);
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status = snd_pcm_hw_params_any(pcm_handle, hwparams);
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status = snd_pcm_hw_params_any(pcm_handle, hwparams);
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CHECK_FAIL( status<0 );
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status = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
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CHECK_FAIL( status<0 );
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//not interested in anything else
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status = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE);
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CHECK_FAIL( status<0 );
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//todo: support 4 and 6
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status = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2);
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CHECK_FAIL( status<0 );
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status = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &mix_rate, NULL);
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CHECK_FAIL( status<0 );
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int latency = GLOBAL_DEF("audio/output_latency",25);
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buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
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status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &buffer_size, NULL);
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// set buffer size from project settings
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status = snd_pcm_hw_params_set_buffer_size_near(pcm_handle, hwparams, &buffer_size);
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CHECK_FAIL( status<0 );
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// make period size 1/8
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period_size = buffer_size >> 3;
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status = snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, &period_size, NULL);
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CHECK_FAIL( status<0 );
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unsigned int periods=2;
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status = snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, &periods, NULL);
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CHECK_FAIL( status<0 );
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status = snd_pcm_hw_params(pcm_handle,hwparams);
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CHECK_FAIL( status<0 );
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//snd_pcm_hw_params_free(&hwparams);
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snd_pcm_sw_params_alloca(&swparams);
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status = snd_pcm_sw_params_current(pcm_handle, swparams);
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CHECK_FAIL( status<0 );
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status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, buffer_size);
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status = snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, period_size);
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CHECK_FAIL( status<0 );
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status = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 1);
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CHECK_FAIL( status<0 );
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status = snd_pcm_sw_params(pcm_handle, swparams);
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CHECK_FAIL( status<0 );
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samples_in = memnew_arr(int32_t, buffer_size*channels);
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samples_out = memnew_arr(int16_t, buffer_size*channels);
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samples_in = memnew_arr(int32_t, period_size*channels);
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samples_out = memnew_arr(int16_t, period_size*channels);
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snd_pcm_nonblock(pcm_handle, 0);
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@ -144,36 +139,28 @@ void AudioDriverALSA::thread_func(void* p_udata) {
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AudioDriverALSA* ad = (AudioDriverALSA*)p_udata;
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while (!ad->exit_thread) {
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if (!ad->active) {
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for (unsigned int i=0; i < ad->buffer_size*ad->channels; i++) {
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for (unsigned int i=0; i < ad->period_size*ad->channels; i++) {
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ad->samples_out[i] = 0;
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};
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} else {
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ad->lock();
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ad->audio_server_process(ad->buffer_size, ad->samples_in);
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ad->audio_server_process(ad->period_size, ad->samples_in);
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ad->unlock();
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for(unsigned int i=0;i<ad->buffer_size*ad->channels;i++) {
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for(unsigned int i=0;i<ad->period_size*ad->channels;i++) {
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ad->samples_out[i]=ad->samples_in[i]>>16;
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}
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};
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int todo = ad->buffer_size; // * ad->channels * 2;
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int todo = ad->period_size;
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int total = 0;
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while (todo) {
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if (ad->exit_thread)
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break;
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uint8_t* src = (uint8_t*)ad->samples_out;
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@ -184,7 +171,8 @@ void AudioDriverALSA::thread_func(void* p_udata) {
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break;
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if ( wrote == -EAGAIN ) {
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usleep(1000); //can't write yet (though this is blocking..)
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//can't write yet (though this is blocking..)
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usleep(1000);
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continue;
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}
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wrote = snd_pcm_recover(ad->pcm_handle, wrote, 0);
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@ -197,9 +185,9 @@ void AudioDriverALSA::thread_func(void* p_udata) {
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}
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continue;
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};
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total += wrote;
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todo -= wrote;
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};
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};
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@ -51,6 +51,7 @@ class AudioDriverALSA : public AudioDriverSW {
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OutputFormat output_format;
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snd_pcm_uframes_t buffer_size;
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snd_pcm_uframes_t period_size;
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int channels;
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bool active;
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@ -36,48 +36,56 @@
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Error AudioDriverPulseAudio::init() {
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active = false;
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thread_exited = false;
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exit_thread = false;
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active = false;
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thread_exited = false;
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exit_thread = false;
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pcm_open = false;
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samples_in = NULL;
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samples_out = NULL;
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mix_rate = 44100;
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mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
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output_format = OUTPUT_STEREO;
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channels = 2;
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pa_sample_spec spec;
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spec.format = PA_SAMPLE_S16LE;
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spec.channels = channels;
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spec.rate = mix_rate;
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pa_sample_spec spec;
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spec.format = PA_SAMPLE_S16LE;
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spec.channels = channels;
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spec.rate = mix_rate;
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int error_code;
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pulse = pa_simple_new(NULL, // default server
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"Godot", // application name
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PA_STREAM_PLAYBACK,
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NULL, // default device
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"Sound", // stream description
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&spec,
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NULL, // use default channel map
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NULL, // use default buffering attributes
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&error_code
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);
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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buffer_size = nearest_power_of_2(latency * mix_rate / 1000);
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if (pulse == NULL) {
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pa_buffer_attr attr;
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// set to appropriate buffer size from global settings
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attr.tlength = buffer_size;
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// set them to be automatically chosen
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attr.prebuf = (uint32_t)-1;
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attr.maxlength = (uint32_t)-1;
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attr.minreq = (uint32_t)-1;
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fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
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ERR_FAIL_COND_V(pulse == NULL, ERR_CANT_OPEN);
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}
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int error_code;
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pulse = pa_simple_new( NULL, // default server
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"Godot", // application name
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PA_STREAM_PLAYBACK,
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NULL, // default device
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"Sound", // stream description
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&spec,
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NULL, // use default channel map
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&attr, // use buffering attributes from above
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&error_code
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);
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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buffer_size = nearest_power_of_2(latency * mix_rate / 1000);
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if (pulse == NULL) {
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fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
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ERR_FAIL_COND_V(pulse == NULL, ERR_CANT_OPEN);
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}
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samples_in = memnew_arr(int32_t, buffer_size * channels);
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samples_out = memnew_arr(int16_t, buffer_size * channels);
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mutex = Mutex::create();
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thread = Thread::create(AudioDriverPulseAudio::thread_func, this);
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samples_in = memnew_arr(int32_t, buffer_size * channels);
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samples_out = memnew_arr(int16_t, buffer_size * channels);
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mutex = Mutex::create();
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thread = Thread::create(AudioDriverPulseAudio::thread_func, this);
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return OK;
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}
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@ -95,47 +103,40 @@ float AudioDriverPulseAudio::get_latency() {
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void AudioDriverPulseAudio::thread_func(void* p_udata) {
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AudioDriverPulseAudio* ad = (AudioDriverPulseAudio*)p_udata;
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AudioDriverPulseAudio* ad = (AudioDriverPulseAudio*)p_udata;
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while (!ad->exit_thread) {
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if (!ad->active) {
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for (unsigned int i=0; i < ad->buffer_size * ad->channels; i++) {
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for (unsigned int i=0; i < ad->buffer_size * ad->channels; i++) {
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ad->samples_out[i] = 0;
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}
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}
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} else {
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ad->lock();
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ad->audio_server_process(ad->buffer_size, ad->samples_in);
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ad->unlock();
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for (unsigned int i=0; i < ad->buffer_size * ad->channels;i ++) {
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ad->samples_out[i] = ad->samples_in[i] >> 16;
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for (unsigned int i=0; i < ad->buffer_size * ad->channels;i ++) {
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ad->samples_out[i] = ad->samples_in[i] >> 16;
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}
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}
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}
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// pa_simple_write always consumes the entire buffer
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// pa_simple_write always consumes the entire buffer
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int error_code;
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int byte_size = ad->buffer_size * sizeof(int16_t) * ad->channels;
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if (pa_simple_write(ad->pulse, ad->samples_out, byte_size, &error_code) < 0) {
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int error_code;
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int byte_size = ad->buffer_size * sizeof(int16_t) * ad->channels;
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if (pa_simple_write(ad->pulse, ad->samples_out, byte_size, &error_code) < 0) {
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// can't recover here
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fprintf(stderr, "PulseAudio failed and can't recover: %s\n", pa_strerror(error_code));
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ad->active = false;
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ad->exit_thread = true;
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break;
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}
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}
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// can't recover here
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fprintf(stderr, "PulseAudio failed and can't recover: %s\n", pa_strerror(error_code));
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ad->active = false;
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ad->exit_thread = true;
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break;
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}
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}
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ad->thread_exited = true;
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ad->thread_exited = true;
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}
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void AudioDriverPulseAudio::start() {
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@ -184,10 +185,10 @@ void AudioDriverPulseAudio::finish() {
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};
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memdelete(thread);
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if (mutex) {
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if (mutex) {
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memdelete(mutex);
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mutex = NULL;
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}
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mutex = NULL;
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}
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thread = NULL;
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}
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@ -195,9 +196,9 @@ void AudioDriverPulseAudio::finish() {
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AudioDriverPulseAudio::AudioDriverPulseAudio() {
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mutex = NULL;
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thread = NULL;
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pulse = NULL;
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latency=0;
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thread = NULL;
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pulse = NULL;
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latency=0;
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}
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AudioDriverPulseAudio::~AudioDriverPulseAudio() {
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@ -47,9 +47,7 @@ const char* AudioDriverRtAudio::get_name() const {
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}
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// Two-channel sawtooth wave generator.
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int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames,
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double streamTime, RtAudioStreamStatus status, void *userData ) {
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int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status, void *userData ) {
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if (status) {
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if (status & RTAUDIO_INPUT_OVERFLOW) {
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@ -64,8 +62,6 @@ int AudioDriverRtAudio::callback( void *outputBuffer, void *inputBuffer, unsigne
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AudioDriverRtAudio *self = (AudioDriverRtAudio*)userData;
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if (self->mutex->try_lock()!=OK) {
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// what should i do..
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for(unsigned int i=0;i<nBufferFrames;i++)
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buffer[i]=0;
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@ -100,61 +96,89 @@ Error AudioDriverRtAudio::init() {
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else
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output_format=OUTPUT_STEREO;
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RtAudio::StreamParameters parameters;
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parameters.deviceId = dac->getDefaultOutputDevice();
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RtAudio::StreamOptions options;
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// set the desired numberOfBuffers
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unsigned int target_number_of_buffers = 4;
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options.numberOfBuffers = target_number_of_buffers;
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// options.
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// RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). *///
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// unsigned int numberOfBuffers; /*!< Number of stream buffers. */
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// std::string streamName; /*!< A stream name (currently used only in Jack). */
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// int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */
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parameters.firstChannel = 0;
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mix_rate = GLOBAL_DEF("audio/mix_rate",44100);
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int latency = GLOBAL_DEF("audio/output_latency",25);
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unsigned int buffer_size = nearest_power_of_2( latency * mix_rate / 1000 );
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// calculate desired buffer_size, taking the desired numberOfBuffers into account (latency depends on numberOfBuffers*buffer_size)
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unsigned int buffer_size = nearest_power_of_2( latency * mix_rate / 1000 / target_number_of_buffers);
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if (OS::get_singleton()->is_stdout_verbose()) {
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print_line("audio buffer size: "+itos(buffer_size));
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}
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// bool success=false;
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while( true) {
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switch(output_format) {
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case OUTPUT_MONO: parameters.nChannels = 1; break;
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case OUTPUT_STEREO: parameters.nChannels = 2; break;
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case OUTPUT_QUAD: parameters.nChannels = 4; break;
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case OUTPUT_5_1: parameters.nChannels = 6; break;
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};
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try {
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dac->openStream( ¶meters, NULL, RTAUDIO_SINT32,
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mix_rate, &buffer_size, &callback, this,&options );
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mutex = Mutex::create(true);
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active=true;
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break;
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} catch ( RtAudioError& e ) {
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// try with less channels
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ERR_PRINT("Unable to open audio, retrying with fewer channels..");
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short int tries = 2;
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while(true) {
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while( true) {
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switch(output_format) {
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case OUTPUT_MONO: ERR_EXPLAIN("Unable to open audio."); ERR_FAIL_V( ERR_UNAVAILABLE ); break;
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case OUTPUT_STEREO: output_format=OUTPUT_MONO; break;
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case OUTPUT_QUAD: output_format=OUTPUT_STEREO; break;
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case OUTPUT_5_1: output_format=OUTPUT_QUAD; break;
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case OUTPUT_MONO: parameters.nChannels = 1; break;
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case OUTPUT_STEREO: parameters.nChannels = 2; break;
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case OUTPUT_QUAD: parameters.nChannels = 4; break;
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case OUTPUT_5_1: parameters.nChannels = 6; break;
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};
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}
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}
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try {
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dac->openStream( ¶meters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_size, &callback, this,&options );
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mutex = Mutex::create(true);
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active=true;
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break;
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} catch ( RtAudioError& e ) {
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// try with less channels
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ERR_PRINT("Unable to open audio, retrying with fewer channels..");
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switch(output_format) {
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case OUTPUT_MONO: ERR_EXPLAIN("Unable to open audio."); ERR_FAIL_V( ERR_UNAVAILABLE ); break;
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case OUTPUT_STEREO: output_format=OUTPUT_MONO; break;
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case OUTPUT_QUAD: output_format=OUTPUT_STEREO; break;
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case OUTPUT_5_1: output_format=OUTPUT_QUAD; break;
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};
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}
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}
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// 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
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if(target_number_of_buffers != options.numberOfBuffers) {
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if(tries <= 0) {
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ERR_EXPLAIN("RtAudio: Unable to set correct number of buffers.");
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ERR_FAIL_V( ERR_UNAVAILABLE );
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break;
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}
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try {
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dac->closeStream();
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} catch ( RtAudioError& e ) {
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ERR_PRINT(e.what());
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||||
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;
|
||||
}
|
||||
|
||||
|
||||
}
|
||||
|
||||
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;
|
||||
|
|
|
|||
Loading…
Reference in New Issue