Corrections to audio buffer size calculations
This commit is contained in:
parent
06d7e36898
commit
f231eadc9e
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@ -31,6 +31,7 @@
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#ifdef ALSA_ENABLED
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#include "os/os.h"
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#include "project_settings.h"
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#include <errno.h>
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@ -44,7 +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 = GLOBAL_DEF("audio/mix_rate", 44100);
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mix_rate = GLOBAL_DEF("audio/mix_rate", DEFAULT_MIX_RATE);
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speaker_mode = SPEAKER_MODE_STEREO;
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channels = 2;
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@ -86,19 +87,25 @@ Error AudioDriverALSA::init() {
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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 = closest_power_of_2(latency * mix_rate / 1000);
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// In ALSA the period size seems to be the one that will determine the actual latency
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// Ref: https://www.alsa-project.org/main/index.php/FramesPeriods
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unsigned int periods = 2;
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int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
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buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
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buffer_size = buffer_frames * periods;
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period_size = buffer_frames;
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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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if (OS::get_singleton()->is_stdout_verbose()) {
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print_line("audio buffer frames: " + itos(period_size) + " calculated latency: " + itos(period_size * 1000 / mix_rate) + "ms");
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}
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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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@ -51,6 +51,7 @@ class AudioDriverALSA : public AudioDriver {
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unsigned int mix_rate;
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SpeakerMode speaker_mode;
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snd_pcm_uframes_t buffer_frames;
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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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@ -33,6 +33,7 @@
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#include <pulse/error.h>
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#include "os/os.h"
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#include "project_settings.h"
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Error AudioDriverPulseAudio::init() {
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@ -44,7 +45,7 @@ Error AudioDriverPulseAudio::init() {
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samples_in = NULL;
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samples_out = NULL;
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mix_rate = GLOBAL_DEF("audio/mix_rate", 44100);
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mix_rate = GLOBAL_DEF("audio/mix_rate", DEFAULT_MIX_RATE);
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speaker_mode = SPEAKER_MODE_STEREO;
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channels = 2;
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@ -53,12 +54,17 @@ Error AudioDriverPulseAudio::init() {
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spec.channels = channels;
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spec.rate = mix_rate;
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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buffer_size = closest_power_of_2(latency * mix_rate / 1000);
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int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
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buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
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buffer_size = buffer_frames * channels;
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if (OS::get_singleton()->is_stdout_verbose()) {
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print_line("audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
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}
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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 to appropriate buffer length (in bytes) from global settings
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attr.tlength = buffer_size * sizeof(int16_t);
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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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@ -80,8 +86,8 @@ Error AudioDriverPulseAudio::init() {
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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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samples_in = memnew_arr(int32_t, buffer_size);
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samples_out = memnew_arr(int16_t, buffer_size);
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mutex = Mutex::create();
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thread = Thread::create(AudioDriverPulseAudio::thread_func, this);
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@ -106,18 +112,18 @@ void AudioDriverPulseAudio::thread_func(void *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; 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->buffer_frames, 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->buffer_size; 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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@ -125,7 +131,7 @@ void AudioDriverPulseAudio::thread_func(void *p_udata) {
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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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int byte_size = ad->buffer_size * sizeof(int16_t);
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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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@ -175,13 +181,20 @@ void AudioDriverPulseAudio::finish() {
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exit_thread = true;
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Thread::wait_to_finish(thread);
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if (pulse)
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if (pulse) {
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pa_simple_free(pulse);
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pulse = NULL;
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}
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if (samples_in) {
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memdelete_arr(samples_in);
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samples_in = NULL;
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}
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if (samples_out) {
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memdelete_arr(samples_out);
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};
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samples_out = NULL;
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}
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memdelete(thread);
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if (mutex) {
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@ -194,10 +207,15 @@ void AudioDriverPulseAudio::finish() {
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AudioDriverPulseAudio::AudioDriverPulseAudio() {
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samples_in = NULL;
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samples_out = NULL;
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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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buffer_frames = 0;
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buffer_size = 0;
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channels = 0;
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}
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AudioDriverPulseAudio::~AudioDriverPulseAudio() {
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@ -51,6 +51,7 @@ class AudioDriverPulseAudio : public AudioDriver {
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unsigned int mix_rate;
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SpeakerMode speaker_mode;
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unsigned int buffer_frames;
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unsigned int buffer_size;
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int channels;
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@ -107,14 +107,13 @@ Error AudioDriverRtAudio::init() {
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options.numberOfBuffers = 4;
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parameters.firstChannel = 0;
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mix_rate = GLOBAL_DEF("audio/mix_rate", 44100);
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mix_rate = GLOBAL_DEF("audio/mix_rate", DEFAULT_MIX_RATE);
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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// calculate desired buffer_size
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unsigned int buffer_size = closest_power_of_2(latency * mix_rate / 1000);
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int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
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unsigned int buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
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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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print_line("audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
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}
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short int tries = 2;
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@ -127,7 +126,7 @@ Error AudioDriverRtAudio::init() {
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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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dac->openStream(¶meters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_frames, &callback, this, &options);
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active = true;
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break;
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@ -199,7 +198,7 @@ AudioDriverRtAudio::AudioDriverRtAudio() {
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active = false;
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mutex = NULL;
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dac = NULL;
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mix_rate = 44100;
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mix_rate = DEFAULT_MIX_RATE;
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speaker_mode = SPEAKER_MODE_STEREO;
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}
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@ -65,6 +65,18 @@ Error AudioDriverWASAPI::init_device() {
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format_tag = pwfex->wFormatTag;
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bits_per_sample = pwfex->wBitsPerSample;
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switch (channels) {
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case 2: // Stereo
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case 6: // Surround 5.1
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case 8: // Surround 7.1
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break;
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default:
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ERR_PRINT("WASAPI: Unsupported number of channels");
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ERR_FAIL_V(ERR_CANT_OPEN);
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break;
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}
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if (format_tag == WAVE_FORMAT_EXTENSIBLE) {
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WAVEFORMATEXTENSIBLE *wfex = (WAVEFORMATEXTENSIBLE *)pwfex;
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@ -83,13 +95,6 @@ Error AudioDriverWASAPI::init_device() {
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}
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}
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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buffer_size = closest_power_of_2(latency * mix_rate / 1000);
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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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hr = audio_client->Initialize(AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, 0, 0, pwfex, NULL);
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ERR_FAIL_COND_V(hr != S_OK, ERR_CANT_OPEN);
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@ -102,11 +107,20 @@ Error AudioDriverWASAPI::init_device() {
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hr = audio_client->GetService(IID_IAudioRenderClient, (void **)&render_client);
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ERR_FAIL_COND_V(hr != S_OK, ERR_CANT_OPEN);
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UINT32 max_frames;
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hr = audio_client->GetBufferSize(&max_frames);
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ERR_FAIL_COND_V(hr != S_OK, ERR_CANT_OPEN);
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// Due to WASAPI Shared Mode we have no control of the buffer size
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buffer_frames = max_frames;
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// Sample rate is independent of channels (ref: https://stackoverflow.com/questions/11048825/audio-sample-frequency-rely-on-channels)
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buffer_size = buffer_frames * channels;
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samples_in.resize(buffer_size);
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buffer_frames = buffer_size / channels;
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if (OS::get_singleton()->is_stdout_verbose()) {
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print_line("audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
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}
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return OK;
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}
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@ -200,7 +214,7 @@ void AudioDriverWASAPI::thread_func(void *p_udata) {
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HRESULT hr = ad->audio_client->GetCurrentPadding(&cur_frames);
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if (hr == S_OK) {
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// Check how much frames are available on the WASAPI buffer
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UINT32 avail_frames = ad->max_frames - cur_frames;
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UINT32 avail_frames = ad->buffer_frames - cur_frames;
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UINT32 write_frames = avail_frames > left_frames ? left_frames : avail_frames;
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BYTE *buffer = NULL;
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@ -332,7 +346,6 @@ AudioDriverWASAPI::AudioDriverWASAPI() {
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mutex = NULL;
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thread = NULL;
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max_frames = 0;
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format_tag = 0;
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bits_per_sample = 0;
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@ -48,7 +48,6 @@ class AudioDriverWASAPI : public AudioDriver {
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Mutex *mutex;
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Thread *thread;
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UINT32 max_frames;
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WORD format_tag;
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WORD bits_per_sample;
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@ -77,7 +77,7 @@ Error AudioDriverOSX::initDevice() {
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break;
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}*/
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mix_rate = GLOBAL_DEF("audio/mix_rate", 44100);
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mix_rate = GLOBAL_DEF("audio/mix_rate", DEFAULT_MIX_RATE);
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zeromem(&strdesc, sizeof(strdesc));
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strdesc.mFormatID = kAudioFormatLinearPCM;
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result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, kOutputBus, &strdesc, sizeof(strdesc));
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ERR_FAIL_COND_V(result != noErr, FAILED);
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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unsigned int buffer_size = closest_power_of_2(latency * mix_rate / 1000);
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int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
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// Sample rate is independent of channels (ref: https://stackoverflow.com/questions/11048825/audio-sample-frequency-rely-on-channels)
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buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
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result = AudioUnitSetProperty(audio_unit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, kOutputBus, &buffer_frames, sizeof(UInt32));
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ERR_FAIL_COND_V(result != noErr, FAILED);
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buffer_size = buffer_frames * channels;
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samples_in.resize(buffer_size);
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if (OS::get_singleton()->is_stdout_verbose()) {
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print_line("audio buffer size: " + itos(buffer_size) + " calculated latency: " + itos(buffer_size * 1000 / mix_rate));
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print_line("audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");
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}
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samples_in.resize(buffer_size);
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buffer_frames = buffer_size / channels;
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AURenderCallbackStruct callback;
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zeromem(&callback, sizeof(AURenderCallbackStruct));
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callback.inputProc = &AudioDriverOSX::output_callback;
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@ -234,7 +238,7 @@ void AudioDriverOSX::start() {
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};
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int AudioDriverOSX::get_mix_rate() const {
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return 44100;
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return mix_rate;
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};
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AudioDriver::SpeakerMode AudioDriverOSX::get_speaker_mode() const {
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@ -282,8 +286,12 @@ AudioDriverOSX::AudioDriverOSX() {
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active = false;
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mutex = NULL;
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mix_rate = 44100;
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mix_rate = 0;
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channels = 2;
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buffer_size = 0;
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buffer_frames = 0;
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samples_in.clear();
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};
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@ -45,8 +45,9 @@ class AudioDriverOSX : public AudioDriver {
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Mutex *mutex;
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int mix_rate;
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int channels;
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int buffer_frames;
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unsigned int channels;
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unsigned int buffer_frames;
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unsigned int buffer_size;
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Vector<int32_t> samples_in;
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@ -37,17 +37,16 @@ Error AudioDriverDummy::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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pcm_open = false;
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samples_in = NULL;
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mix_rate = 44100;
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mix_rate = DEFAULT_MIX_RATE;
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speaker_mode = SPEAKER_MODE_STEREO;
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channels = 2;
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int latency = GLOBAL_DEF("audio/output_latency", 25);
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buffer_size = next_power_of_2(latency * mix_rate / 1000);
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int latency = GLOBAL_DEF("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
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buffer_frames = closest_power_of_2(latency * mix_rate / 1000);
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samples_in = memnew_arr(int32_t, buffer_size * channels);
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samples_in = memnew_arr(int32_t, buffer_frames * channels);
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mutex = Mutex::create();
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thread = Thread::create(AudioDriverDummy::thread_func, this);
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@ -59,17 +58,15 @@ void AudioDriverDummy::thread_func(void *p_udata) {
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AudioDriverDummy *ad = (AudioDriverDummy *)p_udata;
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uint64_t usdelay = (ad->buffer_size / float(ad->mix_rate)) * 1000000;
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uint64_t usdelay = (ad->buffer_frames / float(ad->mix_rate)) * 1000000;
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while (!ad->exit_thread) {
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if (!ad->active) {
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} else {
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if (ad->active) {
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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->buffer_frames, ad->samples_in);
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ad->unlock();
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};
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@ -43,8 +43,8 @@ class AudioDriverDummy : public AudioDriver {
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int32_t *samples_in;
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static void thread_func(void *p_udata);
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int buffer_size;
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unsigned int buffer_frames;
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unsigned int mix_rate;
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SpeakerMode speaker_mode;
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@ -53,7 +53,6 @@ class AudioDriverDummy : public AudioDriver {
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bool active;
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bool thread_exited;
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mutable bool exit_thread;
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bool pcm_open;
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public:
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const char *get_name() const {
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@ -155,6 +155,29 @@ void AudioServer::_driver_process(int p_frames, int32_t *p_buffer) {
|
|||
todo -= to_copy;
|
||||
to_mix -= to_copy;
|
||||
}
|
||||
|
||||
#ifdef DEBUG_ENABLED
|
||||
if (OS::get_singleton() && OS::get_singleton()->is_stdout_verbose()) {
|
||||
static uint64_t first_ticks = 0;
|
||||
static uint64_t last_ticks = 0;
|
||||
static uint64_t ticks = 0;
|
||||
static int count = 0;
|
||||
static int total = 0;
|
||||
|
||||
ticks = OS::get_singleton()->get_ticks_msec();
|
||||
if ((ticks - first_ticks) > 10 * 1000) {
|
||||
print_line("Audio Driver " + String(AudioDriver::get_singleton()->get_name()) + " average latency: " + itos(total / count) + "ms (frame=" + itos(p_frames) + ")");
|
||||
first_ticks = ticks;
|
||||
total = 0;
|
||||
count = 0;
|
||||
}
|
||||
|
||||
total += ticks - last_ticks;
|
||||
count++;
|
||||
|
||||
last_ticks = ticks;
|
||||
}
|
||||
#endif
|
||||
}
|
||||
|
||||
void AudioServer::_mix_step() {
|
||||
|
|
|
@ -54,6 +54,9 @@ public:
|
|||
SPEAKER_SURROUND_71,
|
||||
};
|
||||
|
||||
static const int DEFAULT_MIX_RATE = 44100;
|
||||
static const int DEFAULT_OUTPUT_LATENCY = 15;
|
||||
|
||||
static AudioDriver *get_singleton();
|
||||
void set_singleton();
|
||||
|
||||
|
|
Loading…
Reference in New Issue