maxmustermann/godot
1
0
Fork 0
Code Issues Pull requests Projects Releases Wiki Activity

Fix for the huge audio latency of the SamplePlayer (>200 ms)

Browse source 
- 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 2017-01-03 19:00:31 +01:00
parent 495d059a74
commit da6b6c2dd7
4 changed files with 145 additions and 131 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

50
drivers/alsa/audio_driver_alsa.cpp
View file

@ -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;
};
};

1
drivers/alsa/audio_driver_alsa.h
View file

@ -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;

121
drivers/pulseaudio/audio_driver_pulseaudio.cpp
View file

@ -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;
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
);
int latency = GLOBAL_DEF("audio/output_latency", 25);
buffer_size = nearest_power_of_2(latency * mix_rate / 1000);
if (pulse == NULL) {
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;
fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
ERR_FAIL_COND_V(pulse == NULL, ERR_CANT_OPEN);
}
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
);
int latency = GLOBAL_DEF("audio/output_latency", 25);
buffer_size = nearest_power_of_2(latency * mix_rate / 1000);
if (pulse == NULL) {
fprintf(stderr, "PulseAudio ERR: %s\n", pa_strerror(error_code));\
ERR_FAIL_COND_V(pulse == NULL, ERR_CANT_OPEN);
}
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);
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);
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
// 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) {
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;
}
}
// 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;
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() {

104
drivers/rtaudio/audio_driver_rtaudio.cpp
View file

@ -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;
};
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..");
short int tries = 2;
while(true) {
while( true) {
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;
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;
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;
}
}
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;