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Merge pull request #10683 from marcelofg55/rtaudio_buffer_fix

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Fix RtAudio driver buffer_size incorrect calculation
This commit is contained in:
Rémi Verschelde 2017-08-28 23:03:04 +02:00 committed by GitHub
parent 3ce6972d2c 7e6b015769
commit ddbd133097
1 changed files with 22 additions and 57 deletions
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79
drivers/rtaudio/audio_driver_rtaudio.cpp
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@ -104,21 +104,14 @@ Error AudioDriverRtAudio::init() {
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). */
options.numberOfBuffers = 4;
parameters.firstChannel = 0;
mix_rate = GLOBAL_DEF("audio/mix_rate", 44100);
int latency = GLOBAL_DEF("audio/output_latency", 25);
// calculate desired buffer_size, taking the desired numberOfBuffers into account (latency depends on numberOfBuffers*buffer_size)
unsigned int buffer_size = closest_power_of_2(latency * mix_rate / 1000 / target_number_of_buffers);
// calculate desired buffer_size
unsigned int buffer_size = closest_power_of_2(latency * mix_rate / 1000);
if (OS::get_singleton()->is_stdout_verbose()) {
print_line("audio buffer size: " + itos(buffer_size));
@ -126,56 +119,28 @@ Error AudioDriverRtAudio::init() {
short int tries = 2;
while (true) {
while (true) {
switch (speaker_mode) {
case SPEAKER_MODE_STEREO: parameters.nChannels = 2; break;
case SPEAKER_SURROUND_51: parameters.nChannels = 6; break;
case SPEAKER_SURROUND_71: parameters.nChannels = 8; break;
};
while (tries >= 0) {
switch (speaker_mode) {
case SPEAKER_MODE_STEREO: parameters.nChannels = 2; break;
case SPEAKER_SURROUND_51: parameters.nChannels = 6; break;
case SPEAKER_SURROUND_71: parameters.nChannels = 8; break;
};
try {
dac->openStream(&parameters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_size, &callback, this, &options);
active = true;
try {
dac->openStream(&parameters, NULL, RTAUDIO_SINT32, mix_rate, &buffer_size, &callback, this, &options);
active = true;
break;
} catch (RtAudioError &e) {
// try with less channels
ERR_PRINT("Unable to open audio, retrying with fewer channels..");
switch (speaker_mode) {
case SPEAKER_MODE_STEREO: speaker_mode = SPEAKER_MODE_STEREO; break;
case SPEAKER_SURROUND_51: speaker_mode = SPEAKER_SURROUND_51; break;
case SPEAKER_SURROUND_71: speaker_mode = SPEAKER_SURROUND_71; 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();
active = false;
} 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 (speaker_mode) {
case SPEAKER_SURROUND_51: speaker_mode = SPEAKER_MODE_STEREO; break;
case SPEAKER_SURROUND_71: speaker_mode = SPEAKER_SURROUND_51; break;
}
tries--;
}
}