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Update RtAudio backend to overwrite default samplerate to 48 kHz, if available

This commit is contained in:
Anne de Jong 2022-11-11 12:35:52 +01:00
parent fbb14f475c
commit 0c0a86dc64
1 changed files with 143 additions and 128 deletions
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271
src/lasp/device/lasp_rtaudiodaq.cpp
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@ -38,35 +38,50 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
// "Our device info struct"
DeviceInfo d;
switch (api) {
case RtAudio::LINUX_ALSA:
d.api = rtaudioAlsaApi;
break;
case RtAudio::LINUX_PULSE:
d.api = rtaudioPulseaudioApi;
break;
case RtAudio::WINDOWS_WASAPI:
d.api = rtaudioWasapiApi;
break;
case RtAudio::WINDOWS_DS:
d.api = rtaudioDsApi;
break;
case RtAudio::WINDOWS_ASIO:
d.api = rtaudioAsioApi;
break;
default:
cerr << "Not implemented RtAudio API, skipping." << endl;
continue;
break;
case RtAudio::LINUX_ALSA:
d.api = rtaudioAlsaApi;
break;
case RtAudio::LINUX_PULSE:
d.api = rtaudioPulseaudioApi;
break;
case RtAudio::WINDOWS_WASAPI:
d.api = rtaudioWasapiApi;
break;
case RtAudio::WINDOWS_DS:
d.api = rtaudioDsApi;
break;
case RtAudio::WINDOWS_ASIO:
d.api = rtaudioAsioApi;
break;
default:
cerr << "Not implemented RtAudio API, skipping." << endl;
continue;
break;
}
d.device_name = devinfo.name;
d.api_specific_devindex = devno;
/// We overwrite the default sample rate with the 48 kHz value, which
/// is our preffered rate.
bool rate_48k_found = false;
for (us j = 0; j < devinfo.sampleRates.size(); j++) {
us rate = devinfo.sampleRates[j];
d.availableSampleRates.push_back((double)rate);
if (devinfo.preferredSampleRate == rate) {
d.prefSampleRateIndex = j;
us rate_int = devinfo.sampleRates[j];
d.availableSampleRates.push_back((double)rate_int);
if(!rate_48k_found) {
if (devinfo.preferredSampleRate == rate_int) {
d.prefSampleRateIndex = j;
}
if(rate_int == 48000) {
d.prefSampleRateIndex = j;
rate_48k_found = true;
}
}
}
@ -86,7 +101,7 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
}
/* if (formats & RTAUDIO_SINT24) { *1/ */
/* d.availableDataTypes.push_back(DataTypeDescriptor::DataType::dtype_int24);
*/
*/
/* } */
if (formats & RTAUDIO_SINT32) {
d.availableDataTypes.push_back(
@ -103,7 +118,7 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
d.prefDataTypeIndex = d.availableDataTypes.size() - 1;
d.availableFramesPerBlock = {512, 1024, 2048, 4096, 8192};
d.prefFramesPerBlockIndex = 1;
d.prefFramesPerBlockIndex = 2;
devinfolist.push_back(d);
}
@ -111,8 +126,8 @@ void fillRtAudioDeviceInfo(vector<DeviceInfo> &devinfolist) {
}
static int mycallback(void *outputBuffer, void *inputBuffer,
unsigned int nFrames, double streamTime,
RtAudioStreamStatus status, void *userData);
unsigned int nFrames, double streamTime,
RtAudioStreamStatus status, void *userData);
static void myerrorcallback(RtAudioError::Type, const string &errorText);
@ -129,101 +144,101 @@ class RtAudioDaq : public Daq {
std::atomic<StreamStatus> _streamStatus{};
public:
public:
RtAudioDaq(const DeviceInfo &devinfo, const DaqConfiguration &config)
: Daq(devinfo, config),
rtaudio(static_cast<RtAudio::Api>(devinfo.api.api_specific_subcode)),
nFramesPerBlock(Daq::framesPerBlock()) {
: Daq(devinfo, config),
rtaudio(static_cast<RtAudio::Api>(devinfo.api.api_specific_subcode)),
nFramesPerBlock(Daq::framesPerBlock()) {
DEBUGTRACE_ENTER;
DEBUGTRACE_ENTER;
// We make sure not to run RtAudio in duplex mode. This seems to be buggy
// and untested. Better to use a hardware-type loopback into the system.
if (duplexMode()) {
throw rte("RtAudio backend cannot run in duplex mode.");
}
assert(!monitorOutput);
std::unique_ptr<RtAudio::StreamParameters> inParams, outParams;
if (neninchannels() > 0) {
inParams = std::make_unique<RtAudio::StreamParameters>();
// +1 to get the count.
inParams->nChannels = getHighestEnabledInChannel() + 1;
if (inParams->nChannels < 1) {
throw rte("Invalid input number of channels");
// We make sure not to run RtAudio in duplex mode. This seems to be buggy
// and untested. Better to use a hardware-type loopback into the system.
if (duplexMode()) {
throw rte("RtAudio backend cannot run in duplex mode.");
}
inParams->firstChannel = 0;
inParams->deviceId = devinfo.api_specific_devindex;
assert(!monitorOutput);
} else {
std::unique_ptr<RtAudio::StreamParameters> inParams, outParams;
outParams = std::make_unique<RtAudio::StreamParameters>();
if (neninchannels() > 0) {
outParams->nChannels = getHighestEnabledOutChannel() + 1;
if (outParams->nChannels < 1) {
throw rte("Invalid output number of channels");
inParams = std::make_unique<RtAudio::StreamParameters>();
// +1 to get the count.
inParams->nChannels = getHighestEnabledInChannel() + 1;
if (inParams->nChannels < 1) {
throw rte("Invalid input number of channels");
}
inParams->firstChannel = 0;
inParams->deviceId = devinfo.api_specific_devindex;
} else {
outParams = std::make_unique<RtAudio::StreamParameters>();
outParams->nChannels = getHighestEnabledOutChannel() + 1;
if (outParams->nChannels < 1) {
throw rte("Invalid output number of channels");
}
outParams->firstChannel = 0;
outParams->deviceId = devinfo.api_specific_devindex;
}
RtAudio::StreamOptions streamoptions;
streamoptions.flags = RTAUDIO_HOG_DEVICE | RTAUDIO_NONINTERLEAVED;
streamoptions.numberOfBuffers = 2;
streamoptions.streamName = "LASP RtAudio DAQ stream";
streamoptions.priority = 0;
RtAudioFormat format;
using Dtype = DataTypeDescriptor::DataType;
const Dtype dtype = dataType();
switch (dtype) {
case Dtype::dtype_fl32:
DEBUGTRACE_PRINT("Datatype float32");
format = RTAUDIO_FLOAT32;
break;
case Dtype::dtype_fl64:
DEBUGTRACE_PRINT("Datatype float64");
format = RTAUDIO_FLOAT64;
break;
case Dtype::dtype_int8:
DEBUGTRACE_PRINT("Datatype int8");
format = RTAUDIO_SINT8;
break;
case Dtype::dtype_int16:
DEBUGTRACE_PRINT("Datatype int16");
format = RTAUDIO_SINT16;
break;
case Dtype::dtype_int32:
DEBUGTRACE_PRINT("Datatype int32");
format = RTAUDIO_SINT32;
break;
default:
throw rte("Invalid data type specified for DAQ stream.");
break;
}
// Copy here, as it is used to return the *actual* number of frames per
// block.
unsigned int nFramesPerBlock_copy = nFramesPerBlock;
// Final step: open the stream.
rtaudio.openStream(outParams.get(), inParams.get(), format,
static_cast<us>(samplerate()), &nFramesPerBlock_copy,
mycallback, (void *)this, &streamoptions,
&myerrorcallback);
if (nFramesPerBlock_copy != nFramesPerBlock) {
throw rte("Got different number of frames per block back from RtAudio "
"backend. Do not know what to do");
}
outParams->firstChannel = 0;
outParams->deviceId = devinfo.api_specific_devindex;
}
RtAudio::StreamOptions streamoptions;
streamoptions.flags = RTAUDIO_HOG_DEVICE | RTAUDIO_NONINTERLEAVED;
streamoptions.numberOfBuffers = 2;
streamoptions.streamName = "LASP RtAudio DAQ stream";
streamoptions.priority = 0;
RtAudioFormat format;
using Dtype = DataTypeDescriptor::DataType;
const Dtype dtype = dataType();
switch (dtype) {
case Dtype::dtype_fl32:
DEBUGTRACE_PRINT("Datatype float32");
format = RTAUDIO_FLOAT32;
break;
case Dtype::dtype_fl64:
DEBUGTRACE_PRINT("Datatype float64");
format = RTAUDIO_FLOAT64;
break;
case Dtype::dtype_int8:
DEBUGTRACE_PRINT("Datatype int8");
format = RTAUDIO_SINT8;
break;
case Dtype::dtype_int16:
DEBUGTRACE_PRINT("Datatype int16");
format = RTAUDIO_SINT16;
break;
case Dtype::dtype_int32:
DEBUGTRACE_PRINT("Datatype int32");
format = RTAUDIO_SINT32;
break;
default:
throw rte("Invalid data type specified for DAQ stream.");
break;
}
// Copy here, as it is used to return the *actual* number of frames per
// block.
unsigned int nFramesPerBlock_copy = nFramesPerBlock;
// Final step: open the stream.
rtaudio.openStream(outParams.get(), inParams.get(), format,
static_cast<us>(samplerate()), &nFramesPerBlock_copy,
mycallback, (void *)this, &streamoptions,
&myerrorcallback);
if (nFramesPerBlock_copy != nFramesPerBlock) {
throw rte("Got different number of frames per block back from RtAudio "
"backend. Do not know what to do");
}
}
virtual void start(InDaqCallback inCallback,
OutDaqCallback outCallback) override final {
OutDaqCallback outCallback) override final {
DEBUGTRACE_ENTER;
@ -236,7 +251,7 @@ public:
// Logical XOR
if (inCallback && outCallback) {
throw rte("Either input or output stream possible for RtAudio. "
"Stream duplex mode not provided.");
"Stream duplex mode not provided.");
}
if (neninchannels() > 0) {
@ -279,7 +294,7 @@ public:
}
int streamCallback(void *outputBuffer, void *inputBuffer,
unsigned int nFrames, RtAudioStreamStatus status) {
unsigned int nFrames, RtAudioStreamStatus status) {
DEBUGTRACE_ENTER;
@ -296,16 +311,16 @@ public:
};
switch (status) {
case RTAUDIO_INPUT_OVERFLOW:
stopWithError(se::inputXRun);
return 1;
break;
case RTAUDIO_OUTPUT_UNDERFLOW:
stopWithError(se::outputXRun);
return 1;
break;
default:
break;
case RTAUDIO_INPUT_OVERFLOW:
stopWithError(se::inputXRun);
return 1;
break;
case RTAUDIO_OUTPUT_UNDERFLOW:
stopWithError(se::outputXRun);
return 1;
break;
default:
break;
}
const auto &dtype_descr = dtypeDescr();
@ -316,7 +331,7 @@ public:
us sw = dtype_descr.sw;
if (nFrames != nFramesPerBlock) {
cerr << "RtAudio backend error: nFrames does not match block size!"
<< endl;
<< endl;
stopWithError(se::logicError);
return 1;
}
@ -332,7 +347,7 @@ public:
for (us ch = ch_min; ch <= ch_max; ch++) {
if (inchannel_config.at(ch).enabled) {
byte_t *ptr =
static_cast<byte_t *>(inputBuffer) + sw * i * nFramesPerBlock;
static_cast<byte_t *>(inputBuffer) + sw * i * nFramesPerBlock;
DEBUGTRACE_PRINT((us)ptr);
ptrs.push_back(ptr);
}
@ -362,7 +377,7 @@ public:
if (outchannel_config.at(ch).enabled) {
ptrs.push_back(static_cast<byte_t *>(outputBuffer) +
sw * i * nFramesPerBlock);
sw * i * nFramesPerBlock);
}
i++;
}
@ -390,7 +405,7 @@ public:
};
std::unique_ptr<Daq> createRtAudioDevice(const DeviceInfo &devinfo,
const DaqConfiguration &config) {
const DaqConfiguration &config) {
return std::make_unique<RtAudioDaq>(devinfo, config);
}
@ -398,7 +413,7 @@ void myerrorcallback(RtAudioError::Type, const string &errorText) {
cerr << "RtAudio backend stream error: " << errorText << endl;
}
int mycallback(void *outputBuffer, void *inputBuffer, unsigned int nFrames,
double streamTime, RtAudioStreamStatus status, void *userData) {
double streamTime, RtAudioStreamStatus status, void *userData) {
return static_cast<RtAudioDaq *>(userData)->streamCallback(
outputBuffer, inputBuffer, nFrames, status);