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ASCEE/lasp
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Merge branch 'RtAudioV6' into develop

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This commit is contained in:
Anne de Jong 2024-01-19 12:37:16 +01:00
commit 292a9d5938
3 changed files with 121 additions and 71 deletions
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2
cpp_src/device/CMakeLists.txt
View File

@ -34,7 +34,7 @@ if(LASP_HAS_RTAUDIO)
target_link_libraries(lasp_device_lib rtaudio)
endif()
if(LASP_HAS_PORTAUDIO)
target_link_libraries(lasp_device_lib portaudio)
target_link_libraries(lasp_device_lib PortAudio)
if(WIN32)
else()
target_link_libraries(lasp_device_lib asound)

184
cpp_src/device/lasp_rtaudiodaq.cpp
View File

@ -17,37 +17,46 @@ using rte = std::runtime_error;
using std::vector;
using lck = std::scoped_lock<std::mutex>;
class RtAudioDeviceInfo : public DeviceInfo {
const unsigned RTAUDIO_MAX_CHANNELS = 8;
class RtAudioDeviceInfo : public DeviceInfo
{
public:
/**
* @brief Specific for the device (Sub-API). Important for the RtAudio
* backend, as RtAudio is able to handle different API's.
*/
int _api_devindex;
virtual std::unique_ptr<DeviceInfo> clone() const override {
int ID; // Copy of RtAudio::DeviceInfo::ID
virtual std::unique_ptr<DeviceInfo> clone() const override
{
return std::make_unique<DeviceInfo>(*this);
}
};
void fillRtAudioDeviceInfo(DeviceInfoList &devinfolist) {
void fillRtAudioDeviceInfo(DeviceInfoList &devinfolist)
{
DEBUGTRACE_ENTER;
vector<RtAudio::Api> apis;
RtAudio::getCompiledApi(apis);
for (auto api : apis) {
for (auto api : apis)
{
RtAudio rtaudio(api);
us count = rtaudio.getDeviceCount();
for (us devno = 0; devno < count; devno++) {
const us count = rtaudio.getDeviceCount();
const auto ids = rtaudio.getDeviceIds();
for (us i = 0; i < count; i++)
{
us id = ids.at(i);
RtAudio::DeviceInfo devinfo = rtaudio.getDeviceInfo(id);
RtAudio::DeviceInfo devinfo = rtaudio.getDeviceInfo(devno);
if (!devinfo.probed) {
// Device capabilities not successfully probed. Continue to next
continue;
}
// "Our device info struct"
RtAudioDeviceInfo d;
switch (api) {
switch (api)
{
case RtAudio::LINUX_ALSA:
d.api = rtaudioAlsaApi;
break;
@ -70,43 +79,49 @@ void fillRtAudioDeviceInfo(DeviceInfoList &devinfolist) {
}
d.device_name = devinfo.name;
d._api_devindex = devno;
d.ID = id;
/// When 48k is available 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++) {
for (us j = 0; j < devinfo.sampleRates.size(); j++)
{
us rate_int = devinfo.sampleRates[j];
d.availableSampleRates.push_back((double)rate_int);
if (!rate_48k_found) {
if (!rate_48k_found)
{
if (devinfo.preferredSampleRate == rate_int) {
if (devinfo.preferredSampleRate == rate_int)
{
d.prefSampleRateIndex = j;
}
if (rate_int == 48000) {
if (rate_int == 48000)
{
d.prefSampleRateIndex = j;
rate_48k_found = true;
}
}
}
d.noutchannels = devinfo.outputChannels;
d.ninchannels = devinfo.inputChannels;
d.noutchannels = std::min(devinfo.outputChannels, RTAUDIO_MAX_CHANNELS);
d.ninchannels = std::min(devinfo.inputChannels, RTAUDIO_MAX_CHANNELS);
d.availableInputRanges = {1.0};
d.availableOutputRanges = {1.0};
RtAudioFormat formats = devinfo.nativeFormats;
if (formats & RTAUDIO_SINT8) {
if (formats & RTAUDIO_SINT8)
{
d.availableDataTypes.push_back(
DataTypeDescriptor::DataType::dtype_int8);
}
if (formats & RTAUDIO_SINT16) {
if (formats & RTAUDIO_SINT16)
{
d.availableDataTypes.push_back(
DataTypeDescriptor::DataType::dtype_int16);
}
@ -114,15 +129,18 @@ void fillRtAudioDeviceInfo(DeviceInfoList &devinfolist) {
/* d.availableDataTypes.push_back(DataTypeDescriptor::DataType::dtype_int24);
*/
/* } */
if (formats & RTAUDIO_SINT32) {
if (formats & RTAUDIO_SINT32)
{
d.availableDataTypes.push_back(
DataTypeDescriptor::DataType::dtype_fl32);
}
if (formats & RTAUDIO_FLOAT64) {
if (formats & RTAUDIO_FLOAT64)
{
d.availableDataTypes.push_back(
DataTypeDescriptor::DataType::dtype_fl64);
}
if (d.availableDataTypes.size() == 0) {
if (d.availableDataTypes.size() == 0)
{
std::cerr << "RtAudio: No data types found in device!" << endl;
}
@ -140,9 +158,8 @@ static int mycallback(void *outputBuffer, void *inputBuffer,
unsigned int nFrames, double streamTime,
RtAudioStreamStatus status, void *userData);
static void myerrorcallback(RtAudioError::Type, const string &errorText);
class RtAudioDaq : public Daq {
class RtAudioDaq : public Daq
{
RtAudio rtaudio;
const us nFramesPerBlock;
@ -182,10 +199,11 @@ public:
/// 0. For now, our fix is to shift out the channels we want, and let
/// RtAudio pass on all channels.
inParams->firstChannel = 0;
inParams->nChannels = devinfo_gen.ninchannels;
inParams->deviceId = devinfo._api_devindex;
} else {
inParams->nChannels = devinfo.ninchannels;
inParams->deviceId = devinfo.ID;
}
else
{
outParams = std::make_unique<RtAudio::StreamParameters>();
@ -193,8 +211,8 @@ public:
/// 0. For now, our fix is to shift out the channels we want, and let
/// RtAudio pass on all channels.
outParams->firstChannel = 0;
outParams->nChannels = devinfo_gen.noutchannels;
outParams->deviceId = devinfo._api_devindex;
outParams->nChannels = devinfo.noutchannels;
outParams->deviceId = devinfo.ID;
}
RtAudio::StreamOptions streamoptions;
@ -207,7 +225,8 @@ public:
RtAudioFormat format;
using Dtype = DataTypeDescriptor::DataType;
const Dtype dtype = dataType();
switch (dtype) {
switch (dtype)
{
case Dtype::dtype_fl32:
DEBUGTRACE_PRINT("Datatype float32");
format = RTAUDIO_FLOAT32;
@ -238,36 +257,46 @@ public:
unsigned int nFramesPerBlock_copy = nFramesPerBlock;
// Final step: open the stream.
rtaudio.openStream(outParams.get(), inParams.get(), format,
RtAudioErrorType err = rtaudio.openStream(outParams.get(), inParams.get(), format,
static_cast<us>(samplerate()), &nFramesPerBlock_copy,
mycallback, (void *)this, &streamoptions,
&myerrorcallback);
mycallback, (void *)this, &streamoptions);
if (err != RTAUDIO_NO_ERROR)
{
throw std::runtime_error(string("Error opening stream: ") + rtaudio.getErrorText());
}
if (nFramesPerBlock_copy != nFramesPerBlock) {
if (nFramesPerBlock_copy != nFramesPerBlock)
{
throw rte(string("Got different number of frames per block back from RtAudio "
"backend: ") + std::to_string(nFramesPerBlock_copy) + ". I do not know what to do.");
"backend: ") +
std::to_string(nFramesPerBlock_copy) + ". I do not know what to do.");
}
}
virtual void start(InDaqCallback inCallback,
OutDaqCallback outCallback) override final {
OutDaqCallback outCallback) override final
{
DEBUGTRACE_ENTER;
assert(!monitorOutput);
if (getStreamStatus().runningOK()) {
if (getStreamStatus().runningOK())
{
throw rte("Stream already running");
}
// Logical XOR
if (inCallback && outCallback) {
if (inCallback && outCallback)
{
throw rte("Either input or output stream possible for RtAudio. "
"Stream duplex mode not provided.");
}
if (neninchannels() > 0) {
if (!inCallback) {
if (neninchannels() > 0)
{
if (!inCallback)
{
throw rte(
"Input callback given, but stream does not provide input data");
@ -275,8 +304,10 @@ public:
_incallback = inCallback;
}
if (nenoutchannels() > 0) {
if (!outCallback) {
if (nenoutchannels() > 0)
{
if (!outCallback)
{
throw rte(
"Output callback given, but stream does not provide output data");
}
@ -284,7 +315,11 @@ public:
}
// Start the stream. Throws on error.
rtaudio.startStream();
const auto err = rtaudio.startStream();
if (err != RTAUDIO_NO_ERROR)
{
throw std::runtime_error(string("Error starting stream: ") + rtaudio.getErrorText());
}
// If we are here, we are running without errors.
StreamStatus status;
@ -294,10 +329,15 @@ public:
StreamStatus getStreamStatus() const override final { return _streamStatus; }
void stop() override final {
void stop() override final
{
DEBUGTRACE_ENTER;
if (getStreamStatus().runningOK()) {
rtaudio.stopStream();
if (getStreamStatus().runningOK())
{
const auto err = rtaudio.stopStream();
if(err != RTAUDIO_NO_ERROR) {
std::cerr << "Error occured while stopping the stream: " << rtaudio.getErrorText() << endl;
}
}
StreamStatus s = _streamStatus;
s.isRunning = false;
@ -306,14 +346,16 @@ public:
}
int streamCallback(void *outputBuffer, void *inputBuffer,
unsigned int nFrames, RtAudioStreamStatus status) {
unsigned int nFrames, RtAudioStreamStatus status)
{
DEBUGTRACE_ENTER;
using se = StreamStatus::StreamError;
int rval = 0;
auto stopWithError = [&](se e) {
auto stopWithError = [&](se e)
{
DEBUGTRACE_PRINT("stopWithError");
StreamStatus stat = _streamStatus;
stat.errorType = e;
@ -322,7 +364,8 @@ public:
rval = 1;
};
switch (status) {
switch (status)
{
case RTAUDIO_INPUT_OVERFLOW:
stopWithError(se::inputXRun);
return 1;
@ -341,14 +384,16 @@ public:
const us neninchannels = this->neninchannels();
const us nenoutchannels = this->nenoutchannels();
const us sw = dtype_descr.sw;
if (nFrames != nFramesPerBlock) {
if (nFrames != nFramesPerBlock)
{
cerr << "RtAudio backend error: nFrames does not match block size!"
<< endl;
stopWithError(se::logicError);
return 1;
}
if (inputBuffer) {
if (inputBuffer)
{
assert(_incallback);
std::vector<byte_t *> ptrs;
ptrs.reserve(neninchannels);
@ -359,8 +404,10 @@ public:
assert(ch_max < ninchannels);
/// Only pass on the pointers of the channels we want
for (us ch = ch_min; ch <= ch_max; ch++) {
if (inchannel_config.at(ch).enabled) {
for (us ch = ch_min; ch <= ch_max; ch++)
{
if (inchannel_config.at(ch).enabled)
{
byte_t *ptr =
static_cast<byte_t *>(inputBuffer) + sw * ch * nFramesPerBlock;
ptrs.push_back(ptr);
@ -372,7 +419,8 @@ public:
_incallback(d);
}
if (outputBuffer) {
if (outputBuffer)
{
assert(_outcallback);
std::vector<byte_t *> ptrs;
ptrs.reserve(nenoutchannels);
@ -384,8 +432,10 @@ public:
assert(ch_min < noutchannels);
assert(ch_max < noutchannels);
/// Only pass on the pointers of the channels we want
for (us ch = ch_min; ch <= ch_max; ch++) {
if (outchannel_config.at(ch).enabled) {
for (us ch = ch_min; ch <= ch_max; ch++)
{
if (outchannel_config.at(ch).enabled)
{
ptrs.push_back(static_cast<byte_t *>(outputBuffer) +
sw * ch * nFramesPerBlock);
}
@ -395,7 +445,8 @@ public:
_outcallback(d);
// Copy over the buffer
us j = 0;
for (auto ptr : ptrs) {
for (auto ptr : ptrs)
{
d.copyToRaw(j, ptr);
j++;
}
@ -411,17 +462,16 @@ public:
};
std::unique_ptr<Daq> createRtAudioDevice(const DeviceInfo &devinfo,
const DaqConfiguration &config) {
const DaqConfiguration &config)
{
return std::make_unique<RtAudioDaq>(devinfo, config);
}
void myerrorcallback(RtAudioError::Type, const string &errorText) {
cerr << "RtAudio backend stream error: " << errorText << endl;
}
int mycallback(
void *outputBuffer, void *inputBuffer, unsigned int nFrames,
__attribute__((unused)) double streamTime, // Not used parameter streamTime
RtAudioStreamStatus status, void *userData) {
RtAudioStreamStatus status, void *userData)
{
return static_cast<RtAudioDaq *>(userData)->streamCallback(
outputBuffer, inputBuffer, nFrames, status);

2
third_party/rtaudio vendored

@ -1 +1 @@
Subproject commit 46b01b5b134f33d8ddc3dab76829d4b1350e0522
Subproject commit b4f04903312e0e0efffbe77655172e0f060dc085