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Bumped electron version and adding the possibility to dynamically change the device output sample rate

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This commit is contained in:
WolverinDEV 2021-04-19 19:08:44 +02:00
parent a604cdf77d
commit b3d4bc89f1
11 changed files with 59 additions and 37 deletions
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2
native/CMakeLists.txt
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@ -23,7 +23,7 @@ message("Module path: ${CMAKE_MODULE_PATH}")
function(setup_nodejs)
set(NodeJS_DIR "${CMAKE_CURRENT_SOURCE_DIR}/cmake/")
set(NODEJS_URL "https://atom.io/download/atom-shell")
set(NODEJS_VERSION "v8.0.0")
set(NODEJS_VERSION "v8.5.5")
#set(NODEJS_URL "https://nodejs.org/download/release/")
#set(NODEJS_VERSION "v12.13.0")

12
native/serverconnection/src/audio/AudioLevelMeter.cpp
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@ -83,11 +83,13 @@ bool InputDeviceAudioLevelMeter::running() const {
}
void InputDeviceAudioLevelMeter::stop() {
std::lock_guard lock{this->mutex};
if(this->recorder_instance) {
this->recorder_instance->remove_consumer(this);
this->recorder_instance->stop_if_possible();
this->recorder_instance = nullptr;
std::unique_lock lock{this->mutex};
auto recorder_instance_ = std::exchange(this->recorder_instance, nullptr);
if(recorder_instance_) {
lock.unlock();
/* The recorder might wait for us in this right moment */
recorder_instance_->remove_consumer(this);
recorder_instance_->stop_if_possible();
}
}

36
native/serverconnection/src/audio/AudioOutput.cpp
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@ -72,8 +72,22 @@ void AudioOutput::ensure_chunk_buffer_space(size_t output_samples) {
}
}
void AudioOutput::fill_buffer(void *output, size_t request_sample_count, size_t out_channels) {
void AudioOutput::fill_buffer(void *output, size_t request_sample_count, size_t out_sample_rate, size_t out_channels) {
assert(output);
assert(this->playback_);
if(!this->resampler_ || this->current_output_sample_rate != out_sample_rate) {
log_info(category::audio, tr("Output sample rate changed from {} to {}"), this->resampler_ ? this->resampler_->output_rate() : 0, out_sample_rate);
this->current_output_sample_rate = out_sample_rate;
this->resampler_ = std::make_unique<AudioResampler>(this->sample_rate(), out_sample_rate, this->channel_count());
if(!this->resampler_->valid()) {
log_critical(category::audio, tr("Failed to allocate a new resampler. Audio output will be silent."));
}
}
if(!this->resampler_->valid()) {
return;
}
if(out_channels != this->current_output_channels) {
log_info(category::audio, tr("Output channel count changed from {} to {}"), this->current_output_channels, out_channels);
@ -86,6 +100,10 @@ void AudioOutput::fill_buffer(void *output, size_t request_sample_count, size_t
this->chunk_buffer_sample_offset = 0;
}
return this->fill_buffer_nochecks(output, request_sample_count, out_channels);
}
void AudioOutput::fill_buffer_nochecks(void *output, size_t request_sample_count, size_t out_channels) {
auto remaining_samples{request_sample_count};
auto remaining_buffer{output};
@ -109,8 +127,7 @@ void AudioOutput::fill_buffer(void *output, size_t request_sample_count, size_t
this->fill_chunk_buffer();
this->chunk_buffer_sample_offset = 0;
return this->fill_buffer(remaining_buffer, remaining_samples, out_channels);
return this->fill_buffer_nochecks(remaining_buffer, remaining_samples, out_channels);
}
constexpr static auto kTempChunkBufferSize{64 * 1024};
@ -222,7 +239,7 @@ void AudioOutput::fill_chunk_buffer() {
assert(kTempChunkBufferSize >= this->current_output_channels * this->chunk_buffer_sample_length * sizeof(float));
audio::deinterleave(temp_chunk_buffer, (const float*) this->chunk_buffer, this->current_output_channels, this->chunk_buffer_sample_length);
webrtc::StreamConfig stream_config{(int) this->playback_->sample_rate(), this->current_output_channels};
webrtc::StreamConfig stream_config{(int) this->current_output_sample_rate, this->current_output_channels};
float* channel_ptr[kMaxChannelCount];
for(size_t channel{0}; channel < this->current_output_channels; channel++) {
@ -237,7 +254,7 @@ void AudioOutput::fill_chunk_buffer() {
return;
zero_chunk_exit: {
this->chunk_buffer_sample_length = (this->playback_->sample_rate() * kChunkTimeMs) / 1000;
this->chunk_buffer_sample_length = (this->current_output_sample_rate * kChunkTimeMs) / 1000;
this->ensure_chunk_buffer_space(this->chunk_buffer_sample_length);
memset(this->chunk_buffer, 0, this->chunk_buffer_sample_length * this->current_output_channels * sizeof(float));
return;
@ -282,15 +299,6 @@ bool AudioOutput::playback(std::string& error) {
return false;
}
if(this->playback_->sample_rate() != this->sample_rate()) {
this->resampler_ = std::make_unique<AudioResampler>(this->sample_rate(), this->playback_->sample_rate(), this->channel_count());
if(!this->resampler_->valid()) {
error = "failed to allocate a resampler";
this->playback_ = nullptr;
return false;
}
}
this->ensure_chunk_buffer_space(0);
this->playback_->register_source(this);
return this->playback_->start(error);

4
native/serverconnection/src/audio/AudioOutput.h
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@ -153,7 +153,8 @@ namespace tc::audio {
/* One audio chunk should be 10ms long */
constexpr static auto kChunkTimeMs{10};
void fill_buffer(void *, size_t request_sample_count, size_t out_channels) override;
void fill_buffer(void *, size_t request_sample_count, size_t sample_rate, size_t out_channels) override;
void fill_buffer_nochecks(void *, size_t request_sample_count, size_t out_channels);
void fill_chunk_buffer();
size_t const channel_count_;
@ -183,6 +184,7 @@ namespace tc::audio {
size_t chunk_buffer_sample_length{0};
size_t chunk_buffer_sample_offset{0};
size_t current_output_sample_rate{0};
size_t current_output_channels{0};
void ensure_chunk_buffer_space(size_t /* output samples */);

10
native/serverconnection/src/audio/driver/AudioDriver.cpp
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@ -185,21 +185,21 @@ namespace tc::audio {
}
#define TMP_BUFFER_SIZE (4096 * 16) /* 64k */
void AudioDevicePlayback::fill_buffer(void *buffer, size_t samples, size_t channels) {
void AudioDevicePlayback::fill_buffer(void *buffer, size_t samples, size_t sample_rate, size_t channels) {
std::lock_guard lock{this->source_lock};
if(!buffer) {
for(auto& source : this->_sources) {
source->fill_buffer(nullptr, samples, channels);
source->fill_buffer(nullptr, samples, sample_rate, channels);
}
return;
}
const auto size = this->_sources.size();
if(size == 1) {
this->_sources.front()->fill_buffer(buffer, samples, channels);
this->_sources.front()->fill_buffer(buffer, samples, sample_rate, channels);
} else if(size > 1) {
this->_sources.front()->fill_buffer(buffer, samples, channels);
this->_sources.front()->fill_buffer(buffer, samples, sample_rate, channels);
uint8_t tmp_buffer[TMP_BUFFER_SIZE];
if(sizeof(float) * samples * channels > TMP_BUFFER_SIZE) {
log_warn(category::audio, tr("Dropping input source data because of too small merge buffer"));
@ -207,7 +207,7 @@ namespace tc::audio {
}
for(auto it = this->_sources.begin() + 1; it != this->_sources.end(); it++) {
(*it)->fill_buffer(tmp_buffer, samples, channels);
(*it)->fill_buffer(tmp_buffer, samples, sample_rate, channels);
merge::merge_sources(buffer, buffer, tmp_buffer, channels, samples);
}
} else {

10
native/serverconnection/src/audio/driver/AudioDriver.h
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@ -43,10 +43,14 @@ namespace tc::audio {
public:
class Source {
public:
virtual void fill_buffer(void* /* target */, size_t /* samples */, size_t /* channel count */) = 0;
virtual void fill_buffer(void* /* target */, size_t /* samples */, size_t /* sample rate */, size_t /* channel count */) = 0;
};
[[nodiscard]] virtual size_t sample_rate() const = 0;
/**
* Get the current playback sample rate.
* Note: If the playback hasn't been started it might be zero.
*/
[[nodiscard]] virtual size_t current_sample_rate() const = 0;
[[nodiscard]] bool start(std::string& /* error */);
void stop_if_possible();
@ -65,7 +69,7 @@ namespace tc::audio {
virtual bool impl_start(std::string& /* error */) = 0;
virtual void impl_stop() = 0;
void fill_buffer(void* /* target */, size_t /* samples */, size_t /* channel count */);
void fill_buffer(void* /* target */, size_t /* samples */, size_t /* sample rate */, size_t /* channel count */);
std::mutex state_lock{};
bool running{false};

2
native/serverconnection/src/audio/driver/PortAudio.h
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@ -16,7 +16,7 @@ namespace tc::audio::pa {
explicit PortAudioPlayback(PaDeviceIndex index, const PaDeviceInfo* info);
virtual ~PortAudioPlayback();
[[nodiscard]] size_t sample_rate() const override;
[[nodiscard]] size_t current_sample_rate() const override;
protected:
bool impl_start(std::string& /* error */) override;
void impl_stop() override;

10
native/serverconnection/src/audio/driver/PortAudioPlayback.cpp
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@ -18,7 +18,7 @@ PortAudioPlayback::~PortAudioPlayback() {
}
bool PortAudioPlayback::impl_start(std::string &error) {
//TODO: Detect correct sample rate
//TODO: Detect a supported sample rate and use that
{
auto device_info = Pa_GetDeviceInfo(this->index);
if(this->info != device_info) {
@ -56,6 +56,7 @@ bool PortAudioPlayback::impl_start(std::string &error) {
parameters.channelCount = (int) kChannelCount;
this->source_channel_count = kChannelCount;
}
log_debug(category::audio, "Opening playback device {} (MaxChannels: {}, Channels: {})", this->info->name, this->info->maxOutputChannels, this->source_channel_count);
auto err = Pa_OpenStream(
@ -99,11 +100,14 @@ void PortAudioPlayback::impl_stop() {
this->stream = nullptr;
}
size_t PortAudioPlayback::sample_rate() const {
size_t PortAudioPlayback::current_sample_rate() const {
/* We currently only support one sample rate */
return (size_t) kSampleRate;
}
void PortAudioPlayback::write_callback(void *output, unsigned long frameCount,
const PaStreamCallbackTimeInfo *timeInfo, PaStreamCallbackFlags statusFlags) {
this->fill_buffer(output, frameCount, this->source_channel_count);
(void) timeInfo;
(void) statusFlags;
this->fill_buffer(output, frameCount, kSampleRate, this->source_channel_count);
}

8
native/serverconnection/src/audio/js/AudioProcessor.cpp
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@ -211,13 +211,13 @@ inline bool load_config_value(
return false;
}
if(value < (double) min_value) {
Nan::ThrowError(Nan::LocalStringUTF8("property " + std::string{key} + " exceeds min value of " + std::to_string(min_value)));
if((T) value < min_value) {
Nan::ThrowError(Nan::LocalStringUTF8("property " + std::string{key} + " exceeds min value of " + std::to_string((T) min_value) + " (value: " + std::to_string((T) value) + ")"));
return false;
}
if(value > (double) max_value) {
Nan::ThrowError(Nan::LocalStringUTF8("property " + std::string{key} + " exceeds max value of " + std::to_string(max_value)));
if((T) value > (double) max_value) {
Nan::ThrowError(Nan::LocalStringUTF8("property " + std::string{key} + " exceeds max value of " + std::to_string((T) max_value) + " (value: " + std::to_string((T) value) + ")"));
return false;
}

1
native/serverconnection/src/audio/js/AudioRecorder.cpp
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@ -50,6 +50,7 @@ NAN_MODULE_INIT(AudioRecorderWrapper::Init) {
Nan::SetPrototypeMethod(klass, "delete_consumer", AudioRecorderWrapper::_delete_consumer);
Nan::SetPrototypeMethod(klass, "get_audio_processor", AudioRecorderWrapper::get_audio_processor);
Nan::SetPrototypeMethod(klass, "create_level_meter", AudioRecorderWrapper::create_level_meter);
constructor().Reset(Nan::GetFunction(klass).ToLocalChecked());
}

1
native/serverconnection/src/audio/processing/AudioProcessor.cpp
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@ -125,6 +125,7 @@ void AudioProcessor::apply_config_unlocked(const Config &config) {
if(!this->current_config.rnnoise.enabled) {
this->rnnoise_volume = absl::nullopt;
}
log_trace(category::audio, tr("Applying process config:\n{}\nRNNoise: "), config.ToString(), this->current_config.rnnoise.enabled);
}
AudioProcessor::Stats AudioProcessor::get_statistics() const {