initial hooks for stereo audio
This commit is contained in:
parent
c950a264b0
commit
7da091d2b4
@ -68,6 +68,7 @@ Audio::Audio(int16_t initialJitterBufferSamples, QObject* parent) :
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_proceduralOutputDevice(NULL),
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_proceduralOutputDevice(NULL),
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_inputRingBuffer(0),
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_inputRingBuffer(0),
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_ringBuffer(NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL),
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_ringBuffer(NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL),
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_isStereoInput(false),
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_averagedLatency(0.0),
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_averagedLatency(0.0),
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_measuredJitter(0),
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_measuredJitter(0),
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_jitterBufferSamples(initialJitterBufferSamples),
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_jitterBufferSamples(initialJitterBufferSamples),
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@ -405,12 +406,12 @@ bool Audio::switchOutputToAudioDevice(const QString& outputDeviceName) {
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}
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}
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void Audio::handleAudioInput() {
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void Audio::handleAudioInput() {
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static char monoAudioDataPacket[MAX_PACKET_SIZE];
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static char audioDataPacket[MAX_PACKET_SIZE];
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static int numBytesPacketHeader = numBytesForPacketHeaderGivenPacketType(PacketTypeMicrophoneAudioNoEcho);
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static int numBytesPacketHeader = numBytesForPacketHeaderGivenPacketType(PacketTypeMicrophoneAudioNoEcho);
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static int leadingBytes = numBytesPacketHeader + sizeof(glm::vec3) + sizeof(glm::quat);
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static int leadingBytes = numBytesPacketHeader + sizeof(glm::vec3) + sizeof(glm::quat);
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static int16_t* monoAudioSamples = (int16_t*) (monoAudioDataPacket + leadingBytes);
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static int16_t* networkAudioSamples = (int16_t*) (audioDataPacket + leadingBytes);
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float inputToNetworkInputRatio = calculateDeviceToNetworkInputRatio(_numInputCallbackBytes);
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float inputToNetworkInputRatio = calculateDeviceToNetworkInputRatio(_numInputCallbackBytes);
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@ -452,125 +453,130 @@ void Audio::handleAudioInput() {
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int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
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int16_t* inputAudioSamples = new int16_t[inputSamplesRequired];
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_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
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_inputRingBuffer.readSamples(inputAudioSamples, inputSamplesRequired);
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int numNetworkBytes = _isStereoInput ? NETWORK_BUFFER_LENGTH_BYTES_STEREO : NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL;
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int numNetworkSamples = _isStereoInput ? NETWORK_BUFFER_LENGTH_SAMPLES_STEREO : NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
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// zero out the monoAudioSamples array and the locally injected audio
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// zero out the monoAudioSamples array and the locally injected audio
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memset(monoAudioSamples, 0, NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL);
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memset(networkAudioSamples, 0, numNetworkBytes);
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if (!_muted) {
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if (!_muted) {
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// we aren't muted, downsample the input audio
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// we aren't muted, downsample the input audio
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linearResampling((int16_t*) inputAudioSamples,
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linearResampling((int16_t*) inputAudioSamples, networkAudioSamples,
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monoAudioSamples,
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inputSamplesRequired, numNetworkSamples,
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inputSamplesRequired,
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NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL,
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_inputFormat, _desiredInputFormat);
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_inputFormat, _desiredInputFormat);
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//
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// only impose the noise gate and perform tone injection if we sending mono audio
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// Impose Noise Gate
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if (!_isStereoInput) {
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//
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// The Noise Gate is used to reject constant background noise by measuring the noise
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//
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// floor observed at the microphone and then opening the 'gate' to allow microphone
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// Impose Noise Gate
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// signals to be transmitted when the microphone samples average level exceeds a multiple
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//
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// of the noise floor.
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// The Noise Gate is used to reject constant background noise by measuring the noise
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//
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// floor observed at the microphone and then opening the 'gate' to allow microphone
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// NOISE_GATE_HEIGHT: How loud you have to speak relative to noise background to open the gate.
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// signals to be transmitted when the microphone samples average level exceeds a multiple
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// Make this value lower for more sensitivity and less rejection of noise.
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// of the noise floor.
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// NOISE_GATE_WIDTH: The number of samples in an audio frame for which the height must be exceeded
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//
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// to open the gate.
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// NOISE_GATE_HEIGHT: How loud you have to speak relative to noise background to open the gate.
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// NOISE_GATE_CLOSE_FRAME_DELAY: Once the noise is below the gate height for the frame, how many frames
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// Make this value lower for more sensitivity and less rejection of noise.
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// will we wait before closing the gate.
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// NOISE_GATE_WIDTH: The number of samples in an audio frame for which the height must be exceeded
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// NOISE_GATE_FRAMES_TO_AVERAGE: How many audio frames should we average together to compute noise floor.
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// to open the gate.
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// More means better rejection but also can reject continuous things like singing.
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// NOISE_GATE_CLOSE_FRAME_DELAY: Once the noise is below the gate height for the frame, how many frames
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// NUMBER_OF_NOISE_SAMPLE_FRAMES: How often should we re-evaluate the noise floor?
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// will we wait before closing the gate.
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// NOISE_GATE_FRAMES_TO_AVERAGE: How many audio frames should we average together to compute noise floor.
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// More means better rejection but also can reject continuous things like singing.
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float loudness = 0;
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// NUMBER_OF_NOISE_SAMPLE_FRAMES: How often should we re-evaluate the noise floor?
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float thisSample = 0;
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int samplesOverNoiseGate = 0;
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float loudness = 0;
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const float NOISE_GATE_HEIGHT = 7.0f;
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float thisSample = 0;
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const int NOISE_GATE_WIDTH = 5;
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int samplesOverNoiseGate = 0;
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const int NOISE_GATE_CLOSE_FRAME_DELAY = 5;
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const int NOISE_GATE_FRAMES_TO_AVERAGE = 5;
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const float NOISE_GATE_HEIGHT = 7.0f;
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const float DC_OFFSET_AVERAGING = 0.99f;
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const int NOISE_GATE_WIDTH = 5;
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const float CLIPPING_THRESHOLD = 0.90f;
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const int NOISE_GATE_CLOSE_FRAME_DELAY = 5;
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const int NOISE_GATE_FRAMES_TO_AVERAGE = 5;
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//
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const float DC_OFFSET_AVERAGING = 0.99f;
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// Check clipping, adjust DC offset, and check if should open noise gate
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const float CLIPPING_THRESHOLD = 0.90f;
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//
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float measuredDcOffset = 0.0f;
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//
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// Increment the time since the last clip
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// Check clipping, adjust DC offset, and check if should open noise gate
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if (_timeSinceLastClip >= 0.0f) {
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//
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_timeSinceLastClip += (float) NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL / (float) SAMPLE_RATE;
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float measuredDcOffset = 0.0f;
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}
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// Increment the time since the last clip
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if (_timeSinceLastClip >= 0.0f) {
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for (int i = 0; i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; i++) {
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_timeSinceLastClip += (float) NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL / (float) SAMPLE_RATE;
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measuredDcOffset += monoAudioSamples[i];
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monoAudioSamples[i] -= (int16_t) _dcOffset;
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thisSample = fabsf(monoAudioSamples[i]);
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if (thisSample >= (32767.0f * CLIPPING_THRESHOLD)) {
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_timeSinceLastClip = 0.0f;
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}
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}
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loudness += thisSample;
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// Noise Reduction: Count peaks above the average loudness
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if (_noiseGateEnabled && (thisSample > (_noiseGateMeasuredFloor * NOISE_GATE_HEIGHT))) {
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samplesOverNoiseGate++;
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}
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}
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measuredDcOffset /= NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
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if (_dcOffset == 0.0f) {
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// On first frame, copy over measured offset
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_dcOffset = measuredDcOffset;
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} else {
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_dcOffset = DC_OFFSET_AVERAGING * _dcOffset + (1.0f - DC_OFFSET_AVERAGING) * measuredDcOffset;
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}
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// Add tone injection if enabled
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const float TONE_FREQ = 220.0f / SAMPLE_RATE * TWO_PI;
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const float QUARTER_VOLUME = 8192.0f;
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if (_toneInjectionEnabled) {
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loudness = 0.0f;
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for (int i = 0; i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; i++) {
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for (int i = 0; i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; i++) {
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monoAudioSamples[i] = QUARTER_VOLUME * sinf(TONE_FREQ * (float)(i + _proceduralEffectSample));
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measuredDcOffset += networkAudioSamples[i];
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loudness += fabsf(monoAudioSamples[i]);
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networkAudioSamples[i] -= (int16_t) _dcOffset;
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}
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thisSample = fabsf(networkAudioSamples[i]);
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}
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if (thisSample >= (32767.0f * CLIPPING_THRESHOLD)) {
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_lastInputLoudness = fabs(loudness / NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
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_timeSinceLastClip = 0.0f;
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}
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// If Noise Gate is enabled, check and turn the gate on and off
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loudness += thisSample;
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if (!_toneInjectionEnabled && _noiseGateEnabled) {
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// Noise Reduction: Count peaks above the average loudness
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float averageOfAllSampleFrames = 0.0f;
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if (_noiseGateEnabled && (thisSample > (_noiseGateMeasuredFloor * NOISE_GATE_HEIGHT))) {
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_noiseSampleFrames[_noiseGateSampleCounter++] = _lastInputLoudness;
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samplesOverNoiseGate++;
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if (_noiseGateSampleCounter == NUMBER_OF_NOISE_SAMPLE_FRAMES) {
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float smallestSample = FLT_MAX;
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for (int i = 0; i <= NUMBER_OF_NOISE_SAMPLE_FRAMES - NOISE_GATE_FRAMES_TO_AVERAGE; i += NOISE_GATE_FRAMES_TO_AVERAGE) {
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float thisAverage = 0.0f;
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for (int j = i; j < i + NOISE_GATE_FRAMES_TO_AVERAGE; j++) {
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thisAverage += _noiseSampleFrames[j];
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averageOfAllSampleFrames += _noiseSampleFrames[j];
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}
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thisAverage /= NOISE_GATE_FRAMES_TO_AVERAGE;
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if (thisAverage < smallestSample) {
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smallestSample = thisAverage;
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}
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}
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}
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averageOfAllSampleFrames /= NUMBER_OF_NOISE_SAMPLE_FRAMES;
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_noiseGateMeasuredFloor = smallestSample;
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_noiseGateSampleCounter = 0;
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}
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}
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if (samplesOverNoiseGate > NOISE_GATE_WIDTH) {
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_noiseGateOpen = true;
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measuredDcOffset /= NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
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_noiseGateFramesToClose = NOISE_GATE_CLOSE_FRAME_DELAY;
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if (_dcOffset == 0.0f) {
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// On first frame, copy over measured offset
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_dcOffset = measuredDcOffset;
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} else {
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} else {
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if (--_noiseGateFramesToClose == 0) {
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_dcOffset = DC_OFFSET_AVERAGING * _dcOffset + (1.0f - DC_OFFSET_AVERAGING) * measuredDcOffset;
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_noiseGateOpen = false;
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}
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// Add tone injection if enabled
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const float TONE_FREQ = 220.0f / SAMPLE_RATE * TWO_PI;
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const float QUARTER_VOLUME = 8192.0f;
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if (_toneInjectionEnabled) {
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loudness = 0.0f;
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for (int i = 0; i < NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; i++) {
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networkAudioSamples[i] = QUARTER_VOLUME * sinf(TONE_FREQ * (float)(i + _proceduralEffectSample));
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loudness += fabsf(networkAudioSamples[i]);
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}
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}
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}
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}
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if (!_noiseGateOpen) {
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_lastInputLoudness = fabs(loudness / NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
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memset(monoAudioSamples, 0, NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL);
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_lastInputLoudness = 0;
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// If Noise Gate is enabled, check and turn the gate on and off
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if (!_toneInjectionEnabled && _noiseGateEnabled) {
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float averageOfAllSampleFrames = 0.0f;
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_noiseSampleFrames[_noiseGateSampleCounter++] = _lastInputLoudness;
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if (_noiseGateSampleCounter == NUMBER_OF_NOISE_SAMPLE_FRAMES) {
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float smallestSample = FLT_MAX;
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for (int i = 0; i <= NUMBER_OF_NOISE_SAMPLE_FRAMES - NOISE_GATE_FRAMES_TO_AVERAGE; i += NOISE_GATE_FRAMES_TO_AVERAGE) {
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float thisAverage = 0.0f;
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for (int j = i; j < i + NOISE_GATE_FRAMES_TO_AVERAGE; j++) {
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thisAverage += _noiseSampleFrames[j];
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averageOfAllSampleFrames += _noiseSampleFrames[j];
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}
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thisAverage /= NOISE_GATE_FRAMES_TO_AVERAGE;
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if (thisAverage < smallestSample) {
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smallestSample = thisAverage;
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}
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}
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averageOfAllSampleFrames /= NUMBER_OF_NOISE_SAMPLE_FRAMES;
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_noiseGateMeasuredFloor = smallestSample;
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_noiseGateSampleCounter = 0;
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}
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if (samplesOverNoiseGate > NOISE_GATE_WIDTH) {
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_noiseGateOpen = true;
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_noiseGateFramesToClose = NOISE_GATE_CLOSE_FRAME_DELAY;
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} else {
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if (--_noiseGateFramesToClose == 0) {
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_noiseGateOpen = false;
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}
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}
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if (!_noiseGateOpen) {
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memset(networkAudioSamples, 0, NETWORK_BUFFER_LENGTH_BYTES_PER_CHANNEL);
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_lastInputLoudness = 0;
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}
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}
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}
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}
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}
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} else {
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} else {
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@ -580,19 +586,19 @@ void Audio::handleAudioInput() {
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// at this point we have clean monoAudioSamples, which match our target output...
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// at this point we have clean monoAudioSamples, which match our target output...
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// this is what we should send to our interested listeners
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// this is what we should send to our interested listeners
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if (_processSpatialAudio && !_muted && _audioOutput) {
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if (_processSpatialAudio && !_muted && !_isStereoInput && _audioOutput) {
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QByteArray monoInputData((char*)monoAudioSamples, NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL * sizeof(int16_t));
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QByteArray monoInputData((char*)networkAudioSamples, NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL * sizeof(int16_t));
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emit processLocalAudio(_spatialAudioStart, monoInputData, _desiredInputFormat);
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emit processLocalAudio(_spatialAudioStart, monoInputData, _desiredInputFormat);
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}
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}
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if (_proceduralAudioOutput) {
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if (!_isStereoInput && _proceduralAudioOutput) {
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processProceduralAudio(monoAudioSamples, NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
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processProceduralAudio(networkAudioSamples, NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL);
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}
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}
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if (_scopeEnabled && !_scopeEnabledPause) {
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if (!_isStereoInput && _scopeEnabled && !_scopeEnabledPause) {
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unsigned int numMonoAudioChannels = 1;
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unsigned int numMonoAudioChannels = 1;
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unsigned int monoAudioChannel = 0;
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unsigned int monoAudioChannel = 0;
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addBufferToScope(_scopeInput, _scopeInputOffset, monoAudioSamples, monoAudioChannel, numMonoAudioChannels);
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addBufferToScope(_scopeInput, _scopeInputOffset, networkAudioSamples, monoAudioChannel, numMonoAudioChannels);
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_scopeInputOffset += NETWORK_SAMPLES_PER_FRAME;
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_scopeInputOffset += NETWORK_SAMPLES_PER_FRAME;
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_scopeInputOffset %= _samplesPerScope;
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_scopeInputOffset %= _samplesPerScope;
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}
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}
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@ -615,7 +621,7 @@ void Audio::handleAudioInput() {
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packetType = PacketTypeSilentAudioFrame;
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packetType = PacketTypeSilentAudioFrame;
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// we need to indicate how many silent samples this is to the audio mixer
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// we need to indicate how many silent samples this is to the audio mixer
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monoAudioSamples[0] = NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
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networkAudioSamples[0] = NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
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numAudioBytes = sizeof(int16_t);
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numAudioBytes = sizeof(int16_t);
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} else {
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} else {
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@ -628,7 +634,7 @@ void Audio::handleAudioInput() {
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}
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}
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}
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}
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char* currentPacketPtr = monoAudioDataPacket + populatePacketHeader(monoAudioDataPacket, packetType);
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char* currentPacketPtr = audioDataPacket + populatePacketHeader(audioDataPacket, packetType);
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// memcpy the three float positions
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// memcpy the three float positions
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memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
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memcpy(currentPacketPtr, &headPosition, sizeof(headPosition));
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@ -638,7 +644,7 @@ void Audio::handleAudioInput() {
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memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
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memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
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currentPacketPtr += sizeof(headOrientation);
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currentPacketPtr += sizeof(headOrientation);
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nodeList->writeDatagram(monoAudioDataPacket, numAudioBytes + leadingBytes, audioMixer);
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nodeList->writeDatagram(audioDataPacket, numAudioBytes + leadingBytes, audioMixer);
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Application::getInstance()->getBandwidthMeter()->outputStream(BandwidthMeter::AUDIO)
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Application::getInstance()->getBandwidthMeter()->outputStream(BandwidthMeter::AUDIO)
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.updateValue(numAudioBytes + leadingBytes);
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.updateValue(numAudioBytes + leadingBytes);
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@ -761,6 +767,24 @@ void Audio::toggleAudioNoiseReduction() {
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_noiseGateEnabled = !_noiseGateEnabled;
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_noiseGateEnabled = !_noiseGateEnabled;
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}
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}
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void Audio::toggleStereoInput() {
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int oldChannelCount = _desiredInputFormat.channelCount();
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QAction* stereoAudioOption = Menu::getInstance()->getActionForOption(MenuOption::StereoAudio);
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if (stereoAudioOption->isChecked()) {
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_desiredInputFormat.setChannelCount(2);
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_isStereoInput = true;
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} else {
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_desiredInputFormat.setChannelCount(1);
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_isStereoInput = false;
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}
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if (oldChannelCount != _desiredInputFormat.channelCount()) {
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// change in channel count for desired input format, restart the input device
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switchInputToAudioDevice(_inputAudioDeviceName);
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}
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}
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void Audio::processReceivedAudio(const QByteArray& audioByteArray) {
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void Audio::processReceivedAudio(const QByteArray& audioByteArray) {
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_ringBuffer.parseData(audioByteArray);
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_ringBuffer.parseData(audioByteArray);
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@ -1300,18 +1324,21 @@ bool Audio::switchInputToAudioDevice(const QAudioDeviceInfo& inputDeviceInfo) {
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if (adjustedFormatForAudioDevice(inputDeviceInfo, _desiredInputFormat, _inputFormat)) {
|
if (adjustedFormatForAudioDevice(inputDeviceInfo, _desiredInputFormat, _inputFormat)) {
|
||||||
qDebug() << "The format to be used for audio input is" << _inputFormat;
|
qDebug() << "The format to be used for audio input is" << _inputFormat;
|
||||||
|
|
||||||
_audioInput = new QAudioInput(inputDeviceInfo, _inputFormat, this);
|
// if the user wants stereo but this device can't provide then bail
|
||||||
_numInputCallbackBytes = calculateNumberOfInputCallbackBytes(_inputFormat);
|
if (!_isStereoInput || _inputFormat.channelCount() == 2) {
|
||||||
_audioInput->setBufferSize(_numInputCallbackBytes);
|
_audioInput = new QAudioInput(inputDeviceInfo, _inputFormat, this);
|
||||||
|
_numInputCallbackBytes = calculateNumberOfInputCallbackBytes(_inputFormat);
|
||||||
// how do we want to handle input working, but output not working?
|
_audioInput->setBufferSize(_numInputCallbackBytes);
|
||||||
int numFrameSamples = calculateNumberOfFrameSamples(_numInputCallbackBytes);
|
|
||||||
_inputRingBuffer.resizeForFrameSize(numFrameSamples);
|
// how do we want to handle input working, but output not working?
|
||||||
_inputDevice = _audioInput->start();
|
int numFrameSamples = calculateNumberOfFrameSamples(_numInputCallbackBytes);
|
||||||
connect(_inputDevice, SIGNAL(readyRead()), this, SLOT(handleAudioInput()));
|
_inputRingBuffer.resizeForFrameSize(numFrameSamples);
|
||||||
|
_inputDevice = _audioInput->start();
|
||||||
supportedFormat = true;
|
connect(_inputDevice, SIGNAL(readyRead()), this, SLOT(handleAudioInput()));
|
||||||
|
|
||||||
|
supportedFormat = true;
|
||||||
|
}
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
return supportedFormat;
|
return supportedFormat;
|
||||||
|
@ -85,6 +85,7 @@ public slots:
|
|||||||
void toggleScope();
|
void toggleScope();
|
||||||
void toggleScopePause();
|
void toggleScopePause();
|
||||||
void toggleAudioSpatialProcessing();
|
void toggleAudioSpatialProcessing();
|
||||||
|
void toggleStereoInput();
|
||||||
void selectAudioScopeFiveFrames();
|
void selectAudioScopeFiveFrames();
|
||||||
void selectAudioScopeTwentyFrames();
|
void selectAudioScopeTwentyFrames();
|
||||||
void selectAudioScopeFiftyFrames();
|
void selectAudioScopeFiftyFrames();
|
||||||
@ -127,6 +128,7 @@ private:
|
|||||||
QIODevice* _proceduralOutputDevice;
|
QIODevice* _proceduralOutputDevice;
|
||||||
AudioRingBuffer _inputRingBuffer;
|
AudioRingBuffer _inputRingBuffer;
|
||||||
AudioRingBuffer _ringBuffer;
|
AudioRingBuffer _ringBuffer;
|
||||||
|
bool _isStereoInput;
|
||||||
|
|
||||||
QString _inputAudioDeviceName;
|
QString _inputAudioDeviceName;
|
||||||
QString _outputAudioDeviceName;
|
QString _outputAudioDeviceName;
|
||||||
|
@ -432,6 +432,8 @@ Menu::Menu() :
|
|||||||
SLOT(toggleAudioNoiseReduction()));
|
SLOT(toggleAudioNoiseReduction()));
|
||||||
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::EchoServerAudio);
|
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::EchoServerAudio);
|
||||||
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::EchoLocalAudio);
|
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::EchoLocalAudio);
|
||||||
|
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::StereoAudio, 0, false,
|
||||||
|
appInstance->getAudio(), SLOT(toggleStereoInput()));
|
||||||
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::MuteAudio,
|
addCheckableActionToQMenuAndActionHash(audioDebugMenu, MenuOption::MuteAudio,
|
||||||
Qt::CTRL | Qt::Key_M,
|
Qt::CTRL | Qt::Key_M,
|
||||||
false,
|
false,
|
||||||
|
@ -402,6 +402,7 @@ namespace MenuOption {
|
|||||||
const QString StandOnNearbyFloors = "Stand on nearby floors";
|
const QString StandOnNearbyFloors = "Stand on nearby floors";
|
||||||
const QString Stars = "Stars";
|
const QString Stars = "Stars";
|
||||||
const QString Stats = "Stats";
|
const QString Stats = "Stats";
|
||||||
|
const QString StereoAudio = "Stereo Audio";
|
||||||
const QString StopAllScripts = "Stop All Scripts";
|
const QString StopAllScripts = "Stop All Scripts";
|
||||||
const QString SuppressShortTimings = "Suppress Timings Less than 10ms";
|
const QString SuppressShortTimings = "Suppress Timings Less than 10ms";
|
||||||
const QString TestPing = "Test Ping";
|
const QString TestPing = "Test Ping";
|
||||||
|
@ -20,14 +20,15 @@
|
|||||||
|
|
||||||
#include "PositionalAudioRingBuffer.h"
|
#include "PositionalAudioRingBuffer.h"
|
||||||
|
|
||||||
PositionalAudioRingBuffer::PositionalAudioRingBuffer(PositionalAudioRingBuffer::Type type) :
|
PositionalAudioRingBuffer::PositionalAudioRingBuffer(PositionalAudioRingBuffer::Type type, bool isStereo) :
|
||||||
AudioRingBuffer(NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL),
|
AudioRingBuffer(isStereo ? NETWORK_BUFFER_LENGTH_SAMPLES_STEREO : NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL),
|
||||||
_type(type),
|
_type(type),
|
||||||
_position(0.0f, 0.0f, 0.0f),
|
_position(0.0f, 0.0f, 0.0f),
|
||||||
_orientation(0.0f, 0.0f, 0.0f, 0.0f),
|
_orientation(0.0f, 0.0f, 0.0f, 0.0f),
|
||||||
_willBeAddedToMix(false),
|
_willBeAddedToMix(false),
|
||||||
_shouldLoopbackForNode(false),
|
_shouldLoopbackForNode(false),
|
||||||
_shouldOutputStarveDebug(true)
|
_shouldOutputStarveDebug(true),
|
||||||
|
_isStereo(isStereo)
|
||||||
{
|
{
|
||||||
|
|
||||||
}
|
}
|
||||||
|
@ -24,7 +24,7 @@ public:
|
|||||||
Injector
|
Injector
|
||||||
};
|
};
|
||||||
|
|
||||||
PositionalAudioRingBuffer(PositionalAudioRingBuffer::Type type);
|
PositionalAudioRingBuffer(PositionalAudioRingBuffer::Type type, bool isStereo = false);
|
||||||
~PositionalAudioRingBuffer();
|
~PositionalAudioRingBuffer();
|
||||||
|
|
||||||
int parseData(const QByteArray& packet);
|
int parseData(const QByteArray& packet);
|
||||||
@ -56,6 +56,7 @@ protected:
|
|||||||
bool _willBeAddedToMix;
|
bool _willBeAddedToMix;
|
||||||
bool _shouldLoopbackForNode;
|
bool _shouldLoopbackForNode;
|
||||||
bool _shouldOutputStarveDebug;
|
bool _shouldOutputStarveDebug;
|
||||||
|
bool _isStereo;
|
||||||
|
|
||||||
float _nextOutputTrailingLoudness;
|
float _nextOutputTrailingLoudness;
|
||||||
};
|
};
|
||||||
|
@ -47,6 +47,9 @@ int packArithmeticallyCodedValue(int value, char* destination) {
|
|||||||
|
|
||||||
PacketVersion versionForPacketType(PacketType type) {
|
PacketVersion versionForPacketType(PacketType type) {
|
||||||
switch (type) {
|
switch (type) {
|
||||||
|
case PacketTypeMicrophoneAudioNoEcho:
|
||||||
|
case PacketTypeMicrophoneAudioWithEcho:
|
||||||
|
return 1;
|
||||||
case PacketTypeAvatarData:
|
case PacketTypeAvatarData:
|
||||||
return 3;
|
return 3;
|
||||||
case PacketTypeAvatarIdentity:
|
case PacketTypeAvatarIdentity:
|
||||||
|
Loading…
x
Reference in New Issue
Block a user