// // Audio.cpp // interface // // Created by Stephen Birarda on 1/22/13. // Copyright (c) 2013 High Fidelity, Inc.. All rights reserved. // #include #include #include #include #include "Audio.h" #include "Util.h" #include "AudioSource.h" #include "UDPSocket.h" const int BUFFER_LENGTH_BYTES = 1024; const int BUFFER_LENGTH_SAMPLES = BUFFER_LENGTH_BYTES / sizeof(int16_t); const int PHASE_DELAY_AT_90 = 20; const int AMPLITUDE_RATIO_AT_90 = 0.5; const int NUM_AUDIO_SOURCES = 1; const int ECHO_SERVER_TEST = 1; const int AUDIO_UDP_LISTEN_PORT = 55444; #define LOG_SAMPLE_DELAY 1 bool Audio::initialized; PaError Audio::err; PaStream *Audio::inputStream; PaStream *Audio::outputStream; AudioData *Audio::data; std::ofstream logFile; /** * Audio callback used by portaudio. * Communicates with Audio via a shared pointer to Audio::data. * Writes input audio channels (if they exist) into Audio::data->buffer, multiplied by Audio::data->inputGain. * Then writes Audio::data->buffer into output audio channels, and clears the portion of Audio::data->buffer that has been read from for reuse. * * @param[in] inputBuffer A pointer to an internal portaudio data buffer containing data read by portaudio. * @param[out] outputBuffer A pointer to an internal portaudio data buffer to be read by the configured output device. * @param[in] frames Number of frames that portaudio requests to be read/written. (Valid size of input/output buffers = frames * number of channels (2) * sizeof data type (float)). * @param[in] timeInfo Portaudio time info. Currently unused. * @param[in] statusFlags Portaudio status flags. Currently unused. * @param[in] userData Pointer to supplied user data (in this case, a pointer to Audio::data). Used to communicate with external code (since portaudio calls this function from another thread). * @return Should be of type PaStreamCallbackResult. Return paComplete to end the stream, or paContinue to continue (default). Can be used to end the stream from within the callback. */ int inputCallback(const void *inputBuffer, void *outputBuffer, unsigned long frames, const PaStreamCallbackTimeInfo *timeInfo, PaStreamCallbackFlags statusFlags, void *userData) { UDPSocket *audioSocket = (UDPSocket *) userData; int16_t *inputLeft = ((int16_t **) inputBuffer)[0]; // int16_t *inputRight = ((int16_t **) inputBuffer)[1]; if (inputLeft != NULL) { audioSocket->send((char *) "192.168.1.19", 55443, (void *)inputLeft, BUFFER_LENGTH_BYTES); } return paContinue; } int outputCallback(const void *inputBuffer, void *outputBuffer, unsigned long frames, const PaStreamCallbackTimeInfo *timeInfo, PaStreamCallbackFlags statusFlags, void *userData) { AudioData *data = (AudioData *) userData; int16_t *outputLeft = ((int16_t **) outputBuffer)[0]; int16_t *outputRight = ((int16_t **) outputBuffer)[1]; for (int s = 0; s < NUM_AUDIO_SOURCES; s++) { AudioSource *source = data->sources[s]; if (ECHO_SERVER_TEST) { // copy whatever is source->sourceData to the left and right output channels memcpy(outputLeft, source->sourceData, BUFFER_LENGTH_BYTES); memcpy(outputRight, source->sourceData, BUFFER_LENGTH_BYTES); } else { memset(outputLeft, 0, BUFFER_LENGTH_BYTES); memset(outputRight, 0, BUFFER_LENGTH_BYTES); glm::vec3 headPos = data->linkedHead->getPos(); glm::vec3 sourcePos = source->position; int startPointer = source->samplePointer; int wrapAroundSamples = (BUFFER_LENGTH_SAMPLES) - (source->lengthInSamples - source->samplePointer); if (wrapAroundSamples <= 0) { memcpy(data->samplesToQueue, source->sourceData + source->samplePointer, BUFFER_LENGTH_BYTES); source->samplePointer += (BUFFER_LENGTH_SAMPLES); } else { memcpy(data->samplesToQueue, source->sourceData + source->samplePointer, (source->lengthInSamples - source->samplePointer) * sizeof(int16_t)); memcpy(data->samplesToQueue + (source->lengthInSamples - source->samplePointer), source->sourceData, wrapAroundSamples * sizeof(int16_t)); source->samplePointer = wrapAroundSamples; } float distance = sqrtf(powf(-headPos[0] - sourcePos[0], 2) + powf(-headPos[2] - sourcePos[2], 2)); float distanceAmpRatio = powf(0.5, cbrtf(distance * 10)); float angleToSource = angle_to(headPos * -1.f, sourcePos, data->linkedHead->getRenderYaw(), data->linkedHead->getYaw()) * M_PI/180; float sinRatio = sqrt(fabsf(sinf(angleToSource))); int numSamplesDelay = PHASE_DELAY_AT_90 * sinRatio; float phaseAmpRatio = 1.f - (AMPLITUDE_RATIO_AT_90 * sinRatio); // std::cout << "S: " << numSamplesDelay << " A: " << angleToSource << " S: " << sinRatio << " AR: " << phaseAmpRatio << "\n"; int16_t *leadingOutput = angleToSource > 0 ? outputLeft : outputRight; int16_t *trailingOutput = angleToSource > 0 ? outputRight : outputLeft; for (int i = 0; i < BUFFER_LENGTH_SAMPLES; i++) { data->samplesToQueue[i] *= distanceAmpRatio / NUM_AUDIO_SOURCES; leadingOutput[i] += data->samplesToQueue[i]; if (i >= numSamplesDelay) { trailingOutput[i] += data->samplesToQueue[i - numSamplesDelay]; } else { int sampleIndex = startPointer - numSamplesDelay + i; if (sampleIndex < 0) { sampleIndex += source->lengthInSamples; } trailingOutput[i] += source->sourceData[sampleIndex] * (distanceAmpRatio * phaseAmpRatio / NUM_AUDIO_SOURCES); } } } } return paContinue; } struct AudioRecThreadStruct { AudioData *sharedAudioData; }; void *receiveAudioViaUDP(void *args) { AudioRecThreadStruct *threadArgs = (AudioRecThreadStruct *) args; AudioData *sharedAudioData = threadArgs->sharedAudioData; int16_t *receivedData = new int16_t[BUFFER_LENGTH_SAMPLES]; int *receivedBytes = new int; timeval previousReceiveTime, currentReceiveTime; if (LOG_SAMPLE_DELAY) { gettimeofday(&previousReceiveTime, NULL); char *filename = new char[50]; sprintf(filename, "%s/Desktop/%ld.csv", getenv("HOME"), previousReceiveTime.tv_sec); logFile.open(filename, std::ios::out); delete[] filename; } while (true) { if (sharedAudioData->audioSocket->receive((void *)receivedData, receivedBytes)) { if (LOG_SAMPLE_DELAY) { // write time difference (in microseconds) between packet receipts to file gettimeofday(¤tReceiveTime, NULL); double timeDiff = diffclock(previousReceiveTime, currentReceiveTime); logFile << timeDiff << std::endl; } // add the received data to the shared memory memcpy(sharedAudioData->sources[0]->sourceData, receivedData, *receivedBytes); if (LOG_SAMPLE_DELAY) { gettimeofday(&previousReceiveTime, NULL); } } } } /** * Initialize portaudio and start an audio stream. * Should be called at the beginning of program exection. * @seealso Audio::terminate * @return Returns true if successful or false if an error occurred. Use Audio::getError() to retrieve the error code. */ bool Audio::init() { Head *deadHead = new Head(); return Audio::init(deadHead); } bool Audio::init(Head *mainHead) { err = Pa_Initialize(); if (err != paNoError) goto error; if (ECHO_SERVER_TEST) { data = new AudioData(1, BUFFER_LENGTH_BYTES); // setup a UDPSocket data->audioSocket = new UDPSocket(AUDIO_UDP_LISTEN_PORT); // setup the ring buffer source for the streamed audio data->sources[0]->sourceData = new int16_t[BUFFER_LENGTH_SAMPLES]; memset(data->sources[0]->sourceData, 0, BUFFER_LENGTH_SAMPLES * sizeof(int16_t)); pthread_t audioReceiveThread; AudioRecThreadStruct threadArgs; threadArgs.sharedAudioData = data; pthread_create(&audioReceiveThread, NULL, receiveAudioViaUDP, (void *) &threadArgs); } else { data = new AudioData(NUM_AUDIO_SOURCES, BUFFER_LENGTH_BYTES); data->sources[0]->position = glm::vec3(6, 0, -1); data->sources[0]->loadDataFromFile("jeska.raw"); data->sources[1]->position = glm::vec3(6, 0, 6); data->sources[1]->loadDataFromFile("grayson.raw"); } data->linkedHead = mainHead; err = Pa_OpenDefaultStream(&inputStream, 2, // input channels NULL, // output channels (paInt16 | paNonInterleaved), // sample format 22050, // sample rate (hz) 512, // frames per buffer inputCallback, // callback function (void *) data->audioSocket); // user data to be passed to callback if (err != paNoError) goto error; err = Pa_OpenDefaultStream(&outputStream, NULL, // input channels 2, // output channels (paInt16 | paNonInterleaved), // sample format 22050, // sample rate (hz) 512, // frames per buffer outputCallback, // callback function (void *) data); // user data to be passed to callback if (err != paNoError) goto error; initialized = true; // start the streams now that sources are good to go Pa_StartStream(inputStream); if (err != paNoError) goto error; Pa_StartStream(outputStream); if (err != paNoError) goto error; return paNoError; error: fprintf(stderr, "-- Failed to initialize portaudio --\n"); fprintf(stderr, "PortAudio error (%d): %s\n", err, Pa_GetErrorText(err)); initialized = false; delete[] data; return false; } void Audio::render() { if (initialized && !ECHO_SERVER_TEST) { for (int s = 0; s < NUM_AUDIO_SOURCES; s++) { // render gl objects on screen for our sources glPushMatrix(); glTranslatef(data->sources[s]->position[0], data->sources[s]->position[1], data->sources[s]->position[2]); glColor3f((s == 0 ? 1 : 0), (s == 1 ? 1 : 0), (s == 2 ? 1 : 0)); glutSolidCube(0.5); glPopMatrix(); } } } /** * Close the running audio stream, and deinitialize portaudio. * Should be called at the end of program execution. * @return Returns true if the initialization was successful, or false if an error occured. The error code may be retrieved by Audio::getError(). */ bool Audio::terminate () { if (!initialized) { return true; } else { initialized = false; err = Pa_CloseStream(inputStream); if (err != paNoError) goto error; err = Pa_CloseStream(outputStream); if (err != paNoError) goto error; delete data; err = Pa_Terminate(); if (err != paNoError) goto error; logFile.close(); return true; } error: fprintf(stderr, "-- portaudio termination error --\n"); fprintf(stderr, "PortAudio error (%d): %s\n", err, Pa_GetErrorText(err)); return false; }