add UDP send/receive of audio data for comm with mixer

This commit is contained in:
Stephen Birarda 2013-01-30 12:37:11 -08:00
parent 0ff60fa892
commit efd5f2cdc0
8 changed files with 173 additions and 88 deletions

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@ -15,7 +15,6 @@ AudioData::AudioData(int numberOfSources, int bufferLength) {
for(int s = 0; s < numberOfSources; s++) { for(int s = 0; s < numberOfSources; s++) {
sources[s] = new AudioSource(); sources[s] = new AudioSource();
std::cout << "Created a new audio source!\n";
} }
samplesToQueue = new int16_t[bufferLength / sizeof(int16_t)]; samplesToQueue = new int16_t[bufferLength / sizeof(int16_t)];

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@ -12,11 +12,13 @@
#include <iostream> #include <iostream>
#include "AudioSource.h" #include "AudioSource.h"
#include "head.h" #include "head.h"
#include "UDPSocket.h"
class AudioData { class AudioData {
public: public:
Head *linkedHead; Head *linkedHead;
AudioSource **sources; AudioSource **sources;
UDPSocket *audioSocket;
int16_t *samplesToQueue; int16_t *samplesToQueue;

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@ -19,7 +19,7 @@ class AudioSource {
int lengthInSamples; int lengthInSamples;
int samplePointer; int samplePointer;
AudioSource() { samplePointer = 0; sourceData = NULL; } AudioSource() { samplePointer = 0; sourceData = NULL; lengthInSamples = 0; }
~AudioSource(); ~AudioSource();
int loadDataFromFile(const char *filename); int loadDataFromFile(const char *filename);

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@ -11,17 +11,8 @@
#include <arpa/inet.h> #include <arpa/inet.h>
#include <fcntl.h> #include <fcntl.h>
struct sockaddr_in UDPSocket::sockaddr_util(char* hostname, int port) { sockaddr_in destSockaddr, senderAddress;
sockaddr_in dest_address; socklen_t addLength = sizeof(senderAddress);
dest_address.sin_family = AF_INET;
dest_address.sin_addr.s_addr = inet_addr(hostname);
dest_address.sin_port = htons((uint16_t)port);
return dest_address;
}
struct sockaddr_in dest_sockaddr;
UDPSocket::UDPSocket(int listeningPort) { UDPSocket::UDPSocket(int listeningPort) {
// create the socket // create the socket
@ -33,10 +24,16 @@ UDPSocket::UDPSocket(int listeningPort) {
} }
// instantiate the re-usable dest_sockaddr with a dummy IP and port // instantiate the re-usable dest_sockaddr with a dummy IP and port
dest_sockaddr = UDPSocket::sockaddr_util((char *) "1.0.0.0", 1); sockaddr_in dest_sockaddr;
dest_sockaddr.sin_family = AF_INET;
dest_sockaddr.sin_addr.s_addr = inet_addr("1.0.0.0");
dest_sockaddr.sin_port = htons((uint16_t) 1);
// bind the socket to the passed listeningPort // bind the socket to the passed listeningPort
sockaddr_in bind_address = UDPSocket::sockaddr_util(INADDR_ANY, listeningPort); sockaddr_in bind_address;
bind_address.sin_family = AF_INET;
bind_address.sin_addr.s_addr = INADDR_ANY;
bind_address.sin_port = htons((uint16_t) listeningPort);
if (bind(handle, (const sockaddr*) &bind_address, sizeof(sockaddr_in)) < 0) { if (bind(handle, (const sockaddr*) &bind_address, sizeof(sockaddr_in)) < 0) {
printf("Failed to bind socket to port %d.\n", listeningPort); printf("Failed to bind socket to port %d.\n", listeningPort);
@ -49,19 +46,29 @@ UDPSocket::UDPSocket(int listeningPort) {
printf("Failed to set non-blocking socket\n"); printf("Failed to set non-blocking socket\n");
return; return;
} }
printf("Created UDP socket listening on port %d.\n", listeningPort);
} }
int UDPSocket::send(char * dest_address, int dest_port, const void *data, int length_in_bytes) { bool UDPSocket::receive(void *receivedData, int *receivedBytes) {
*receivedBytes = recvfrom(handle, receivedData, MAX_BUFFER_LENGTH_BYTES,
0, (sockaddr *)&senderAddress, &addLength);
return (*receivedBytes > 0);
}
int UDPSocket::send(char * destAddress, int destPort, const void *data, int byteLength) {
// change address and port on reusable global to passed variables // change address and port on reusable global to passed variables
dest_sockaddr.sin_addr.s_addr = inet_addr(dest_address); destSockaddr.sin_addr.s_addr = inet_addr(destAddress);
dest_sockaddr.sin_port = htons((uint16_t)dest_port); destSockaddr.sin_port = htons((uint16_t)destPort);
// send data via UDP // send data via UDP
int sent_bytes = sendto(handle, (const char*)data, length_in_bytes, int sent_bytes = sendto(handle, (const char*)data, byteLength,
0, (sockaddr*)&dest_address, sizeof(sockaddr_in)); 0, (sockaddr *)&destSockaddr, sizeof(sockaddr_in));
if (sent_bytes != length_in_bytes) { if (sent_bytes != byteLength) {
printf("Failed to send packet: return value = %d\n", sent_bytes); printf("Failed to send packet: return value = %d\n", sent_bytes);
return false; return false;
} }

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@ -12,16 +12,18 @@
#include <iostream> #include <iostream>
#include <netinet/in.h> #include <netinet/in.h>
#define MAX_BUFFER_LENGTH_BYTES 1024
class UDPSocket { class UDPSocket {
public: public:
static struct sockaddr_in sockaddr_util(char *address, int port); UDPSocket(int listening_port);
int send(char *destAddress, int destPort, const void *data, int byteLength);
UDPSocket(int listening_port); bool receive(void *receivedData, int *receivedBytes);
int send(char * dest_address, int dest_port, const void *data, int length_in_bytes); private:
private:
UDPSocket(); // private default constructor
int handle; int handle;
UDPSocket(); // private default constructor
struct AgentData { struct AgentData {
char * address; char * address;
int listening_port; int listening_port;

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@ -8,9 +8,23 @@
#include <iostream> #include <iostream>
#include <fstream> #include <fstream>
#include <pthread.h>
#include "audio.h" #include "audio.h"
#include "util.h" #include "util.h"
#include "AudioSource.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 RING_BUFFER_LENGTH_SAMPLES = BUFFER_LENGTH_SAMPLES * 10;
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;
bool Audio::initialized; bool Audio::initialized;
PaError Audio::err; PaError Audio::err;
@ -46,6 +60,12 @@ int audioCallback (const void *inputBuffer,
{ {
AudioData *data = (AudioData *) userData; AudioData *data = (AudioData *) userData;
int16_t *inBuffer = (int16_t *) inputBuffer;
if (inBuffer != NULL) {
data->audioSocket->send((char *) "0.0.0.0", 55443, (void *)inBuffer, BUFFER_LENGTH_BYTES);
}
int16_t *outputLeft = ((int16_t **) outputBuffer)[0]; int16_t *outputLeft = ((int16_t **) outputBuffer)[0];
int16_t *outputRight = ((int16_t **) outputBuffer)[1]; int16_t *outputRight = ((int16_t **) outputBuffer)[1];
@ -56,49 +76,64 @@ int audioCallback (const void *inputBuffer,
AudioSource *source = data->sources[s]; AudioSource *source = data->sources[s];
glm::vec3 headPos = data->linkedHead->getPos(); if (ECHO_SERVER_TEST) {
glm::vec3 sourcePos = source->position;
int startPointer = source->samplePointer;
int wrapAroundSamples = (BUFFER_LENGTH_BYTES / sizeof(int16_t)) - (source->lengthInSamples - source->samplePointer);
if (wrapAroundSamples <= 0) {
memcpy(data->samplesToQueue, source->sourceData + source->samplePointer, BUFFER_LENGTH_BYTES);
source->samplePointer += (BUFFER_LENGTH_BYTES / sizeof(int16_t));
} 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_BYTES / sizeof(int16_t); i++) {
data->samplesToQueue[i] *= distanceAmpRatio / NUM_AUDIO_SOURCES;
leadingOutput[i] += data->samplesToQueue[i];
if (i >= numSamplesDelay) { // copy whatever is source->sourceData to the left and right output channels
trailingOutput[i] += data->samplesToQueue[i - numSamplesDelay]; memcpy(outputLeft, source->sourceData + source->samplePointer, BUFFER_LENGTH_BYTES);
memcpy(outputRight, source->sourceData + source->samplePointer, BUFFER_LENGTH_BYTES);
if (source->samplePointer < RING_BUFFER_LENGTH_SAMPLES - BUFFER_LENGTH_SAMPLES) {
source->samplePointer += BUFFER_LENGTH_SAMPLES;
} else { } else {
int sampleIndex = startPointer - numSamplesDelay + i; source->samplePointer = 0;
}
} else {
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 (sampleIndex < 0) { if (i >= numSamplesDelay) {
sampleIndex += source->lengthInSamples; 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);
} }
trailingOutput[i] += source->sourceData[sampleIndex] * (distanceAmpRatio * phaseAmpRatio / NUM_AUDIO_SOURCES);
} }
} }
} }
@ -106,6 +141,32 @@ int audioCallback (const void *inputBuffer,
return paContinue; 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;
int streamSamplePointer = 0;
while (true) {
if (sharedAudioData->audioSocket->receive((void *)receivedData, receivedBytes)) {
// add the received data to the shared memory
memcpy(sharedAudioData->sources[0]->sourceData + streamSamplePointer, receivedData, *receivedBytes);
if (streamSamplePointer < RING_BUFFER_LENGTH_SAMPLES - BUFFER_LENGTH_SAMPLES) {
streamSamplePointer += BUFFER_LENGTH_SAMPLES;
} else {
streamSamplePointer = 0;
}
}
}
}
/** /**
* Initialize portaudio and start an audio stream. * Initialize portaudio and start an audio stream.
* Should be called at the beginning of program exection. * Should be called at the beginning of program exection.
@ -115,26 +176,45 @@ Use Audio::getError() to retrieve the error code.
*/ */
bool Audio::init() bool Audio::init()
{ {
Head deadHead = Head(); Head *deadHead = new Head();
return Audio::init(&deadHead); return Audio::init(deadHead);
} }
bool Audio::init(Head* mainHead) bool Audio::init(Head *mainHead)
{ {
data = new AudioData(NUM_AUDIO_SOURCES, BUFFER_LENGTH_BYTES);
data->linkedHead = mainHead;
err = Pa_Initialize(); err = Pa_Initialize();
if (err != paNoError) goto error; if (err != paNoError) goto error;
data->sources[0]->position = glm::vec3(6, 0, -1); if (ECHO_SERVER_TEST) {
data->sources[0]->loadDataFromFile("jeska.raw"); 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[RING_BUFFER_LENGTH_SAMPLES];
memset(data->sources[0]->sourceData, 0, RING_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->sources[1]->position = glm::vec3(6, 0, 6); data->linkedHead = mainHead;
data->sources[1]->loadDataFromFile("grayson.raw");
err = Pa_OpenDefaultStream(&stream, err = Pa_OpenDefaultStream(&stream,
NULL, // input channels 1, // input channels
2, // output channels 2, // output channels
(paInt16 | paNonInterleaved), // sample format (paInt16 | paNonInterleaved), // sample format
22050, // sample rate (hz) 22050, // sample rate (hz)
@ -158,9 +238,9 @@ error:
return false; return false;
} }
void Audio::sourceSetup() void Audio::render()
{ {
if (initialized) { if (initialized && !ECHO_SERVER_TEST) {
for (int s = 0; s < NUM_AUDIO_SOURCES; s++) { for (int s = 0; s < NUM_AUDIO_SOURCES; s++) {
// render gl objects on screen for our sources // render gl objects on screen for our sources

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@ -14,18 +14,13 @@
#include "head.h" #include "head.h"
#include "AudioData.h" #include "AudioData.h"
#define BUFFER_LENGTH_BYTES 1024
#define PHASE_DELAY_AT_90 20
#define AMPLITUDE_RATIO_AT_90 0.5
#define NUM_AUDIO_SOURCES 2
class Audio { class Audio {
public: public:
// initializes audio I/O // initializes audio I/O
static bool init(); static bool init();
static bool init(Head* mainHead); static bool init(Head* mainHead);
static void sourceSetup(); static void render();
// terminates audio I/O // terminates audio I/O
static bool terminate(); static bool terminate();

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@ -610,7 +610,7 @@ void display(void)
// render audio sources and start them // render audio sources and start them
if (audio_on) { if (audio_on) {
Audio::sourceSetup(); Audio::render();
} }
//glm::vec3 test(0.5, 0.5, 0.5); //glm::vec3 test(0.5, 0.5, 0.5);