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tweak resample to handle 48 to 24 stereo, add stereo/mono byte

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This commit is contained in:
Stephen Birarda 2014-06-06 10:27:59 -07:00
parent 7da091d2b4
commit 681ce247d6
1 changed files with 20 additions and 12 deletions
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32
interface/src/Audio.cpp
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@ -290,20 +290,27 @@ void linearResampling(int16_t* sourceSamples, int16_t* destinationSamples,
if (sourceToDestinationFactor >= 2) { if (sourceToDestinationFactor >= 2) {
// we need to downsample from 48 to 24 // we need to downsample from 48 to 24
// for now this only supports a mono output - this would be the case for audio input // for now this only supports a mono output - this would be the case for audio input
if (destinationAudioFormat.channelCount() == 1) {
for (unsigned int i = sourceAudioFormat.channelCount(); i < numSourceSamples; i += 2 * sourceAudioFormat.channelCount()) { for (unsigned int i = sourceAudioFormat.channelCount(); i < numSourceSamples; i += 2 * sourceAudioFormat.channelCount()) {
if (i + (sourceAudioFormat.channelCount()) >= numSourceSamples) { if (i + (sourceAudioFormat.channelCount()) >= numSourceSamples) {
destinationSamples[(i - sourceAudioFormat.channelCount()) / (int) sourceToDestinationFactor] = destinationSamples[(i - sourceAudioFormat.channelCount()) / (int) sourceToDestinationFactor] =
(sourceSamples[i - sourceAudioFormat.channelCount()] / 2) (sourceSamples[i - sourceAudioFormat.channelCount()] / 2)
+ (sourceSamples[i] / 2); + (sourceSamples[i] / 2);
} else { } else {
destinationSamples[(i - sourceAudioFormat.channelCount()) / (int) sourceToDestinationFactor] = destinationSamples[(i - sourceAudioFormat.channelCount()) / (int) sourceToDestinationFactor] =
(sourceSamples[i - sourceAudioFormat.channelCount()] / 4) (sourceSamples[i - sourceAudioFormat.channelCount()] / 4)
+ (sourceSamples[i] / 2) + (sourceSamples[i] / 2)
+ (sourceSamples[i + sourceAudioFormat.channelCount()] / 4); + (sourceSamples[i + sourceAudioFormat.channelCount()] / 4);
}
}
} else {
// this is a 48 to 24 resampling but both source and destination are two channels
// squish two samples into one in each channel
for (int i = 0; i < numSourceSamples; i += 2) {
destinationSamples[i / 2] = (sourceSamples[i] / 2) + (sourceSamples[i + 2] / 2);
destinationSamples[(i / 2) + 1] = (sourceSamples[i + 1] / 2) + (sourceSamples[i + 3] / 2);
} }
} }
} else { } else {
if (sourceAudioFormat.sampleRate() == destinationAudioFormat.sampleRate()) { if (sourceAudioFormat.sampleRate() == destinationAudioFormat.sampleRate()) {
// mono to stereo, same sample rate // mono to stereo, same sample rate
@ -409,7 +416,7 @@ void Audio::handleAudioInput() {
static char audioDataPacket[MAX_PACKET_SIZE]; static char audioDataPacket[MAX_PACKET_SIZE];
static int numBytesPacketHeader = numBytesForPacketHeaderGivenPacketType(PacketTypeMicrophoneAudioNoEcho); static int numBytesPacketHeader = numBytesForPacketHeaderGivenPacketType(PacketTypeMicrophoneAudioNoEcho);
static int leadingBytes = numBytesPacketHeader + sizeof(glm::vec3) + sizeof(glm::quat); static int leadingBytes = numBytesPacketHeader + sizeof(glm::vec3) + sizeof(glm::quat) + sizeof(quint8);
static int16_t* networkAudioSamples = (int16_t*) (audioDataPacket + leadingBytes); static int16_t* networkAudioSamples = (int16_t*) (audioDataPacket + leadingBytes);
@ -610,9 +617,7 @@ void Audio::handleAudioInput() {
MyAvatar* interfaceAvatar = Application::getInstance()->getAvatar(); MyAvatar* interfaceAvatar = Application::getInstance()->getAvatar();
glm::vec3 headPosition = interfaceAvatar->getHead()->getPosition(); glm::vec3 headPosition = interfaceAvatar->getHead()->getPosition();
glm::quat headOrientation = interfaceAvatar->getHead()->getFinalOrientation(); glm::quat headOrientation = interfaceAvatar->getHead()->getFinalOrientation();
quint8 isStereo = _isStereoInput ? 1 : 0;
// we need the amount of bytes in the buffer + 1 for type
// + 12 for 3 floats for position + float for bearing + 1 attenuation byte
int numAudioBytes = 0; int numAudioBytes = 0;
@ -621,7 +626,7 @@ void Audio::handleAudioInput() {
packetType = PacketTypeSilentAudioFrame; packetType = PacketTypeSilentAudioFrame;
// we need to indicate how many silent samples this is to the audio mixer // we need to indicate how many silent samples this is to the audio mixer
networkAudioSamples[0] = NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL; audioDataPacket[0] = NETWORK_BUFFER_LENGTH_SAMPLES_PER_CHANNEL;
numAudioBytes = sizeof(int16_t); numAudioBytes = sizeof(int16_t);
} else { } else {
@ -644,6 +649,9 @@ void Audio::handleAudioInput() {
memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation)); memcpy(currentPacketPtr, &headOrientation, sizeof(headOrientation));
currentPacketPtr += sizeof(headOrientation); currentPacketPtr += sizeof(headOrientation);
// set the mono/stereo byte
*currentPacketPtr++ = isStereo;
nodeList->writeDatagram(audioDataPacket, numAudioBytes + leadingBytes, audioMixer); nodeList->writeDatagram(audioDataPacket, numAudioBytes + leadingBytes, audioMixer);
Application::getInstance()->getBandwidthMeter()->outputStream(BandwidthMeter::AUDIO) Application::getInstance()->getBandwidthMeter()->outputStream(BandwidthMeter::AUDIO)