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some spacing cleanup in Audio class

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This commit is contained in:
Stephen Birarda 2013-04-26 10:58:01 -07:00
parent f18942d016
commit 1884bfc17e
1 changed files with 12 additions and 66 deletions
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78
interface/src/Audio.cpp
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@ -114,15 +114,12 @@ int audioCallback (const void *inputBuffer,
AudioData *data = (AudioData *) userData;
int16_t *inputLeft = ((int16_t **) inputBuffer)[0];
// int16_t *inputRight = ((int16_t **) inputBuffer)[1];
//printLog("Audio callback at %6.0f\n", usecTimestampNow()/1000);
// printLog("Audio callback at %6.0f\n", usecTimestampNow()/1000);
if (inputLeft != NULL) {
//
// Measure the loudness of the signal from the microphone and store in audio object
//
float loudness = 0;
for (int i = 0; i < BUFFER_LENGTH_SAMPLES; i++) {
loudness += abs(inputLeft[i]);
@ -130,10 +127,8 @@ int audioCallback (const void *inputBuffer,
loudness /= BUFFER_LENGTH_SAMPLES;
data->lastInputLoudness = loudness;
//
// If scope is turned on, copy input buffer to scope
//
if (scope->getState()) {
for (int i = 0; i < BUFFER_LENGTH_SAMPLES; i++) {
scope->addData((float)inputLeft[i]/32767.0, 1, i);
@ -174,42 +169,13 @@ int audioCallback (const void *inputBuffer,
}
if (data->mixerLoopbackFlag) {
correctedYaw = correctedYaw > 0 ? correctedYaw + AGENT_LOOPBACK_MODIFIER : correctedYaw - AGENT_LOOPBACK_MODIFIER;
correctedYaw = correctedYaw > 0
? correctedYaw + AGENT_LOOPBACK_MODIFIER
: correctedYaw - AGENT_LOOPBACK_MODIFIER;
}
memcpy(currentPacketPtr, &correctedYaw, sizeof(float));
currentPacketPtr += sizeof(float);
// if (samplesLeftForWalk == 0) {
// sampleWalkPointer = walkingSoundArray;
// }
//
// if (data->playWalkSound) {
// // if this boolean is true and we aren't currently playing the walk sound
// // set the number of samples left for walk
// samplesLeftForWalk = walkingSoundSamples;
// data->playWalkSound = false;
// }
//
// if (samplesLeftForWalk > 0) {
// // we need to play part of the walking sound
// // so add it in
// int affectedSamples = std::min(samplesLeftForWalk, BUFFER_LENGTH_SAMPLES);
// for (int i = 0; i < affectedSamples; i++) {
// inputLeft[i] += *sampleWalkPointer;
// inputLeft[i] = std::max(inputLeft[i], std::numeric_limits<int16_t>::min());
// inputLeft[i] = std::min(inputLeft[i], std::numeric_limits<int16_t>::max());
//
// sampleWalkPointer++;
// samplesLeftForWalk--;
//
// if (sampleWalkPointer - walkingSoundArray > walkingSoundSamples) {
// sampleWalkPointer = walkingSoundArray;
// };
// }
// }
//
currentPacketPtr += sizeof(float);
// copy the audio data to the last BUFFER_LENGTH_BYTES bytes of the data packet
memcpy(currentPacketPtr, inputLeft, BUFFER_LENGTH_BYTES);
@ -239,26 +205,21 @@ int audioCallback (const void *inputBuffer,
starve_counter++;
packetsReceivedThisPlayback = 0;
//printLog("Starved #%d\n", starve_counter);
// printLog("Starved #%d\n", starve_counter);
data->wasStarved = 10; // Frames to render the indication that the system was starved.
} else {
if (!ringBuffer->isStarted()) {
ringBuffer->setStarted(true);
//printLog("starting playback %3.1f msecs delayed \n", (usecTimestampNow() - usecTimestamp(&firstPlaybackTimer))/1000.0);
// printLog("starting playback %3.1f msecs delayed \n", (usecTimestampNow() - usecTimestamp(&firstPlaybackTimer))/1000.0);
} else {
//printLog("pushing buffer\n");
// printLog("pushing buffer\n");
}
// play whatever we have in the audio buffer
//
// if we haven't fired off the flange effect, check if we should
//
//
// NOTE: PER - LastMeasuredHeadYaw is now relative to body position, represents the local
// rotation of the head relative to body, this may effect flange effect!
//
//
// if we haven't fired off the flange effect, check if we should
// TODO: lastMeasuredHeadYaw is now relative to body - check if this still works.
int lastYawMeasured = fabsf(data->linkedAvatar->getLastMeasuredHeadYaw());
if (!samplesLeftForFlange && lastYawMeasured > MIN_FLANGE_EFFECT_THRESHOLD) {
@ -277,21 +238,6 @@ int audioCallback (const void *inputBuffer,
}
}
// check if we have more than we need to play out
// int thresholdFrames = ceilf((PACKET_LENGTH_SAMPLES + JITTER_BUFFER_SAMPLES) / (float)PACKET_LENGTH_SAMPLES);
// int thresholdSamples = thresholdFrames * PACKET_LENGTH_SAMPLES;
//
// if (ringBuffer->diffLastWriteNextOutput() > thresholdSamples) {
// // we need to push the next output forwards
// int samplesToPush = ringBuffer->diffLastWriteNextOutput() - thresholdSamples;
//
// if (ringBuffer->getNextOutput() + samplesToPush > ringBuffer->getBuffer()) {
// ringBuffer->setNextOutput(ringBuffer->getBuffer() + (samplesToPush - (ringBuffer->getBuffer() + RING_BUFFER_SAMPLES - ringBuffer->getNextOutput())));
// } else {
// ringBuffer->setNextOutput(ringBuffer->getNextOutput() + samplesToPush);
// }
// }
for (int s = 0; s < PACKET_LENGTH_SAMPLES_PER_CHANNEL; s++) {
int leftSample = ringBuffer->getNextOutput()[s];