Audiomixer now working (added call to updateNextOutputTrailingLoudness())

This commit is contained in:
wangyix 2014-07-24 14:43:29 -07:00
parent 3d22a11e28
commit 83ba4b9a1c
5 changed files with 39 additions and 21 deletions

View file

@ -98,19 +98,15 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
// if the frame to be mixed is silent, don't mix it
if (bufferToAdd->getNextOutputTrailingLoudness() == 0.0f) {
bufferToAdd->popFrames(1);
printf("trailing loudness too soft: not mixing!\n");
return;
}
// get pointer to frame to be mixed. If the stream cannot provide a frame (is starved), bail
AudioRingBuffer::ConstIterator nextOutputStart;
if (!bufferToAdd->popFrames(&nextOutputStart, 1)) {
printf("stream is starved! not mixing!\n");
return;
}
printf("mixing stream\n");
float bearingRelativeAngleToSource = 0.0f;
float attenuationCoefficient = 1.0f;
int numSamplesDelay = 0;
@ -221,7 +217,7 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
if (!bufferToAdd->isStereo() && shouldAttenuate) {
if (!bufferToAdd->isStereo() && shouldAttenuate && false) {
// this is a mono buffer, which means it gets full attenuation and spatialization
// if the bearing relative angle to source is > 0 then the delayed channel is the right one
@ -269,7 +265,20 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
}
}
} else {
// this is a stereo buffer or an unattenuated buffer, don't perform spatialization
int stereoDivider = bufferToAdd->isStereo() ? 1 : 2;
if (!shouldAttenuate) {
attenuationCoefficient = 1.0f;
}
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s++) {
_clientSamples[s] = glm::clamp(_clientSamples[s] + (int)(nextOutputStart[s / stereoDivider] * attenuationCoefficient),
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
}
/*// this is a stereo buffer or an unattenuated buffer, don't perform spatialization
for (int s = 0; s < NETWORK_BUFFER_LENGTH_SAMPLES_STEREO; s += 4) {
int stereoDivider = bufferToAdd->isStereo() ? 1 : 2;
@ -293,7 +302,7 @@ void AudioMixer::addBufferToMixForListeningNodeWithBuffer(PositionalAudioRingBuf
+ (int) (nextOutputStart[(s / stereoDivider) + (3 / stereoDivider)]
* attenuationCoefficient),
MIN_SAMPLE_VALUE, MAX_SAMPLE_VALUE);
}
}*/
}
}
@ -318,6 +327,8 @@ void AudioMixer::prepareMixForListeningNode(Node* node) {
if (*otherNode != *node || otherNodeBuffer->shouldLoopbackForNode()) {
addBufferToMixForListeningNodeWithBuffer(otherNodeBuffer, nodeRingBuffer);
} else {
otherNodeBuffer->popFrames(1);
}
}
}

View file

@ -86,7 +86,6 @@ int InboundAudioStream::parseData(const QByteArray& packet) {
}
if (_isStarved && _ringBuffer.samplesAvailable() >= _desiredJitterBufferFrames * _ringBuffer.getNumFrameSamples()) {
printf("\nstream refilled from starve!\n");
_isStarved = false;
}
@ -130,26 +129,25 @@ bool InboundAudioStream::popFrames(AudioRingBuffer::ConstIterator* nextOutput, i
}
bool InboundAudioStream::shouldPop(int numSamples, bool starveOnFail) {
printf("\nshouldPop()\n");
if (_isStarved) {
printf("\t we're starved, not popping\n");
// we're still refilling; don't mix
_consecutiveNotMixedCount++;
return false;
}
if (_ringBuffer.samplesAvailable() >= numSamples) {
printf("have requested samples and not starved, popping\n");
// we have enough samples to pop, so we're good to mix
_hasStarted = true;
return true;
} else {
if (starveOnFail) {
printf("don't have enough samples; starved!\n");
setToStarved();
_consecutiveNotMixedCount++;
}
return false;
}
// we don't have enough samples, so set this stream to starve
// if starveOnFail is true
if (starveOnFail) {
setToStarved();
_consecutiveNotMixedCount++;
}
return false;
}
void InboundAudioStream::setToStarved() {

View file

@ -51,7 +51,7 @@ public:
void resetSequenceNumberStats() { _incomingSequenceNumberStats.reset(); }
int parseData(const QByteArray& packet);
virtual int parseData(const QByteArray& packet);
bool popFrames(int numFrames, bool starveOnFail = true);
bool popFrames(int16_t* dest, int numFrames, bool starveOnFail = true);
@ -62,6 +62,7 @@ public:
/// this function should be called once per second to ensure the seq num stats history spans ~30 seconds
AudioStreamStats updateSeqHistoryAndGetAudioStreamStats();
virtual AudioStreamStats getAudioStreamStats() const;
int getCalculatedDesiredJitterBufferFrames() const;

View file

@ -34,6 +34,12 @@ PositionalAudioRingBuffer::PositionalAudioRingBuffer(PositionalAudioRingBuffer::
{
}
int PositionalAudioRingBuffer::parseData(const QByteArray& packet) {
int bytesRead = InboundAudioStream::parseData(packet);
updateNextOutputTrailingLoudness();
return bytesRead;
}
void PositionalAudioRingBuffer::updateNextOutputTrailingLoudness() {
float nextLoudness = _ringBuffer.getNextOutputFrameLoudness();

View file

@ -28,7 +28,9 @@ public:
};
PositionalAudioRingBuffer(PositionalAudioRingBuffer::Type type, bool isStereo = false, bool dynamicJitterBuffers = false);
int parseData(const QByteArray& packet);
virtual AudioStreamStats getAudioStreamStats() const;
void updateNextOutputTrailingLoudness();