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This commit is contained in:
Brad Hefta-Gaub 2017-03-08 20:29:32 -08:00
parent 60daaba525
commit 2d595c1f87
2 changed files with 18 additions and 19 deletions
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7
libraries/audio-client/src/AudioNoiseGate.cpp
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@ -77,8 +77,6 @@ void AudioNoiseGate::gateSamples(int16_t* samples, int numSamples) {
// More means better rejection but also can reject continuous things like singing.
// NUMBER_OF_NOISE_SAMPLE_FRAMES: How often should we re-evaluate the noise floor?
_closedInLastFrame = false;
float loudness = 0;
int thisSample = 0;
int samplesOverNoiseGate = 0;
@ -142,14 +140,13 @@ void AudioNoiseGate::gateSamples(int16_t* samples, int numSamples) {
_sampleCounter = 0;
}
if (samplesOverNoiseGate > NOISE_GATE_WIDTH) {
_isOpen = true;
_framesToClose = NOISE_GATE_CLOSE_FRAME_DELAY;
} else {
if (--_framesToClose == 0) {
if (_isOpen) {
_closedInLastFrame = true;
}
_closedInLastFrame = !_isOpen;
_isOpen = false;
}
}

30
libraries/audio/src/InboundAudioStream.cpp
View file

@ -136,9 +136,9 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
break;
}
case SequenceNumberStats::Early: {
// Packet is early treat the packets as if all the packets between the last
// Packet is early. Treat the packets as if all the packets between the last
// OnTime packet and this packet was lost. If we're using a codec this will
// also result in allowing the codec to flush its internal state. Then
// also result in allowing the codec to interpolate lost data. Then
// fall through to the "on time" logic to actually handle this packet
int packetsDropped = arrivalInfo._seqDiffFromExpected;
lostAudioData(packetsDropped);
@ -150,8 +150,6 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
if (message.getType() == PacketType::SilentAudioFrame) {
// If we recieved a SilentAudioFrame from our sender, we might want to drop
// some of the samples in order to catch up to our desired jitter buffer size.
// NOTE: If we're using a codec we will be calling the codec's lostFrame()
// method to allow the codec to flush its internal state.
writeDroppableSilentFrames(networkFrames);
} else {
// note: PCM and no codec are identical
@ -249,17 +247,21 @@ int InboundAudioStream::parseAudioData(PacketType type, const QByteArray& packet
int InboundAudioStream::writeDroppableSilentFrames(int silentFrames) {
// if we have a decoder, we still want to tell the decoder about our
// lost frame. this will flush the internal state of the decoder
// we can safely ignore the output of the codec in this case, because
// we've enforced that on the sending side, the encoder ran at least
// one frame of truly silent audio before we sent the "droppable" silent
// frame. Technically we could leave this out, if we know for certain
// that the sender has really sent us an encoded packet of zeros, but
// since we can trust all encoders to always encode at least one silent
// frame (open source, someone code modify it), we will go ahead and
// tell our decoder about the lost frame.
// We can't guarentee that all clients have faded the stream down
// to silence and encode that silence before sending us a
// SilentAudioFrame. The encoder may have truncated the stream and
// left the decoder holding some loud state. To handle this case
// we will call the decoder's lostFrame() method, which indicates
// that it should interpolate from it's last known state (which may
// be some unknown loud state) down toward silence.
if (_decoder) {
// FIXME - We could potentially use the output from the codec, in which
// case we might get a cleaner fade toward silence. NOTE: The below logic
// attempts to catch up in the event that the jitter buffers have grown.
// The better long term fix is to use the output from the decode, detect
// when it actually reaches silence, and then delete the silent portions
// of the jitter buffers. Or petentially do a cross fade from the decode
// output to silence.
QByteArray decodedBuffer;
_decoder->lostFrame(decodedBuffer);
}