From 4a4a92c009662daced888d38a26cb141025a17d0 Mon Sep 17 00:00:00 2001
From: Roxanne Skelly <roxanne@highfidelity.io>
Date: Mon, 6 May 2019 15:46:06 -0700
Subject: [PATCH] BUGZ-85 - audio pipeline interpolation on ring buffer starve
 Kick the PLC so that it generates a more pleasing 'fade' frame when ring
 buffer starves.

---
 libraries/audio/src/AudioSRC.cpp           |  4 ++--
 libraries/audio/src/AudioSRC.h             |  3 +++
 libraries/audio/src/InboundAudioStream.cpp | 11 +++++++----
 3 files changed, 12 insertions(+), 6 deletions(-)

diff --git a/libraries/audio/src/AudioSRC.cpp b/libraries/audio/src/AudioSRC.cpp
index d488eccb6a..1accc8565d 100644
--- a/libraries/audio/src/AudioSRC.cpp
+++ b/libraries/audio/src/AudioSRC.cpp
@@ -1491,7 +1491,7 @@ AudioSRC::~AudioSRC() {
 //
 int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames) {
     int outputFrames = 0;
-
+    QMutexLocker lock(&_renderMutex);
     while (inputFrames) {
         int ni = MIN(inputFrames, _inputBlock);
 
@@ -1516,7 +1516,7 @@ int AudioSRC::render(const int16_t* input, int16_t* output, int inputFrames) {
 //
 int AudioSRC::render(const float* input, float* output, int inputFrames) {
     int outputFrames = 0;
-
+    QMutexLocker lock(&_renderMutex);
     while (inputFrames) {
         int ni = MIN(inputFrames, _inputBlock);
 
diff --git a/libraries/audio/src/AudioSRC.h b/libraries/audio/src/AudioSRC.h
index d5971e5838..5299d9da7f 100644
--- a/libraries/audio/src/AudioSRC.h
+++ b/libraries/audio/src/AudioSRC.h
@@ -13,6 +13,7 @@
 #define hifi_AudioSRC_h
 
 #include <stdint.h>
+#include <QMutex>
 
 static const int SRC_MAX_CHANNELS = 4;
 
@@ -55,6 +56,8 @@ public:
     int getMaxInput(int outputFrames);
 
 private:
+    QMutex _renderMutex;
+
     float* _polyphaseFilter;
     int* _stepTable;
 
diff --git a/libraries/audio/src/InboundAudioStream.cpp b/libraries/audio/src/InboundAudioStream.cpp
index 8c5388e222..9844159c1a 100644
--- a/libraries/audio/src/InboundAudioStream.cpp
+++ b/libraries/audio/src/InboundAudioStream.cpp
@@ -215,7 +215,6 @@ int InboundAudioStream::parseData(ReceivedMessage& message) {
     if (framesAvailable > _desiredJitterBufferFrames + MAX_FRAMES_OVER_DESIRED) {
         int framesToDrop = framesAvailable - (_desiredJitterBufferFrames + DESIRED_JITTER_BUFFER_FRAMES_PADDING);
         _ringBuffer.shiftReadPosition(framesToDrop * _ringBuffer.getNumFrameSamples());
-        
         _framesAvailableStat.reset();
         _currentJitterBufferFrames = 0;
 
@@ -250,7 +249,7 @@ int InboundAudioStream::lostAudioData(int numPackets) {
         if (_decoder) {
             _decoder->lostFrame(decodedBuffer);
         } else {
-            decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_STEREO);
+            decodedBuffer.resize(AudioConstants::NETWORK_FRAME_BYTES_PER_CHANNEL * _numChannels);
             memset(decodedBuffer.data(), 0, decodedBuffer.size());
         }
         _ringBuffer.writeData(decodedBuffer.data(), decodedBuffer.size());
@@ -338,10 +337,14 @@ int InboundAudioStream::popSamples(int maxSamples, bool allOrNothing) {
             popSamplesNoCheck(samplesAvailable);
             samplesPopped = samplesAvailable;
         } else {
-            // we can't pop any samples, set this stream to starved
+            // we can't pop any samples, set this stream to starved for jitter 
+            // buffer calculations.
             setToStarved();
             _consecutiveNotMixedCount++;
-            _lastPopSucceeded = false;
+            //Kick PLC to generate a filler frame, reducing 'click'
+            lostAudioData(allOrNothing ? (maxSamples - samplesAvailable) / _ringBuffer.getNumFrameSamples() : 1);
+            samplesPopped = _ringBuffer.samplesAvailable();
+            popSamplesNoCheck(samplesPopped);
         }
     }
     return samplesPopped;