From adfb4287962f908ea35d9fed29729a95fc03f75a Mon Sep 17 00:00:00 2001
From: David Rowe <david@ctrlaltstudio.com>
Date: Thu, 20 May 2021 09:43:18 +1200
Subject: [PATCH] Distinguish use of WebRTC features - audio versus data
 channel

---
 libraries/audio-client/src/AudioClient.cpp | 11 ++++++-----
 libraries/audio-client/src/AudioClient.h   |  8 +++++++-
 libraries/shared/src/shared/WebRTC.h       | 22 +++++++++++++---------
 3 files changed, 26 insertions(+), 15 deletions(-)

diff --git a/libraries/audio-client/src/AudioClient.cpp b/libraries/audio-client/src/AudioClient.cpp
index 38635870fd..c03f37576d 100644
--- a/libraries/audio-client/src/AudioClient.cpp
+++ b/libraries/audio-client/src/AudioClient.cpp
@@ -4,6 +4,7 @@
 //
 //  Created by Stephen Birarda on 1/22/13.
 //  Copyright 2013 High Fidelity, Inc.
+//  Copyright 2021 Vircadia contributors.
 //
 //  Distributed under the Apache License, Version 2.0.
 //  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
@@ -363,7 +364,7 @@ AudioClient::AudioClient() {
 
     configureReverb();
 
-#if defined(WEBRTC_ENABLED)
+#if defined(WEBRTC_AUDIO)
     configureWebrtc();
 #endif
 
@@ -1142,7 +1143,7 @@ void AudioClient::setReverbOptions(const AudioEffectOptions* options) {
     }
 }
 
-#if defined(WEBRTC_ENABLED)
+#if defined(WEBRTC_AUDIO)
 
 static void deinterleaveToFloat(const int16_t* src, float* const* dst, int numFrames, int numChannels) {
     for (int i = 0; i < numFrames; i++) {
@@ -1261,7 +1262,7 @@ void AudioClient::processWebrtcNearEnd(int16_t* samples, int numFrames, int numC
     }
 }
 
-#endif // WEBRTC_ENABLED
+#endif // WEBRTC_AUDIO
 
 void AudioClient::handleLocalEchoAndReverb(QByteArray& inputByteArray) {
     // If there is server echo, reverb will be applied to the recieved audio stream so no need to have it here.
@@ -1462,7 +1463,7 @@ void AudioClient::handleMicAudioInput() {
         }
         isClipping = (_timeSinceLastClip >= 0.0f) && (_timeSinceLastClip < 2.0f);  // 2 second hold time
 
-#if defined(WEBRTC_ENABLED)
+#if defined(WEBRTC_AUDIO)
         if (_isAECEnabled) {
             processWebrtcNearEnd(inputAudioSamples.get(), inputSamplesRequired / _inputFormat.channelCount(),
                                  _inputFormat.channelCount(), _inputFormat.sampleRate());
@@ -2420,7 +2421,7 @@ qint64 AudioClient::AudioOutputIODevice::readData(char * data, qint64 maxSize) {
     // limit the audio
     _audio->_audioLimiter.render(mixBuffer, scratchBuffer, framesPopped);
 
-#if defined(WEBRTC_ENABLED)
+#if defined(WEBRTC_AUDIO)
     if (_audio->_isAECEnabled) {
         _audio->processWebrtcFarEnd(scratchBuffer, framesPopped, OUTPUT_CHANNEL_COUNT, _audio->_outputFormat.sampleRate());
     }
diff --git a/libraries/audio-client/src/AudioClient.h b/libraries/audio-client/src/AudioClient.h
index a5de9bd4ca..19ccb587a4 100644
--- a/libraries/audio-client/src/AudioClient.h
+++ b/libraries/audio-client/src/AudioClient.h
@@ -4,6 +4,7 @@
 //
 //  Created by Stephen Birarda on 1/22/13.
 //  Copyright 2013 High Fidelity, Inc.
+//  Copyright 2021 Vircadia contributors.
 //
 //  Distributed under the Apache License, Version 2.0.
 //  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
@@ -56,6 +57,11 @@
 #include "AudioFileWav.h"
 #include "HifiAudioDeviceInfo.h"
 
+#if defined(WEBRTC_AUDIO)
+#  include <modules/audio_processing/include/audio_processing.h>
+#  include "modules/audio_processing/audio_processing_impl.h"
+#endif
+
 #ifdef _WIN32
 #pragma warning( push )
 #pragma warning( disable : 4273 )
@@ -450,7 +456,7 @@ private:
     void updateReverbOptions();
     void handleLocalEchoAndReverb(QByteArray& inputByteArray);
 
-#if defined(WEBRTC_ENABLED)
+#if defined(WEBRTC_AUDIO)
     static const int WEBRTC_SAMPLE_RATE_MAX = 96000;
     static const int WEBRTC_CHANNELS_MAX = 2;
     static const int WEBRTC_FRAMES_MAX = webrtc::AudioProcessing::kChunkSizeMs * WEBRTC_SAMPLE_RATE_MAX / 1000;
diff --git a/libraries/shared/src/shared/WebRTC.h b/libraries/shared/src/shared/WebRTC.h
index 2f0e444bff..e99c643045 100644
--- a/libraries/shared/src/shared/WebRTC.h
+++ b/libraries/shared/src/shared/WebRTC.h
@@ -3,6 +3,7 @@
 //  libraries/shared/src/shared/
 //
 //  Copyright 2019 High Fidelity, Inc.
+//  Copyright 2021 Vircadia contributors.
 //
 //  Distributed under the Apache License, Version 2.0.
 //  See the accompanying file LICENSE or http://www.apache.org/licenses/LICENSE-2.0.html
@@ -11,26 +12,29 @@
 #ifndef hifi_WebRTC_h
 #define hifi_WebRTC_h
 
+#ifndef QSYSTEMDETECTION_H
+#include <qsystemdetection.h>
+#endif
+
+// WEBRTC_AUDIO: WebRTC audio features, e.g., echo canceling.
+// WEBRTC_DATA_CHANNEL: WebRTC client-server connections in parallel with UDP.
+
 #if defined(Q_OS_MAC)
-#  define WEBRTC_ENABLED 1
+#  define WEBRTC_AUDIO 1
 #  define WEBRTC_POSIX 1
 #elif defined(Q_OS_WIN)
-#  define WEBRTC_ENABLED 1
+#  define WEBRTC_AUDIO 1
+#  define WEBRTC_DATA_CHANNEL 1
 #  define WEBRTC_WIN 1
 #  define NOMINMAX 1
 #  define WIN32_LEAN_AND_MEAN 1
 #elif defined(Q_OS_ANDROID)
 // I don't yet have a working libwebrtc for android
-// #  define WEBRTC_ENABLED 1
+// #  define WEBRTC_AUDIO 1
 // #  define WEBRTC_POSIX 1
 #elif defined(Q_OS_LINUX)
-#  define WEBRTC_ENABLED 1
+#  define WEBRTC_AUDIO 1
 #  define WEBRTC_POSIX 1
 #endif
 
-#if defined(WEBRTC_ENABLED)
-#  include <modules/audio_processing/include/audio_processing.h>
-#  include "modules/audio_processing/audio_processing_impl.h"
-#endif
-
 #endif // hifi_WebRTC_h