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Distinguish use of WebRTC features - audio versus data channel

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This commit is contained in:
David Rowe 2021-05-20 09:43:18 +12:00
parent 97352b8858
commit adfb428796
3 changed files with 26 additions and 15 deletions
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11
libraries/audio-client/src/AudioClient.cpp
View file

@ -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());
}

8
libraries/audio-client/src/AudioClient.h
View file

@ -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;

22
libraries/shared/src/shared/WebRTC.h
View file

@ -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