OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration. Review-Url: https://codereview.webrtc.org/2111293003 Cr-Commit-Position: refs/heads/master@{#13550}
434 lines
14 KiB
C++
434 lines
14 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_mixer/audio_mixer.h"
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#include "webrtc/base/format_macros.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/modules/utility/include/audio_frame_operations.h"
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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#include "webrtc/system_wrappers/include/trace.h"
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#include "webrtc/voice_engine/include/voe_external_media.h"
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#include "webrtc/voice_engine/statistics.h"
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#include "webrtc/voice_engine/utility.h"
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namespace webrtc {
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namespace voe {
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void AudioMixer::PlayNotification(int32_t id, uint32_t durationMs) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::PlayNotification(id=%d, durationMs=%d)", id,
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durationMs);
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// Not implement yet
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}
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void AudioMixer::RecordNotification(int32_t id, uint32_t durationMs) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::RecordNotification(id=%d, durationMs=%d)", id,
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durationMs);
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// Not implement yet
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}
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void AudioMixer::PlayFileEnded(int32_t id) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::PlayFileEnded(id=%d)", id);
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// not needed
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}
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void AudioMixer::RecordFileEnded(int32_t id) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::RecordFileEnded(id=%d)", id);
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RTC_DCHECK_EQ(id, _instanceId);
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rtc::CritScope cs(&_fileCritSect);
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_outputFileRecording = false;
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WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::RecordFileEnded() =>"
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"output file recorder module is shutdown");
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}
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int32_t AudioMixer::Create(AudioMixer*& mixer, uint32_t instanceId) {
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WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
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"AudioMixer::Create(instanceId=%d)", instanceId);
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mixer = new AudioMixer(instanceId);
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if (mixer == NULL) {
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WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
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"AudioMixer::Create() unable to allocate memory for"
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"mixer");
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return -1;
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}
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return 0;
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}
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AudioMixer::AudioMixer(uint32_t instanceId)
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: _mixerModule(*NewAudioConferenceMixer::Create(instanceId)),
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_audioLevel(),
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_instanceId(instanceId),
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_externalMediaCallbackPtr(NULL),
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_externalMedia(false),
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_panLeft(1.0f),
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_panRight(1.0f),
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_mixingFrequencyHz(8000),
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_outputFileRecorderPtr(NULL),
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_outputFileRecording(false) {
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WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::AudioMixer() - ctor");
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}
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void AudioMixer::Destroy(AudioMixer*& mixer) {
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if (mixer) {
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delete mixer;
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mixer = NULL;
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}
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}
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AudioMixer::~AudioMixer() {
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WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::~AudioMixer() - dtor");
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if (_externalMedia) {
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DeRegisterExternalMediaProcessing();
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}
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{
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rtc::CritScope cs(&_fileCritSect);
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if (_outputFileRecorderPtr) {
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_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
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_outputFileRecorderPtr->StopRecording();
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FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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_outputFileRecorderPtr = NULL;
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}
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}
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delete &_mixerModule;
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}
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int32_t AudioMixer::SetEngineInformation(voe::Statistics& engineStatistics) {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::SetEngineInformation()");
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_engineStatisticsPtr = &engineStatistics;
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return 0;
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}
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int32_t AudioMixer::SetAudioProcessingModule(
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AudioProcessing* audioProcessingModule) {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::SetAudioProcessingModule("
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"audioProcessingModule=0x%x)",
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audioProcessingModule);
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_audioProcessingModulePtr = audioProcessingModule;
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return 0;
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}
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int AudioMixer::RegisterExternalMediaProcessing(
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VoEMediaProcess& proccess_object) {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::RegisterExternalMediaProcessing()");
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rtc::CritScope cs(&_callbackCritSect);
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_externalMediaCallbackPtr = &proccess_object;
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_externalMedia = true;
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return 0;
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}
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int AudioMixer::DeRegisterExternalMediaProcessing() {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::DeRegisterExternalMediaProcessing()");
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rtc::CritScope cs(&_callbackCritSect);
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_externalMedia = false;
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_externalMediaCallbackPtr = NULL;
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return 0;
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}
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int32_t AudioMixer::SetMixabilityStatus(MixerAudioSource& audio_source,
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bool mixable) {
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return _mixerModule.SetMixabilityStatus(&audio_source, mixable);
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}
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int32_t AudioMixer::SetAnonymousMixabilityStatus(MixerAudioSource& audio_source,
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bool mixable) {
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return _mixerModule.SetAnonymousMixabilityStatus(&audio_source, mixable);
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}
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int32_t AudioMixer::MixActiveChannels() {
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_mixerModule.Mix(&_audioFrame);
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return 0;
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}
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int AudioMixer::GetSpeechOutputLevel(uint32_t& level) {
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int8_t currentLevel = _audioLevel.Level();
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level = static_cast<uint32_t>(currentLevel);
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WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
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"GetSpeechOutputLevel() => level=%u", level);
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return 0;
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}
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int AudioMixer::GetSpeechOutputLevelFullRange(uint32_t& level) {
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int16_t currentLevel = _audioLevel.LevelFullRange();
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level = static_cast<uint32_t>(currentLevel);
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WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
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"GetSpeechOutputLevelFullRange() => level=%u", level);
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return 0;
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}
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int AudioMixer::SetOutputVolumePan(float left, float right) {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::SetOutputVolumePan()");
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_panLeft = left;
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_panRight = right;
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return 0;
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}
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int AudioMixer::GetOutputVolumePan(float& left, float& right) {
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left = _panLeft;
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right = _panRight;
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WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
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"GetOutputVolumePan() => left=%2.1f, right=%2.1f", left, right);
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return 0;
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}
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int AudioMixer::StartRecordingPlayout(const char* fileName,
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const CodecInst* codecInst) {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::StartRecordingPlayout(fileName=%s)", fileName);
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if (_outputFileRecording) {
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WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
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"StartRecordingPlayout() is already recording");
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return 0;
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}
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FileFormats format;
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const uint32_t notificationTime(0);
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CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
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if ((codecInst != NULL) &&
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((codecInst->channels < 1) || (codecInst->channels > 2))) {
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_engineStatisticsPtr->SetLastError(
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VE_BAD_ARGUMENT, kTraceError,
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"StartRecordingPlayout() invalid compression");
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return (-1);
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}
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if (codecInst == NULL) {
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format = kFileFormatPcm16kHzFile;
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codecInst = &dummyCodec;
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} else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
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(STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
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(STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
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format = kFileFormatWavFile;
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} else {
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format = kFileFormatCompressedFile;
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}
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rtc::CritScope cs(&_fileCritSect);
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// Destroy the old instance
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if (_outputFileRecorderPtr) {
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_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
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FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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_outputFileRecorderPtr = NULL;
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}
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_outputFileRecorderPtr =
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FileRecorder::CreateFileRecorder(_instanceId, (const FileFormats)format);
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if (_outputFileRecorderPtr == NULL) {
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_engineStatisticsPtr->SetLastError(
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VE_INVALID_ARGUMENT, kTraceError,
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"StartRecordingPlayout() fileRecorder format isnot correct");
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return -1;
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}
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if (_outputFileRecorderPtr->StartRecordingAudioFile(
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fileName, (const CodecInst&)*codecInst, notificationTime) != 0) {
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_engineStatisticsPtr->SetLastError(
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VE_BAD_FILE, kTraceError,
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"StartRecordingAudioFile() failed to start file recording");
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_outputFileRecorderPtr->StopRecording();
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FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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_outputFileRecorderPtr = NULL;
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return -1;
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}
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_outputFileRecorderPtr->RegisterModuleFileCallback(this);
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_outputFileRecording = true;
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return 0;
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}
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int AudioMixer::StartRecordingPlayout(OutStream* stream,
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const CodecInst* codecInst) {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::StartRecordingPlayout()");
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if (_outputFileRecording) {
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WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
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"StartRecordingPlayout() is already recording");
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return 0;
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}
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FileFormats format;
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const uint32_t notificationTime(0);
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CodecInst dummyCodec = {100, "L16", 16000, 320, 1, 320000};
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if (codecInst != NULL && codecInst->channels != 1) {
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_engineStatisticsPtr->SetLastError(
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VE_BAD_ARGUMENT, kTraceError,
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"StartRecordingPlayout() invalid compression");
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return (-1);
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}
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if (codecInst == NULL) {
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format = kFileFormatPcm16kHzFile;
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codecInst = &dummyCodec;
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} else if ((STR_CASE_CMP(codecInst->plname, "L16") == 0) ||
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(STR_CASE_CMP(codecInst->plname, "PCMU") == 0) ||
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(STR_CASE_CMP(codecInst->plname, "PCMA") == 0)) {
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format = kFileFormatWavFile;
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} else {
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format = kFileFormatCompressedFile;
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}
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rtc::CritScope cs(&_fileCritSect);
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// Destroy the old instance
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if (_outputFileRecorderPtr) {
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_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
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FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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_outputFileRecorderPtr = NULL;
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}
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_outputFileRecorderPtr =
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FileRecorder::CreateFileRecorder(_instanceId, (const FileFormats)format);
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if (_outputFileRecorderPtr == NULL) {
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_engineStatisticsPtr->SetLastError(
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VE_INVALID_ARGUMENT, kTraceError,
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"StartRecordingPlayout() fileRecorder format isnot correct");
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return -1;
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}
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if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
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notificationTime) != 0) {
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_engineStatisticsPtr->SetLastError(
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VE_BAD_FILE, kTraceError,
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"StartRecordingAudioFile() failed to start file recording");
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_outputFileRecorderPtr->StopRecording();
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FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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_outputFileRecorderPtr = NULL;
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return -1;
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}
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_outputFileRecorderPtr->RegisterModuleFileCallback(this);
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_outputFileRecording = true;
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return 0;
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}
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int AudioMixer::StopRecordingPlayout() {
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WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::StopRecordingPlayout()");
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if (!_outputFileRecording) {
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WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
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"StopRecordingPlayout() file isnot recording");
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return -1;
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}
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rtc::CritScope cs(&_fileCritSect);
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if (_outputFileRecorderPtr->StopRecording() != 0) {
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_engineStatisticsPtr->SetLastError(
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VE_STOP_RECORDING_FAILED, kTraceError,
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"StopRecording(), could not stop recording");
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return -1;
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}
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_outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
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FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
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_outputFileRecorderPtr = NULL;
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_outputFileRecording = false;
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return 0;
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}
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int AudioMixer::GetMixedAudio(int sample_rate_hz,
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size_t num_channels,
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AudioFrame* frame) {
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WEBRTC_TRACE(
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kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
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sample_rate_hz, num_channels);
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// --- Record playout if enabled
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{
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rtc::CritScope cs(&_fileCritSect);
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if (_outputFileRecording && _outputFileRecorderPtr)
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_outputFileRecorderPtr->RecordAudioToFile(_audioFrame);
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}
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frame->num_channels_ = num_channels;
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frame->sample_rate_hz_ = sample_rate_hz;
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// TODO(andrew): Ideally the downmixing would occur much earlier, in
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// AudioCodingModule.
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RemixAndResample(_audioFrame, &resampler_, frame);
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return 0;
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}
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int32_t AudioMixer::DoOperationsOnCombinedSignal(bool feed_data_to_apm) {
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if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz) {
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WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
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"AudioMixer::DoOperationsOnCombinedSignal() => "
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"mixing frequency = %d",
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_audioFrame.sample_rate_hz_);
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_mixingFrequencyHz = _audioFrame.sample_rate_hz_;
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}
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// Scale left and/or right channel(s) if balance is active
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if (_panLeft != 1.0 || _panRight != 1.0) {
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if (_audioFrame.num_channels_ == 1) {
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AudioFrameOperations::MonoToStereo(&_audioFrame);
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} else {
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// Pure stereo mode (we are receiving a stereo signal).
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}
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RTC_DCHECK_EQ(_audioFrame.num_channels_, static_cast<size_t>(2));
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AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame);
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}
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// --- Far-end Voice Quality Enhancement (AudioProcessing Module)
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if (feed_data_to_apm) {
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if (_audioProcessingModulePtr->ProcessReverseStream(&_audioFrame) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
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"AudioProcessingModule::ProcessReverseStream() => error");
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RTC_DCHECK(false);
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}
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}
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// --- External media processing
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{
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rtc::CritScope cs(&_callbackCritSect);
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if (_externalMedia) {
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const bool is_stereo = (_audioFrame.num_channels_ == 2);
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if (_externalMediaCallbackPtr) {
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_externalMediaCallbackPtr->Process(
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-1, kPlaybackAllChannelsMixed,
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reinterpret_cast<int16_t*>(_audioFrame.data_),
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_audioFrame.samples_per_channel_, _audioFrame.sample_rate_hz_,
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is_stereo);
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}
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}
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}
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// --- Measure audio level (0-9) for the combined signal
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_audioLevel.ComputeLevel(_audioFrame);
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return 0;
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}
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} // namespace voe
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} // namespace webrtc
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