a9cc40b7d2
[This CL is a rebase of an original CL by solenberg@: https://codereview.webrtc.org/2948763002/ which in turn was a rebase of an original CL by peah@: https://chromium-review.googlesource.com/c/527032/] Allow an external audio processing module to be used in WebRTC This CL adds support for optionally using an externally created audio processing module in a peerconnection. The ownership is shared between the peerconnection and the external creator of the module. As part of this the internal ownership of the audio processing module is moved from VoiceEngine to WebRtcVoiceEngine. BUG=webrtc:7775 Review-Url: https://codereview.webrtc.org/2961723004 Cr-Commit-Position: refs/heads/master@{#18837}
111 lines
3.2 KiB
C++
111 lines
3.2 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/voice_engine/shared_data.h"
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#include "webrtc/modules/audio_processing/include/audio_processing.h"
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#include "webrtc/system_wrappers/include/trace.h"
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#include "webrtc/voice_engine/channel.h"
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#include "webrtc/voice_engine/output_mixer.h"
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#include "webrtc/voice_engine/transmit_mixer.h"
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namespace webrtc {
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namespace voe {
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static int32_t _gInstanceCounter = 0;
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SharedData::SharedData()
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: _instanceId(++_gInstanceCounter),
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_channelManager(_gInstanceCounter),
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_engineStatistics(_gInstanceCounter),
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_audioDevicePtr(NULL),
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_moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")),
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encoder_queue_("AudioEncoderQueue") {
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Trace::CreateTrace();
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if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0) {
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_outputMixerPtr->SetEngineInformation(_engineStatistics);
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}
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if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0) {
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_transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
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_engineStatistics, _channelManager);
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}
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}
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SharedData::~SharedData()
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{
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OutputMixer::Destroy(_outputMixerPtr);
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TransmitMixer::Destroy(_transmitMixerPtr);
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if (_audioDevicePtr) {
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_audioDevicePtr->Release();
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}
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_moduleProcessThreadPtr->Stop();
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Trace::ReturnTrace();
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}
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rtc::TaskQueue* SharedData::encoder_queue() {
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RTC_DCHECK_RUN_ON(&construction_thread_);
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return &encoder_queue_;
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}
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void SharedData::set_audio_device(
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const rtc::scoped_refptr<AudioDeviceModule>& audio_device) {
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_audioDevicePtr = audio_device;
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}
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void SharedData::set_audio_processing(AudioProcessing* audioproc) {
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_transmitMixerPtr->SetAudioProcessingModule(audioproc);
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_outputMixerPtr->SetAudioProcessingModule(audioproc);
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}
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int SharedData::NumOfSendingChannels() {
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ChannelManager::Iterator it(&_channelManager);
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int sending_channels = 0;
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for (ChannelManager::Iterator it(&_channelManager); it.IsValid();
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it.Increment()) {
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if (it.GetChannel()->Sending())
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++sending_channels;
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}
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return sending_channels;
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}
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int SharedData::NumOfPlayingChannels() {
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ChannelManager::Iterator it(&_channelManager);
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int playout_channels = 0;
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for (ChannelManager::Iterator it(&_channelManager); it.IsValid();
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it.Increment()) {
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if (it.GetChannel()->Playing())
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++playout_channels;
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}
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return playout_channels;
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}
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void SharedData::SetLastError(int32_t error) const {
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_engineStatistics.SetLastError(error);
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}
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void SharedData::SetLastError(int32_t error,
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TraceLevel level) const {
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_engineStatistics.SetLastError(error, level);
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}
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void SharedData::SetLastError(int32_t error, TraceLevel level,
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const char* msg) const {
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_engineStatistics.SetLastError(error, level, msg);
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}
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} // namespace voe
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} // namespace webrtc
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