92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
111 lines
3.1 KiB
C++
111 lines
3.1 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 "voice_engine/shared_data.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "system_wrappers/include/trace.h"
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#include "voice_engine/channel.h"
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#include "voice_engine/output_mixer.h"
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#include "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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