ec6fbd2776
First approach to remove parts of the heavy load done for encoding, and preparation for sending, from native audio thread to separate task queue. With this change we will give the native input audio thread more time to "relax" between successive audio captures. Separate profiling done on Android has verified that the change works well; the load is now redistributed and the load of the native AudioRecordThread is reduced. Similar conclusions should be valid for all other OS:es as well. BUG=NONE CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng Review-Url: https://codereview.webrtc.org/2665693002 Cr-Commit-Position: refs/heads/master@{#17488}
112 lines
3.2 KiB
C++
112 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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audioproc_.reset(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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