voice_engine/voe_base_impl.cc - src - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "voice_engine/voe_base_impl.h"

 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_device/audio_device_impl.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "rtc_base/format_macros.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 #include "voice_engine/channel.h"
 #include "voice_engine/include/voe_errors.h"
 #include "voice_engine/transmit_mixer.h"
 #include "voice_engine/voice_engine_impl.h"

 namespace webrtc {

 VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) {
   if (nullptr == voiceEngine) {
     return nullptr;
   }
   VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
   s->AddRef();
   return s;
 }

 VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared)
     : shared_(shared) {}

 VoEBaseImpl::~VoEBaseImpl() {
   TerminateInternal();
 }

 int32_t VoEBaseImpl::RecordedDataIsAvailable(
     const void* audio_data,
     const size_t number_of_frames,
     const size_t bytes_per_sample,
     const size_t number_of_channels,
     const uint32_t sample_rate,
     const uint32_t audio_delay_milliseconds,
     const int32_t clock_drift,
     const uint32_t volume,
     const bool key_pressed,
     uint32_t& new_mic_volume) {
   RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
   RTC_DCHECK(shared_->transmit_mixer() != nullptr);
   RTC_DCHECK(shared_->audio_device() != nullptr);

   uint32_t max_volume = 0;
   uint16_t voe_mic_level = 0;
   // Check for zero to skip this calculation; the consumer may use this to
   // indicate no volume is available.
   if (volume != 0) {
     // Scale from ADM to VoE level range
     if (shared_->audio_device()->MaxMicrophoneVolume(&max_volume) == 0) {
       if (max_volume) {
         voe_mic_level = static_cast<uint16_t>(
             (volume * kMaxVolumeLevel + static_cast<int>(max_volume / 2)) /
             max_volume);
       }
     }
     // We learned that on certain systems (e.g Linux) the voe_mic_level
     // can be greater than the maxVolumeLevel therefore
     // we are going to cap the voe_mic_level to the maxVolumeLevel
     // and change the maxVolume to volume if it turns out that
     // the voe_mic_level is indeed greater than the maxVolumeLevel.
     if (voe_mic_level > kMaxVolumeLevel) {
       voe_mic_level = kMaxVolumeLevel;
       max_volume = volume;
     }
   }

   // Perform channel-independent operations
   // (APM, mix with file, record to file, mute, etc.)
   shared_->transmit_mixer()->PrepareDemux(
       audio_data, number_of_frames, number_of_channels, sample_rate,
       static_cast<uint16_t>(audio_delay_milliseconds), clock_drift,
       voe_mic_level, key_pressed);

   // Copy the audio frame to each sending channel and perform
   // channel-dependent operations (file mixing, mute, etc.), encode and
   // packetize+transmit the RTP packet.
   shared_->transmit_mixer()->ProcessAndEncodeAudio();

   // Scale from VoE to ADM level range.
   uint32_t new_voe_mic_level = shared_->transmit_mixer()->CaptureLevel();
   if (new_voe_mic_level != voe_mic_level) {
     // Return the new volume if AGC has changed the volume.
     return static_cast<int>((new_voe_mic_level * max_volume +
                              static_cast<int>(kMaxVolumeLevel / 2)) /
                             kMaxVolumeLevel);
   }

   return 0;
 }

 int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
                                       const size_t nBytesPerSample,
                                       const size_t nChannels,
                                       const uint32_t samplesPerSec,
                                       void* audioSamples,
                                       size_t& nSamplesOut,
                                       int64_t* elapsed_time_ms,
                                       int64_t* ntp_time_ms) {
   RTC_NOTREACHED();
   return 0;
 }

 void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
                                   int bits_per_sample, int sample_rate,
                                   size_t number_of_channels,
                                   size_t number_of_frames) {
   voe::ChannelOwner ch = shared_->channel_manager().GetChannel(voe_channel);
   voe::Channel* channel = ch.channel();
   if (!channel)
     return;
   if (channel->Sending()) {
     // Send the audio to each channel directly without using the APM in the
     // transmit mixer.
     channel->ProcessAndEncodeAudio(static_cast<const int16_t*>(audio_data),
                                    sample_rate, number_of_frames,
                                    number_of_channels);
   }
 }

 void VoEBaseImpl::PullRenderData(int bits_per_sample,
                                  int sample_rate,
                                  size_t number_of_channels,
                                  size_t number_of_frames,
                                  void* audio_data, int64_t* elapsed_time_ms,
                                  int64_t* ntp_time_ms) {
   RTC_NOTREACHED();
 }

 int VoEBaseImpl::Init(
     AudioDeviceModule* audio_device,
     AudioProcessing* audio_processing,
     const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
   RTC_DCHECK(audio_device);
   RTC_DCHECK(audio_processing);
   rtc::CritScope cs(shared_->crit_sec());
   if (shared_->process_thread()) {
     shared_->process_thread()->Start();
   }

   shared_->set_audio_device(audio_device);
   shared_->set_audio_processing(audio_processing);

   // Configure AudioProcessing components.
   // TODO(peah): Move this initialization to webrtcvoiceengine.cc.
   if (audio_processing->high_pass_filter()->Enable(true) != 0) {
     RTC_LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
     return -1;
   }
   if (audio_processing->echo_cancellation()->enable_drift_compensation(false) !=
       0) {
     RTC_LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
     return -1;
   }
   if (audio_processing->noise_suppression()->set_level(kDefaultNsMode) != 0) {
     RTC_LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
                         << kDefaultNsMode;
     return -1;
   }
   GainControl* agc = audio_processing->gain_control();
   if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
     RTC_LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: "
                         << kMinVolumeLevel
                         << " and maximum: " << kMaxVolumeLevel;
     return -1;
   }
   if (agc->set_mode(kDefaultAgcMode) != 0) {
     RTC_LOG_F(LS_ERROR) << "Failed to set mode: " << kDefaultAgcMode;
     return -1;
   }
   if (agc->Enable(kDefaultAgcState) != 0) {
     RTC_LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState;
     return -1;
   }

 #ifdef WEBRTC_VOICE_ENGINE_AGC
   bool agc_enabled =
       agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled();
   if (shared_->audio_device()->SetAGC(agc_enabled) != 0) {
     RTC_LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled;
     // TODO(ajm): No error return here due to
     // https://code.google.com/p/webrtc/issues/detail?id=1464
   }
 #endif

   RTC_DCHECK(decoder_factory);
   decoder_factory_ = decoder_factory;

   return 0;
 }

 int VoEBaseImpl::Terminate() {
   rtc::CritScope cs(shared_->crit_sec());
   return TerminateInternal();
 }

 int VoEBaseImpl::CreateChannel() {
   return CreateChannel(ChannelConfig());
 }

 int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
   rtc::CritScope cs(shared_->crit_sec());
   ChannelConfig config_copy(config);
   config_copy.acm_config.decoder_factory = decoder_factory_;
   voe::ChannelOwner channel_owner =
       shared_->channel_manager().CreateChannel(config_copy);
   return InitializeChannel(&channel_owner);
 }

 int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
   if (channel_owner->channel()->SetEngineInformation(
           *shared_->process_thread(), *shared_->audio_device(),
           shared_->encoder_queue()) != 0) {
     RTC_LOG(LS_ERROR)
         << "CreateChannel() failed to associate engine and channel."
            " Destroying channel.";
     shared_->channel_manager().DestroyChannel(
         channel_owner->channel()->ChannelId());
     return -1;
   } else if (channel_owner->channel()->Init() != 0) {
     RTC_LOG(LS_ERROR)
         << "CreateChannel() failed to initialize channel. Destroying"
            " channel.";
     shared_->channel_manager().DestroyChannel(
         channel_owner->channel()->ChannelId());
     return -1;
   }
   return channel_owner->channel()->ChannelId();
 }

 int VoEBaseImpl::DeleteChannel(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
   {
     voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
     voe::Channel* channelPtr = ch.channel();
     if (channelPtr == nullptr) {
       RTC_LOG(LS_ERROR) << "DeleteChannel() failed to locate channel";
       return -1;
     }
   }

   shared_->channel_manager().DestroyChannel(channel);
   if (StopSend() != 0) {
     return -1;
   }
   if (StopPlayout() != 0) {
     return -1;
   }
   return 0;
 }

 int VoEBaseImpl::StartPlayout(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
   voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == nullptr) {
     RTC_LOG(LS_ERROR) << "StartPlayout() failed to locate channel";
     return -1;
   }
   if (channelPtr->Playing()) {
     return 0;
   }
   if (StartPlayout() != 0) {
     RTC_LOG(LS_ERROR) << "StartPlayout() failed to start playout";
     return -1;
   }
   return channelPtr->StartPlayout();
 }

 int VoEBaseImpl::StopPlayout(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
   voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == nullptr) {
     RTC_LOG(LS_ERROR) << "StopPlayout() failed to locate channel";
     return -1;
   }
   if (channelPtr->StopPlayout() != 0) {
     RTC_LOG_F(LS_WARNING) << "StopPlayout() failed to stop playout for channel "
                           << channel;
   }
   return StopPlayout();
 }

 int VoEBaseImpl::StartSend(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
   voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == nullptr) {
     RTC_LOG(LS_ERROR) << "StartSend() failed to locate channel";
     return -1;
   }
   if (channelPtr->Sending()) {
     return 0;
   }
   if (StartSend() != 0) {
     RTC_LOG(LS_ERROR) << "StartSend() failed to start recording";
     return -1;
   }
   return channelPtr->StartSend();
 }

 int VoEBaseImpl::StopSend(int channel) {
   rtc::CritScope cs(shared_->crit_sec());
   voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
   voe::Channel* channelPtr = ch.channel();
   if (channelPtr == nullptr) {
     RTC_LOG(LS_ERROR) << "StopSend() failed to locate channel";
     return -1;
   }
   channelPtr->StopSend();
   return StopSend();
 }

 int32_t VoEBaseImpl::StartPlayout() {
   if (!shared_->audio_device()->Playing()) {
     if (shared_->audio_device()->InitPlayout() != 0) {
       RTC_LOG_F(LS_ERROR) << "Failed to initialize playout";
       return -1;
     }
     if (playout_enabled_ && shared_->audio_device()->StartPlayout() != 0) {
       RTC_LOG_F(LS_ERROR) << "Failed to start playout";
       return -1;
     }
   }
   return 0;
 }

 int32_t VoEBaseImpl::StopPlayout() {
   if (!playout_enabled_) {
     return 0;
   }
   // Stop audio-device playing if no channel is playing out.
   if (shared_->NumOfPlayingChannels() == 0) {
     if (shared_->audio_device()->StopPlayout() != 0) {
       RTC_LOG(LS_ERROR) << "StopPlayout() failed to stop playout";
       return -1;
     }
   }
   return 0;
 }

 int32_t VoEBaseImpl::StartSend() {
   if (!shared_->audio_device()->Recording()) {
     if (shared_->audio_device()->InitRecording() != 0) {
       RTC_LOG_F(LS_ERROR) << "Failed to initialize recording";
       return -1;
     }
     if (recording_enabled_ && shared_->audio_device()->StartRecording() != 0) {
       RTC_LOG_F(LS_ERROR) << "Failed to start recording";
       return -1;
     }
   }
   return 0;
 }

 int32_t VoEBaseImpl::StopSend() {
   if (!recording_enabled_) {
     return 0;
   }
   // Stop audio-device recording if no channel is recording.
   if (shared_->NumOfSendingChannels() == 0) {
     if (shared_->audio_device()->StopRecording() != 0) {
       RTC_LOG(LS_ERROR) << "StopSend() failed to stop recording";
       return -1;
     }
     shared_->transmit_mixer()->StopSend();
   }

   return 0;
 }

 int32_t VoEBaseImpl::SetPlayout(bool enabled) {
   RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
   if (playout_enabled_ == enabled) {
     return 0;
   }
   playout_enabled_ = enabled;
   if (shared_->NumOfPlayingChannels() == 0) {
     // If there are no channels attempting to play out yet, there's nothing to
     // be done; we should be in a "not playing out" state either way.
     return 0;
   }
   int32_t ret;
   if (enabled) {
     ret = shared_->audio_device()->StartPlayout();
     if (ret != 0) {
       RTC_LOG(LS_ERROR) << "SetPlayout(true) failed to start playout";
     }
   } else {
     ret = shared_->audio_device()->StopPlayout();
     if (ret != 0) {
       RTC_LOG(LS_ERROR) << "SetPlayout(false) failed to stop playout";
     }
   }
   return ret;
 }

 int32_t VoEBaseImpl::SetRecording(bool enabled) {
   RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
   if (recording_enabled_ == enabled) {
     return 0;
   }
   recording_enabled_ = enabled;
   if (shared_->NumOfSendingChannels() == 0) {
     // If there are no channels attempting to record out yet, there's nothing to
     // be done; we should be in a "not recording" state either way.
     return 0;
   }
   int32_t ret;
   if (enabled) {
     ret = shared_->audio_device()->StartRecording();
     if (ret != 0) {
       RTC_LOG(LS_ERROR) << "SetRecording(true) failed to start recording";
     }
   } else {
     ret = shared_->audio_device()->StopRecording();
     if (ret != 0) {
       RTC_LOG(LS_ERROR) << "SetRecording(false) failed to stop recording";
     }
   }
   return ret;
 }

 int32_t VoEBaseImpl::TerminateInternal() {
   // Delete any remaining channel objects
   shared_->channel_manager().DestroyAllChannels();

   if (shared_->process_thread()) {
     shared_->process_thread()->Stop();
   }

   shared_->set_audio_device(nullptr);
   shared_->set_audio_processing(nullptr);

   return 0;
 }
 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "voice_engine/voe_base_impl.h"

	#include "common_audio/signal_processing/include/signal_processing_library.h"
	#include "modules/audio_coding/include/audio_coding_module.h"
	#include "modules/audio_device/audio_device_impl.h"
	#include "modules/audio_processing/include/audio_processing.h"
	#include "rtc_base/format_macros.h"
	#include "rtc_base/location.h"
	#include "rtc_base/logging.h"
	#include "voice_engine/channel.h"
	#include "voice_engine/include/voe_errors.h"
	#include "voice_engine/transmit_mixer.h"
	#include "voice_engine/voice_engine_impl.h"

	namespace webrtc {

	VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) {
	if (nullptr == voiceEngine) {
	return nullptr;
	}
	VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
	s->AddRef();
	return s;
	}

	VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared)
	: shared_(shared) {}

	VoEBaseImpl::~VoEBaseImpl() {
	TerminateInternal();
	}

	int32_t VoEBaseImpl::RecordedDataIsAvailable(
	const void* audio_data,
	const size_t number_of_frames,
	const size_t bytes_per_sample,
	const size_t number_of_channels,
	const uint32_t sample_rate,
	const uint32_t audio_delay_milliseconds,
	const int32_t clock_drift,
	const uint32_t volume,
	const bool key_pressed,
	uint32_t& new_mic_volume) {
	RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
	RTC_DCHECK(shared_->transmit_mixer() != nullptr);
	RTC_DCHECK(shared_->audio_device() != nullptr);

	uint32_t max_volume = 0;
	uint16_t voe_mic_level = 0;
	// Check for zero to skip this calculation; the consumer may use this to
	// indicate no volume is available.
	if (volume != 0) {
	// Scale from ADM to VoE level range
	if (shared_->audio_device()->MaxMicrophoneVolume(&max_volume) == 0) {
	if (max_volume) {
	voe_mic_level = static_cast<uint16_t>(
	(volume * kMaxVolumeLevel + static_cast<int>(max_volume / 2)) /
	max_volume);
	}
	}
	// We learned that on certain systems (e.g Linux) the voe_mic_level
	// can be greater than the maxVolumeLevel therefore
	// we are going to cap the voe_mic_level to the maxVolumeLevel
	// and change the maxVolume to volume if it turns out that
	// the voe_mic_level is indeed greater than the maxVolumeLevel.
	if (voe_mic_level > kMaxVolumeLevel) {
	voe_mic_level = kMaxVolumeLevel;
	max_volume = volume;
	}
	}

	// Perform channel-independent operations
	// (APM, mix with file, record to file, mute, etc.)
	shared_->transmit_mixer()->PrepareDemux(
	audio_data, number_of_frames, number_of_channels, sample_rate,
	static_cast<uint16_t>(audio_delay_milliseconds), clock_drift,
	voe_mic_level, key_pressed);

	// Copy the audio frame to each sending channel and perform
	// channel-dependent operations (file mixing, mute, etc.), encode and
	// packetize+transmit the RTP packet.
	shared_->transmit_mixer()->ProcessAndEncodeAudio();

	// Scale from VoE to ADM level range.
	uint32_t new_voe_mic_level = shared_->transmit_mixer()->CaptureLevel();
	if (new_voe_mic_level != voe_mic_level) {
	// Return the new volume if AGC has changed the volume.
	return static_cast<int>((new_voe_mic_level * max_volume +
	static_cast<int>(kMaxVolumeLevel / 2)) /
	kMaxVolumeLevel);
	}

	return 0;
	}

	int32_t VoEBaseImpl::NeedMorePlayData(const size_t nSamples,
	const size_t nBytesPerSample,
	const size_t nChannels,
	const uint32_t samplesPerSec,
	void* audioSamples,
	size_t& nSamplesOut,
	int64_t* elapsed_time_ms,
	int64_t* ntp_time_ms) {
	RTC_NOTREACHED();
	return 0;
	}

	void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data,
	int bits_per_sample, int sample_rate,
	size_t number_of_channels,
	size_t number_of_frames) {
	voe::ChannelOwner ch = shared_->channel_manager().GetChannel(voe_channel);
	voe::Channel* channel = ch.channel();
	if (!channel)
	return;
	if (channel->Sending()) {
	// Send the audio to each channel directly without using the APM in the
	// transmit mixer.
	channel->ProcessAndEncodeAudio(static_cast<const int16_t*>(audio_data),
	sample_rate, number_of_frames,
	number_of_channels);
	}
	}

	void VoEBaseImpl::PullRenderData(int bits_per_sample,
	int sample_rate,
	size_t number_of_channels,
	size_t number_of_frames,
	void* audio_data, int64_t* elapsed_time_ms,
	int64_t* ntp_time_ms) {
	RTC_NOTREACHED();
	}

	int VoEBaseImpl::Init(
	AudioDeviceModule* audio_device,
	AudioProcessing* audio_processing,
	const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
	RTC_DCHECK(audio_device);
	RTC_DCHECK(audio_processing);
	rtc::CritScope cs(shared_->crit_sec());
	if (shared_->process_thread()) {
	shared_->process_thread()->Start();
	}

	shared_->set_audio_device(audio_device);
	shared_->set_audio_processing(audio_processing);

	// Configure AudioProcessing components.
	// TODO(peah): Move this initialization to webrtcvoiceengine.cc.
	if (audio_processing->high_pass_filter()->Enable(true) != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to enable high pass filter.";
	return -1;
	}
	if (audio_processing->echo_cancellation()->enable_drift_compensation(false) !=
	0) {
	RTC_LOG_F(LS_ERROR) << "Failed to disable drift compensation.";
	return -1;
	}
	if (audio_processing->noise_suppression()->set_level(kDefaultNsMode) != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to set noise suppression level: "
	<< kDefaultNsMode;
	return -1;
	}
	GainControl* agc = audio_processing->gain_control();
	if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: "
	<< kMinVolumeLevel
	<< " and maximum: " << kMaxVolumeLevel;
	return -1;
	}
	if (agc->set_mode(kDefaultAgcMode) != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to set mode: " << kDefaultAgcMode;
	return -1;
	}
	if (agc->Enable(kDefaultAgcState) != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState;
	return -1;
	}

	#ifdef WEBRTC_VOICE_ENGINE_AGC
	bool agc_enabled =
	agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled();
	if (shared_->audio_device()->SetAGC(agc_enabled) != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled;
	// TODO(ajm): No error return here due to
	// https://code.google.com/p/webrtc/issues/detail?id=1464
	}
	#endif

	RTC_DCHECK(decoder_factory);
	decoder_factory_ = decoder_factory;

	return 0;
	}

	int VoEBaseImpl::Terminate() {
	rtc::CritScope cs(shared_->crit_sec());
	return TerminateInternal();
	}

	int VoEBaseImpl::CreateChannel() {
	return CreateChannel(ChannelConfig());
	}

	int VoEBaseImpl::CreateChannel(const ChannelConfig& config) {
	rtc::CritScope cs(shared_->crit_sec());
	ChannelConfig config_copy(config);
	config_copy.acm_config.decoder_factory = decoder_factory_;
	voe::ChannelOwner channel_owner =
	shared_->channel_manager().CreateChannel(config_copy);
	return InitializeChannel(&channel_owner);
	}

	int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) {
	if (channel_owner->channel()->SetEngineInformation(
	shared_->process_thread(), shared_->audio_device(),
	shared_->encoder_queue()) != 0) {
	RTC_LOG(LS_ERROR)
	<< "CreateChannel() failed to associate engine and channel."
	" Destroying channel.";
	shared_->channel_manager().DestroyChannel(
	channel_owner->channel()->ChannelId());
	return -1;
	} else if (channel_owner->channel()->Init() != 0) {
	RTC_LOG(LS_ERROR)
	<< "CreateChannel() failed to initialize channel. Destroying"
	" channel.";
	shared_->channel_manager().DestroyChannel(
	channel_owner->channel()->ChannelId());
	return -1;
	}
	return channel_owner->channel()->ChannelId();
	}

	int VoEBaseImpl::DeleteChannel(int channel) {
	rtc::CritScope cs(shared_->crit_sec());
	{
	voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
	voe::Channel* channelPtr = ch.channel();
	if (channelPtr == nullptr) {
	RTC_LOG(LS_ERROR) << "DeleteChannel() failed to locate channel";
	return -1;
	}
	}

	shared_->channel_manager().DestroyChannel(channel);
	if (StopSend() != 0) {
	return -1;
	}
	if (StopPlayout() != 0) {
	return -1;
	}
	return 0;
	}

	int VoEBaseImpl::StartPlayout(int channel) {
	rtc::CritScope cs(shared_->crit_sec());
	voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
	voe::Channel* channelPtr = ch.channel();
	if (channelPtr == nullptr) {
	RTC_LOG(LS_ERROR) << "StartPlayout() failed to locate channel";
	return -1;
	}
	if (channelPtr->Playing()) {
	return 0;
	}
	if (StartPlayout() != 0) {
	RTC_LOG(LS_ERROR) << "StartPlayout() failed to start playout";
	return -1;
	}
	return channelPtr->StartPlayout();
	}

	int VoEBaseImpl::StopPlayout(int channel) {
	rtc::CritScope cs(shared_->crit_sec());
	voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
	voe::Channel* channelPtr = ch.channel();
	if (channelPtr == nullptr) {
	RTC_LOG(LS_ERROR) << "StopPlayout() failed to locate channel";
	return -1;
	}
	if (channelPtr->StopPlayout() != 0) {
	RTC_LOG_F(LS_WARNING) << "StopPlayout() failed to stop playout for channel "
	<< channel;
	}
	return StopPlayout();
	}

	int VoEBaseImpl::StartSend(int channel) {
	rtc::CritScope cs(shared_->crit_sec());
	voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
	voe::Channel* channelPtr = ch.channel();
	if (channelPtr == nullptr) {
	RTC_LOG(LS_ERROR) << "StartSend() failed to locate channel";
	return -1;
	}
	if (channelPtr->Sending()) {
	return 0;
	}
	if (StartSend() != 0) {
	RTC_LOG(LS_ERROR) << "StartSend() failed to start recording";
	return -1;
	}
	return channelPtr->StartSend();
	}

	int VoEBaseImpl::StopSend(int channel) {
	rtc::CritScope cs(shared_->crit_sec());
	voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel);
	voe::Channel* channelPtr = ch.channel();
	if (channelPtr == nullptr) {
	RTC_LOG(LS_ERROR) << "StopSend() failed to locate channel";
	return -1;
	}
	channelPtr->StopSend();
	return StopSend();
	}

	int32_t VoEBaseImpl::StartPlayout() {
	if (!shared_->audio_device()->Playing()) {
	if (shared_->audio_device()->InitPlayout() != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to initialize playout";
	return -1;
	}
	if (playout_enabled_ && shared_->audio_device()->StartPlayout() != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to start playout";
	return -1;
	}
	}
	return 0;
	}

	int32_t VoEBaseImpl::StopPlayout() {
	if (!playout_enabled_) {
	return 0;
	}
	// Stop audio-device playing if no channel is playing out.
	if (shared_->NumOfPlayingChannels() == 0) {
	if (shared_->audio_device()->StopPlayout() != 0) {
	RTC_LOG(LS_ERROR) << "StopPlayout() failed to stop playout";
	return -1;
	}
	}
	return 0;
	}

	int32_t VoEBaseImpl::StartSend() {
	if (!shared_->audio_device()->Recording()) {
	if (shared_->audio_device()->InitRecording() != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to initialize recording";
	return -1;
	}
	if (recording_enabled_ && shared_->audio_device()->StartRecording() != 0) {
	RTC_LOG_F(LS_ERROR) << "Failed to start recording";
	return -1;
	}
	}
	return 0;
	}

	int32_t VoEBaseImpl::StopSend() {
	if (!recording_enabled_) {
	return 0;
	}
	// Stop audio-device recording if no channel is recording.
	if (shared_->NumOfSendingChannels() == 0) {
	if (shared_->audio_device()->StopRecording() != 0) {
	RTC_LOG(LS_ERROR) << "StopSend() failed to stop recording";
	return -1;
	}
	shared_->transmit_mixer()->StopSend();
	}

	return 0;
	}

	int32_t VoEBaseImpl::SetPlayout(bool enabled) {
	RTC_LOG(INFO) << "SetPlayout(" << enabled << ")";
	if (playout_enabled_ == enabled) {
	return 0;
	}
	playout_enabled_ = enabled;
	if (shared_->NumOfPlayingChannels() == 0) {
	// If there are no channels attempting to play out yet, there's nothing to
	// be done; we should be in a "not playing out" state either way.
	return 0;
	}
	int32_t ret;
	if (enabled) {
	ret = shared_->audio_device()->StartPlayout();
	if (ret != 0) {
	RTC_LOG(LS_ERROR) << "SetPlayout(true) failed to start playout";
	}
	} else {
	ret = shared_->audio_device()->StopPlayout();
	if (ret != 0) {
	RTC_LOG(LS_ERROR) << "SetPlayout(false) failed to stop playout";
	}
	}
	return ret;
	}

	int32_t VoEBaseImpl::SetRecording(bool enabled) {
	RTC_LOG(INFO) << "SetRecording(" << enabled << ")";
	if (recording_enabled_ == enabled) {
	return 0;
	}
	recording_enabled_ = enabled;
	if (shared_->NumOfSendingChannels() == 0) {
	// If there are no channels attempting to record out yet, there's nothing to
	// be done; we should be in a "not recording" state either way.
	return 0;
	}
	int32_t ret;
	if (enabled) {
	ret = shared_->audio_device()->StartRecording();
	if (ret != 0) {
	RTC_LOG(LS_ERROR) << "SetRecording(true) failed to start recording";
	}
	} else {
	ret = shared_->audio_device()->StopRecording();
	if (ret != 0) {
	RTC_LOG(LS_ERROR) << "SetRecording(false) failed to stop recording";
	}
	}
	return ret;
	}

	int32_t VoEBaseImpl::TerminateInternal() {
	// Delete any remaining channel objects
	shared_->channel_manager().DestroyAllChannels();

	if (shared_->process_thread()) {
	shared_->process_thread()->Stop();
	}

	shared_->set_audio_device(nullptr);
	shared_->set_audio_processing(nullptr);

	return 0;
	}
	} // namespace webrtc