| /* |
| * 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 "api/audio_codecs/builtin_audio_decoder_factory.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(); |
| } |
| |
| void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) { |
| rtc::CritScope cs(&callbackCritSect_); |
| if (error == AudioDeviceObserver::kRecordingError) { |
| LOG_F(LS_ERROR) << "VE_RUNTIME_REC_ERROR"; |
| } else if (error == AudioDeviceObserver::kPlayoutError) { |
| LOG_F(LS_ERROR) << "VE_RUNTIME_PLAY_ERROR"; |
| } |
| } |
| |
| void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) { |
| rtc::CritScope cs(&callbackCritSect_); |
| if (warning == AudioDeviceObserver::kRecordingWarning) { |
| LOG_F(LS_WARNING) << "VE_RUNTIME_REC_WARNING"; |
| } else if (warning == AudioDeviceObserver::kPlayoutWarning) { |
| LOG_F(LS_WARNING) << "VE_RUNTIME_PLAY_WARNING"; |
| } |
| } |
| |
| 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* external_adm, |
| AudioProcessing* audio_processing, |
| const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) { |
| RTC_DCHECK(audio_processing); |
| rtc::CritScope cs(shared_->crit_sec()); |
| WebRtcSpl_Init(); |
| if (shared_->statistics().Initialized()) { |
| return 0; |
| } |
| if (shared_->process_thread()) { |
| shared_->process_thread()->Start(); |
| } |
| |
| // Create an internal ADM if the user has not added an external |
| // ADM implementation as input to Init(). |
| if (external_adm == nullptr) { |
| #if !defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE) |
| return -1; |
| #else |
| // Create the internal ADM implementation. |
| shared_->set_audio_device(AudioDeviceModule::Create( |
| VoEId(shared_->instance_id(), -1), |
| AudioDeviceModule::kPlatformDefaultAudio)); |
| if (shared_->audio_device() == nullptr) { |
| shared_->SetLastError(VE_NO_MEMORY, kTraceCritical, |
| "Init() failed to create the ADM"); |
| return -1; |
| } |
| #endif // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE |
| } else { |
| // Use the already existing external ADM implementation. |
| shared_->set_audio_device(external_adm); |
| LOG_F(LS_INFO) |
| << "An external ADM implementation will be used in VoiceEngine"; |
| } |
| |
| // Register the ADM to the process thread, which will drive the error |
| // callback mechanism |
| if (shared_->process_thread()) { |
| shared_->process_thread()->RegisterModule(shared_->audio_device(), |
| RTC_FROM_HERE); |
| } |
| |
| bool available = false; |
| |
| // -------------------- |
| // Reinitialize the ADM |
| |
| // Register the AudioObserver implementation |
| if (shared_->audio_device()->RegisterEventObserver(this) != 0) { |
| shared_->SetLastError( |
| VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning, |
| "Init() failed to register event observer for the ADM"); |
| } |
| |
| // Register the AudioTransport implementation |
| if (shared_->audio_device()->RegisterAudioCallback(this) != 0) { |
| shared_->SetLastError( |
| VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning, |
| "Init() failed to register audio callback for the ADM"); |
| } |
| |
| // ADM initialization |
| if (shared_->audio_device()->Init() != 0) { |
| shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError, |
| "Init() failed to initialize the ADM"); |
| return -1; |
| } |
| |
| // Initialize the default speaker |
| if (shared_->audio_device()->SetPlayoutDevice( |
| WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) { |
| shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo, |
| "Init() failed to set the default output device"); |
| } |
| if (shared_->audio_device()->InitSpeaker() != 0) { |
| shared_->SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo, |
| "Init() failed to initialize the speaker"); |
| } |
| |
| // Initialize the default microphone |
| if (shared_->audio_device()->SetRecordingDevice( |
| WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) != 0) { |
| shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo, |
| "Init() failed to set the default input device"); |
| } |
| if (shared_->audio_device()->InitMicrophone() != 0) { |
| shared_->SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo, |
| "Init() failed to initialize the microphone"); |
| } |
| |
| // Set number of channels |
| if (shared_->audio_device()->StereoPlayoutIsAvailable(&available) != 0) { |
| shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "Init() failed to query stereo playout mode"); |
| } |
| if (shared_->audio_device()->SetStereoPlayout(available) != 0) { |
| shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "Init() failed to set mono/stereo playout mode"); |
| } |
| |
| // TODO(andrew): These functions don't tell us whether stereo recording |
| // is truly available. We simply set the AudioProcessing input to stereo |
| // here, because we have to wait until receiving the first frame to |
| // determine the actual number of channels anyway. |
| // |
| // These functions may be changed; tracked here: |
| // http://code.google.com/p/webrtc/issues/detail?id=204 |
| shared_->audio_device()->StereoRecordingIsAvailable(&available); |
| if (shared_->audio_device()->SetStereoRecording(available) != 0) { |
| shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "Init() failed to set mono/stereo recording mode"); |
| } |
| |
| shared_->set_audio_processing(audio_processing); |
| |
| // Set the error state for any failures in this block. |
| shared_->SetLastError(VE_APM_ERROR); |
| // Configure AudioProcessing components. |
| // TODO(peah): Move this initialization to webrtcvoiceengine.cc. |
| if (audio_processing->high_pass_filter()->Enable(true) != 0) { |
| LOG_F(LS_ERROR) << "Failed to enable high pass filter."; |
| return -1; |
| } |
| if (audio_processing->echo_cancellation()->enable_drift_compensation(false) != |
| 0) { |
| LOG_F(LS_ERROR) << "Failed to disable drift compensation."; |
| return -1; |
| } |
| if (audio_processing->noise_suppression()->set_level(kDefaultNsMode) != 0) { |
| 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) { |
| LOG_F(LS_ERROR) << "Failed to set analog level limits with minimum: " |
| << kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel; |
| return -1; |
| } |
| if (agc->set_mode(kDefaultAgcMode) != 0) { |
| LOG_F(LS_ERROR) << "Failed to set mode: " << kDefaultAgcMode; |
| return -1; |
| } |
| if (agc->Enable(kDefaultAgcState) != 0) { |
| LOG_F(LS_ERROR) << "Failed to set agc state: " << kDefaultAgcState; |
| return -1; |
| } |
| shared_->SetLastError(0); // Clear error state. |
| |
| #ifdef WEBRTC_VOICE_ENGINE_AGC |
| bool agc_enabled = |
| agc->mode() == GainControl::kAdaptiveAnalog && agc->is_enabled(); |
| if (shared_->audio_device()->SetAGC(agc_enabled) != 0) { |
| LOG_F(LS_ERROR) << "Failed to set agc to enabled: " << agc_enabled; |
| shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR); |
| // TODO(ajm): No error return here due to |
| // https://code.google.com/p/webrtc/issues/detail?id=1464 |
| } |
| #endif |
| |
| if (decoder_factory) |
| decoder_factory_ = decoder_factory; |
| else |
| decoder_factory_ = CreateBuiltinAudioDecoderFactory(); |
| |
| return shared_->statistics().SetInitialized(); |
| } |
| |
| 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()); |
| if (!shared_->statistics().Initialized()) { |
| shared_->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| |
| 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_->statistics(), |
| *shared_->process_thread(), *shared_->audio_device(), |
| &callbackCritSect_, shared_->encoder_queue()) != 0) { |
| shared_->SetLastError( |
| VE_CHANNEL_NOT_CREATED, kTraceError, |
| "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) { |
| shared_->SetLastError( |
| VE_CHANNEL_NOT_CREATED, kTraceError, |
| "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()); |
| if (!shared_->statistics().Initialized()) { |
| shared_->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| |
| { |
| voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel); |
| voe::Channel* channelPtr = ch.channel(); |
| if (channelPtr == nullptr) { |
| shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "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()); |
| if (!shared_->statistics().Initialized()) { |
| shared_->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel); |
| voe::Channel* channelPtr = ch.channel(); |
| if (channelPtr == nullptr) { |
| shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StartPlayout() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->Playing()) { |
| return 0; |
| } |
| if (StartPlayout() != 0) { |
| shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError, |
| "StartPlayout() failed to start playout"); |
| return -1; |
| } |
| return channelPtr->StartPlayout(); |
| } |
| |
| int VoEBaseImpl::StopPlayout(int channel) { |
| rtc::CritScope cs(shared_->crit_sec()); |
| if (!shared_->statistics().Initialized()) { |
| shared_->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel); |
| voe::Channel* channelPtr = ch.channel(); |
| if (channelPtr == nullptr) { |
| shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StopPlayout() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->StopPlayout() != 0) { |
| 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()); |
| if (!shared_->statistics().Initialized()) { |
| shared_->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel); |
| voe::Channel* channelPtr = ch.channel(); |
| if (channelPtr == nullptr) { |
| shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StartSend() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->Sending()) { |
| return 0; |
| } |
| if (StartSend() != 0) { |
| shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError, |
| "StartSend() failed to start recording"); |
| return -1; |
| } |
| return channelPtr->StartSend(); |
| } |
| |
| int VoEBaseImpl::StopSend(int channel) { |
| rtc::CritScope cs(shared_->crit_sec()); |
| if (!shared_->statistics().Initialized()) { |
| shared_->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ChannelOwner ch = shared_->channel_manager().GetChannel(channel); |
| voe::Channel* channelPtr = ch.channel(); |
| if (channelPtr == nullptr) { |
| shared_->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "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) { |
| LOG_F(LS_ERROR) << "Failed to initialize playout"; |
| return -1; |
| } |
| if (shared_->audio_device()->StartPlayout() != 0) { |
| LOG_F(LS_ERROR) << "Failed to start playout"; |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::StopPlayout() { |
| // Stop audio-device playing if no channel is playing out |
| if (shared_->NumOfPlayingChannels() == 0) { |
| if (shared_->audio_device()->StopPlayout() != 0) { |
| shared_->SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError, |
| "StopPlayout() failed to stop playout"); |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::StartSend() { |
| if (!shared_->audio_device()->RecordingIsInitialized() && |
| !shared_->audio_device()->Recording()) { |
| if (shared_->audio_device()->InitRecording() != 0) { |
| LOG_F(LS_ERROR) << "Failed to initialize recording"; |
| return -1; |
| } |
| } |
| if (!shared_->audio_device()->Recording()) { |
| if (shared_->audio_device()->StartRecording() != 0) { |
| LOG_F(LS_ERROR) << "Failed to start recording"; |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::StopSend() { |
| if (shared_->NumOfSendingChannels() == 0) { |
| // Stop audio-device recording if no channel is recording |
| if (shared_->audio_device()->StopRecording() != 0) { |
| shared_->SetLastError(VE_CANNOT_STOP_RECORDING, kTraceError, |
| "StopSend() failed to stop recording"); |
| return -1; |
| } |
| shared_->transmit_mixer()->StopSend(); |
| } |
| |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::TerminateInternal() { |
| // Delete any remaining channel objects |
| shared_->channel_manager().DestroyAllChannels(); |
| |
| if (shared_->process_thread()) { |
| if (shared_->audio_device()) { |
| shared_->process_thread()->DeRegisterModule(shared_->audio_device()); |
| } |
| shared_->process_thread()->Stop(); |
| } |
| |
| if (shared_->audio_device()) { |
| if (shared_->audio_device()->StopPlayout() != 0) { |
| shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "TerminateInternal() failed to stop playout"); |
| } |
| if (shared_->audio_device()->StopRecording() != 0) { |
| shared_->SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "TerminateInternal() failed to stop recording"); |
| } |
| if (shared_->audio_device()->RegisterEventObserver(nullptr) != 0) { |
| shared_->SetLastError( |
| VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning, |
| "TerminateInternal() failed to de-register event observer " |
| "for the ADM"); |
| } |
| if (shared_->audio_device()->RegisterAudioCallback(nullptr) != 0) { |
| shared_->SetLastError( |
| VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning, |
| "TerminateInternal() failed to de-register audio callback " |
| "for the ADM"); |
| } |
| if (shared_->audio_device()->Terminate() != 0) { |
| shared_->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError, |
| "TerminateInternal() failed to terminate the ADM"); |
| } |
| shared_->set_audio_device(nullptr); |
| } |
| |
| shared_->set_audio_processing(nullptr); |
| |
| return shared_->statistics().SetUnInitialized(); |
| } |
| } // namespace webrtc |