| /* |
| * 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 "webrtc/voice_engine/voe_base_impl.h" |
| |
| #include "webrtc/common.h" |
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
| #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" |
| #include "webrtc/modules/audio_device/audio_device_impl.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/file_wrapper.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| #include "webrtc/voice_engine/channel.h" |
| #include "webrtc/voice_engine/include/voe_errors.h" |
| #include "webrtc/voice_engine/output_mixer.h" |
| #include "webrtc/voice_engine/transmit_mixer.h" |
| #include "webrtc/voice_engine/utility.h" |
| #include "webrtc/voice_engine/voice_engine_impl.h" |
| |
| namespace webrtc |
| { |
| |
| VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) |
| { |
| if (NULL == voiceEngine) |
| { |
| return NULL; |
| } |
| VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine); |
| s->AddRef(); |
| return s; |
| } |
| |
| VoEBaseImpl::VoEBaseImpl(voe::SharedData* shared) : |
| _voiceEngineObserverPtr(NULL), |
| _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _voiceEngineObserver(false), _shared(shared) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl() - ctor"); |
| } |
| |
| VoEBaseImpl::~VoEBaseImpl() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "~VoEBaseImpl() - dtor"); |
| |
| TerminateInternal(); |
| |
| delete &_callbackCritSect; |
| } |
| |
| void VoEBaseImpl::OnErrorIsReported(ErrorCode error) |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (_voiceEngineObserver) |
| { |
| if (_voiceEngineObserverPtr) |
| { |
| int errCode(0); |
| if (error == AudioDeviceObserver::kRecordingError) |
| { |
| errCode = VE_RUNTIME_REC_ERROR; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::OnErrorIsReported() => VE_RUNTIME_REC_ERROR"); |
| } |
| else if (error == AudioDeviceObserver::kPlayoutError) |
| { |
| errCode = VE_RUNTIME_PLAY_ERROR; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_PLAY_ERROR"); |
| } |
| // Deliver callback (-1 <=> no channel dependency) |
| _voiceEngineObserverPtr->CallbackOnError(-1, errCode); |
| } |
| } |
| } |
| |
| void VoEBaseImpl::OnWarningIsReported(WarningCode warning) |
| { |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (_voiceEngineObserver) |
| { |
| if (_voiceEngineObserverPtr) |
| { |
| int warningCode(0); |
| if (warning == AudioDeviceObserver::kRecordingWarning) |
| { |
| warningCode = VE_RUNTIME_REC_WARNING; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_REC_WARNING"); |
| } |
| else if (warning == AudioDeviceObserver::kPlayoutWarning) |
| { |
| warningCode = VE_RUNTIME_PLAY_WARNING; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_PLAY_WARNING"); |
| } |
| // Deliver callback (-1 <=> no channel dependency) |
| _voiceEngineObserverPtr->CallbackOnError(-1, warningCode); |
| } |
| } |
| } |
| |
| int32_t VoEBaseImpl::RecordedDataIsAvailable( |
| const void* audioSamples, |
| uint32_t nSamples, |
| uint8_t nBytesPerSample, |
| uint8_t nChannels, |
| uint32_t samplesPerSec, |
| uint32_t totalDelayMS, |
| int32_t clockDrift, |
| uint32_t micLevel, |
| bool keyPressed, |
| uint32_t& newMicLevel) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, " |
| "nBytesPerSample=%u, nChannels=%u, samplesPerSec=%u, " |
| "totalDelayMS=%u, clockDrift=%d, micLevel=%u)", |
| nSamples, nBytesPerSample, nChannels, samplesPerSec, |
| totalDelayMS, clockDrift, micLevel); |
| newMicLevel = static_cast<uint32_t>(ProcessRecordedDataWithAPM( |
| NULL, 0, audioSamples, samplesPerSec, nChannels, nSamples, |
| totalDelayMS, clockDrift, micLevel, keyPressed)); |
| |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::NeedMorePlayData( |
| uint32_t nSamples, |
| uint8_t nBytesPerSample, |
| uint8_t nChannels, |
| uint32_t samplesPerSec, |
| void* audioSamples, |
| uint32_t& nSamplesOut, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::NeedMorePlayData(nSamples=%u, " |
| "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)", |
| nSamples, nBytesPerSample, nChannels, samplesPerSec); |
| |
| GetPlayoutData(static_cast<int>(samplesPerSec), |
| static_cast<int>(nChannels), |
| static_cast<int>(nSamples), true, audioSamples, |
| elapsed_time_ms, ntp_time_ms); |
| |
| nSamplesOut = _audioFrame.samples_per_channel_; |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::OnDataAvailable(const int voe_channels[], |
| int number_of_voe_channels, |
| const int16_t* audio_data, |
| int sample_rate, |
| int number_of_channels, |
| int number_of_frames, |
| int audio_delay_milliseconds, |
| int volume, |
| bool key_pressed, |
| bool need_audio_processing) { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::OnDataAvailable(number_of_voe_channels=%d, " |
| "sample_rate=%d, number_of_channels=%d, number_of_frames=%d, " |
| "audio_delay_milliseconds=%d, volume=%d, " |
| "key_pressed=%d, need_audio_processing=%d)", |
| number_of_voe_channels, sample_rate, number_of_channels, |
| number_of_frames, audio_delay_milliseconds, volume, |
| key_pressed, need_audio_processing); |
| if (number_of_voe_channels == 0) |
| return 0; |
| |
| if (need_audio_processing) { |
| return ProcessRecordedDataWithAPM( |
| voe_channels, number_of_voe_channels, audio_data, sample_rate, |
| number_of_channels, number_of_frames, audio_delay_milliseconds, |
| 0, volume, key_pressed); |
| } |
| |
| // No need to go through the APM, demultiplex the data to each VoE channel, |
| // encode and send to the network. |
| for (int i = 0; i < number_of_voe_channels; ++i) { |
| // TODO(ajm): In the case where multiple channels are using the same codec |
| // rate, this path needlessly does extra conversions. We should convert once |
| // and share between channels. |
| PushCaptureData(voe_channels[i], audio_data, 16, sample_rate, |
| number_of_channels, number_of_frames); |
| } |
| |
| // Return 0 to indicate no need to change the volume. |
| return 0; |
| } |
| |
| void VoEBaseImpl::OnData(int voe_channel, const void* audio_data, |
| int bits_per_sample, int sample_rate, |
| int number_of_channels, |
| int number_of_frames) { |
| PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate, |
| number_of_channels, number_of_frames); |
| } |
| |
| void VoEBaseImpl::PushCaptureData(int voe_channel, const void* audio_data, |
| int bits_per_sample, int sample_rate, |
| int number_of_channels, |
| int number_of_frames) { |
| voe::ChannelOwner ch = _shared->channel_manager().GetChannel(voe_channel); |
| voe::Channel* channel_ptr = ch.channel(); |
| if (!channel_ptr) |
| return; |
| |
| if (channel_ptr->Sending()) { |
| channel_ptr->Demultiplex(static_cast<const int16_t*>(audio_data), |
| sample_rate, number_of_frames, number_of_channels); |
| channel_ptr->PrepareEncodeAndSend(sample_rate); |
| channel_ptr->EncodeAndSend(); |
| } |
| } |
| |
| void VoEBaseImpl::PullRenderData(int bits_per_sample, int sample_rate, |
| int number_of_channels, int number_of_frames, |
| void* audio_data, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) { |
| assert(bits_per_sample == 16); |
| assert(number_of_frames == static_cast<int>(sample_rate / 100)); |
| |
| GetPlayoutData(sample_rate, number_of_channels, number_of_frames, false, |
| audio_data, elapsed_time_ms, ntp_time_ms); |
| } |
| |
| int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "RegisterVoiceEngineObserver(observer=0x%d)", &observer); |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (_voiceEngineObserverPtr) |
| { |
| _shared->SetLastError(VE_INVALID_OPERATION, kTraceError, |
| "RegisterVoiceEngineObserver() observer already enabled"); |
| return -1; |
| } |
| |
| // Register the observer in all active channels |
| for (voe::ChannelManager::Iterator it(&_shared->channel_manager()); |
| it.IsValid(); |
| it.Increment()) { |
| it.GetChannel()->RegisterVoiceEngineObserver(observer); |
| } |
| |
| _shared->transmit_mixer()->RegisterVoiceEngineObserver(observer); |
| |
| _voiceEngineObserverPtr = &observer; |
| _voiceEngineObserver = true; |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::DeRegisterVoiceEngineObserver() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "DeRegisterVoiceEngineObserver()"); |
| CriticalSectionScoped cs(&_callbackCritSect); |
| if (!_voiceEngineObserverPtr) |
| { |
| _shared->SetLastError(VE_INVALID_OPERATION, kTraceError, |
| "DeRegisterVoiceEngineObserver() observer already disabled"); |
| return 0; |
| } |
| |
| _voiceEngineObserver = false; |
| _voiceEngineObserverPtr = NULL; |
| |
| // Deregister the observer in all active channels |
| for (voe::ChannelManager::Iterator it(&_shared->channel_manager()); |
| it.IsValid(); |
| it.Increment()) { |
| it.GetChannel()->DeRegisterVoiceEngineObserver(); |
| } |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::Init(AudioDeviceModule* external_adm, |
| AudioProcessing* audioproc) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "Init(external_adm=0x%p)", external_adm); |
| CriticalSectionScoped cs(_shared->crit_sec()); |
| |
| WebRtcSpl_Init(); |
| |
| if (_shared->statistics().Initialized()) |
| { |
| return 0; |
| } |
| |
| if (_shared->process_thread()) |
| { |
| if (_shared->process_thread()->Start() != 0) |
| { |
| _shared->SetLastError(VE_THREAD_ERROR, kTraceError, |
| "Init() failed to start module process thread"); |
| return -1; |
| } |
| } |
| |
| // Create an internal ADM if the user has not added an external |
| // ADM implementation as input to Init(). |
| if (external_adm == NULL) |
| { |
| // Create the internal ADM implementation. |
| _shared->set_audio_device(AudioDeviceModuleImpl::Create( |
| VoEId(_shared->instance_id(), -1), _shared->audio_device_layer())); |
| |
| if (_shared->audio_device() == NULL) |
| { |
| _shared->SetLastError(VE_NO_MEMORY, kTraceCritical, |
| "Init() failed to create the ADM"); |
| return -1; |
| } |
| } |
| else |
| { |
| // Use the already existing external ADM implementation. |
| _shared->set_audio_device(external_adm); |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "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()) != 0) |
| { |
| _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError, |
| "Init() failed to register the ADM"); |
| return -1; |
| } |
| |
| 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"); |
| } |
| |
| if (!audioproc) { |
| audioproc = AudioProcessing::Create(VoEId(_shared->instance_id(), -1)); |
| if (!audioproc) { |
| LOG(LS_ERROR) << "Failed to create AudioProcessing."; |
| _shared->SetLastError(VE_NO_MEMORY); |
| return -1; |
| } |
| } |
| _shared->set_audio_processing(audioproc); |
| |
| // Set the error state for any failures in this block. |
| _shared->SetLastError(VE_APM_ERROR); |
| // Configure AudioProcessing components. |
| if (audioproc->high_pass_filter()->Enable(true) != 0) { |
| LOG_FERR1(LS_ERROR, high_pass_filter()->Enable, true); |
| return -1; |
| } |
| if (audioproc->echo_cancellation()->enable_drift_compensation(false) != 0) { |
| LOG_FERR1(LS_ERROR, enable_drift_compensation, false); |
| return -1; |
| } |
| if (audioproc->noise_suppression()->set_level(kDefaultNsMode) != 0) { |
| LOG_FERR1(LS_ERROR, noise_suppression()->set_level, kDefaultNsMode); |
| return -1; |
| } |
| GainControl* agc = audioproc->gain_control(); |
| if (agc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) { |
| LOG_FERR2(LS_ERROR, agc->set_analog_level_limits, kMinVolumeLevel, |
| kMaxVolumeLevel); |
| return -1; |
| } |
| if (agc->set_mode(kDefaultAgcMode) != 0) { |
| LOG_FERR1(LS_ERROR, agc->set_mode, kDefaultAgcMode); |
| return -1; |
| } |
| if (agc->Enable(kDefaultAgcState) != 0) { |
| LOG_FERR1(LS_ERROR, agc->Enable, 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_FERR1(LS_ERROR, audio_device()->SetAGC, 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 |
| |
| return _shared->statistics().SetInitialized(); |
| } |
| |
| int VoEBaseImpl::Terminate() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "Terminate()"); |
| CriticalSectionScoped cs(_shared->crit_sec()); |
| return TerminateInternal(); |
| } |
| |
| int VoEBaseImpl::CreateChannel() { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "CreateChannel()"); |
| CriticalSectionScoped cs(_shared->crit_sec()); |
| if (!_shared->statistics().Initialized()) { |
| _shared->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| |
| voe::ChannelOwner channel_owner = _shared->channel_manager().CreateChannel(); |
| |
| return InitializeChannel(&channel_owner); |
| } |
| |
| int VoEBaseImpl::CreateChannel(const Config& config) { |
| CriticalSectionScoped cs(_shared->crit_sec()); |
| if (!_shared->statistics().Initialized()) { |
| _shared->SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ChannelOwner channel_owner = _shared->channel_manager().CreateChannel( |
| config); |
| return InitializeChannel(&channel_owner); |
| } |
| |
| int VoEBaseImpl::InitializeChannel(voe::ChannelOwner* channel_owner) |
| { |
| if (channel_owner->channel()->SetEngineInformation( |
| _shared->statistics(), |
| *_shared->output_mixer(), |
| *_shared->transmit_mixer(), |
| *_shared->process_thread(), |
| *_shared->audio_device(), |
| _voiceEngineObserverPtr, |
| &_callbackCritSect) != 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; |
| } |
| |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "CreateChannel() => %d", channel_owner->channel()->ChannelId()); |
| return channel_owner->channel()->ChannelId(); |
| } |
| |
| int VoEBaseImpl::DeleteChannel(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "DeleteChannel(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _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::StartReceive(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "StartReceive(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StartReceive() failed to locate channel"); |
| return -1; |
| } |
| return channelPtr->StartReceiving(); |
| } |
| |
| int VoEBaseImpl::StopReceive(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "StopListen(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetLocalReceiver() failed to locate channel"); |
| return -1; |
| } |
| return channelPtr->StopReceiving(); |
| } |
| |
| int VoEBaseImpl::StartPlayout(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "StartPlayout(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _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) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "StopPlayout(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StopPlayout() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->StopPlayout() != 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "StopPlayout() failed to stop playout for channel %d", channel); |
| } |
| return StopPlayout(); |
| } |
| |
| int VoEBaseImpl::StartSend(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "StartSend(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _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) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "StopSend(channel=%d)", channel); |
| CriticalSectionScoped 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 == NULL) |
| { |
| _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StopSend() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->StopSend() != 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "StopSend() failed to stop sending for channel %d", channel); |
| } |
| return StopSend(); |
| } |
| |
| int VoEBaseImpl::GetVersion(char version[1024]) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "GetVersion(version=?)"); |
| assert(kVoiceEngineVersionMaxMessageSize == 1024); |
| |
| if (version == NULL) |
| { |
| _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError); |
| return (-1); |
| } |
| |
| char versionBuf[kVoiceEngineVersionMaxMessageSize]; |
| char* versionPtr = versionBuf; |
| |
| int32_t len = 0; |
| int32_t accLen = 0; |
| |
| len = AddVoEVersion(versionPtr); |
| if (len == -1) |
| { |
| return -1; |
| } |
| versionPtr += len; |
| accLen += len; |
| assert(accLen < kVoiceEngineVersionMaxMessageSize); |
| |
| len = AddBuildInfo(versionPtr); |
| if (len == -1) |
| { |
| return -1; |
| } |
| versionPtr += len; |
| accLen += len; |
| assert(accLen < kVoiceEngineVersionMaxMessageSize); |
| |
| #ifdef WEBRTC_EXTERNAL_TRANSPORT |
| len = AddExternalTransportBuild(versionPtr); |
| if (len == -1) |
| { |
| return -1; |
| } |
| versionPtr += len; |
| accLen += len; |
| assert(accLen < kVoiceEngineVersionMaxMessageSize); |
| #endif |
| |
| memcpy(version, versionBuf, accLen); |
| version[accLen] = '\0'; |
| |
| // to avoid the truncation in the trace, split the string into parts |
| char partOfVersion[256]; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), "GetVersion() =>"); |
| for (int partStart = 0; partStart < accLen;) |
| { |
| memset(partOfVersion, 0, sizeof(partOfVersion)); |
| int partEnd = partStart + 180; |
| while (version[partEnd] != '\n' && version[partEnd] != '\0') |
| { |
| partEnd--; |
| } |
| if (partEnd < accLen) |
| { |
| memcpy(partOfVersion, &version[partStart], partEnd - partStart); |
| } |
| else |
| { |
| memcpy(partOfVersion, &version[partStart], accLen - partStart); |
| } |
| partStart = partEnd; |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), "%s", partOfVersion); |
| } |
| |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::AddBuildInfo(char* str) const |
| { |
| return sprintf(str, "Build: %s\n", BUILDINFO); |
| } |
| |
| int32_t VoEBaseImpl::AddVoEVersion(char* str) const |
| { |
| return sprintf(str, "VoiceEngine 4.1.0\n"); |
| } |
| |
| #ifdef WEBRTC_EXTERNAL_TRANSPORT |
| int32_t VoEBaseImpl::AddExternalTransportBuild(char* str) const |
| { |
| return sprintf(str, "External transport build\n"); |
| } |
| #endif |
| |
| int VoEBaseImpl::LastError() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "LastError()"); |
| return (_shared->statistics().LastError()); |
| } |
| |
| int32_t VoEBaseImpl::StartPlayout() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::StartPlayout()"); |
| if (_shared->audio_device()->Playing()) |
| { |
| return 0; |
| } |
| if (!_shared->ext_playout()) |
| { |
| if (_shared->audio_device()->InitPlayout() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "StartPlayout() failed to initialize playout"); |
| return -1; |
| } |
| if (_shared->audio_device()->StartPlayout() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "StartPlayout() failed to start playout"); |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::StopPlayout() { |
| WEBRTC_TRACE(kTraceInfo, |
| kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "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() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::StartSend()"); |
| if (_shared->audio_device()->Recording()) |
| { |
| return 0; |
| } |
| if (!_shared->ext_recording()) |
| { |
| if (_shared->audio_device()->InitRecording() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "StartSend() failed to initialize recording"); |
| return -1; |
| } |
| if (_shared->audio_device()->StartRecording() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, |
| VoEId(_shared->instance_id(), -1), |
| "StartSend() failed to start recording"); |
| return -1; |
| } |
| } |
| |
| return 0; |
| } |
| |
| int32_t VoEBaseImpl::StopSend() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::StopSend()"); |
| |
| if (_shared->NumOfSendingChannels() == 0 && |
| !_shared->transmit_mixer()->IsRecordingMic()) |
| { |
| // 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() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_shared->instance_id(), -1), |
| "VoEBaseImpl::TerminateInternal()"); |
| |
| // Delete any remaining channel objects |
| _shared->channel_manager().DestroyAllChannels(); |
| |
| if (_shared->process_thread()) |
| { |
| if (_shared->audio_device()) |
| { |
| if (_shared->process_thread()-> |
| DeRegisterModule(_shared->audio_device()) != 0) |
| { |
| _shared->SetLastError(VE_THREAD_ERROR, kTraceError, |
| "TerminateInternal() failed to deregister ADM"); |
| } |
| } |
| if (_shared->process_thread()->Stop() != 0) |
| { |
| _shared->SetLastError(VE_THREAD_ERROR, kTraceError, |
| "TerminateInternal() failed to stop module process thread"); |
| } |
| } |
| |
| 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(NULL) != 0) { |
| _shared->SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning, |
| "TerminateInternal() failed to de-register event observer " |
| "for the ADM"); |
| } |
| if (_shared->audio_device()->RegisterAudioCallback(NULL) != 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(NULL); |
| } |
| |
| if (_shared->audio_processing()) { |
| _shared->set_audio_processing(NULL); |
| } |
| |
| return _shared->statistics().SetUnInitialized(); |
| } |
| |
| int VoEBaseImpl::ProcessRecordedDataWithAPM( |
| const int voe_channels[], |
| int number_of_voe_channels, |
| const void* audio_data, |
| uint32_t sample_rate, |
| uint8_t number_of_channels, |
| uint32_t number_of_frames, |
| uint32_t audio_delay_milliseconds, |
| int32_t clock_drift, |
| uint32_t volume, |
| bool key_pressed) { |
| assert(_shared->transmit_mixer() != NULL); |
| assert(_shared->audio_device() != NULL); |
| |
| 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. When |number_of_voe_channels| == 0, |
| // do the operations on all the existing VoE channels; otherwise the |
| // operations will be done on specific channels. |
| if (number_of_voe_channels == 0) { |
| _shared->transmit_mixer()->DemuxAndMix(); |
| _shared->transmit_mixer()->EncodeAndSend(); |
| } else { |
| _shared->transmit_mixer()->DemuxAndMix(voe_channels, |
| number_of_voe_channels); |
| _shared->transmit_mixer()->EncodeAndSend(voe_channels, |
| number_of_voe_channels); |
| } |
| |
| // 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 to indicate no change on the volume. |
| return 0; |
| } |
| |
| void VoEBaseImpl::GetPlayoutData(int sample_rate, int number_of_channels, |
| int number_of_frames, bool feed_data_to_apm, |
| void* audio_data, |
| int64_t* elapsed_time_ms, |
| int64_t* ntp_time_ms) { |
| assert(_shared->output_mixer() != NULL); |
| |
| // TODO(andrew): if the device is running in mono, we should tell the mixer |
| // here so that it will only request mono from AudioCodingModule. |
| // Perform mixing of all active participants (channel-based mixing) |
| _shared->output_mixer()->MixActiveChannels(); |
| |
| // Additional operations on the combined signal |
| _shared->output_mixer()->DoOperationsOnCombinedSignal(feed_data_to_apm); |
| |
| // Retrieve the final output mix (resampled to match the ADM) |
| _shared->output_mixer()->GetMixedAudio(sample_rate, number_of_channels, |
| &_audioFrame); |
| |
| assert(number_of_frames == _audioFrame.samples_per_channel_); |
| assert(sample_rate == _audioFrame.sample_rate_hz_); |
| |
| // Deliver audio (PCM) samples to the ADM |
| memcpy(audio_data, _audioFrame.data_, |
| sizeof(int16_t) * number_of_frames * number_of_channels); |
| |
| *elapsed_time_ms = _audioFrame.elapsed_time_ms_; |
| *ntp_time_ms = _audioFrame.ntp_time_ms_; |
| } |
| |
| } // namespace webrtc |
