| /* |
| * 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 "voe_base_impl.h" |
| |
| #include "audio_coding_module.h" |
| #include "audio_device_impl.h" |
| #include "audio_processing.h" |
| #include "channel.h" |
| #include "critical_section_wrapper.h" |
| #include "file_wrapper.h" |
| #include "output_mixer.h" |
| #include "signal_processing_library.h" |
| #include "trace.h" |
| #include "transmit_mixer.h" |
| #include "utility.h" |
| #include "voe_errors.h" |
| #include "voice_engine_impl.h" |
| |
| #if (defined(_WIN32) && defined(_DLL) && (_MSC_VER == 1400)) |
| // Fix for VS 2005 MD/MDd link problem |
| #include <stdio.h> |
| extern "C" |
| { FILE _iob[3] = { __iob_func()[0], __iob_func()[1], __iob_func()[2]}; } |
| #endif |
| |
| namespace webrtc |
| { |
| |
| VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine) |
| { |
| if (NULL == voiceEngine) |
| { |
| return NULL; |
| } |
| VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine); |
| VoEBaseImpl* d = s; |
| (*d)++; |
| return (d); |
| } |
| |
| VoEBaseImpl::VoEBaseImpl() : |
| _voiceEngineObserverPtr(NULL), |
| _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()), |
| _voiceEngineObserver(false), _oldVoEMicLevel(0), _oldMicLevel(0) |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl() - ctor"); |
| } |
| |
| VoEBaseImpl::~VoEBaseImpl() |
| { |
| WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1), |
| "~VoEBaseImpl() - dtor"); |
| |
| TerminateInternal(); |
| |
| delete &_callbackCritSect; |
| } |
| |
| int VoEBaseImpl::Release() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::Release()"); |
| (*this)--; |
| int refCount = GetCount(); |
| if (refCount < 0) |
| { |
| Reset(); |
| _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND, kTraceWarning); |
| return (-1); |
| } |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl reference counter = %d", refCount); |
| return (refCount); |
| } |
| |
| void VoEBaseImpl::OnErrorIsReported(const 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(_instanceId, -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_REC_ERROR"); |
| } |
| else if (error == AudioDeviceObserver::kPlayoutError) |
| { |
| errCode = VE_RUNTIME_PLAY_ERROR; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_PLAY_ERROR"); |
| } |
| // Deliver callback (-1 <=> no channel dependency) |
| _voiceEngineObserverPtr->CallbackOnError(-1, errCode); |
| } |
| } |
| } |
| |
| void VoEBaseImpl::OnWarningIsReported(const 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(_instanceId, -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_REC_WARNING"); |
| } |
| else if (warning == AudioDeviceObserver::kPlayoutWarning) |
| { |
| warningCode = VE_RUNTIME_PLAY_WARNING; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::OnErrorIsReported() => " |
| "VE_RUNTIME_PLAY_WARNING"); |
| } |
| // Deliver callback (-1 <=> no channel dependency) |
| _voiceEngineObserverPtr->CallbackOnError(-1, warningCode); |
| } |
| } |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::RecordedDataIsAvailable( |
| const WebRtc_Word8* audioSamples, |
| const WebRtc_UWord32 nSamples, |
| const WebRtc_UWord8 nBytesPerSample, |
| const WebRtc_UWord8 nChannels, |
| const WebRtc_UWord32 samplesPerSec, |
| const WebRtc_UWord32 totalDelayMS, |
| const WebRtc_Word32 clockDrift, |
| const WebRtc_UWord32 currentMicLevel, |
| WebRtc_UWord32& newMicLevel) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, " |
| "nBytesPerSample=%u, nChannels=%u, samplesPerSec=%u, " |
| "totalDelayMS=%u, clockDrift=%d, currentMicLevel=%u)", |
| nSamples, nBytesPerSample, nChannels, samplesPerSec, |
| totalDelayMS, clockDrift, currentMicLevel); |
| |
| assert(_transmitMixerPtr != NULL); |
| assert(_audioDevicePtr != NULL); |
| |
| bool isAnalogAGC(false); |
| WebRtc_UWord32 maxVolume(0); |
| WebRtc_UWord16 currentVoEMicLevel(0); |
| WebRtc_UWord32 newVoEMicLevel(0); |
| |
| if (_audioProcessingModulePtr |
| && (_audioProcessingModulePtr->gain_control()->mode() |
| == GainControl::kAdaptiveAnalog)) |
| { |
| isAnalogAGC = true; |
| } |
| |
| // Will only deal with the volume in adaptive analog mode |
| if (isAnalogAGC) |
| { |
| // Scale from ADM to VoE level range |
| if (_audioDevicePtr->MaxMicrophoneVolume(&maxVolume) == 0) |
| { |
| if (0 != maxVolume) |
| { |
| currentVoEMicLevel = (WebRtc_UWord16) ((currentMicLevel |
| * kMaxVolumeLevel + (int) (maxVolume / 2)) |
| / (maxVolume)); |
| } |
| } |
| // We learned that on certain systems (e.g Linux) the currentVoEMicLevel |
| // can be greater than the maxVolumeLevel therefore |
| // we are going to cap the currentVoEMicLevel to the maxVolumeLevel |
| // and change the maxVolume to currentMicLevel if it turns out that |
| // the currentVoEMicLevel is indeed greater than the maxVolumeLevel. |
| if (currentVoEMicLevel > kMaxVolumeLevel) |
| { |
| currentVoEMicLevel = kMaxVolumeLevel; |
| maxVolume = currentMicLevel; |
| } |
| } |
| |
| // Keep track if the MicLevel has been changed by the AGC, if not, |
| // use the old value AGC returns to let AGC continue its trend, |
| // so eventually the AGC is able to change the mic level. This handles |
| // issues with truncation introduced by the scaling. |
| if (_oldMicLevel == currentMicLevel) |
| { |
| currentVoEMicLevel = (WebRtc_UWord16) _oldVoEMicLevel; |
| } |
| |
| // Perform channel-independent operations |
| // (APM, mix with file, record to file, mute, etc.) |
| _transmitMixerPtr->PrepareDemux(audioSamples, nSamples, nChannels, |
| samplesPerSec, |
| (WebRtc_UWord16) totalDelayMS, clockDrift, |
| currentVoEMicLevel); |
| |
| // Copy the audio frame to each sending channel and perform |
| // channel-dependent operations (file mixing, mute, etc.) to prepare |
| // for encoding. |
| _transmitMixerPtr->DemuxAndMix(); |
| // Do the encoding and packetize+transmit the RTP packet when encoding |
| // is done. |
| _transmitMixerPtr->EncodeAndSend(); |
| |
| // Will only deal with the volume in adaptive analog mode |
| if (isAnalogAGC) |
| { |
| // Scale from VoE to ADM level range |
| newVoEMicLevel = _transmitMixerPtr->CaptureLevel(); |
| if (newVoEMicLevel != currentVoEMicLevel) |
| { |
| // Add (kMaxVolumeLevel/2) to round the value |
| newMicLevel = (WebRtc_UWord32) ((newVoEMicLevel * maxVolume |
| + (int) (kMaxVolumeLevel / 2)) / (kMaxVolumeLevel)); |
| } |
| else |
| { |
| // Pass zero if the level is unchanged |
| newMicLevel = 0; |
| } |
| |
| // Keep track of the value AGC returns |
| _oldVoEMicLevel = newVoEMicLevel; |
| _oldMicLevel = currentMicLevel; |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::NeedMorePlayData( |
| const WebRtc_UWord32 nSamples, |
| const WebRtc_UWord8 nBytesPerSample, |
| const WebRtc_UWord8 nChannels, |
| const WebRtc_UWord32 samplesPerSec, |
| WebRtc_Word8* audioSamples, |
| WebRtc_UWord32& nSamplesOut) |
| { |
| WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::NeedMorePlayData(nSamples=%u, " |
| "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)", |
| nSamples, nBytesPerSample, nChannels, samplesPerSec); |
| |
| assert(_outputMixerPtr != NULL); |
| |
| // Perform mixing of all active participants (channel-based mixing) |
| _outputMixerPtr->MixActiveChannels(); |
| |
| // Additional operations on the combined signal |
| _outputMixerPtr->DoOperationsOnCombinedSignal(); |
| |
| // Retrieve the final output mix (resampled to match the ADM) |
| _outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, _audioFrame); |
| |
| assert(nSamples == _audioFrame._payloadDataLengthInSamples); |
| assert(samplesPerSec == |
| static_cast<WebRtc_UWord32>(_audioFrame._frequencyInHz)); |
| |
| // Deliver audio (PCM) samples to the ADM |
| memcpy( |
| (WebRtc_Word16*) audioSamples, |
| (const WebRtc_Word16*) _audioFrame._payloadData, |
| sizeof(WebRtc_Word16) * (_audioFrame._payloadDataLengthInSamples |
| * _audioFrame._audioChannel)); |
| |
| nSamplesOut = _audioFrame._payloadDataLengthInSamples; |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "RegisterVoiceEngineObserver(observer=0x%d)", &observer); |
| CriticalSectionScoped cs(_callbackCritSect); |
| if (_voiceEngineObserverPtr) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError, |
| "RegisterVoiceEngineObserver() observer" |
| " already enabled"); |
| return -1; |
| } |
| |
| // Register the observer in all active channels |
| voe::ScopedChannel sc(_channelManager); |
| void* iterator(NULL); |
| voe::Channel* channelPtr = sc.GetFirstChannel(iterator); |
| while (channelPtr != NULL) |
| { |
| channelPtr->RegisterVoiceEngineObserver(observer); |
| channelPtr = sc.GetNextChannel(iterator); |
| } |
| _transmitMixerPtr->RegisterVoiceEngineObserver(observer); |
| |
| _voiceEngineObserverPtr = &observer; |
| _voiceEngineObserver = true; |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::DeRegisterVoiceEngineObserver() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "DeRegisterVoiceEngineObserver()"); |
| CriticalSectionScoped cs(_callbackCritSect); |
| if (!_voiceEngineObserverPtr) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError, |
| "DeRegisterVoiceEngineObserver() observer already disabled"); |
| return 0; |
| } |
| |
| _voiceEngineObserver = false; |
| _voiceEngineObserverPtr = NULL; |
| |
| // Deregister the observer in all active channels |
| voe::ScopedChannel sc(_channelManager); |
| void* iterator(NULL); |
| voe::Channel* channelPtr = sc.GetFirstChannel(iterator); |
| while (channelPtr != NULL) |
| { |
| channelPtr->DeRegisterVoiceEngineObserver(); |
| channelPtr = sc.GetNextChannel(iterator); |
| } |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::Init(AudioDeviceModule* external_adm) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "Init(external_adm=0x%p)", external_adm); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| |
| if (_engineStatistics.Initialized()) |
| { |
| return 0; |
| } |
| |
| if (_moduleProcessThreadPtr) |
| { |
| if (_moduleProcessThreadPtr->Start() != 0) |
| { |
| _engineStatistics.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. |
| _audioDevicePtr = AudioDeviceModuleImpl::Create( |
| VoEId(_instanceId, -1), _audioDeviceLayer); |
| |
| if (_audioDevicePtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical, |
| "Init() failed to create the ADM"); |
| return -1; |
| } |
| } |
| else |
| { |
| // Use the already existing external ADM implementation. |
| _audioDevicePtr = external_adm; |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "An external ADM implementation will be used in VoiceEngine"); |
| } |
| |
| // Increase the reference counter for both external and internal usage. |
| _audioDevicePtr->AddRef(); |
| |
| // Register the ADM to the process thread, which will drive the error |
| // callback mechanism |
| if (_moduleProcessThreadPtr && |
| _moduleProcessThreadPtr->RegisterModule(_audioDevicePtr) != 0) |
| { |
| _engineStatistics.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 (_audioDevicePtr->RegisterEventObserver(this) != 0) { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceWarning, |
| "Init() failed to register event observer " |
| "for the ADM"); |
| } |
| |
| // Register the AudioTransport implementation |
| if (_audioDevicePtr->RegisterAudioCallback(this) != 0) { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceWarning, |
| "Init() failed to register audio callback " |
| "for the ADM"); |
| } |
| |
| // ADM initialization |
| if (_audioDevicePtr->Init() != 0) |
| { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceError, |
| "Init() failed to initialize the ADM"); |
| return -1; |
| } |
| |
| // Initialize the default speaker |
| if (_audioDevicePtr->SetPlayoutDevice(WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) |
| != 0) |
| { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo, |
| "Init() failed to set the default output device"); |
| } |
| if (_audioDevicePtr->SpeakerIsAvailable(&available) != 0) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo, |
| "Init() failed to check speaker availability, trying to " |
| "initialize speaker anyway"); |
| } |
| else if (!available) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo, |
| "Init() speaker not available, trying to initialize speaker " |
| "anyway"); |
| } |
| if (_audioDevicePtr->InitSpeaker() != 0) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo, |
| "Init() failed to initialize the speaker"); |
| } |
| |
| // Initialize the default microphone |
| if (_audioDevicePtr->SetRecordingDevice(WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE) |
| != 0) |
| { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo, |
| "Init() failed to set the default input device"); |
| } |
| if (_audioDevicePtr->MicrophoneIsAvailable(&available) != 0) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo, |
| "Init() failed to check microphone availability, trying to " |
| "initialize microphone anyway"); |
| } |
| else if (!available) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo, |
| "Init() microphone not available, trying to initialize " |
| "microphone anyway"); |
| } |
| if (_audioDevicePtr->InitMicrophone() != 0) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo, |
| "Init() failed to initialize the microphone"); |
| } |
| |
| // Set number of channels |
| if (_audioDevicePtr->StereoPlayoutIsAvailable(&available) != 0) { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "Init() failed to query stereo playout " |
| "mode"); |
| } |
| if (_audioDevicePtr->SetStereoPlayout(available) != 0) |
| { |
| _engineStatistics.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 |
| _audioDevicePtr->StereoRecordingIsAvailable(&available); |
| if (_audioDevicePtr->SetStereoRecording(available) != 0) |
| { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "Init() failed to set mono/stereo recording mode"); |
| } |
| |
| // APM initialization done after sound card since we need |
| // to know if we support stereo recording or not. |
| |
| // Create the AudioProcessing Module if it does not exist. |
| |
| if (_audioProcessingModulePtr == NULL) |
| { |
| _audioProcessingModulePtr = AudioProcessing::Create( |
| VoEId(_instanceId, -1)); |
| if (_audioProcessingModulePtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical, |
| "Init() failed to create the AP module"); |
| return -1; |
| } |
| // Ensure that mixers in both directions has access to the created APM |
| _transmitMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr); |
| _outputMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr); |
| |
| if (_audioProcessingModulePtr->echo_cancellation()-> |
| set_device_sample_rate_hz( |
| kVoiceEngineAudioProcessingDeviceSampleRateHz)) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set the device sample rate to 48K for AP " |
| " module"); |
| return -1; |
| } |
| // Using 8 kHz as inital Fs. Might be changed already at first call. |
| if (_audioProcessingModulePtr->set_sample_rate_hz(8000)) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set the sample rate to 8K for AP module"); |
| return -1; |
| } |
| |
| // Assume mono output until a send codec is set, and stereo input until |
| // we receive the first captured frame. We set stereo input here to |
| // avoid triggering a possible error in SetSendCodec when a stereo |
| // codec is selected. |
| if (_audioProcessingModulePtr->set_num_channels(2, 1) != 0) |
| { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError, |
| "Init() failed to set channels for the primary audio stream"); |
| return -1; |
| } |
| |
| if (_audioProcessingModulePtr->set_num_reverse_channels(1) != 0) |
| { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError, |
| "Init() failed to set channels for the primary audio stream"); |
| return -1; |
| } |
| // high-pass filter |
| if (_audioProcessingModulePtr->high_pass_filter()->Enable( |
| WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE) != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set the high-pass filter for AP module"); |
| return -1; |
| } |
| // Echo Cancellation |
| if (_audioProcessingModulePtr->echo_cancellation()-> |
| enable_drift_compensation(false) != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set drift compensation for AP module"); |
| return -1; |
| } |
| if (_audioProcessingModulePtr->echo_cancellation()->Enable( |
| WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE)) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set echo cancellation state for AP module"); |
| return -1; |
| } |
| // Noise Reduction |
| if (_audioProcessingModulePtr->noise_suppression()->set_level( |
| (NoiseSuppression::Level) WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE) |
| != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set noise reduction level for AP module"); |
| return -1; |
| } |
| if (_audioProcessingModulePtr->noise_suppression()->Enable( |
| WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE) != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set noise reduction state for AP module"); |
| return -1; |
| } |
| // Automatic Gain control |
| if (_audioProcessingModulePtr->gain_control()->set_analog_level_limits( |
| kMinVolumeLevel,kMaxVolumeLevel) != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set AGC analog level for AP module"); |
| return -1; |
| } |
| if (_audioProcessingModulePtr->gain_control()->set_mode( |
| (GainControl::Mode) WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE) |
| != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set AGC mode for AP module"); |
| return -1; |
| } |
| if (_audioProcessingModulePtr->gain_control()->Enable( |
| WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE) |
| != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set AGC state for AP module"); |
| return -1; |
| } |
| // VAD |
| if (_audioProcessingModulePtr->voice_detection()->Enable( |
| WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE) |
| != 0) |
| { |
| _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError, |
| "Init() failed to set VAD state for AP module"); |
| return -1; |
| } |
| } |
| |
| // Set default AGC mode for the ADM |
| #ifdef WEBRTC_VOICE_ENGINE_AGC |
| bool enable(false); |
| if (_audioProcessingModulePtr->gain_control()->mode() |
| != GainControl::kFixedDigital) |
| { |
| enable = _audioProcessingModulePtr->gain_control()->is_enabled(); |
| // Only set the AGC mode for the ADM when Adaptive AGC mode is selected |
| if (_audioDevicePtr->SetAGC(enable) != 0) |
| { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceError, "Init() failed to set default AGC mode in ADM 0"); |
| } |
| } |
| #endif |
| |
| return _engineStatistics.SetInitialized(); |
| } |
| |
| int VoEBaseImpl::Terminate() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "Terminate()"); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| return TerminateInternal(); |
| } |
| |
| int VoEBaseImpl::MaxNumOfChannels() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "MaxNumOfChannels()"); |
| WebRtc_Word32 maxNumOfChannels = _channelManager.MaxNumOfChannels(); |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "MaxNumOfChannels() => %d", maxNumOfChannels); |
| return (maxNumOfChannels); |
| } |
| |
| int VoEBaseImpl::CreateChannel() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "CreateChannel()"); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| |
| WebRtc_Word32 channelId = -1; |
| |
| if (!_channelManager.CreateChannel(channelId)) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError, |
| "CreateChannel() failed to allocate " |
| "memory for channel"); |
| return -1; |
| } |
| |
| bool destroyChannel(false); |
| { |
| voe::ScopedChannel sc(_channelManager, channelId); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError, |
| "CreateChannel() failed to allocate" |
| " memory for channel"); |
| return -1; |
| } |
| else if (channelPtr->SetEngineInformation(_engineStatistics, |
| *_outputMixerPtr, |
| *_transmitMixerPtr, |
| *_moduleProcessThreadPtr, |
| *_audioDevicePtr, |
| _voiceEngineObserverPtr, |
| &_callbackCritSect) != 0) |
| { |
| destroyChannel = true; |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError, |
| "CreateChannel() failed to " |
| "associate engine and channel." |
| " Destroying channel."); |
| } |
| else if (channelPtr->Init() != 0) |
| { |
| destroyChannel = true; |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError, |
| "CreateChannel() failed to " |
| "initialize channel. Destroying" |
| " channel."); |
| } |
| } |
| if (destroyChannel) |
| { |
| _channelManager.DestroyChannel(channelId); |
| return -1; |
| } |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "CreateChannel() => %d", channelId); |
| return channelId; |
| } |
| |
| int VoEBaseImpl::DeleteChannel(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "DeleteChannel(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| |
| { |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "DeleteChannel() failed to locate " |
| "channel"); |
| return -1; |
| } |
| } |
| |
| if (_channelManager.DestroyChannel(channel) != 0) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "DeleteChannel() failed to destroy " |
| "channel"); |
| return -1; |
| } |
| |
| if (StopSend() != 0) |
| { |
| return -1; |
| } |
| |
| if (StopPlayout() != 0) |
| { |
| return -1; |
| } |
| |
| return 0; |
| } |
| |
| int VoEBaseImpl::SetLocalReceiver(int channel, int port, int RTCPport, |
| const char ipAddr[64], |
| const char multiCastAddr[64]) |
| { |
| // Inititialize local receive sockets (RTP and RTCP). |
| // |
| // The sockets are always first closed and then created again by this |
| // function call. The created sockets are by default also used for |
| // transmission (unless source port is set in SetSendDestination). |
| // |
| // Note that, sockets can also be created automatically if a user calls |
| // SetSendDestination and StartSend without having called SetLocalReceiver |
| // first. The sockets are then created at the first packet transmission. |
| |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (ipAddr == NULL && multiCastAddr == NULL) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d)", |
| channel, port, RTCPport); |
| } |
| else if (ipAddr != NULL && multiCastAddr == NULL) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, " |
| "ipAddr=%s)", channel, port, RTCPport, ipAddr); |
| } |
| else if (ipAddr == NULL && multiCastAddr != NULL) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, " |
| "multiCastAddr=%s)", channel, port, RTCPport, |
| multiCastAddr); |
| } |
| else |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, " |
| "ipAddr=%s, multiCastAddr=%s)", channel, port, |
| RTCPport, ipAddr, multiCastAddr); |
| } |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| if ((port < 0) || (port > 65535)) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError, |
| "SetLocalReceiver() invalid RTP port"); |
| return -1; |
| } |
| if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport |
| != kVoEDefault) && (RTCPport > 65535))) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError, |
| "SetLocalReceiver() invalid RTCP port"); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetLocalReceiver() failed to locate " |
| "channel"); |
| return -1; |
| } |
| |
| // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in |
| // the module, which is the default. |
| WebRtc_UWord16 rtcpPortUW16(0); |
| if (RTCPport != kVoEDefault) |
| { |
| rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport); |
| } |
| |
| return channelPtr->SetLocalReceiver(port, rtcpPortUW16, ipAddr, |
| multiCastAddr); |
| #else |
| _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, |
| kTraceWarning, |
| "SetLocalReceiver() VoE is built for " |
| "external transport"); |
| return -1; |
| #endif |
| } |
| |
| int VoEBaseImpl::GetLocalReceiver(int channel, int& port, int& RTCPport, |
| char ipAddr[64]) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "GetLocalReceiver(channel=%d, ipAddr[]=?)", channel); |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetLocalReceiver() failed to locate " |
| "channel"); |
| return -1; |
| } |
| WebRtc_Word32 ret = channelPtr->GetLocalReceiver(port, RTCPport, ipAddr); |
| if (ipAddr != NULL) |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "GetLocalReceiver() => port=%d, RTCPport=%d, ipAddr=%s", |
| port, RTCPport, ipAddr); |
| } |
| else |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "GetLocalReceiver() => port=%d, RTCPport=%d", port, |
| RTCPport); |
| } |
| return ret; |
| #else |
| _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, |
| kTraceWarning, |
| "SetLocalReceiver() VoE is built for " |
| "external transport"); |
| return -1; |
| #endif |
| } |
| |
| int VoEBaseImpl::SetSendDestination(int channel, int port, const char* ipaddr, |
| int sourcePort, int RTCPport) |
| { |
| WEBRTC_TRACE( |
| kTraceApiCall, |
| kTraceVoice, |
| VoEId(_instanceId, -1), |
| "SetSendDestination(channel=%d, port=%d, ipaddr=%s," |
| "sourcePort=%d, RTCPport=%d)", |
| channel, port, ipaddr, sourcePort, RTCPport); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetSendDestination() failed to locate " |
| "channel"); |
| return -1; |
| } |
| if ((port < 0) || (port > 65535)) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError, |
| "SetSendDestination() invalid RTP port"); |
| return -1; |
| } |
| if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport |
| != kVoEDefault) && (RTCPport > 65535))) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError, |
| "SetSendDestination() invalid RTCP " |
| "port"); |
| return -1; |
| } |
| if (((sourcePort != kVoEDefault) && (sourcePort < 0)) || ((sourcePort |
| != kVoEDefault) && (sourcePort > 65535))) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError, |
| "SetSendDestination() invalid source " |
| "port"); |
| return -1; |
| } |
| |
| // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in the |
| // module, which is the default. |
| WebRtc_UWord16 rtcpPortUW16(0); |
| if (RTCPport != kVoEDefault) |
| { |
| rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport); |
| WEBRTC_TRACE( |
| kTraceInfo, |
| kTraceVoice, |
| VoEId(_instanceId, channel), |
| "SetSendDestination() non default RTCP port %u will be " |
| "utilized", |
| rtcpPortUW16); |
| } |
| |
| return channelPtr->SetSendDestination(port, ipaddr, sourcePort, |
| rtcpPortUW16); |
| #else |
| _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, |
| kTraceWarning, |
| "SetSendDestination() VoE is built for " |
| "external transport"); |
| return -1; |
| #endif |
| } |
| |
| int VoEBaseImpl::GetSendDestination(int channel, int& port, char ipAddr[64], |
| int& sourcePort, int& RTCPport) |
| { |
| WEBRTC_TRACE( |
| kTraceApiCall, |
| kTraceVoice, |
| VoEId(_instanceId, -1), |
| "GetSendDestination(channel=%d, ipAddr[]=?, sourcePort=?," |
| "RTCPport=?)", |
| channel); |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "GetSendDestination() failed to locate " |
| "channel"); |
| return -1; |
| } |
| WebRtc_Word32 ret = channelPtr->GetSendDestination(port, ipAddr, |
| sourcePort, RTCPport); |
| if (ipAddr != NULL) |
| { |
| WEBRTC_TRACE( |
| kTraceStateInfo, |
| kTraceVoice, |
| VoEId(_instanceId, -1), |
| "GetSendDestination() => port=%d, RTCPport=%d, ipAddr=%s, " |
| "sourcePort=%d, RTCPport=%d", |
| port, RTCPport, ipAddr, sourcePort, RTCPport); |
| } |
| else |
| { |
| WEBRTC_TRACE( |
| kTraceStateInfo, |
| kTraceVoice, |
| VoEId(_instanceId, -1), |
| "GetSendDestination() => port=%d, RTCPport=%d, " |
| "sourcePort=%d, RTCPport=%d", |
| port, RTCPport, sourcePort, RTCPport); |
| } |
| return ret; |
| #else |
| _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED, |
| kTraceWarning, |
| "GetSendDestination() VoE is built for " |
| "external transport"); |
| return -1; |
| #endif |
| } |
| |
| int VoEBaseImpl::StartReceive(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "StartReceive(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.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(_instanceId, -1), |
| "StopListen(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.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(_instanceId, -1), |
| "StartPlayout(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StartPlayout() failed to locate " |
| "channel"); |
| return -1; |
| } |
| if (channelPtr->Playing()) |
| { |
| return 0; |
| } |
| if (StartPlayout() != 0) |
| { |
| _engineStatistics.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(_instanceId, -1), |
| "StopPlayout(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StopPlayout() failed to locate " |
| "channel"); |
| return -1; |
| } |
| if (channelPtr->StopPlayout() != 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StopPlayout() failed to stop playout for channel %d", |
| channel); |
| } |
| return StopPlayout(); |
| } |
| |
| int VoEBaseImpl::StartSend(int channel) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "StartSend(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StartSend() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->Sending()) |
| { |
| return 0; |
| } |
| #ifndef WEBRTC_EXTERNAL_TRANSPORT |
| if (!channelPtr->ExternalTransport() |
| && !channelPtr->SendSocketsInitialized()) |
| { |
| _engineStatistics.SetLastError(VE_DESTINATION_NOT_INITED, kTraceError, |
| "StartSend() must set send destination " |
| "first"); |
| return -1; |
| } |
| #endif |
| if (StartSend() != 0) |
| { |
| _engineStatistics.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(_instanceId, -1), |
| "StopSend(channel=%d)", channel); |
| CriticalSectionScoped cs(*_apiCritPtr); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "StopSend() failed to locate channel"); |
| return -1; |
| } |
| if (channelPtr->StopSend() != 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1), |
| "StopSend() failed to stop sending for channel %d", |
| channel); |
| } |
| return StopSend(); |
| } |
| |
| int VoEBaseImpl::GetVersion(char version[1024]) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "GetVersion(version=?)"); |
| assert(kVoiceEngineVersionMaxMessageSize == 1024); |
| |
| if (version == NULL) |
| { |
| _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError); |
| return (-1); |
| } |
| |
| char versionBuf[kVoiceEngineVersionMaxMessageSize]; |
| char* versionPtr = versionBuf; |
| |
| WebRtc_Word32 len = 0; |
| WebRtc_Word32 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 |
| #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT |
| len = AddExternalRecAndPlayoutBuild(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(_instanceId, -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(_instanceId, -1), |
| "%s", partOfVersion); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::AddBuildInfo(char* str) const |
| { |
| return sprintf(str, "Build: %s\n", BUILDINFO); |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::AddVoEVersion(char* str) const |
| { |
| return sprintf(str, "VoiceEngine 4.1.0\n"); |
| } |
| |
| #ifdef WEBRTC_EXTERNAL_TRANSPORT |
| WebRtc_Word32 VoEBaseImpl::AddExternalTransportBuild(char* str) const |
| { |
| return sprintf(str, "External transport build\n"); |
| } |
| #endif |
| |
| #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT |
| WebRtc_Word32 VoEBaseImpl::AddExternalRecAndPlayoutBuild(char* str) const |
| { |
| return sprintf(str, "External recording and playout build\n"); |
| } |
| #endif |
| |
| int VoEBaseImpl::LastError() |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "LastError()"); |
| return (_engineStatistics.LastError()); |
| } |
| |
| |
| int VoEBaseImpl::SetNetEQPlayoutMode(int channel, NetEqModes mode) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetNetEQPlayoutMode(channel=%i, mode=%i)", channel, mode); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetNetEQPlayoutMode() failed to locate" |
| " channel"); |
| return -1; |
| } |
| return channelPtr->SetNetEQPlayoutMode(mode); |
| } |
| |
| int VoEBaseImpl::GetNetEQPlayoutMode(int channel, NetEqModes& mode) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "GetNetEQPlayoutMode(channel=%i, mode=?)", channel); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "GetNetEQPlayoutMode() failed to locate" |
| " channel"); |
| return -1; |
| } |
| return channelPtr->GetNetEQPlayoutMode(mode); |
| } |
| |
| int VoEBaseImpl::SetNetEQBGNMode(int channel, NetEqBgnModes mode) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetNetEQBGNMode(channel=%i, mode=%i)", channel, mode); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetNetEQBGNMode() failed to locate " |
| "channel"); |
| return -1; |
| } |
| return channelPtr->SetNetEQBGNMode(mode); |
| } |
| |
| int VoEBaseImpl::GetNetEQBGNMode(int channel, NetEqBgnModes& mode) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "GetNetEQBGNMode(channel=%i, mode=?)", channel); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "GetNetEQBGNMode() failed to locate " |
| "channel"); |
| return -1; |
| } |
| return channelPtr->GetNetEQBGNMode(mode); |
| } |
| |
| int VoEBaseImpl::SetOnHoldStatus(int channel, bool enable, OnHoldModes mode) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "SetOnHoldStatus(channel=%d, enable=%d, mode=%d)", channel, |
| enable, mode); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "SetOnHoldStatus() failed to locate " |
| "channel"); |
| return -1; |
| } |
| return channelPtr->SetOnHoldStatus(enable, mode); |
| } |
| |
| int VoEBaseImpl::GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode) |
| { |
| WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), |
| "GetOnHoldStatus(channel=%d, enabled=?, mode=?)", channel); |
| if (!_engineStatistics.Initialized()) |
| { |
| _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError); |
| return -1; |
| } |
| voe::ScopedChannel sc(_channelManager, channel); |
| voe::Channel* channelPtr = sc.ChannelPtr(); |
| if (channelPtr == NULL) |
| { |
| _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError, |
| "GetOnHoldStatus() failed to locate " |
| "channel"); |
| return -1; |
| } |
| return channelPtr->GetOnHoldStatus(enabled, mode); |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::StartPlayout() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::StartPlayout()"); |
| if (_audioDevicePtr->Playing()) |
| { |
| return 0; |
| } |
| if (!_externalPlayout) |
| { |
| if (_audioDevicePtr->InitPlayout() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "StartPlayout() failed to initialize playout"); |
| return -1; |
| } |
| if (_audioDevicePtr->StartPlayout() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "StartPlayout() failed to start playout"); |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::StopPlayout() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::StopPlayout()"); |
| |
| WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels(); |
| if (numOfChannels <= 0) |
| { |
| return 0; |
| } |
| |
| WebRtc_UWord16 nChannelsPlaying(0); |
| WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels]; |
| |
| // Get number of playing channels |
| _channelManager.GetChannelIds(channelsArray, numOfChannels); |
| for (int i = 0; i < numOfChannels; i++) |
| { |
| voe::ScopedChannel sc(_channelManager, channelsArray[i]); |
| voe::Channel* chPtr = sc.ChannelPtr(); |
| if (chPtr) |
| { |
| if (chPtr->Playing()) |
| { |
| nChannelsPlaying++; |
| } |
| } |
| } |
| delete[] channelsArray; |
| |
| // Stop audio-device playing if no channel is playing out |
| if (nChannelsPlaying == 0) |
| { |
| if (_audioDevicePtr->StopPlayout() != 0) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError, |
| "StopPlayout() failed to stop " |
| "playout"); |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::StartSend() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::StartSend()"); |
| if (_audioDevicePtr->Recording()) |
| { |
| return 0; |
| } |
| if (!_externalRecording) |
| { |
| if (_audioDevicePtr->InitRecording() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "StartSend() failed to initialize recording"); |
| return -1; |
| } |
| if (_audioDevicePtr->StartRecording() != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1), |
| "StartSend() failed to start recording"); |
| return -1; |
| } |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::StopSend() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::StopSend()"); |
| |
| if ((NumOfSendingChannels() == 0) && !_transmitMixerPtr->IsRecordingMic()) |
| { |
| // Stop audio-device recording if no channel is recording |
| if (_audioDevicePtr->StopRecording() != 0) |
| { |
| _engineStatistics.SetLastError(VE_CANNOT_STOP_RECORDING, |
| kTraceError, |
| "StopSend() failed to stop " |
| "recording"); |
| return -1; |
| } |
| _transmitMixerPtr->StopSend(); |
| } |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 VoEBaseImpl::TerminateInternal() |
| { |
| WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), |
| "VoEBaseImpl::TerminateInternal()"); |
| |
| // Delete any remaining channel objects |
| WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels(); |
| if (numOfChannels > 0) |
| { |
| WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels]; |
| _channelManager.GetChannelIds(channelsArray, numOfChannels); |
| for (int i = 0; i < numOfChannels; i++) |
| { |
| DeleteChannel(channelsArray[i]); |
| } |
| delete[] channelsArray; |
| } |
| |
| if (_moduleProcessThreadPtr) |
| { |
| if (_audioDevicePtr) |
| { |
| if (_moduleProcessThreadPtr->DeRegisterModule(_audioDevicePtr) != 0) |
| { |
| _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError, |
| "TerminateInternal() failed to " |
| "deregister ADM"); |
| } |
| } |
| if (_moduleProcessThreadPtr->Stop() != 0) |
| { |
| _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError, |
| "TerminateInternal() failed to stop " |
| "module process thread"); |
| } |
| } |
| |
| // Audio Device Module |
| |
| if (_audioDevicePtr != NULL) |
| { |
| if (_audioDevicePtr->StopPlayout() != 0) |
| { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "TerminateInternal() failed to stop " |
| "playout"); |
| } |
| if (_audioDevicePtr->StopRecording() != 0) |
| { |
| _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning, |
| "TerminateInternal() failed to stop " |
| "recording"); |
| } |
| if (_audioDevicePtr->RegisterEventObserver(NULL) != 0) { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceWarning, |
| "TerminateInternal() failed to de-" |
| "register event observer for the ADM"); |
| } |
| if (_audioDevicePtr->RegisterAudioCallback(NULL) != 0) { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceWarning, |
| "TerminateInternal() failed to de-" |
| "register audio callback for the ADM"); |
| } |
| if (_audioDevicePtr->Terminate() != 0) |
| { |
| _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, |
| kTraceError, |
| "TerminateInternal() failed to " |
| "terminate the ADM"); |
| } |
| |
| _audioDevicePtr->Release(); |
| _audioDevicePtr = NULL; |
| } |
| |
| // AP module |
| |
| if (_audioProcessingModulePtr != NULL) |
| { |
| _transmitMixerPtr->SetAudioProcessingModule(NULL); |
| AudioProcessing::Destroy(_audioProcessingModulePtr); |
| _audioProcessingModulePtr = NULL; |
| } |
| |
| return _engineStatistics.SetUnInitialized(); |
| } |
| |
| } // namespace webrtc |
