| /* |
| * Copyright (c) 2011 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. |
| */ |
| |
| #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H |
| |
| #include "audio_device.h" |
| #include "audio_device_buffer.h" |
| |
| namespace webrtc |
| { |
| |
| class AudioDeviceGeneric; |
| class AudioDeviceUtility; |
| class CriticalSectionWrapper; |
| |
| class AudioDeviceModuleImpl : public AudioDeviceModule |
| { |
| public: |
| enum PlatformType |
| { |
| kPlatformNotSupported = 0, |
| kPlatformWin32 = 1, |
| kPlatformWinCe = 2, |
| kPlatformLinux = 3, |
| kPlatformMac = 4, |
| kPlatformAndroid = 5 |
| }; |
| |
| WebRtc_Word32 CheckPlatform(); |
| WebRtc_Word32 CreatePlatformSpecificObjects(); |
| WebRtc_Word32 AttachAudioBuffer(); |
| |
| AudioDeviceModuleImpl(const WebRtc_Word32 id, const AudioLayer audioLayer); |
| virtual ~AudioDeviceModuleImpl(); |
| |
| public: // RefCountedModule |
| virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id); |
| virtual WebRtc_Word32 TimeUntilNextProcess(); |
| virtual WebRtc_Word32 Process(); |
| |
| public: |
| // Factory methods (resource allocation/deallocation) |
| static AudioDeviceModule* Create( |
| const WebRtc_Word32 id, |
| const AudioLayer audioLayer = kPlatformDefaultAudio); |
| |
| // Retrieve the currently utilized audio layer |
| virtual WebRtc_Word32 ActiveAudioLayer(AudioLayer* audioLayer) const; |
| |
| // Error handling |
| virtual ErrorCode LastError() const; |
| virtual WebRtc_Word32 RegisterEventObserver( |
| AudioDeviceObserver* eventCallback); |
| |
| // Full-duplex transportation of PCM audio |
| virtual WebRtc_Word32 RegisterAudioCallback( |
| AudioTransport* audioCallback); |
| |
| // Main initializaton and termination |
| virtual WebRtc_Word32 Init(); |
| virtual WebRtc_Word32 Terminate(); |
| virtual bool Initialized() const; |
| |
| // Device enumeration |
| virtual WebRtc_Word16 PlayoutDevices(); |
| virtual WebRtc_Word16 RecordingDevices(); |
| virtual WebRtc_Word32 PlayoutDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]); |
| virtual WebRtc_Word32 RecordingDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]); |
| |
| // Device selection |
| virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index); |
| virtual WebRtc_Word32 SetPlayoutDevice(WindowsDeviceType device); |
| virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index); |
| virtual WebRtc_Word32 SetRecordingDevice(WindowsDeviceType device); |
| |
| // Audio transport initialization |
| virtual WebRtc_Word32 PlayoutIsAvailable(bool* available); |
| virtual WebRtc_Word32 InitPlayout(); |
| virtual bool PlayoutIsInitialized() const; |
| virtual WebRtc_Word32 RecordingIsAvailable(bool* available); |
| virtual WebRtc_Word32 InitRecording(); |
| virtual bool RecordingIsInitialized() const; |
| |
| // Audio transport control |
| virtual WebRtc_Word32 StartPlayout(); |
| virtual WebRtc_Word32 StopPlayout(); |
| virtual bool Playing() const; |
| virtual WebRtc_Word32 StartRecording(); |
| virtual WebRtc_Word32 StopRecording(); |
| virtual bool Recording() const; |
| |
| // Microphone Automatic Gain Control (AGC) |
| virtual WebRtc_Word32 SetAGC(bool enable); |
| virtual bool AGC() const; |
| |
| // Volume control based on the Windows Wave API (Windows only) |
| virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, |
| WebRtc_UWord16 volumeRight); |
| virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16* volumeLeft, |
| WebRtc_UWord16* volumeRight) const; |
| |
| // Audio mixer initialization |
| virtual WebRtc_Word32 SpeakerIsAvailable(bool* available); |
| virtual WebRtc_Word32 InitSpeaker(); |
| virtual bool SpeakerIsInitialized() const; |
| virtual WebRtc_Word32 MicrophoneIsAvailable(bool* available); |
| virtual WebRtc_Word32 InitMicrophone(); |
| virtual bool MicrophoneIsInitialized() const; |
| |
| // Speaker volume controls |
| virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool* available); |
| virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32* volume) const; |
| virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32* maxVolume) const; |
| virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32* minVolume) const; |
| virtual WebRtc_Word32 SpeakerVolumeStepSize( |
| WebRtc_UWord16* stepSize) const; |
| |
| // Microphone volume controls |
| virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool* available); |
| virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32* volume) const; |
| virtual WebRtc_Word32 MaxMicrophoneVolume( |
| WebRtc_UWord32* maxVolume) const; |
| virtual WebRtc_Word32 MinMicrophoneVolume( |
| WebRtc_UWord32* minVolume) const; |
| virtual WebRtc_Word32 MicrophoneVolumeStepSize( |
| WebRtc_UWord16* stepSize) const; |
| |
| // Speaker mute control |
| virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool* available); |
| virtual WebRtc_Word32 SetSpeakerMute(bool enable); |
| virtual WebRtc_Word32 SpeakerMute(bool* enabled) const; |
| |
| // Microphone mute control |
| virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool* available); |
| virtual WebRtc_Word32 SetMicrophoneMute(bool enable); |
| virtual WebRtc_Word32 MicrophoneMute(bool* enabled) const; |
| |
| // Microphone boost control |
| virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool* available); |
| virtual WebRtc_Word32 SetMicrophoneBoost(bool enable); |
| virtual WebRtc_Word32 MicrophoneBoost(bool* enabled) const; |
| |
| // Stereo support |
| virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool* available) const; |
| virtual WebRtc_Word32 SetStereoPlayout(bool enable); |
| virtual WebRtc_Word32 StereoPlayout(bool* enabled) const; |
| virtual WebRtc_Word32 StereoRecordingIsAvailable(bool* available) const; |
| virtual WebRtc_Word32 SetStereoRecording(bool enable); |
| virtual WebRtc_Word32 StereoRecording(bool* enabled) const; |
| virtual WebRtc_Word32 SetRecordingChannel(const ChannelType channel); |
| virtual WebRtc_Word32 RecordingChannel(ChannelType* channel) const; |
| |
| // Delay information and control |
| virtual WebRtc_Word32 SetPlayoutBuffer(const BufferType type, |
| WebRtc_UWord16 sizeMS = 0); |
| virtual WebRtc_Word32 PlayoutBuffer(BufferType* type, |
| WebRtc_UWord16* sizeMS) const; |
| virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16* delayMS) const; |
| virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16* delayMS) const; |
| |
| // CPU load |
| virtual WebRtc_Word32 CPULoad(WebRtc_UWord16* load) const; |
| |
| // Recording of raw PCM data |
| virtual WebRtc_Word32 StartRawOutputFileRecording( |
| const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize]); |
| virtual WebRtc_Word32 StopRawOutputFileRecording(); |
| virtual WebRtc_Word32 StartRawInputFileRecording( |
| const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize]); |
| virtual WebRtc_Word32 StopRawInputFileRecording(); |
| |
| // Native sample rate controls (samples/sec) |
| virtual WebRtc_Word32 SetRecordingSampleRate( |
| const WebRtc_UWord32 samplesPerSec); |
| virtual WebRtc_Word32 RecordingSampleRate( |
| WebRtc_UWord32* samplesPerSec) const; |
| virtual WebRtc_Word32 SetPlayoutSampleRate( |
| const WebRtc_UWord32 samplesPerSec); |
| virtual WebRtc_Word32 PlayoutSampleRate( |
| WebRtc_UWord32* samplesPerSec) const; |
| |
| // Mobile device specific functions |
| virtual WebRtc_Word32 ResetAudioDevice(); |
| virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable); |
| virtual WebRtc_Word32 GetLoudspeakerStatus(bool* enabled) const; |
| |
| virtual int32_t EnableBuiltInAEC(bool enable); |
| virtual bool BuiltInAECIsEnabled() const; |
| |
| public: |
| WebRtc_Word32 Id() {return _id;} |
| |
| private: |
| PlatformType Platform() const; |
| AudioLayer PlatformAudioLayer() const; |
| |
| private: |
| CriticalSectionWrapper& _critSect; |
| CriticalSectionWrapper& _critSectEventCb; |
| CriticalSectionWrapper& _critSectAudioCb; |
| |
| AudioDeviceObserver* _ptrCbAudioDeviceObserver; |
| |
| AudioDeviceUtility* _ptrAudioDeviceUtility; |
| AudioDeviceGeneric* _ptrAudioDevice; |
| |
| AudioDeviceBuffer _audioDeviceBuffer; |
| |
| WebRtc_Word32 _id; |
| AudioLayer _platformAudioLayer; |
| WebRtc_UWord32 _lastProcessTime; |
| PlatformType _platformType; |
| bool _initialized; |
| mutable ErrorCode _lastError; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_INTERFACE_AUDIO_DEVICE_IMPL_H_ |
