| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ |
| |
| #include <AudioUnit/AudioUnit.h> |
| |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/modules/audio_device/audio_device_generic.h" |
| |
| namespace webrtc { |
| |
| class FineAudioBuffer; |
| |
| // Implements full duplex 16-bit mono PCM audio support for iOS using a |
| // Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit |
| // supports audio echo cancellation. It also adds automatic gain control, |
| // adjustment of voice-processing quality and muting. |
| // |
| // An instance must be created and destroyed on one and the same thread. |
| // All supported public methods must also be called on the same thread. |
| // A thread checker will RTC_DCHECK if any supported method is called on an |
| // invalid thread. |
| // |
| // Recorded audio will be delivered on a real-time internal I/O thread in the |
| // audio unit. The audio unit will also ask for audio data to play out on this |
| // same thread. |
| class AudioDeviceIOS : public AudioDeviceGeneric { |
| public: |
| AudioDeviceIOS(); |
| ~AudioDeviceIOS(); |
| |
| void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; |
| |
| int32_t Init() override; |
| int32_t Terminate() override; |
| bool Initialized() const override { return initialized_; } |
| |
| int32_t InitPlayout() override; |
| bool PlayoutIsInitialized() const override { return play_is_initialized_; } |
| |
| int32_t InitRecording() override; |
| bool RecordingIsInitialized() const override { return rec_is_initialized_; } |
| |
| int32_t StartPlayout() override; |
| int32_t StopPlayout() override; |
| bool Playing() const override { return playing_; } |
| |
| int32_t StartRecording() override; |
| int32_t StopRecording() override; |
| bool Recording() const override { return recording_; } |
| |
| int32_t SetLoudspeakerStatus(bool enable) override; |
| int32_t GetLoudspeakerStatus(bool& enabled) const override; |
| |
| // These methods returns hard-coded delay values and not dynamic delay |
| // estimates. The reason is that iOS supports a built-in AEC and the WebRTC |
| // AEC will always be disabled in the Libjingle layer to avoid running two |
| // AEC implementations at the same time. And, it saves resources to avoid |
| // updating these delay values continuously. |
| // TODO(henrika): it would be possible to mark these two methods as not |
| // implemented since they are only called for A/V-sync purposes today and |
| // A/V-sync is not supported on iOS. However, we avoid adding error messages |
| // the log by using these dummy implementations instead. |
| int32_t PlayoutDelay(uint16_t& delayMS) const override; |
| int32_t RecordingDelay(uint16_t& delayMS) const override; |
| |
| // Native audio parameters stored during construction. |
| // These methods are unique for the iOS implementation. |
| int GetPlayoutAudioParameters(AudioParameters* params) const override; |
| int GetRecordAudioParameters(AudioParameters* params) const override; |
| |
| // These methods are currently not fully implemented on iOS: |
| |
| // See audio_device_not_implemented.cc for trivial implementations. |
| int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, |
| uint16_t& sizeMS) const override; |
| int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const; |
| int32_t ResetAudioDevice() override; |
| int32_t PlayoutIsAvailable(bool& available) override; |
| int32_t RecordingIsAvailable(bool& available) override; |
| int32_t SetAGC(bool enable) override; |
| bool AGC() const override; |
| int16_t PlayoutDevices() override; |
| int16_t RecordingDevices() override; |
| int32_t PlayoutDeviceName(uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]) override; |
| int32_t RecordingDeviceName(uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]) override; |
| int32_t SetPlayoutDevice(uint16_t index) override; |
| int32_t SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType device) override; |
| int32_t SetRecordingDevice(uint16_t index) override; |
| int32_t SetRecordingDevice( |
| AudioDeviceModule::WindowsDeviceType device) override; |
| int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight) override; |
| int32_t WaveOutVolume(uint16_t& volumeLeft, |
| uint16_t& volumeRight) const override; |
| int32_t InitSpeaker() override; |
| bool SpeakerIsInitialized() const override; |
| int32_t InitMicrophone() override; |
| bool MicrophoneIsInitialized() const override; |
| int32_t SpeakerVolumeIsAvailable(bool& available) override; |
| int32_t SetSpeakerVolume(uint32_t volume) override; |
| int32_t SpeakerVolume(uint32_t& volume) const override; |
| int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override; |
| int32_t MinSpeakerVolume(uint32_t& minVolume) const override; |
| int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const override; |
| int32_t MicrophoneVolumeIsAvailable(bool& available) override; |
| int32_t SetMicrophoneVolume(uint32_t volume) override; |
| int32_t MicrophoneVolume(uint32_t& volume) const override; |
| int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override; |
| int32_t MinMicrophoneVolume(uint32_t& minVolume) const override; |
| int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const override; |
| int32_t MicrophoneMuteIsAvailable(bool& available) override; |
| int32_t SetMicrophoneMute(bool enable) override; |
| int32_t MicrophoneMute(bool& enabled) const override; |
| int32_t SpeakerMuteIsAvailable(bool& available) override; |
| int32_t SetSpeakerMute(bool enable) override; |
| int32_t SpeakerMute(bool& enabled) const override; |
| int32_t MicrophoneBoostIsAvailable(bool& available) override; |
| int32_t SetMicrophoneBoost(bool enable) override; |
| int32_t MicrophoneBoost(bool& enabled) const override; |
| int32_t StereoPlayoutIsAvailable(bool& available) override; |
| int32_t SetStereoPlayout(bool enable) override; |
| int32_t StereoPlayout(bool& enabled) const override; |
| int32_t StereoRecordingIsAvailable(bool& available) override; |
| int32_t SetStereoRecording(bool enable) override; |
| int32_t StereoRecording(bool& enabled) const override; |
| int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type, |
| uint16_t sizeMS) override; |
| int32_t CPULoad(uint16_t& load) const override; |
| bool PlayoutWarning() const override; |
| bool PlayoutError() const override; |
| bool RecordingWarning() const override; |
| bool RecordingError() const override; |
| void ClearPlayoutWarning() override {} |
| void ClearPlayoutError() override {} |
| void ClearRecordingWarning() override {} |
| void ClearRecordingError() override {} |
| |
| private: |
| // Uses current |playout_parameters_| and |record_parameters_| to inform the |
| // audio device buffer (ADB) about our internal audio parameters. |
| void UpdateAudioDeviceBuffer(); |
| |
| // Registers observers for the AVAudioSessionRouteChangeNotification and |
| // AVAudioSessionInterruptionNotification notifications. |
| void RegisterNotificationObservers(); |
| void UnregisterNotificationObservers(); |
| |
| // Since the preferred audio parameters are only hints to the OS, the actual |
| // values may be different once the AVAudioSession has been activated. |
| // This method asks for the current hardware parameters and takes actions |
| // if they should differ from what we have asked for initially. It also |
| // defines |playout_parameters_| and |record_parameters_|. |
| void SetupAudioBuffersForActiveAudioSession(); |
| |
| // Creates a Voice-Processing I/O unit and configures it for full-duplex |
| // audio. The selected stream format is selected to avoid internal resampling |
| // and to match the 10ms callback rate for WebRTC as well as possible. |
| // This method also initializes the created audio unit. |
| bool SetupAndInitializeVoiceProcessingAudioUnit(); |
| |
| // Restarts active audio streams using a new sample rate. Required when e.g. |
| // a BT headset is enabled or disabled. |
| bool RestartAudioUnitWithNewFormat(float sample_rate); |
| |
| // Activates our audio session, creates and initializes the voice-processing |
| // audio unit and verifies that we got the preferred native audio parameters. |
| bool InitPlayOrRecord(); |
| |
| // Closes and deletes the voice-processing I/O unit. |
| bool ShutdownPlayOrRecord(); |
| |
| // Callback function called on a real-time priority I/O thread from the audio |
| // unit. This method is used to signal that recorded audio is available. |
| static OSStatus RecordedDataIsAvailable( |
| void* in_ref_con, |
| AudioUnitRenderActionFlags* io_action_flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 in_bus_number, |
| UInt32 in_number_frames, |
| AudioBufferList* io_data); |
| OSStatus OnRecordedDataIsAvailable( |
| AudioUnitRenderActionFlags* io_action_flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 in_bus_number, |
| UInt32 in_number_frames); |
| |
| // Callback function called on a real-time priority I/O thread from the audio |
| // unit. This method is used to provide audio samples to the audio unit. |
| static OSStatus GetPlayoutData(void* in_ref_con, |
| AudioUnitRenderActionFlags* io_action_flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 in_bus_number, |
| UInt32 in_number_frames, |
| AudioBufferList* io_data); |
| OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags, |
| UInt32 in_number_frames, |
| AudioBufferList* io_data); |
| |
| // Ensures that methods are called from the same thread as this object is |
| // created on. |
| rtc::ThreadChecker thread_checker_; |
| |
| // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the |
| // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create(). |
| // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance |
| // and therefore outlives this object. |
| AudioDeviceBuffer* audio_device_buffer_; |
| |
| // Contains audio parameters (sample rate, #channels, buffer size etc.) for |
| // the playout and recording sides. These structure is set in two steps: |
| // first, native sample rate and #channels are defined in Init(). Next, the |
| // audio session is activated and we verify that the preferred parameters |
| // were granted by the OS. At this stage it is also possible to add a third |
| // component to the parameters; the native I/O buffer duration. |
| // A RTC_CHECK will be hit if we for some reason fail to open an audio session |
| // using the specified parameters. |
| AudioParameters playout_parameters_; |
| AudioParameters record_parameters_; |
| |
| // The Voice-Processing I/O unit has the same characteristics as the |
| // Remote I/O unit (supports full duplex low-latency audio input and output) |
| // and adds AEC for for two-way duplex communication. It also adds AGC, |
| // adjustment of voice-processing quality, and muting. Hence, ideal for |
| // VoIP applications. |
| AudioUnit vpio_unit_; |
| |
| // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data |
| // in chunks of 10ms. It then allows for this data to be pulled in |
| // a finer or coarser granularity. I.e. interacting with this class instead |
| // of directly with the AudioDeviceBuffer one can ask for any number of |
| // audio data samples. Is also supports a similar scheme for the recording |
| // side. |
| // Example: native buffer size can be 128 audio frames at 16kHz sample rate. |
| // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128 |
| // in each callback (one every 8ms). This class can then ask for 128 and the |
| // FineAudioBuffer will ask WebRTC for new data only when needed and also |
| // cache non-utilized audio between callbacks. On the recording side, iOS |
| // can provide audio data frames of size 128 and these are accumulated until |
| // enough data to supply one 10ms call exists. This 10ms chunk is then sent |
| // to WebRTC and the remaining part is stored. |
| rtc::scoped_ptr<FineAudioBuffer> fine_audio_buffer_; |
| |
| // Extra audio buffer to be used by the playout side for rendering audio. |
| // The buffer size is given by FineAudioBuffer::RequiredBufferSizeBytes(). |
| rtc::scoped_ptr<SInt8[]> playout_audio_buffer_; |
| |
| // Provides a mechanism for encapsulating one or more buffers of audio data. |
| // Only used on the recording side. |
| AudioBufferList audio_record_buffer_list_; |
| |
| // Temporary storage for recorded data. AudioUnitRender() renders into this |
| // array as soon as a frame of the desired buffer size has been recorded. |
| rtc::scoped_ptr<SInt8[]> record_audio_buffer_; |
| |
| // Set to 1 when recording is active and 0 otherwise. |
| volatile int recording_; |
| |
| // Set to 1 when playout is active and 0 otherwise. |
| volatile int playing_; |
| |
| // Set to true after successful call to Init(), false otherwise. |
| bool initialized_; |
| |
| // Set to true after successful call to InitRecording(), false otherwise. |
| bool rec_is_initialized_; |
| |
| // Set to true after successful call to InitPlayout(), false otherwise. |
| bool play_is_initialized_; |
| |
| // Audio interruption observer instance. |
| void* audio_interruption_observer_; |
| void* route_change_observer_; |
| |
| // Contains the audio data format specification for a stream of audio. |
| AudioStreamBasicDescription application_format_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_ |