| /* |
| * Copyright 2016 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_VOICE_PROCESSING_AUDIO_UNIT_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_ |
| |
| #include <AudioUnit/AudioUnit.h> |
| |
| namespace webrtc { |
| |
| class VoiceProcessingAudioUnitObserver { |
| public: |
| // 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. |
| virtual OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data) = 0; |
| |
| // 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. |
| virtual OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data) = 0; |
| |
| protected: |
| ~VoiceProcessingAudioUnitObserver() {} |
| }; |
| |
| // Convenience class to abstract away the management of a Voice Processing |
| // I/O Audio Unit. 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. |
| class VoiceProcessingAudioUnit { |
| public: |
| explicit VoiceProcessingAudioUnit(VoiceProcessingAudioUnitObserver* observer); |
| ~VoiceProcessingAudioUnit(); |
| |
| // TODO(tkchin): enum for state and state checking. |
| enum State : int32_t { |
| // Init() should be called. |
| kInitRequired, |
| // Audio unit created but not initialized. |
| kUninitialized, |
| // Initialized but not started. Equivalent to stopped. |
| kInitialized, |
| // Initialized and started. |
| kStarted, |
| }; |
| |
| // Number of bytes per audio sample for 16-bit signed integer representation. |
| static const UInt32 kBytesPerSample; |
| |
| // Initializes this class by creating the underlying audio unit instance. |
| // 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. |
| // Does not intialize the audio unit. |
| bool Init(); |
| |
| VoiceProcessingAudioUnit::State GetState() const; |
| |
| // Initializes the underlying audio unit with the given sample rate. |
| bool Initialize(Float64 sample_rate); |
| |
| // Starts the underlying audio unit. |
| bool Start(); |
| |
| // Stops the underlying audio unit. |
| bool Stop(); |
| |
| // Uninitializes the underlying audio unit. |
| bool Uninitialize(); |
| |
| // Calls render on the underlying audio unit. |
| OSStatus Render(AudioUnitRenderActionFlags* flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 output_bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data); |
| |
| private: |
| // The C API used to set callbacks requires static functions. When these are |
| // called, they will invoke the relevant instance method by casting |
| // in_ref_con to VoiceProcessingAudioUnit*. |
| static OSStatus OnGetPlayoutData(void* in_ref_con, |
| AudioUnitRenderActionFlags* flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data); |
| static OSStatus OnDeliverRecordedData(void* in_ref_con, |
| AudioUnitRenderActionFlags* flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data); |
| |
| // Notifies observer that samples are needed for playback. |
| OSStatus NotifyGetPlayoutData(AudioUnitRenderActionFlags* flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data); |
| // Notifies observer that recorded samples are available for render. |
| OSStatus NotifyDeliverRecordedData(AudioUnitRenderActionFlags* flags, |
| const AudioTimeStamp* time_stamp, |
| UInt32 bus_number, |
| UInt32 num_frames, |
| AudioBufferList* io_data); |
| |
| // Returns the predetermined format with a specific sample rate. See |
| // implementation file for details on format. |
| AudioStreamBasicDescription GetFormat(Float64 sample_rate) const; |
| |
| // Deletes the underlying audio unit. |
| void DisposeAudioUnit(); |
| |
| VoiceProcessingAudioUnitObserver* observer_; |
| AudioUnit vpio_unit_; |
| VoiceProcessingAudioUnit::State state_; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_ |