voice_processing_audio_unit.h

/*

 *  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 SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_

#define SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_

#include <AudioUnit/AudioUnit.h>

namespace webrtc {

namespace ios_adm {

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;

  // Callback function called when a user speaking during muted is detected by

  // system.

  virtual void OnReceivedMutedSpeechActivity(

      AUVoiceIOSpeechActivityEvent event) = 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:

  VoiceProcessingAudioUnit(bool bypass_voice_processing,

                           bool detect_mute_speech,

                           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.

  OSStatus Start();

  // Stops the underlying audio unit.

  bool Stop();

  // Uninitializes the underlying audio unit.

  bool Uninitialize();

  // Mutes the microphone.

  bool SetMicrophoneMute(bool enable);

  // 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();

  const bool bypass_voice_processing_;

  const bool detect_mute_speech_;

  VoiceProcessingAudioUnitObserver* observer_;

  AudioUnit vpio_unit_;

  VoiceProcessingAudioUnit::State state_;

};

}  // namespace ios_adm

}  // namespace webrtc

#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_