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/*
* 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;
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,
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();
// 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_;
VoiceProcessingAudioUnitObserver* observer_;
AudioUnit vpio_unit_;
VoiceProcessingAudioUnit::State state_;
};
} // namespace ios_adm
} // namespace webrtc
#endif // SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_
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