| Index: webrtc/modules/audio_device/ios/voice_processing_audio_unit.h
|
| diff --git a/webrtc/modules/audio_device/ios/voice_processing_audio_unit.h b/webrtc/modules/audio_device/ios/voice_processing_audio_unit.h
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| new file mode 100644
|
| index 0000000000000000000000000000000000000000..c1e5cafb4bbc9500c70c172b9a21064217ddb4c5
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| --- /dev/null
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| +++ b/webrtc/modules/audio_device/ios/voice_processing_audio_unit.h
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| @@ -0,0 +1,124 @@
|
| +/*
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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 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
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| + // unit. This method is used to signal that recorded audio is available.
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| + virtual OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
|
| + const AudioTimeStamp* time_stamp,
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| + UInt32 bus_number,
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| + UInt32 num_frames,
|
| + AudioBufferList* io_data) = 0;
|
| +
|
| + // Callback function called on a real-time priority I/O thread from the audio
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| + // unit. This method is used to provide audio samples to the audio unit.
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| + virtual OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
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| + const AudioTimeStamp* time_stamp,
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| + UInt32 bus_number,
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| + UInt32 num_frames,
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| + AudioBufferList* io_data) = 0;
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| +
|
| + protected:
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| + ~VoiceProcessingAudioUnitObserver() {}
|
| +};
|
| +
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| +// Convenience class to abstract away the management of a Voice Processing
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| +// I/O Audio Unit. The Voice Processing I/O unit has the same characteristics
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| +// as the Remote I/O unit (supports full duplex low-latency audio input and
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| +// output) and adds AEC for for two-way duplex communication. It also adds AGC,
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| +// adjustment of voice-processing quality, and muting. Hence, ideal for
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| +// VoIP applications.
|
| +class VoiceProcessingAudioUnit {
|
| + public:
|
| + explicit VoiceProcessingAudioUnit(VoiceProcessingAudioUnitObserver* observer);
|
| + ~VoiceProcessingAudioUnit();
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| +
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| + // TODO(tkchin): enum for state and state checking.
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| +
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| + // Number of bytes per audio sample for 16-bit signed integer representation.
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| + static const UInt32 kBytesPerSample;
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| +
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| + // Initializes this class by creating the underlying audio unit instance.
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| + // Creates a Voice-Processing I/O unit and configures it for full-duplex
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| + // audio. The selected stream format is selected to avoid internal resampling
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| + // and to match the 10ms callback rate for WebRTC as well as possible.
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| + // Does not intialize the audio unit.
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| + bool Init();
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| +
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| + // Initializes the underlying audio unit with the given sample rate.
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| + bool Initialize(Float64 sample_rate);
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| +
|
| + // Starts the underlying audio unit.
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| + bool Start();
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| +
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| + // Stops the underlying audio unit.
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| + bool Stop();
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| +
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| + // Uninitializes the underlying audio unit.
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| + bool Uninitialize();
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| +
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| + // Calls render on the underlying audio unit.
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| + OSStatus Render(AudioUnitRenderActionFlags* flags,
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| + const AudioTimeStamp* time_stamp,
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| + UInt32 output_bus_number,
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| + UInt32 num_frames,
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| + AudioBufferList* io_data);
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| +
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| + private:
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| + // The C API used to set callbacks requires static functions. When these are
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| + // called, they will invoke the relevant instance method by casting
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| + // in_ref_con to VoiceProcessingAudioUnit*.
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| + static OSStatus OnGetPlayoutData(void* in_ref_con,
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| + AudioUnitRenderActionFlags* flags,
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| + const AudioTimeStamp* time_stamp,
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| + UInt32 bus_number,
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| + UInt32 num_frames,
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| + AudioBufferList* io_data);
|
| + static OSStatus OnDeliverRecordedData(void* in_ref_con,
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| + AudioUnitRenderActionFlags* flags,
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| + const AudioTimeStamp* time_stamp,
|
| + UInt32 bus_number,
|
| + UInt32 num_frames,
|
| + AudioBufferList* io_data);
|
| +
|
| + // Notifies observer that samples are needed for playback.
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| + OSStatus NotifyGetPlayoutData(AudioUnitRenderActionFlags* flags,
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| + const AudioTimeStamp* time_stamp,
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| + UInt32 bus_number,
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| + UInt32 num_frames,
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| + AudioBufferList* io_data);
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| + // Notifies observer that recorded samples are available for render.
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| + OSStatus NotifyDeliverRecordedData(AudioUnitRenderActionFlags* flags,
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| + 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.
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| + AudioStreamBasicDescription GetFormat(Float64 sample_rate) const;
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| +
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| + // Deletes the underlying audio unit.
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| + void DisposeAudioUnit();
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| +
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| + VoiceProcessingAudioUnitObserver* observer_;
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| + AudioUnit vpio_unit_;
|
| +};
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
|
|
|