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Diffstat (limited to 'third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h')
| -rw-r--r-- | third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h | 141 |
1 files changed, 141 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h b/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h new file mode 100644 index 0000000000..ed9dd98568 --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h @@ -0,0 +1,141 @@ +/* + * 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_ |