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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index 0000000000000000000000000000000000000000..c1e5cafb4bbc9500c70c172b9a21064217ddb4c5 |
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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 |
+ // 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. |
+ |
+ // 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(); |
+ |
+ // 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_; |
+}; |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_ |