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Unified Diff: webrtc/modules/audio_device/ios/voice_processing_audio_unit.h

Issue 1809343002: Refactor AudioUnit code into its own class. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: CR comments Created 4 years, 9 months ago
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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
new file mode 100644
index 0000000000000000000000000000000000000000..c1e5cafb4bbc9500c70c172b9a21064217ddb4c5
--- /dev/null
+++ b/webrtc/modules/audio_device/ios/voice_processing_audio_unit.h
@@ -0,0 +1,124 @@
+/*
+ * 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_

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