modules/audio_device/android/aaudio_recorder.h

/*
 *  Copyright (c) 2018 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 MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_

#include <aaudio/AAudio.h>

#include <memory>

#include "modules/audio_device/android/aaudio_wrapper.h"
#include "modules/audio_device/include/audio_device_defines.h"
#include "rtc_base/message_handler.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_checker.h"

namespace webrtc {

class AudioDeviceBuffer;
class FineAudioBuffer;
class AudioManager;

// Implements low-latency 16-bit mono PCM audio input support for Android
// using the C based AAudio API.
//
// An instance must be created and destroyed on one and the same thread.
// All public methods must also be called on the same thread. A thread checker
// will RTC_DCHECK if any method is called on an invalid thread. Audio buffers
// are delivered on a dedicated high-priority thread owned by AAudio.
//
// The existing design forces the user to call InitRecording() after
// StopRecording() to be able to call StartRecording() again. This is in line
// with how the Java- based implementation works.
//
// TODO(henrika): add comments about device changes and adaptive buffer
// management.
class AAudioRecorder : public AAudioObserverInterface,
                       public rtc::MessageHandler {
 public:
  explicit AAudioRecorder(AudioManager* audio_manager);
  ~AAudioRecorder();

  int Init();
  int Terminate();

  int InitRecording();
  bool RecordingIsInitialized() const { return initialized_; }

  int StartRecording();
  int StopRecording();
  bool Recording() const { return recording_; }

  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);

  double latency_millis() const { return latency_millis_; }

  // TODO(henrika): add support using AAudio APIs when available.
  int EnableBuiltInAEC(bool enable);
  int EnableBuiltInAGC(bool enable);
  int EnableBuiltInNS(bool enable);

 protected:
  // AAudioObserverInterface implementation.

  // For an input stream, this function should read |num_frames| of recorded
  // data, in the stream's current data format, from the |audio_data| buffer.
  // Called on a real-time thread owned by AAudio.
  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
                                               int32_t num_frames) override;

  // AAudio calls this function if any error occurs on a callback thread.
  // Called on a real-time thread owned by AAudio.
  void OnErrorCallback(aaudio_result_t error) override;

  // rtc::MessageHandler used for restart messages.
  void OnMessage(rtc::Message* msg) override;

 private:
  // Closes the existing stream and starts a new stream.
  void HandleStreamDisconnected();

  // Ensures that methods are called from the same thread as this object is
  // created on.
  rtc::ThreadChecker thread_checker_;

  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
  // real-time thread owned by AAudio. Detached during construction of this
  // object.
  rtc::ThreadChecker thread_checker_aaudio_;

  // The thread on which this object is created on.
  rtc::Thread* main_thread_;

  // Wraps all AAudio resources. Contains an input stream using the default
  // input audio device.
  AAudioWrapper aaudio_;

  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
  AudioDeviceBuffer* audio_device_buffer_ = nullptr;

  bool initialized_ = false;
  bool recording_ = false;

  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
  // chunks of audio.
  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;

  // Counts number of detected overflow events reported by AAudio.
  int32_t overflow_count_ = 0;

  // Estimated time between an audio frame was recorded by the input device and
  // it can read on the input stream.
  double latency_millis_ = 0;

  // True only for the first data callback in each audio session.
  bool first_data_callback_ = true;
};

}  // namespace webrtc

#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_