text: /*
   *  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 SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
  #define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
  
  #include <memory>
  
  #include "absl/types/optional.h"
  #include "modules/audio_device/audio_device_buffer.h"
  #include "sdk/android/native_api/jni/scoped_java_ref.h"
  
  namespace webrtc {
  namespace jni {
  
  class AudioInput {
   public:
    virtual ~AudioInput() {}
  
    virtual int32_t Init() = 0;
    virtual int32_t Terminate() = 0;
  
    virtual int32_t InitRecording() = 0;
    virtual bool RecordingIsInitialized() const = 0;
  
    virtual int32_t StartRecording() = 0;
    virtual int32_t StopRecording() = 0;
    virtual bool Recording() const = 0;
  
    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
  
    // Returns true if the audio input supports built-in audio effects for AEC and
    // NS.
    virtual bool IsAcousticEchoCancelerSupported() const = 0;
    virtual bool IsNoiseSuppressorSupported() const = 0;
  
    virtual int32_t EnableBuiltInAEC(bool enable) = 0;
    virtual int32_t EnableBuiltInNS(bool enable) = 0;
  };
  
  class AudioOutput {
   public:
    virtual ~AudioOutput() {}
  
    virtual int32_t Init() = 0;
    virtual int32_t Terminate() = 0;
    virtual int32_t InitPlayout() = 0;
    virtual bool PlayoutIsInitialized() const = 0;
    virtual int32_t StartPlayout() = 0;
    virtual int32_t StopPlayout() = 0;
    virtual bool Playing() const = 0;
    virtual bool SpeakerVolumeIsAvailable() = 0;
    virtual int SetSpeakerVolume(uint32_t volume) = 0;
    virtual absl::optional<uint32_t> SpeakerVolume() const = 0;
    virtual absl::optional<uint32_t> MaxSpeakerVolume() const = 0;
    virtual absl::optional<uint32_t> MinSpeakerVolume() const = 0;
    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
  };
  
  // Extract an android.media.AudioManager from an android.content.Context.
  ScopedJavaLocalRef<jobject> GetAudioManager(JNIEnv* env,
                                              const JavaRef<jobject>& j_context);
  
  // Get default audio sample rate by querying an android.media.AudioManager.
  int GetDefaultSampleRate(JNIEnv* env, const JavaRef<jobject>& j_audio_manager);
  
  // Get audio input and output parameters based on a number of settings.
  void GetAudioParameters(JNIEnv* env,
                          const JavaRef<jobject>& j_context,
                          const JavaRef<jobject>& j_audio_manager,
                          int sample_rate,
                          bool use_stereo_input,
                          bool use_stereo_output,
                          AudioParameters* input_parameters,
                          AudioParameters* output_parameters);
  
  // Glue together an audio input and audio output to get an AudioDeviceModule.
  rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
      AudioDeviceModule::AudioLayer audio_layer,
      bool is_stereo_playout_supported,
      bool is_stereo_record_supported,
      uint16_t playout_delay_ms,
      std::unique_ptr<AudioInput> audio_input,
      std::unique_ptr<AudioOutput> audio_output);
  
  }  // namespace jni
  
  }  // namespace webrtc
  
  #endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
  