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/*
* 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/include/audio_device.h"
#include "sdk/android/native_api/jni/scoped_java_ref.h"
namespace webrtc {
class AudioDeviceBuffer;
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;
virtual int GetPlayoutUnderrunCount() = 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 input_sample_rate,
int output_sample_rate,
bool use_stereo_input,
bool use_stereo_output,
AudioParameters* input_parameters,
AudioParameters* output_parameters);
bool IsLowLatencyInputSupported(JNIEnv* env, const JavaRef<jobject>& j_context);
bool IsLowLatencyOutputSupported(JNIEnv* env, const JavaRef<jobject>& j_context);
// 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_
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