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/*
* Copyright (c) 2013 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.
*/
// Android APIs used to access Java functionality needed to enable low latency
// audio.
#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_JNI_H_
#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_JNI_H_
#include <jni.h>
namespace webrtc {
class AudioManagerJni {
public:
AudioManagerJni();
~AudioManagerJni() {}
// SetAndroidAudioDeviceObjects must only be called once unless there has
// been a successive call to ClearAndroidAudioDeviceObjects. For each
// call to ClearAndroidAudioDeviceObjects, SetAndroidAudioDeviceObjects may be
// called once.
// This function must be called by a Java thread as calling it from a thread
// created by the native application will prevent FindClass from working. See
// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
// for more details.
// It has to be called for this class' APIs to be successful. Calling
// ClearAndroidAudioDeviceObjects will prevent this class' APIs to be called
// successfully if SetAndroidAudioDeviceObjects is not called after it.
static void SetAndroidAudioDeviceObjects(void* jvm, void* env,
void* context);
// This function must be called when the AudioManagerJni class is no
// longer needed. It frees up the global references acquired in
// SetAndroidAudioDeviceObjects.
static void ClearAndroidAudioDeviceObjects();
bool low_latency_supported() const { return low_latency_supported_; }
int native_output_sample_rate() const { return native_output_sample_rate_; }
int native_buffer_size() const { return native_buffer_size_; }
private:
bool HasDeviceObjects();
// Following functions assume that the calling thread has been attached.
void SetLowLatencySupported(JNIEnv* env);
void SetNativeOutputSampleRate(JNIEnv* env);
void SetNativeFrameSize(JNIEnv* env);
jmethodID LookUpMethodId(JNIEnv* env, const char* method_name,
const char* method_signature);
void CreateInstance(JNIEnv* env);
// Whether or not low latency audio is supported, the native output sample
// rate and the audio buffer size do not change. I.e the values might as well
// just be cached when initializing.
bool low_latency_supported_;
int native_output_sample_rate_;
int native_buffer_size_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_JNI_H_
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