09bf1a169b
Restores stored audio mode when all streaming stops. TBR=glaznev BUG=NONE TEST=AppRTCDemo Review URL: https://webrtc-codereview.appspot.com/46869005 Cr-Commit-Position: refs/heads/master@{#8970}
135 lines
5.1 KiB
C++
135 lines
5.1 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
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#define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
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#include <jni.h>
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/audio_device/android/audio_common.h"
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#include "webrtc/modules/audio_device/include/audio_device_defines.h"
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#include "webrtc/modules/audio_device/audio_device_generic.h"
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#include "webrtc/modules/utility/interface/helpers_android.h"
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namespace webrtc {
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class AudioParameters {
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public:
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enum { kBitsPerSample = 16 };
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AudioParameters()
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: sample_rate_(0),
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channels_(0),
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frames_per_buffer_(0),
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bits_per_sample_(kBitsPerSample) {}
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AudioParameters(int sample_rate, int channels)
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: sample_rate_(sample_rate),
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channels_(channels),
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frames_per_buffer_(sample_rate / 100),
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bits_per_sample_(kBitsPerSample) {}
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void reset(int sample_rate, int channels) {
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sample_rate_ = sample_rate;
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channels_ = channels;
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// WebRTC uses a fixed buffer size equal to 10ms.
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frames_per_buffer_ = (sample_rate / 100);
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}
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int sample_rate() const { return sample_rate_; }
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int channels() const { return channels_; }
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int frames_per_buffer() const { return frames_per_buffer_; }
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bool is_valid() const {
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return ((sample_rate_ > 0) && (channels_ > 0) && (frames_per_buffer_ > 0));
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}
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int GetBytesPerFrame() const { return channels_ * bits_per_sample_ / 8; }
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int GetBytesPerBuffer() const {
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return frames_per_buffer_ * GetBytesPerFrame();
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}
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private:
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int sample_rate_;
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int channels_;
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int frames_per_buffer_;
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const int bits_per_sample_;
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};
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// Implements support for functions in the WebRTC audio stack for Android that
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// relies on the AudioManager in android.media. It also populates an
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// AudioParameter structure with native audio parameters detected at
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// construction. This class does not make any audio-related modifications
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// unless Init() is called. Caching audio parameters makes no changes but only
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// reads data from the Java side.
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// TODO(henrika): expand this class when adding support for low-latency
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// OpenSL ES. Currently, it only contains very basic functionality.
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class AudioManager {
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public:
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// Use the invocation API to allow the native application to use the JNI
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// interface pointer to access VM features. |jvm| denotes the Java VM and
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// |context| corresponds to android.content.Context in Java.
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// This method also sets a global jclass object, |g_audio_manager_class| for
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// the "org/webrtc/voiceengine/WebRtcAudioManager"-class.
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static void SetAndroidAudioDeviceObjects(void* jvm, void* context);
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// Always call this method after the object has been destructed. It deletes
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// existing global references and enables garbage collection.
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static void ClearAndroidAudioDeviceObjects();
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AudioManager();
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~AudioManager();
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// Initializes the audio manager and stores the current audio mode.
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bool Init();
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// Revert any setting done by Init().
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bool Close();
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// Sets audio mode to AudioManager.MODE_IN_COMMUNICATION if |enable| is true.
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// Restores audio mode that was stored in Init() if |enable| is false.
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void SetCommunicationMode(bool enable);
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// Native audio parameters stored during construction.
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AudioParameters GetPlayoutAudioParameters() const;
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AudioParameters GetRecordAudioParameters() const;
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bool initialized() const { return initialized_; }
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private:
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// Called from Java side so we can cache the native audio parameters.
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// This method will be called by the WebRtcAudioManager constructor, i.e.
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// on the same thread that this object is created on.
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static void JNICALL CacheAudioParameters(JNIEnv* env, jobject obj,
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jint sample_rate, jint channels, jlong nativeAudioManager);
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void OnCacheAudioParameters(JNIEnv* env, jint sample_rate, jint channels);
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// Returns true if SetAndroidAudioDeviceObjects() has been called
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// successfully.
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bool HasDeviceObjects();
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// Called from the constructor. Defines the |j_audio_manager_| member.
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void CreateJavaInstance();
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// Stores thread ID in the constructor.
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// We can then use ThreadChecker::CalledOnValidThread() to ensure that
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// other methods are called from the same thread.
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rtc::ThreadChecker thread_checker_;
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// The Java WebRtcAudioManager instance.
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jobject j_audio_manager_;
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// Set to true by Init() and false by Close().
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bool initialized_;
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// Contains native parameters (e.g. sample rate, channel configuration).
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// Set at construction in OnCacheAudioParameters() which is called from
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// Java on the same thread as this object is created on.
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AudioParameters playout_parameters_;
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AudioParameters record_parameters_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
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