| /* |
| * 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. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_ |
| |
| #include <jni.h> |
| |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/modules/audio_device/include/audio_device_defines.h" |
| #include "webrtc/modules/audio_device/audio_device_generic.h" |
| #include "webrtc/modules/utility/interface/helpers_android.h" |
| |
| namespace webrtc { |
| |
| class PlayoutDelayProvider; |
| |
| // Implements 16-bit mono PCM audio input support for Android using the Java |
| // AudioRecord interface. Most of the work is done by its Java counterpart in |
| // WebRtcAudioRecord.java. This class is created and lives on a thread in |
| // C++-land, but recorded audio buffers are delivered on a high-priority |
| // thread managed by the Java class. |
| // |
| // The Java class makes use of AudioEffect features (mainly AEC) which are |
| // first available in Jelly Bean. If it is instantiated running against earlier |
| // SDKs, the AEC provided by the APM in WebRTC must be used and enabled |
| // separately instead. |
| // |
| // 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 DCHECK if any method is called on an invalid thread. |
| // It is possible to call the two static methods (SetAndroidAudioDeviceObjects |
| // and ClearAndroidAudioDeviceObjects) from a different thread but both will |
| // CHECK that the calling thread is attached to a Java VM. |
| // |
| // All methods use AttachThreadScoped to attach to a Java VM if needed and then |
| // detach when method goes out of scope. We do so because this class does not |
| // own the thread is is created and called on and other objects on the same |
| // thread might put us in a detached state at any time. |
| class AudioRecordJni { |
| public: |
| // Use the invocation API to allow the native application to use the JNI |
| // interface pointer to access VM features. |
| // |jvm| denotes the Java VM and |context| corresponds to |
| // android.content.Context in Java. |
| // This method also sets a global jclass object, |g_audio_record_class| for |
| // the "org/webrtc/voiceengine/WebRtcAudioRecord"-class. |
| static void SetAndroidAudioDeviceObjects(void* jvm, void* context); |
| // Always call this method after the object has been destructed. It deletes |
| // existing global references and enables garbage collection. |
| static void ClearAndroidAudioDeviceObjects(); |
| |
| AudioRecordJni(PlayoutDelayProvider* delay_provider); |
| ~AudioRecordJni(); |
| |
| int32_t Init(); |
| int32_t Terminate(); |
| |
| int32_t InitRecording(); |
| bool RecordingIsInitialized() const { return initialized_; } |
| |
| int32_t StartRecording(); |
| int32_t StopRecording (); |
| bool Recording() const { return recording_; } |
| |
| int32_t RecordingDelay(uint16_t& delayMS) const; |
| |
| void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| |
| bool BuiltInAECIsAvailable() const; |
| int32_t EnableBuiltInAEC(bool enable); |
| |
| private: |
| // Called from Java side so we can cache the address of the Java-manged |
| // |byte_buffer| in |direct_buffer_address_|. The size of the buffer |
| // is also stored in |direct_buffer_capacity_in_bytes_|. |
| // This method will be called by the WebRtcAudioRecord constructor, i.e., |
| // on the same thread that this object is created on. |
| static void JNICALL CacheDirectBufferAddress( |
| JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioRecord); |
| void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer); |
| |
| // Called periodically by the Java based WebRtcAudioRecord object when |
| // recording has started. Each call indicates that there are |length| new |
| // bytes recorded in the memory area |direct_buffer_address_| and it is |
| // now time to send these to the consumer. |
| // This method is called on a high-priority thread from Java. The name of |
| // the thread is 'AudioRecordThread'. |
| static void JNICALL DataIsRecorded( |
| JNIEnv* env, jobject obj, jint length, jlong nativeAudioRecord); |
| void OnDataIsRecorded(int length); |
| |
| // Returns true if SetAndroidAudioDeviceObjects() has been called |
| // successfully. |
| bool HasDeviceObjects(); |
| |
| // Called from the constructor. Defines the |j_audio_record_| member. |
| void CreateJavaInstance(); |
| |
| // Returns the native, or optimal, sample rate reported by the audio input |
| // device. |
| int GetNativeSampleRate(); |
| |
| // Stores thread ID in constructor. |
| // We can then use ThreadChecker::CalledOnValidThread() to ensure that |
| // other methods are called from the same thread. |
| // Currently only does DCHECK(thread_checker_.CalledOnValidThread()). |
| rtc::ThreadChecker thread_checker_; |
| |
| // Stores thread ID in first call to OnDataIsRecorded() from high-priority |
| // thread in Java. Detached during construction of this object. |
| rtc::ThreadChecker thread_checker_java_; |
| |
| // Returns the current playout delay. |
| // TODO(henrika): this value is currently fixed since initial tests have |
| // shown that the estimated delay varies very little over time. It might be |
| // possible to make improvements in this area. |
| PlayoutDelayProvider* delay_provider_; |
| |
| // The Java WebRtcAudioRecord instance. |
| jobject j_audio_record_; |
| |
| // Cached copy of address to direct audio buffer owned by |j_audio_record_|. |
| void* direct_buffer_address_; |
| |
| // Number of bytes in the direct audio buffer owned by |j_audio_record_|. |
| int direct_buffer_capacity_in_bytes_; |
| |
| // Number audio frames per audio buffer. Each audio frame corresponds to |
| // one sample of PCM mono data at 16 bits per sample. Hence, each audio |
| // frame contains 2 bytes (given that the Java layer only supports mono). |
| // Example: 480 for 48000 Hz or 441 for 44100 Hz. |
| int frames_per_buffer_; |
| |
| bool initialized_; |
| |
| bool recording_; |
| |
| // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the |
| // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create(). |
| AudioDeviceBuffer* audio_device_buffer_; |
| |
| // Native sample rate set in AttachAudioBuffer() which uses JNI to ask the |
| // Java layer for the best possible sample rate for this particular device |
| // and audio configuration. |
| int sample_rate_hz_; |
| |
| // Contains a delay estimate from the playout side given by |delay_provider_|. |
| int playout_delay_in_milliseconds_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_ |