| /* |
| * 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_TRACK_JNI_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_ |
| |
| #include <jni.h> |
| |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/modules/audio_device/android/audio_common.h" |
| #include "webrtc/modules/audio_device/include/audio_device_defines.h" |
| #include "webrtc/modules/audio_device/audio_device_generic.h" |
| |
| namespace webrtc { |
| |
| class EventWrapper; |
| class ThreadWrapper; |
| |
| const uint32_t N_PLAY_SAMPLES_PER_SEC = 16000; // Default is 16 kHz |
| const uint32_t N_PLAY_CHANNELS = 1; // default is mono playout |
| |
| class AudioTrackJni : public PlayoutDelayProvider { |
| public: |
| static int32_t SetAndroidAudioDeviceObjects(void* javaVM, void* env, |
| void* context); |
| static void ClearAndroidAudioDeviceObjects(); |
| explicit AudioTrackJni(const int32_t id); |
| virtual ~AudioTrackJni(); |
| |
| // Main initializaton and termination |
| int32_t Init(); |
| int32_t Terminate(); |
| bool Initialized() const { return _initialized; } |
| |
| // Device enumeration |
| int16_t PlayoutDevices() { return 1; } // There is one device only. |
| |
| int32_t PlayoutDeviceName(uint16_t index, |
| char name[kAdmMaxDeviceNameSize], |
| char guid[kAdmMaxGuidSize]); |
| |
| // Device selection |
| int32_t SetPlayoutDevice(uint16_t index); |
| int32_t SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| |
| // Audio transport initialization |
| int32_t PlayoutIsAvailable(bool& available); // NOLINT |
| int32_t InitPlayout(); |
| bool PlayoutIsInitialized() const { return _playIsInitialized; } |
| |
| // Audio transport control |
| int32_t StartPlayout(); |
| int32_t StopPlayout(); |
| bool Playing() const { return _playing; } |
| |
| // Audio mixer initialization |
| int32_t InitSpeaker(); |
| bool SpeakerIsInitialized() const { return _speakerIsInitialized; } |
| |
| // Speaker volume controls |
| int32_t SpeakerVolumeIsAvailable(bool& available); // NOLINT |
| int32_t SetSpeakerVolume(uint32_t volume); |
| int32_t SpeakerVolume(uint32_t& volume) const; // NOLINT |
| int32_t MaxSpeakerVolume(uint32_t& maxVolume) const; // NOLINT |
| int32_t MinSpeakerVolume(uint32_t& minVolume) const; // NOLINT |
| int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const; // NOLINT |
| |
| // Speaker mute control |
| int32_t SpeakerMuteIsAvailable(bool& available); // NOLINT |
| int32_t SetSpeakerMute(bool enable); |
| int32_t SpeakerMute(bool& enabled) const; // NOLINT |
| |
| |
| // Stereo support |
| int32_t StereoPlayoutIsAvailable(bool& available); // NOLINT |
| int32_t SetStereoPlayout(bool enable); |
| int32_t StereoPlayout(bool& enabled) const; // NOLINT |
| |
| // Delay information and control |
| int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type, |
| uint16_t sizeMS); |
| int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type, // NOLINT |
| uint16_t& sizeMS) const; |
| int32_t PlayoutDelay(uint16_t& delayMS) const; // NOLINT |
| |
| // Attach audio buffer |
| void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| |
| int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec); |
| |
| // Error and warning information |
| bool PlayoutWarning() const; |
| bool PlayoutError() const; |
| void ClearPlayoutWarning(); |
| void ClearPlayoutError(); |
| |
| // Speaker audio routing |
| int32_t SetLoudspeakerStatus(bool enable); |
| int32_t GetLoudspeakerStatus(bool& enable) const; // NOLINT |
| |
| protected: |
| virtual int PlayoutDelayMs() { return 0; } |
| |
| private: |
| void Lock() EXCLUSIVE_LOCK_FUNCTION(_critSect) { |
| _critSect.Enter(); |
| } |
| void UnLock() UNLOCK_FUNCTION(_critSect) { |
| _critSect.Leave(); |
| } |
| |
| int32_t InitJavaResources(); |
| int32_t InitSampleRate(); |
| |
| static bool PlayThreadFunc(void*); |
| bool PlayThreadProcess(); |
| |
| // TODO(leozwang): Android holds only one JVM, all these jni handling |
| // will be consolidated into a single place to make it consistant and |
| // reliable. Chromium has a good example at base/android. |
| static JavaVM* globalJvm; |
| static JNIEnv* globalJNIEnv; |
| static jobject globalContext; |
| static jclass globalScClass; |
| |
| JavaVM* _javaVM; // denotes a Java VM |
| JNIEnv* _jniEnvPlay; // The JNI env for playout thread |
| jclass _javaScClass; // AudioDeviceAndroid class |
| jobject _javaScObj; // AudioDeviceAndroid object |
| jobject _javaPlayBuffer; |
| void* _javaDirectPlayBuffer; // Direct buffer pointer to play buffer |
| jmethodID _javaMidPlayAudio; // Method ID of play in AudioDeviceAndroid |
| |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| CriticalSectionWrapper& _critSect; |
| int32_t _id; |
| bool _initialized; |
| |
| EventWrapper& _timeEventPlay; |
| EventWrapper& _playStartStopEvent; |
| ThreadWrapper* _ptrThreadPlay; |
| uint32_t _playThreadID; |
| bool _playThreadIsInitialized; |
| bool _shutdownPlayThread; |
| bool _playoutDeviceIsSpecified; |
| |
| bool _playing; |
| bool _playIsInitialized; |
| bool _speakerIsInitialized; |
| |
| bool _startPlay; |
| |
| uint16_t _playWarning; |
| uint16_t _playError; |
| |
| uint16_t _delayPlayout; |
| |
| uint16_t _samplingFreqOut; // Sampling frequency for Speaker |
| uint32_t _maxSpeakerVolume; // The maximum speaker volume value |
| bool _loudSpeakerOn; |
| |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_ |