| /* |
| ** |
| ** Copyright 2018, The Android Open Source Project |
| ** |
| ** Licensed under the Apache License, Version 2.0 (the "License"); |
| ** you may not use this file except in compliance with the License. |
| ** You may obtain a copy of the License at |
| ** |
| ** http://www.apache.org/licenses/LICENSE-2.0 |
| ** |
| ** Unless required by applicable law or agreed to in writing, software |
| ** distributed under the License is distributed on an "AS IS" BASIS, |
| ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| ** See the License for the specific language governing permissions and |
| ** limitations under the License. |
| */ |
| |
| #ifndef ANDROID_MEDIAPLAYER2AUDIOOUTPUT_H |
| #define ANDROID_MEDIAPLAYER2AUDIOOUTPUT_H |
| |
| #include <mediaplayer2/MediaPlayer2Interface.h> |
| |
| #include <utils/String16.h> |
| #include <utils/Vector.h> |
| |
| namespace android { |
| |
| class AudioTrack; |
| |
| class MediaPlayer2AudioOutput : public MediaPlayer2Interface::AudioSink |
| { |
| class CallbackData; |
| |
| public: |
| MediaPlayer2AudioOutput(audio_session_t sessionId, |
| uid_t uid, |
| int pid, |
| const audio_attributes_t * attr, |
| const sp<AudioSystem::AudioDeviceCallback>& deviceCallback); |
| virtual ~MediaPlayer2AudioOutput(); |
| |
| virtual bool ready() const { |
| return mTrack != 0; |
| } |
| virtual ssize_t bufferSize() const; |
| virtual ssize_t frameCount() const; |
| virtual ssize_t channelCount() const; |
| virtual ssize_t frameSize() const; |
| virtual uint32_t latency() const; |
| virtual float msecsPerFrame() const; |
| virtual status_t getPosition(uint32_t *position) const; |
| virtual status_t getTimestamp(AudioTimestamp &ts) const; |
| virtual int64_t getPlayedOutDurationUs(int64_t nowUs) const; |
| virtual status_t getFramesWritten(uint32_t *frameswritten) const; |
| virtual audio_session_t getSessionId() const; |
| virtual uint32_t getSampleRate() const; |
| virtual int64_t getBufferDurationInUs() const; |
| |
| virtual status_t open( |
| uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, |
| audio_format_t format, |
| AudioCallback cb, void *cookie, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| const audio_offload_info_t *offloadInfo = NULL, |
| bool doNotReconnect = false, |
| uint32_t suggestedFrameCount = 0); |
| |
| virtual status_t start(); |
| virtual ssize_t write(const void* buffer, size_t size, bool blocking = true); |
| virtual void stop(); |
| virtual void flush(); |
| virtual void pause(); |
| virtual void close(); |
| void setAudioStreamType(audio_stream_type_t streamType); |
| virtual audio_stream_type_t getAudioStreamType() const { |
| return mStreamType; |
| } |
| void setAudioAttributes(const audio_attributes_t * attributes); |
| |
| void setVolume(float left, float right); |
| virtual status_t setPlaybackRate(const AudioPlaybackRate& rate); |
| virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */); |
| |
| status_t setAuxEffectSendLevel(float level); |
| status_t attachAuxEffect(int effectId); |
| virtual status_t dump(int fd, const Vector<String16>& args) const; |
| |
| static bool isOnEmulator(); |
| static int getMinBufferCount(); |
| virtual bool needsTrailingPadding() { |
| return true; |
| // TODO: return correct value. |
| //return mNextOutput == NULL; |
| } |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| |
| // AudioRouting |
| virtual status_t setOutputDevice(audio_port_handle_t deviceId); |
| virtual status_t getRoutedDeviceId(audio_port_handle_t* deviceId); |
| virtual status_t enableAudioDeviceCallback(bool enabled); |
| |
| private: |
| static void setMinBufferCount(); |
| static void CallbackWrapper(int event, void *me, void *info); |
| void deleteRecycledTrack_l(); |
| void close_l(); |
| status_t updateTrack_l(); |
| |
| sp<AudioTrack> mTrack; |
| AudioCallback mCallback; |
| void * mCallbackCookie; |
| CallbackData * mCallbackData; |
| audio_stream_type_t mStreamType; |
| audio_attributes_t * mAttributes; |
| float mLeftVolume; |
| float mRightVolume; |
| AudioPlaybackRate mPlaybackRate; |
| uint32_t mSampleRateHz; // sample rate of the content, as set in open() |
| float mMsecsPerFrame; |
| size_t mFrameSize; |
| audio_session_t mSessionId; |
| uid_t mUid; |
| int mPid; |
| float mSendLevel; |
| int mAuxEffectId; |
| audio_output_flags_t mFlags; |
| audio_port_handle_t mSelectedDeviceId; |
| audio_port_handle_t mRoutedDeviceId; |
| bool mDeviceCallbackEnabled; |
| wp<AudioSystem::AudioDeviceCallback> mDeviceCallback; |
| mutable Mutex mLock; |
| |
| // static variables below not protected by mutex |
| static bool mIsOnEmulator; |
| static int mMinBufferCount; // 12 for emulator; otherwise 4 |
| |
| // CallbackData is what is passed to the AudioTrack as the "user" data. |
| // We need to be able to target this to a different Output on the fly, |
| // so we can't use the Output itself for this. |
| class CallbackData { |
| friend MediaPlayer2AudioOutput; |
| public: |
| explicit CallbackData(MediaPlayer2AudioOutput *cookie) { |
| mData = cookie; |
| mSwitching = false; |
| } |
| MediaPlayer2AudioOutput *getOutput() const { |
| return mData; |
| } |
| void setOutput(MediaPlayer2AudioOutput* newcookie) { |
| mData = newcookie; |
| } |
| // lock/unlock are used by the callback before accessing the payload of this object |
| void lock() const { |
| mLock.lock(); |
| } |
| void unlock() const { |
| mLock.unlock(); |
| } |
| |
| // tryBeginTrackSwitch/endTrackSwitch are used when the CallbackData is handed over |
| // to the next sink. |
| |
| // tryBeginTrackSwitch() returns true only if it obtains the lock. |
| bool tryBeginTrackSwitch() { |
| LOG_ALWAYS_FATAL_IF(mSwitching, "tryBeginTrackSwitch() already called"); |
| if (mLock.tryLock() != OK) { |
| return false; |
| } |
| mSwitching = true; |
| return true; |
| } |
| void endTrackSwitch() { |
| if (mSwitching) { |
| mLock.unlock(); |
| } |
| mSwitching = false; |
| } |
| |
| private: |
| MediaPlayer2AudioOutput *mData; |
| mutable Mutex mLock; // a recursive mutex might make this unnecessary. |
| bool mSwitching; |
| DISALLOW_EVIL_CONSTRUCTORS(CallbackData); |
| }; |
| }; |
| |
| }; // namespace android |
| |
| #endif // ANDROID_MEDIAPLAYER2AUDIOOUTPUT_H |