| /* |
| ** |
| ** Copyright 2008, 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_MEDIAPLAYERSERVICE_H |
| #define ANDROID_MEDIAPLAYERSERVICE_H |
| |
| #include <arpa/inet.h> |
| |
| #include <utils/threads.h> |
| #include <utils/Errors.h> |
| #include <utils/KeyedVector.h> |
| #include <utils/String8.h> |
| #include <utils/Vector.h> |
| |
| #include <media/MediaPlayerInterface.h> |
| #include <media/Metadata.h> |
| #include <media/stagefright/foundation/ABase.h> |
| |
| #include <system/audio.h> |
| |
| namespace android { |
| |
| class AudioTrack; |
| class IMediaRecorder; |
| class IMediaMetadataRetriever; |
| class IOMX; |
| class IRemoteDisplay; |
| class IRemoteDisplayClient; |
| class MediaRecorderClient; |
| |
| #define CALLBACK_ANTAGONIZER 0 |
| #if CALLBACK_ANTAGONIZER |
| class Antagonizer { |
| public: |
| Antagonizer(notify_callback_f cb, void* client); |
| void start() { mActive = true; } |
| void stop() { mActive = false; } |
| void kill(); |
| private: |
| static const int interval; |
| Antagonizer(); |
| static int callbackThread(void* cookie); |
| Mutex mLock; |
| Condition mCondition; |
| bool mExit; |
| bool mActive; |
| void* mClient; |
| notify_callback_f mCb; |
| }; |
| #endif |
| |
| class MediaPlayerService : public BnMediaPlayerService |
| { |
| class Client; |
| |
| class AudioOutput : public MediaPlayerBase::AudioSink |
| { |
| class CallbackData; |
| |
| public: |
| AudioOutput(int sessionId, int uid, int pid, |
| const audio_attributes_t * attr); |
| virtual ~AudioOutput(); |
| |
| virtual bool ready() const { return mTrack != 0; } |
| virtual bool realtime() const { return true; } |
| 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 status_t getFramesWritten(uint32_t *frameswritten) const; |
| virtual int getSessionId() const; |
| virtual uint32_t getSampleRate() const; |
| |
| virtual status_t open( |
| uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, |
| audio_format_t format, int bufferCount, |
| AudioCallback cb, void *cookie, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| const audio_offload_info_t *offloadInfo = NULL); |
| |
| virtual status_t start(); |
| virtual ssize_t write(const void* buffer, size_t size); |
| virtual void stop(); |
| virtual void flush(); |
| virtual void pause(); |
| virtual void close(); |
| void setAudioStreamType(audio_stream_type_t streamType) { |
| mStreamType = 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 setPlaybackRatePermille(int32_t ratePermille); |
| 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(); |
| void setNextOutput(const sp<AudioOutput>& nextOutput); |
| void switchToNextOutput(); |
| virtual bool needsTrailingPadding() { return mNextOutput == NULL; } |
| virtual status_t setParameters(const String8& keyValuePairs); |
| virtual String8 getParameters(const String8& keys); |
| |
| private: |
| static void setMinBufferCount(); |
| static void CallbackWrapper( |
| int event, void *me, void *info); |
| void deleteRecycledTrack(); |
| |
| sp<AudioTrack> mTrack; |
| sp<AudioTrack> mRecycledTrack; |
| sp<AudioOutput> mNextOutput; |
| AudioCallback mCallback; |
| void * mCallbackCookie; |
| CallbackData * mCallbackData; |
| uint64_t mBytesWritten; |
| audio_stream_type_t mStreamType; |
| const audio_attributes_t *mAttributes; |
| float mLeftVolume; |
| float mRightVolume; |
| int32_t mPlaybackRatePermille; |
| uint32_t mSampleRateHz; // sample rate of the content, as set in open() |
| float mMsecsPerFrame; |
| int mSessionId; |
| int mUid; |
| int mPid; |
| float mSendLevel; |
| int mAuxEffectId; |
| static bool mIsOnEmulator; |
| static int mMinBufferCount; // 12 for emulator; otherwise 4 |
| audio_output_flags_t mFlags; |
| |
| // 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 { |
| public: |
| CallbackData(AudioOutput *cookie) { |
| mData = cookie; |
| mSwitching = false; |
| } |
| AudioOutput * getOutput() { return mData;} |
| void setOutput(AudioOutput* newcookie) { mData = newcookie; } |
| // lock/unlock are used by the callback before accessing the payload of this object |
| void lock() { mLock.lock(); } |
| void unlock() { mLock.unlock(); } |
| // beginTrackSwitch/endTrackSwitch are used when this object is being handed over |
| // to the next sink. |
| void beginTrackSwitch() { mLock.lock(); mSwitching = true; } |
| void endTrackSwitch() { |
| if (mSwitching) { |
| mLock.unlock(); |
| } |
| mSwitching = false; |
| } |
| private: |
| AudioOutput * mData; |
| mutable Mutex mLock; |
| bool mSwitching; |
| DISALLOW_EVIL_CONSTRUCTORS(CallbackData); |
| }; |
| |
| }; // AudioOutput |
| |
| |
| class AudioCache : public MediaPlayerBase::AudioSink |
| { |
| public: |
| AudioCache(const sp<IMemoryHeap>& heap); |
| virtual ~AudioCache() {} |
| |
| virtual bool ready() const { return (mChannelCount > 0) && (mHeap->getHeapID() > 0); } |
| virtual bool realtime() const { return false; } |
| virtual ssize_t bufferSize() const { return frameSize() * mFrameCount; } |
| virtual ssize_t frameCount() const { return mFrameCount; } |
| virtual ssize_t channelCount() const { return (ssize_t)mChannelCount; } |
| virtual ssize_t frameSize() const { return (ssize_t)mFrameSize; } |
| 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 status_t getFramesWritten(uint32_t *frameswritten) const; |
| virtual int getSessionId() const; |
| virtual uint32_t getSampleRate() const; |
| |
| virtual status_t open( |
| uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, |
| audio_format_t format, int bufferCount = 1, |
| AudioCallback cb = NULL, void *cookie = NULL, |
| audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, |
| const audio_offload_info_t *offloadInfo = NULL); |
| |
| virtual status_t start(); |
| virtual ssize_t write(const void* buffer, size_t size); |
| virtual void stop(); |
| virtual void flush() {} |
| virtual void pause() {} |
| virtual void close() {} |
| void setAudioStreamType(audio_stream_type_t streamType __unused) {} |
| // stream type is not used for AudioCache |
| virtual audio_stream_type_t getAudioStreamType() const { return AUDIO_STREAM_DEFAULT; } |
| |
| void setVolume(float left __unused, float right __unused) {} |
| virtual status_t setPlaybackRatePermille(int32_t ratePermille __unused) { return INVALID_OPERATION; } |
| uint32_t sampleRate() const { return mSampleRate; } |
| audio_format_t format() const { return mFormat; } |
| size_t size() const { return mSize; } |
| status_t wait(); |
| |
| sp<IMemoryHeap> getHeap() const { return mHeap; } |
| |
| static void notify(void* cookie, int msg, |
| int ext1, int ext2, const Parcel *obj); |
| virtual status_t dump(int fd, const Vector<String16>& args) const; |
| |
| private: |
| AudioCache(); |
| |
| Mutex mLock; |
| Condition mSignal; |
| sp<IMemoryHeap> mHeap; |
| float mMsecsPerFrame; |
| uint16_t mChannelCount; |
| audio_format_t mFormat; |
| ssize_t mFrameCount; |
| uint32_t mSampleRate; |
| uint32_t mSize; |
| size_t mFrameSize; |
| int mError; |
| bool mCommandComplete; |
| |
| sp<Thread> mCallbackThread; |
| }; // AudioCache |
| |
| public: |
| static void instantiate(); |
| |
| // IMediaPlayerService interface |
| virtual sp<IMediaRecorder> createMediaRecorder(); |
| void removeMediaRecorderClient(wp<MediaRecorderClient> client); |
| virtual sp<IMediaMetadataRetriever> createMetadataRetriever(); |
| |
| virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId); |
| |
| virtual status_t decode( |
| const sp<IMediaHTTPService> &httpService, |
| const char* url, |
| uint32_t *pSampleRate, |
| int* pNumChannels, |
| audio_format_t* pFormat, |
| const sp<IMemoryHeap>& heap, |
| size_t *pSize); |
| |
| virtual status_t decode(int fd, int64_t offset, int64_t length, |
| uint32_t *pSampleRate, int* pNumChannels, |
| audio_format_t* pFormat, |
| const sp<IMemoryHeap>& heap, size_t *pSize); |
| virtual sp<IMediaCodecList> getCodecList() const; |
| virtual sp<IOMX> getOMX(); |
| virtual sp<ICrypto> makeCrypto(); |
| virtual sp<IDrm> makeDrm(); |
| virtual sp<IHDCP> makeHDCP(bool createEncryptionModule); |
| |
| virtual sp<IRemoteDisplay> listenForRemoteDisplay(const sp<IRemoteDisplayClient>& client, |
| const String8& iface); |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| void removeClient(wp<Client> client); |
| |
| // For battery usage tracking purpose |
| struct BatteryUsageInfo { |
| // how many streams are being played by one UID |
| int refCount; |
| // a temp variable to store the duration(ms) of audio codecs |
| // when we start a audio codec, we minus the system time from audioLastTime |
| // when we pause it, we add the system time back to the audioLastTime |
| // so after the pause, audioLastTime = pause time - start time |
| // if multiple audio streams are played (or recorded), then audioLastTime |
| // = the total playing time of all the streams |
| int32_t audioLastTime; |
| // when all the audio streams are being paused, we assign audioLastTime to |
| // this variable, so this value could be provided to the battery app |
| // in the next pullBatteryData call |
| int32_t audioTotalTime; |
| |
| int32_t videoLastTime; |
| int32_t videoTotalTime; |
| }; |
| KeyedVector<int, BatteryUsageInfo> mBatteryData; |
| |
| enum { |
| SPEAKER, |
| OTHER_AUDIO_DEVICE, |
| SPEAKER_AND_OTHER, |
| NUM_AUDIO_DEVICES |
| }; |
| |
| struct BatteryAudioFlingerUsageInfo { |
| int refCount; // how many audio streams are being played |
| int deviceOn[NUM_AUDIO_DEVICES]; // whether the device is currently used |
| int32_t lastTime[NUM_AUDIO_DEVICES]; // in ms |
| // totalTime[]: total time of audio output devices usage |
| int32_t totalTime[NUM_AUDIO_DEVICES]; // in ms |
| }; |
| |
| // This varialble is used to record the usage of audio output device |
| // for battery app |
| BatteryAudioFlingerUsageInfo mBatteryAudio; |
| |
| // Collect info of the codec usage from media player and media recorder |
| virtual void addBatteryData(uint32_t params); |
| // API for the Battery app to pull the data of codecs usage |
| virtual status_t pullBatteryData(Parcel* reply); |
| private: |
| |
| class Client : public BnMediaPlayer { |
| // IMediaPlayer interface |
| virtual void disconnect(); |
| virtual status_t setVideoSurfaceTexture( |
| const sp<IGraphicBufferProducer>& bufferProducer); |
| virtual status_t prepareAsync(); |
| virtual status_t start(); |
| virtual status_t stop(); |
| virtual status_t pause(); |
| virtual status_t isPlaying(bool* state); |
| virtual status_t seekTo(int msec); |
| virtual status_t getCurrentPosition(int* msec); |
| virtual status_t getDuration(int* msec); |
| virtual status_t reset(); |
| virtual status_t setAudioStreamType(audio_stream_type_t type); |
| virtual status_t setLooping(int loop); |
| virtual status_t setVolume(float leftVolume, float rightVolume); |
| virtual status_t invoke(const Parcel& request, Parcel *reply); |
| virtual status_t setMetadataFilter(const Parcel& filter); |
| virtual status_t getMetadata(bool update_only, |
| bool apply_filter, |
| Parcel *reply); |
| virtual status_t setAuxEffectSendLevel(float level); |
| virtual status_t attachAuxEffect(int effectId); |
| virtual status_t setParameter(int key, const Parcel &request); |
| virtual status_t getParameter(int key, Parcel *reply); |
| virtual status_t setRetransmitEndpoint(const struct sockaddr_in* endpoint); |
| virtual status_t getRetransmitEndpoint(struct sockaddr_in* endpoint); |
| virtual status_t setNextPlayer(const sp<IMediaPlayer>& player); |
| |
| sp<MediaPlayerBase> createPlayer(player_type playerType); |
| |
| virtual status_t setDataSource( |
| const sp<IMediaHTTPService> &httpService, |
| const char *url, |
| const KeyedVector<String8, String8> *headers); |
| |
| virtual status_t setDataSource(int fd, int64_t offset, int64_t length); |
| |
| virtual status_t setDataSource(const sp<IStreamSource> &source); |
| |
| sp<MediaPlayerBase> setDataSource_pre(player_type playerType); |
| void setDataSource_post(const sp<MediaPlayerBase>& p, |
| status_t status); |
| |
| static void notify(void* cookie, int msg, |
| int ext1, int ext2, const Parcel *obj); |
| |
| pid_t pid() const { return mPid; } |
| virtual status_t dump(int fd, const Vector<String16>& args) const; |
| |
| int getAudioSessionId() { return mAudioSessionId; } |
| |
| private: |
| friend class MediaPlayerService; |
| Client( const sp<MediaPlayerService>& service, |
| pid_t pid, |
| int32_t connId, |
| const sp<IMediaPlayerClient>& client, |
| int audioSessionId, |
| uid_t uid); |
| Client(); |
| virtual ~Client(); |
| |
| void deletePlayer(); |
| |
| sp<MediaPlayerBase> getPlayer() const { Mutex::Autolock lock(mLock); return mPlayer; } |
| |
| |
| |
| // @param type Of the metadata to be tested. |
| // @return true if the metadata should be dropped according to |
| // the filters. |
| bool shouldDropMetadata(media::Metadata::Type type) const; |
| |
| // Add a new element to the set of metadata updated. Noop if |
| // the element exists already. |
| // @param type Of the metadata to be recorded. |
| void addNewMetadataUpdate(media::Metadata::Type type); |
| |
| // Disconnect from the currently connected ANativeWindow. |
| void disconnectNativeWindow(); |
| |
| status_t setAudioAttributes_l(const Parcel &request); |
| |
| mutable Mutex mLock; |
| sp<MediaPlayerBase> mPlayer; |
| sp<MediaPlayerService> mService; |
| sp<IMediaPlayerClient> mClient; |
| sp<AudioOutput> mAudioOutput; |
| pid_t mPid; |
| status_t mStatus; |
| bool mLoop; |
| int32_t mConnId; |
| int mAudioSessionId; |
| audio_attributes_t * mAudioAttributes; |
| uid_t mUID; |
| sp<ANativeWindow> mConnectedWindow; |
| sp<IBinder> mConnectedWindowBinder; |
| struct sockaddr_in mRetransmitEndpoint; |
| bool mRetransmitEndpointValid; |
| sp<Client> mNextClient; |
| |
| // Metadata filters. |
| media::Metadata::Filter mMetadataAllow; // protected by mLock |
| media::Metadata::Filter mMetadataDrop; // protected by mLock |
| |
| // Metadata updated. For each MEDIA_INFO_METADATA_UPDATE |
| // notification we try to update mMetadataUpdated which is a |
| // set: no duplicate. |
| // getMetadata clears this set. |
| media::Metadata::Filter mMetadataUpdated; // protected by mLock |
| |
| #if CALLBACK_ANTAGONIZER |
| Antagonizer* mAntagonizer; |
| #endif |
| }; // Client |
| |
| // ---------------------------------------------------------------------------- |
| |
| MediaPlayerService(); |
| virtual ~MediaPlayerService(); |
| |
| mutable Mutex mLock; |
| SortedVector< wp<Client> > mClients; |
| SortedVector< wp<MediaRecorderClient> > mMediaRecorderClients; |
| int32_t mNextConnId; |
| sp<IOMX> mOMX; |
| sp<ICrypto> mCrypto; |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| |
| }; // namespace android |
| |
| #endif // ANDROID_MEDIAPLAYERSERVICE_H |
