/* * Copyright (C) 2007 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_MEDIAPLAYERINTERFACE_H #define ANDROID_MEDIAPLAYERINTERFACE_H #ifdef __cplusplus #include #include #include #include #include #include #include #include #include #include #include // Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is // global, and not in android:: struct sockaddr_in; namespace android { class DataSource; class Parcel; class Surface; class IGraphicBufferProducer; template class SortedVector; enum player_type { STAGEFRIGHT_PLAYER = 3, NU_PLAYER = 4, // Test players are available only in the 'test' and 'eng' builds. // The shared library with the test player is passed passed as an // argument to the 'test:' url in the setDataSource call. TEST_PLAYER = 5, }; #define DEFAULT_AUDIOSINK_BUFFERCOUNT 4 #define DEFAULT_AUDIOSINK_BUFFERSIZE 1200 #define DEFAULT_AUDIOSINK_SAMPLERATE 44100 // when the channel mask isn't known, use the channel count to derive a mask in AudioSink::open() #define CHANNEL_MASK_USE_CHANNEL_ORDER 0 // duration below which we do not allow deep audio buffering #define AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US 5000000 // callback mechanism for passing messages to MediaPlayer object typedef void (*notify_callback_f)(void* cookie, int msg, int ext1, int ext2, const Parcel *obj); // abstract base class - use MediaPlayerInterface class MediaPlayerBase : public RefBase { public: // AudioSink: abstraction layer for audio output class AudioSink : public RefBase { public: enum cb_event_t { CB_EVENT_FILL_BUFFER, // Request to write more data to buffer. CB_EVENT_STREAM_END, // Sent after all the buffers queued in AF and HW are played // back (after stop is called) CB_EVENT_TEAR_DOWN // The AudioTrack was invalidated due to use case change: // Need to re-evaluate offloading options }; // Callback returns the number of bytes actually written to the buffer. typedef size_t (*AudioCallback)( AudioSink *audioSink, void *buffer, size_t size, void *cookie, cb_event_t event); virtual ~AudioSink() {} virtual bool ready() const = 0; // audio output is open and ready virtual ssize_t bufferSize() const = 0; virtual ssize_t frameCount() const = 0; virtual ssize_t channelCount() const = 0; virtual ssize_t frameSize() const = 0; virtual uint32_t latency() const = 0; virtual float msecsPerFrame() const = 0; virtual status_t getPosition(uint32_t *position) const = 0; virtual status_t getTimestamp(AudioTimestamp &ts) const = 0; virtual int64_t getPlayedOutDurationUs(int64_t nowUs) const = 0; virtual status_t getFramesWritten(uint32_t *frameswritten) const = 0; virtual audio_session_t getSessionId() const = 0; virtual audio_stream_type_t getAudioStreamType() const = 0; virtual uint32_t getSampleRate() const = 0; virtual int64_t getBufferDurationInUs() const = 0; // If no callback is specified, use the "write" API below to submit // audio data. virtual status_t open( uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, audio_format_t format=AUDIO_FORMAT_PCM_16_BIT, int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT, AudioCallback cb = NULL, void *cookie = NULL, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL, bool doNotReconnect = false, uint32_t suggestedFrameCount = 0) = 0; virtual status_t start() = 0; /* Input parameter |size| is in byte units stored in |buffer|. * Data is copied over and actual number of bytes written (>= 0) * is returned, or no data is copied and a negative status code * is returned (even when |blocking| is true). * When |blocking| is false, AudioSink will immediately return after * part of or full |buffer| is copied over. * When |blocking| is true, AudioSink will wait to copy the entire * buffer, unless an error occurs or the copy operation is * prematurely stopped. */ virtual ssize_t write(const void* buffer, size_t size, bool blocking = true) = 0; virtual void stop() = 0; virtual void flush() = 0; virtual void pause() = 0; virtual void close() = 0; virtual status_t setPlaybackRate(const AudioPlaybackRate& rate) = 0; virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */) = 0; virtual bool needsTrailingPadding() { return true; } virtual status_t setParameters(const String8& /* keyValuePairs */) { return NO_ERROR; } virtual String8 getParameters(const String8& /* keys */) { return String8::empty(); } }; MediaPlayerBase() : mCookie(0), mNotify(0) {} virtual ~MediaPlayerBase() {} virtual status_t initCheck() = 0; virtual bool hardwareOutput() = 0; virtual status_t setUID(uid_t /* uid */) { return INVALID_OPERATION; } virtual status_t setDataSource( const sp &httpService, const char *url, const KeyedVector *headers = NULL) = 0; virtual status_t setDataSource(int fd, int64_t offset, int64_t length) = 0; virtual status_t setDataSource(const sp& /* source */) { return INVALID_OPERATION; } virtual status_t setDataSource(const sp& /* source */) { return INVALID_OPERATION; } // pass the buffered IGraphicBufferProducer to the media player service virtual status_t setVideoSurfaceTexture( const sp& bufferProducer) = 0; virtual status_t prepare() = 0; virtual status_t prepareAsync() = 0; virtual status_t start() = 0; virtual status_t stop() = 0; virtual status_t pause() = 0; virtual bool isPlaying() = 0; virtual status_t setPlaybackSettings(const AudioPlaybackRate& rate) { // by default, players only support setting rate to the default if (!isAudioPlaybackRateEqual(rate, AUDIO_PLAYBACK_RATE_DEFAULT)) { return BAD_VALUE; } return OK; } virtual status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) { *rate = AUDIO_PLAYBACK_RATE_DEFAULT; return OK; } virtual status_t setSyncSettings(const AVSyncSettings& sync, float /* videoFps */) { // By default, players only support setting sync source to default; all other sync // settings are ignored. There is no requirement for getters to return set values. if (sync.mSource != AVSYNC_SOURCE_DEFAULT) { return BAD_VALUE; } return OK; } virtual status_t getSyncSettings( AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */) { *sync = AVSyncSettings(); *videoFps = -1.f; return OK; } virtual status_t seekTo(int msec) = 0; virtual status_t getCurrentPosition(int *msec) = 0; virtual status_t getDuration(int *msec) = 0; virtual status_t reset() = 0; virtual status_t setLooping(int loop) = 0; virtual player_type playerType() = 0; virtual status_t setParameter(int key, const Parcel &request) = 0; virtual status_t getParameter(int key, Parcel *reply) = 0; // default no-op implementation of optional extensions virtual status_t setRetransmitEndpoint(const struct sockaddr_in* /* endpoint */) { return INVALID_OPERATION; } virtual status_t getRetransmitEndpoint(struct sockaddr_in* /* endpoint */) { return INVALID_OPERATION; } virtual status_t setNextPlayer(const sp& /* next */) { return OK; } // Invoke a generic method on the player by using opaque parcels // for the request and reply. // // @param request Parcel that is positioned at the start of the // data sent by the java layer. // @param[out] reply Parcel to hold the reply data. Cannot be null. // @return OK if the call was successful. virtual status_t invoke(const Parcel& request, Parcel *reply) = 0; // The Client in the MetadataPlayerService calls this method on // the native player to retrieve all or a subset of metadata. // // @param ids SortedList of metadata ID to be fetch. If empty, all // the known metadata should be returned. // @param[inout] records Parcel where the player appends its metadata. // @return OK if the call was successful. virtual status_t getMetadata(const media::Metadata::Filter& /* ids */, Parcel* /* records */) { return INVALID_OPERATION; }; void setNotifyCallback( void* cookie, notify_callback_f notifyFunc) { Mutex::Autolock autoLock(mNotifyLock); mCookie = cookie; mNotify = notifyFunc; } void sendEvent(int msg, int ext1=0, int ext2=0, const Parcel *obj=NULL) { notify_callback_f notifyCB; void* cookie; { Mutex::Autolock autoLock(mNotifyLock); notifyCB = mNotify; cookie = mCookie; } if (notifyCB) notifyCB(cookie, msg, ext1, ext2, obj); } virtual status_t dump(int /* fd */, const Vector& /* args */) const { return INVALID_OPERATION; } private: friend class MediaPlayerService; Mutex mNotifyLock; void* mCookie; notify_callback_f mNotify; }; // Implement this class for media players that use the AudioFlinger software mixer class MediaPlayerInterface : public MediaPlayerBase { public: virtual ~MediaPlayerInterface() { } virtual bool hardwareOutput() { return false; } virtual void setAudioSink(const sp& audioSink) { mAudioSink = audioSink; } protected: sp mAudioSink; }; // Implement this class for media players that output audio directly to hardware class MediaPlayerHWInterface : public MediaPlayerBase { public: virtual ~MediaPlayerHWInterface() {} virtual bool hardwareOutput() { return true; } virtual status_t setVolume(float leftVolume, float rightVolume) = 0; virtual status_t setAudioStreamType(audio_stream_type_t streamType) = 0; }; }; // namespace android #endif // __cplusplus #endif // ANDROID_MEDIAPLAYERINTERFACE_H