| /* |
| * 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 <sys/types.h> |
| #include <utils/Errors.h> |
| #include <utils/KeyedVector.h> |
| #include <utils/String8.h> |
| #include <utils/RefBase.h> |
| |
| #include <media/mediaplayer.h> |
| #include <media/AudioResamplerPublic.h> |
| #include <media/AudioTimestamp.h> |
| #include <media/AVSyncSettings.h> |
| #include <media/BufferingSettings.h> |
| #include <media/Metadata.h> |
| |
| // 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<typename T> 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 AUDIO_CHANNEL_NONE |
| |
| // duration below which we do not allow deep audio buffering |
| #define AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US 5000000 |
| |
| // abstract base class - use MediaPlayerInterface |
| class MediaPlayerBase : public RefBase |
| { |
| public: |
| // callback mechanism for passing messages to MediaPlayer object |
| class Listener : public RefBase { |
| public: |
| virtual void notify(int msg, int ext1, int ext2, const Parcel *obj) = 0; |
| virtual ~Listener() {} |
| }; |
| |
| // 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; |
| virtual audio_output_flags_t getFlags() 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(); } |
| |
| virtual media::VolumeShaper::Status applyVolumeShaper( |
| const sp<media::VolumeShaper::Configuration>& configuration, |
| const sp<media::VolumeShaper::Operation>& operation) = 0; |
| virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) = 0; |
| |
| // AudioRouting |
| virtual status_t setOutputDevice(audio_port_handle_t deviceId) = 0; |
| virtual status_t getRoutedDeviceId(audio_port_handle_t* deviceId) = 0; |
| virtual status_t enableAudioDeviceCallback(bool enabled) = 0; |
| }; |
| |
| MediaPlayerBase() {} |
| 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<IMediaHTTPService> &httpService, |
| const char *url, |
| const KeyedVector<String8, String8> *headers = NULL) = 0; |
| |
| virtual status_t setDataSource(int fd, int64_t offset, int64_t length) = 0; |
| |
| virtual status_t setDataSource(const sp<IStreamSource>& /* source */) { |
| return INVALID_OPERATION; |
| } |
| |
| virtual status_t setDataSource(const sp<DataSource>& /* source */) { |
| return INVALID_OPERATION; |
| } |
| |
| virtual status_t setDataSource(const String8& /* rtpParams */) { |
| return INVALID_OPERATION; |
| } |
| |
| // pass the buffered IGraphicBufferProducer to the media player service |
| virtual status_t setVideoSurfaceTexture( |
| const sp<IGraphicBufferProducer>& bufferProducer) = 0; |
| |
| virtual status_t getBufferingSettings( |
| BufferingSettings* buffering /* nonnull */) { |
| *buffering = BufferingSettings(); |
| return OK; |
| } |
| virtual status_t setBufferingSettings(const BufferingSettings& /* buffering */) { |
| return OK; |
| } |
| |
| 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, MediaPlayerSeekMode mode = MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC) = 0; |
| virtual status_t getCurrentPosition(int *msec) = 0; |
| virtual status_t getDuration(int *msec) = 0; |
| virtual status_t reset() = 0; |
| virtual status_t notifyAt(int64_t /* mediaTimeUs */) { |
| return INVALID_OPERATION; |
| } |
| 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<MediaPlayerBase>& /* 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( |
| const sp<Listener> &listener) { |
| Mutex::Autolock autoLock(mNotifyLock); |
| mListener = listener; |
| } |
| |
| void sendEvent(int msg, int ext1=0, int ext2=0, |
| const Parcel *obj=NULL) { |
| sp<Listener> listener; |
| { |
| Mutex::Autolock autoLock(mNotifyLock); |
| listener = mListener; |
| } |
| |
| if (listener != NULL) { |
| listener->notify(msg, ext1, ext2, obj); |
| } |
| } |
| |
| virtual status_t dump(int /* fd */, const Vector<String16>& /* args */) const { |
| return INVALID_OPERATION; |
| } |
| |
| // Modular DRM |
| virtual status_t prepareDrm(const uint8_t /* uuid */[16], const Vector<uint8_t>& /* drmSessionId */) { |
| return INVALID_OPERATION; |
| } |
| virtual status_t releaseDrm() { |
| return INVALID_OPERATION; |
| } |
| |
| private: |
| friend class MediaPlayerService; |
| |
| Mutex mNotifyLock; |
| sp<Listener> mListener; |
| }; |
| |
| // 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>& audioSink) { mAudioSink = audioSink; } |
| protected: |
| sp<AudioSink> 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 |