1 /* 2 * Copyright 2017 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #ifndef ANDROID_MEDIAPLAYER2INTERFACE_H 18 #define ANDROID_MEDIAPLAYER2INTERFACE_H 19 20 #ifdef __cplusplus 21 22 #include <sys/types.h> 23 #include <utils/Errors.h> 24 #include <utils/String8.h> 25 #include <utils/RefBase.h> 26 27 #include <media/AVSyncSettings.h> 28 #include <media/AudioResamplerPublic.h> 29 #include <media/AudioSystem.h> 30 #include <media/AudioTimestamp.h> 31 #include <media/BufferingSettings.h> 32 #include <media/Metadata.h> 33 #include <media/stagefright/foundation/AHandler.h> 34 #include <mediaplayer2/MediaPlayer2Types.h> 35 36 // Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is 37 // global, and not in android:: 38 struct sockaddr_in; 39 40 namespace android { 41 42 struct DataSourceDesc; 43 class Parcel; 44 struct ANativeWindowWrapper; 45 46 #define DEFAULT_AUDIOSINK_BUFFERCOUNT 4 47 #define DEFAULT_AUDIOSINK_BUFFERSIZE 1200 48 #define DEFAULT_AUDIOSINK_SAMPLERATE 44100 49 50 // when the channel mask isn't known, use the channel count to derive a mask in AudioSink::open() 51 #define CHANNEL_MASK_USE_CHANNEL_ORDER 0 52 53 // duration below which we do not allow deep audio buffering 54 #define AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US 5000000 55 56 class MediaPlayer2InterfaceListener: public RefBase 57 { 58 public: 59 virtual void notify(int64_t srcId, int msg, int ext1, int ext2, const Parcel *obj) = 0; 60 }; 61 62 class MediaPlayer2Interface : public AHandler { 63 public: 64 // AudioSink: abstraction layer for audio output 65 class AudioSink : public RefBase { 66 public: 67 enum cb_event_t { 68 CB_EVENT_FILL_BUFFER, // Request to write more data to buffer. 69 CB_EVENT_STREAM_END, // Sent after all the buffers queued in AF and HW are played 70 // back (after stop is called) 71 CB_EVENT_TEAR_DOWN // The AudioTrack was invalidated due to use case change: 72 // Need to re-evaluate offloading options 73 }; 74 75 // Callback returns the number of bytes actually written to the buffer. 76 typedef size_t (*AudioCallback)( 77 AudioSink *audioSink, void *buffer, size_t size, void *cookie, cb_event_t event); 78 79 virtual ~AudioSink() {} 80 virtual bool ready() const = 0; // audio output is open and ready 81 virtual ssize_t bufferSize() const = 0; 82 virtual ssize_t frameCount() const = 0; 83 virtual ssize_t channelCount() const = 0; 84 virtual ssize_t frameSize() const = 0; 85 virtual uint32_t latency() const = 0; 86 virtual float msecsPerFrame() const = 0; 87 virtual status_t getPosition(uint32_t *position) const = 0; 88 virtual status_t getTimestamp(AudioTimestamp &ts) const = 0; 89 virtual int64_t getPlayedOutDurationUs(int64_t nowUs) const = 0; 90 virtual status_t getFramesWritten(uint32_t *frameswritten) const = 0; 91 virtual audio_session_t getSessionId() const = 0; 92 virtual audio_stream_type_t getAudioStreamType() const = 0; 93 virtual uint32_t getSampleRate() const = 0; 94 virtual int64_t getBufferDurationInUs() const = 0; 95 96 // If no callback is specified, use the "write" API below to submit 97 // audio data. 98 virtual status_t open( 99 uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask, 100 audio_format_t format=AUDIO_FORMAT_PCM_16_BIT, 101 int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT, 102 AudioCallback cb = NULL, 103 void *cookie = NULL, 104 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 105 const audio_offload_info_t *offloadInfo = NULL, 106 bool doNotReconnect = false, 107 uint32_t suggestedFrameCount = 0) = 0; 108 109 virtual status_t start() = 0; 110 111 /* Input parameter |size| is in byte units stored in |buffer|. 112 * Data is copied over and actual number of bytes written (>= 0) 113 * is returned, or no data is copied and a negative status code 114 * is returned (even when |blocking| is true). 115 * When |blocking| is false, AudioSink will immediately return after 116 * part of or full |buffer| is copied over. 117 * When |blocking| is true, AudioSink will wait to copy the entire 118 * buffer, unless an error occurs or the copy operation is 119 * prematurely stopped. 120 */ 121 virtual ssize_t write(const void* buffer, size_t size, bool blocking = true) = 0; 122 123 virtual void stop() = 0; 124 virtual void flush() = 0; 125 virtual void pause() = 0; 126 virtual void close() = 0; 127 128 virtual status_t setPlaybackRate(const AudioPlaybackRate& rate) = 0; 129 virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */) = 0; 130 virtual bool needsTrailingPadding() { 131 return true; 132 } 133 134 virtual status_t setParameters(const String8& /* keyValuePairs */) { 135 return NO_ERROR; 136 } 137 virtual String8 getParameters(const String8& /* keys */) { 138 return String8::empty(); 139 } 140 141 // AudioRouting 142 virtual status_t setOutputDevice(audio_port_handle_t deviceId); 143 virtual status_t getRoutedDeviceId(audio_port_handle_t* deviceId); 144 virtual status_t enableAudioDeviceCallback(bool enabled); 145 }; 146 147 MediaPlayer2Interface() : mListener(NULL) { } 148 virtual ~MediaPlayer2Interface() { } 149 virtual status_t initCheck() = 0; 150 151 virtual void setAudioSink(const sp<AudioSink>& audioSink) { 152 mAudioSink = audioSink; 153 } 154 155 virtual status_t setDataSource(const sp<DataSourceDesc> &dsd) = 0; 156 157 virtual status_t prepareNextDataSource(const sp<DataSourceDesc> &dsd) = 0; 158 159 virtual status_t playNextDataSource(int64_t srcId) = 0; 160 161 // pass the buffered native window to the media player service 162 virtual status_t setVideoSurfaceTexture(const sp<ANativeWindowWrapper>& nww) = 0; 163 164 virtual status_t getBufferingSettings(BufferingSettings* buffering /* nonnull */) { 165 *buffering = BufferingSettings(); 166 return OK; 167 } 168 virtual status_t setBufferingSettings(const BufferingSettings& /* buffering */) { 169 return OK; 170 } 171 172 virtual status_t prepareAsync() = 0; 173 virtual status_t start() = 0; 174 virtual status_t stop() = 0; 175 virtual status_t pause() = 0; 176 virtual bool isPlaying() = 0; 177 virtual status_t setPlaybackSettings(const AudioPlaybackRate& rate) { 178 // by default, players only support setting rate to the default 179 if (!isAudioPlaybackRateEqual(rate, AUDIO_PLAYBACK_RATE_DEFAULT)) { 180 return BAD_VALUE; 181 } 182 return OK; 183 } 184 virtual status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) { 185 *rate = AUDIO_PLAYBACK_RATE_DEFAULT; 186 return OK; 187 } 188 virtual status_t setSyncSettings(const AVSyncSettings& sync, float /* videoFps */) { 189 // By default, players only support setting sync source to default; all other sync 190 // settings are ignored. There is no requirement for getters to return set values. 191 if (sync.mSource != AVSYNC_SOURCE_DEFAULT) { 192 return BAD_VALUE; 193 } 194 return OK; 195 } 196 virtual status_t getSyncSettings( 197 AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */) { 198 *sync = AVSyncSettings(); 199 *videoFps = -1.f; 200 return OK; 201 } 202 virtual status_t seekTo( 203 int64_t msec, MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) = 0; 204 virtual status_t getCurrentPosition(int64_t *msec) = 0; 205 virtual status_t getDuration(int64_t *msec) = 0; 206 virtual status_t reset() = 0; 207 virtual status_t notifyAt(int64_t /* mediaTimeUs */) { 208 return INVALID_OPERATION; 209 } 210 virtual status_t setLooping(int loop) = 0; 211 virtual status_t setParameter(int key, const Parcel &request) = 0; 212 virtual status_t getParameter(int key, Parcel *reply) = 0; 213 214 // Invoke a generic method on the player by using opaque parcels 215 // for the request and reply. 216 // 217 // @param request Parcel that is positioned at the start of the 218 // data sent by the java layer. 219 // @param[out] reply Parcel to hold the reply data. Cannot be null. 220 // @return OK if the call was successful. 221 virtual status_t invoke(const Parcel& request, Parcel *reply) = 0; 222 223 // The Client in the MetadataPlayerService calls this method on 224 // the native player to retrieve all or a subset of metadata. 225 // 226 // @param ids SortedList of metadata ID to be fetch. If empty, all 227 // the known metadata should be returned. 228 // @param[inout] records Parcel where the player appends its metadata. 229 // @return OK if the call was successful. 230 virtual status_t getMetadata(const media::Metadata::Filter& /* ids */, 231 Parcel* /* records */) { 232 return INVALID_OPERATION; 233 }; 234 235 void setListener(const sp<MediaPlayer2InterfaceListener> &listener) { 236 Mutex::Autolock autoLock(mListenerLock); 237 mListener = listener; 238 } 239 240 void sendEvent(int64_t srcId, int msg, int ext1=0, int ext2=0, const Parcel *obj=NULL) { 241 sp<MediaPlayer2InterfaceListener> listener; 242 { 243 Mutex::Autolock autoLock(mListenerLock); 244 listener = mListener; 245 } 246 247 if (listener) { 248 listener->notify(srcId, msg, ext1, ext2, obj); 249 } 250 } 251 252 virtual status_t dump(int /* fd */, const Vector<String16>& /* args */) const { 253 return INVALID_OPERATION; 254 } 255 256 virtual void onMessageReceived(const sp<AMessage> & /* msg */) override { } 257 258 // Modular DRM 259 virtual status_t prepareDrm(const uint8_t /* uuid */[16], 260 const Vector<uint8_t>& /* drmSessionId */) { 261 return INVALID_OPERATION; 262 } 263 virtual status_t releaseDrm() { 264 return INVALID_OPERATION; 265 } 266 267 protected: 268 sp<AudioSink> mAudioSink; 269 270 private: 271 Mutex mListenerLock; 272 sp<MediaPlayer2InterfaceListener> mListener; 273 }; 274 275 }; // namespace android 276 277 #endif // __cplusplus 278 279 280 #endif // ANDROID_MEDIAPLAYER2INTERFACE_H 281
