/* * Copyright (C) 2010 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. */ #include "sles_allinclusive.h" #include "android_prompts.h" #include "android/android_AudioToCbRenderer.h" #include "android/android_StreamPlayer.h" #include "android/android_LocAVPlayer.h" #include "android/include/AacBqToPcmCbRenderer.h" #include #include #include template class android::KeyedVector ; #define KEY_STREAM_TYPE_PARAMSIZE sizeof(SLint32) #define AUDIOTRACK_MIN_PLAYBACKRATE_PERMILLE 500 #define AUDIOTRACK_MAX_PLAYBACKRATE_PERMILLE 2000 #define MEDIAPLAYER_MIN_PLAYBACKRATE_PERMILLE AUDIOTRACK_MIN_PLAYBACKRATE_PERMILLE #define MEDIAPLAYER_MAX_PLAYBACKRATE_PERMILLE AUDIOTRACK_MAX_PLAYBACKRATE_PERMILLE //----------------------------------------------------------------------------- // FIXME this method will be absorbed into android_audioPlayer_setPlayState() once // bufferqueue and uri/fd playback are moved under the GenericPlayer C++ object SLresult aplayer_setPlayState(const android::sp &ap, SLuint32 playState, AndroidObjectState* pObjState) { SLresult result = SL_RESULT_SUCCESS; AndroidObjectState objState = *pObjState; switch (playState) { case SL_PLAYSTATE_STOPPED: SL_LOGV("setting GenericPlayer to SL_PLAYSTATE_STOPPED"); ap->stop(); break; case SL_PLAYSTATE_PAUSED: SL_LOGV("setting GenericPlayer to SL_PLAYSTATE_PAUSED"); switch(objState) { case ANDROID_UNINITIALIZED: *pObjState = ANDROID_PREPARING; ap->prepare(); break; case ANDROID_PREPARING: break; case ANDROID_READY: ap->pause(); break; default: SL_LOGE(ERROR_PLAYERSETPLAYSTATE_INVALID_OBJECT_STATE_D, playState); result = SL_RESULT_INTERNAL_ERROR; break; } break; case SL_PLAYSTATE_PLAYING: { SL_LOGV("setting GenericPlayer to SL_PLAYSTATE_PLAYING"); switch(objState) { case ANDROID_UNINITIALIZED: *pObjState = ANDROID_PREPARING; ap->prepare(); // intended fall through case ANDROID_PREPARING: // intended fall through case ANDROID_READY: ap->play(); break; default: SL_LOGE(ERROR_PLAYERSETPLAYSTATE_INVALID_OBJECT_STATE_D, playState); result = SL_RESULT_INTERNAL_ERROR; break; } } break; default: // checked by caller, should not happen SL_LOGE(ERROR_SHOULDNT_BE_HERE_S, "aplayer_setPlayState"); result = SL_RESULT_INTERNAL_ERROR; break; } return result; } //----------------------------------------------------------------------------- // Callback associated with a AudioToCbRenderer of an SL ES AudioPlayer that gets its data // from a URI or FD, to write the decoded audio data to a buffer queue static size_t adecoder_writeToBufferQueue(const uint8_t *data, size_t size, CAudioPlayer* ap) { if (!android::CallbackProtector::enterCbIfOk(ap->mCallbackProtector)) { // it is not safe to enter the callback (the player is about to go away) return 0; } size_t sizeConsumed = 0; SL_LOGD("received %d bytes from decoder", size); slBufferQueueCallback callback = NULL; void * callbackPContext = NULL; // push decoded data to the buffer queue object_lock_exclusive(&ap->mObject); if (ap->mBufferQueue.mState.count != 0) { assert(ap->mBufferQueue.mFront != ap->mBufferQueue.mRear); BufferHeader *oldFront = ap->mBufferQueue.mFront; BufferHeader *newFront = &oldFront[1]; uint8_t *pDest = (uint8_t *)oldFront->mBuffer + ap->mBufferQueue.mSizeConsumed; if (ap->mBufferQueue.mSizeConsumed + size < oldFront->mSize) { // room to consume the whole or rest of the decoded data in one shot ap->mBufferQueue.mSizeConsumed += size; // consume data but no callback to the BufferQueue interface here memcpy(pDest, data, size); sizeConsumed = size; } else { // push as much as possible of the decoded data into the buffer queue sizeConsumed = oldFront->mSize - ap->mBufferQueue.mSizeConsumed; // the buffer at the head of the buffer queue is full, update the state ap->mBufferQueue.mSizeConsumed = 0; if (newFront == &ap->mBufferQueue.mArray[ap->mBufferQueue.mNumBuffers + 1]) { newFront = ap->mBufferQueue.mArray; } ap->mBufferQueue.mFront = newFront; ap->mBufferQueue.mState.count--; ap->mBufferQueue.mState.playIndex++; // consume data memcpy(pDest, data, sizeConsumed); // data has been copied to the buffer, and the buffer queue state has been updated // we will notify the client if applicable callback = ap->mBufferQueue.mCallback; // save callback data callbackPContext = ap->mBufferQueue.mContext; } } else { // no available buffers in the queue to write the decoded data sizeConsumed = 0; } object_unlock_exclusive(&ap->mObject); // notify client if (NULL != callback) { (*callback)(&ap->mBufferQueue.mItf, callbackPContext); } ap->mCallbackProtector->exitCb(); return sizeConsumed; } //----------------------------------------------------------------------------- #define LEFT_CHANNEL_MASK 0x1 << 0 #define RIGHT_CHANNEL_MASK 0x1 << 1 void android_audioPlayer_volumeUpdate(CAudioPlayer* ap) { assert(ap != NULL); // the source's channel count, where zero means unknown SLuint8 channelCount = ap->mNumChannels; // whether each channel is audible bool leftAudibilityFactor, rightAudibilityFactor; // mute has priority over solo if (channelCount >= STEREO_CHANNELS) { if (ap->mMuteMask & LEFT_CHANNEL_MASK) { // left muted leftAudibilityFactor = false; } else { // left not muted if (ap->mSoloMask & LEFT_CHANNEL_MASK) { // left soloed leftAudibilityFactor = true; } else { // left not soloed if (ap->mSoloMask & RIGHT_CHANNEL_MASK) { // right solo silences left leftAudibilityFactor = false; } else { // left and right are not soloed, and left is not muted leftAudibilityFactor = true; } } } if (ap->mMuteMask & RIGHT_CHANNEL_MASK) { // right muted rightAudibilityFactor = false; } else { // right not muted if (ap->mSoloMask & RIGHT_CHANNEL_MASK) { // right soloed rightAudibilityFactor = true; } else { // right not soloed if (ap->mSoloMask & LEFT_CHANNEL_MASK) { // left solo silences right rightAudibilityFactor = false; } else { // left and right are not soloed, and right is not muted rightAudibilityFactor = true; } } } // channel mute and solo are ignored for mono and unknown channel count sources } else { leftAudibilityFactor = true; rightAudibilityFactor = true; } // compute volumes without setting const bool audibilityFactors[2] = {leftAudibilityFactor, rightAudibilityFactor}; float volumes[2]; android_player_volumeUpdate(volumes, &ap->mVolume, channelCount, ap->mAmplFromDirectLevel, audibilityFactors); float leftVol = volumes[0], rightVol = volumes[1]; // set volume on the underlying media player or audio track if (ap->mAPlayer != 0) { ap->mAPlayer->setVolume(leftVol, rightVol); } else if (ap->mAudioTrack != 0) { ap->mAudioTrack->setVolume(leftVol, rightVol); } // changes in the AudioPlayer volume must be reflected in the send level: // in SLEffectSendItf or in SLAndroidEffectSendItf? // FIXME replace interface test by an internal API once we have one. if (NULL != ap->mEffectSend.mItf) { for (unsigned int i=0 ; imEffectSend.mEnableLevels[i].mEnable) { android_fxSend_setSendLevel(ap, ap->mEffectSend.mEnableLevels[i].mSendLevel + ap->mVolume.mLevel); // there's a single aux bus on Android, so we can stop looking once the first // aux effect is found. break; } } } else if (NULL != ap->mAndroidEffectSend.mItf) { android_fxSend_setSendLevel(ap, ap->mAndroidEffectSend.mSendLevel + ap->mVolume.mLevel); } } // Called by android_audioPlayer_volumeUpdate and android_mediaPlayer_volumeUpdate to compute // volumes, but setting volumes is handled by the caller. void android_player_volumeUpdate(float *pVolumes /*[2]*/, const IVolume *volumeItf, unsigned channelCount, float amplFromDirectLevel, const bool *audibilityFactors /*[2]*/) { assert(pVolumes != NULL); assert(volumeItf != NULL); // OK for audibilityFactors to be NULL bool leftAudibilityFactor, rightAudibilityFactor; // apply player mute factor // note that AudioTrack has mute() but not MediaPlayer, so it's easier to use volume // to mute for both rather than calling mute() for AudioTrack // player is muted if (volumeItf->mMute) { leftAudibilityFactor = false; rightAudibilityFactor = false; // player isn't muted, and channel mute/solo audibility factors are available (AudioPlayer) } else if (audibilityFactors != NULL) { leftAudibilityFactor = audibilityFactors[0]; rightAudibilityFactor = audibilityFactors[1]; // player isn't muted, and channel mute/solo audibility factors aren't available (MediaPlayer) } else { leftAudibilityFactor = true; rightAudibilityFactor = true; } // compute amplification as the combination of volume level and stereo position // amplification (or attenuation) from volume level float amplFromVolLevel = sles_to_android_amplification(volumeItf->mLevel); // amplification from direct level (changed in SLEffectSendtItf and SLAndroidEffectSendItf) float leftVol = amplFromVolLevel * amplFromDirectLevel; float rightVol = leftVol; // amplification from stereo position if (volumeItf->mEnableStereoPosition) { // Left/right amplification (can be attenuations) factors derived for the StereoPosition float amplFromStereoPos[STEREO_CHANNELS]; // panning law depends on content channel count: mono to stereo panning vs stereo balance if (1 == channelCount) { // mono to stereo panning double theta = (1000+volumeItf->mStereoPosition)*M_PI_4/1000.0f; // 0 <= theta <= Pi/2 amplFromStereoPos[0] = cos(theta); amplFromStereoPos[1] = sin(theta); // channel count is 0 (unknown), 2 (stereo), or > 2 (multi-channel) } else { // stereo balance if (volumeItf->mStereoPosition > 0) { amplFromStereoPos[0] = (1000-volumeItf->mStereoPosition)/1000.0f; amplFromStereoPos[1] = 1.0f; } else { amplFromStereoPos[0] = 1.0f; amplFromStereoPos[1] = (1000+volumeItf->mStereoPosition)/1000.0f; } } leftVol *= amplFromStereoPos[0]; rightVol *= amplFromStereoPos[1]; } // apply audibility factors if (!leftAudibilityFactor) { leftVol = 0.0; } if (!rightAudibilityFactor) { rightVol = 0.0; } // return the computed volumes pVolumes[0] = leftVol; pVolumes[1] = rightVol; } //----------------------------------------------------------------------------- void audioTrack_handleMarker_lockPlay(CAudioPlayer* ap) { //SL_LOGV("received event EVENT_MARKER from AudioTrack"); slPlayCallback callback = NULL; void* callbackPContext = NULL; interface_lock_shared(&ap->mPlay); callback = ap->mPlay.mCallback; callbackPContext = ap->mPlay.mContext; interface_unlock_shared(&ap->mPlay); if (NULL != callback) { // getting this event implies SL_PLAYEVENT_HEADATMARKER was set in the event mask (*callback)(&ap->mPlay.mItf, callbackPContext, SL_PLAYEVENT_HEADATMARKER); } } //----------------------------------------------------------------------------- void audioTrack_handleNewPos_lockPlay(CAudioPlayer* ap) { //SL_LOGV("received event EVENT_NEW_POS from AudioTrack"); slPlayCallback callback = NULL; void* callbackPContext = NULL; interface_lock_shared(&ap->mPlay); callback = ap->mPlay.mCallback; callbackPContext = ap->mPlay.mContext; interface_unlock_shared(&ap->mPlay); if (NULL != callback) { // getting this event implies SL_PLAYEVENT_HEADATNEWPOS was set in the event mask (*callback)(&ap->mPlay.mItf, callbackPContext, SL_PLAYEVENT_HEADATNEWPOS); } } //----------------------------------------------------------------------------- void audioTrack_handleUnderrun_lockPlay(CAudioPlayer* ap) { slPlayCallback callback = NULL; void* callbackPContext = NULL; interface_lock_shared(&ap->mPlay); callback = ap->mPlay.mCallback; callbackPContext = ap->mPlay.mContext; bool headStalled = (ap->mPlay.mEventFlags & SL_PLAYEVENT_HEADSTALLED) != 0; interface_unlock_shared(&ap->mPlay); if ((NULL != callback) && headStalled) { (*callback)(&ap->mPlay.mItf, callbackPContext, SL_PLAYEVENT_HEADSTALLED); } } //----------------------------------------------------------------------------- /** * post-condition: play state of AudioPlayer is SL_PLAYSTATE_PAUSED if setPlayStateToPaused is true * * note: a conditional flag, setPlayStateToPaused, is used here to specify whether the play state * needs to be changed when the player reaches the end of the content to play. This is * relative to what the specification describes for buffer queues vs the * SL_PLAYEVENT_HEADATEND event. In the OpenSL ES specification 1.0.1: * - section 8.12 SLBufferQueueItf states "In the case of starvation due to insufficient * buffers in the queue, the playing of audio data stops. The player remains in the * SL_PLAYSTATE_PLAYING state." * - section 9.2.31 SL_PLAYEVENT states "SL_PLAYEVENT_HEADATEND Playback head is at the end * of the current content and the player has paused." */ void audioPlayer_dispatch_headAtEnd_lockPlay(CAudioPlayer *ap, bool setPlayStateToPaused, bool needToLock) { //SL_LOGV("ap=%p, setPlayStateToPaused=%d, needToLock=%d", ap, setPlayStateToPaused, // needToLock); slPlayCallback playCallback = NULL; void * playContext = NULL; // SLPlayItf callback or no callback? if (needToLock) { interface_lock_exclusive(&ap->mPlay); } if (ap->mPlay.mEventFlags & SL_PLAYEVENT_HEADATEND) { playCallback = ap->mPlay.mCallback; playContext = ap->mPlay.mContext; } if (setPlayStateToPaused) { ap->mPlay.mState = SL_PLAYSTATE_PAUSED; } if (needToLock) { interface_unlock_exclusive(&ap->mPlay); } // enqueue callback with no lock held if (NULL != playCallback) { #ifndef USE_ASYNCHRONOUS_PLAY_CALLBACK (*playCallback)(&ap->mPlay.mItf, playContext, SL_PLAYEVENT_HEADATEND); #else SLresult result = EnqueueAsyncCallback_ppi(ap, playCallback, &ap->mPlay.mItf, playContext, SL_PLAYEVENT_HEADATEND); if (SL_RESULT_SUCCESS != result) { ALOGW("Callback %p(%p, %p, SL_PLAYEVENT_HEADATEND) dropped", playCallback, &ap->mPlay.mItf, playContext); } #endif } } //----------------------------------------------------------------------------- SLresult audioPlayer_setStreamType(CAudioPlayer* ap, SLint32 type) { SLresult result = SL_RESULT_SUCCESS; SL_LOGV("type %d", type); audio_stream_type_t newStreamType = ANDROID_DEFAULT_OUTPUT_STREAM_TYPE; switch(type) { case SL_ANDROID_STREAM_VOICE: newStreamType = AUDIO_STREAM_VOICE_CALL; break; case SL_ANDROID_STREAM_SYSTEM: newStreamType = AUDIO_STREAM_SYSTEM; break; case SL_ANDROID_STREAM_RING: newStreamType = AUDIO_STREAM_RING; break; case SL_ANDROID_STREAM_MEDIA: newStreamType = AUDIO_STREAM_MUSIC; break; case SL_ANDROID_STREAM_ALARM: newStreamType = AUDIO_STREAM_ALARM; break; case SL_ANDROID_STREAM_NOTIFICATION: newStreamType = AUDIO_STREAM_NOTIFICATION; break; default: SL_LOGE(ERROR_PLAYERSTREAMTYPE_SET_UNKNOWN_TYPE); result = SL_RESULT_PARAMETER_INVALID; break; } // stream type needs to be set before the object is realized // (ap->mAudioTrack is supposed to be NULL until then) if (SL_OBJECT_STATE_UNREALIZED != ap->mObject.mState) { SL_LOGE(ERROR_PLAYERSTREAMTYPE_REALIZED); result = SL_RESULT_PRECONDITIONS_VIOLATED; } else { ap->mStreamType = newStreamType; } return result; } //----------------------------------------------------------------------------- SLresult audioPlayer_getStreamType(CAudioPlayer* ap, SLint32 *pType) { SLresult result = SL_RESULT_SUCCESS; switch(ap->mStreamType) { case AUDIO_STREAM_VOICE_CALL: *pType = SL_ANDROID_STREAM_VOICE; break; case AUDIO_STREAM_SYSTEM: *pType = SL_ANDROID_STREAM_SYSTEM; break; case AUDIO_STREAM_RING: *pType = SL_ANDROID_STREAM_RING; break; case AUDIO_STREAM_DEFAULT: case AUDIO_STREAM_MUSIC: *pType = SL_ANDROID_STREAM_MEDIA; break; case AUDIO_STREAM_ALARM: *pType = SL_ANDROID_STREAM_ALARM; break; case AUDIO_STREAM_NOTIFICATION: *pType = SL_ANDROID_STREAM_NOTIFICATION; break; default: result = SL_RESULT_INTERNAL_ERROR; *pType = SL_ANDROID_STREAM_MEDIA; break; } return result; } //----------------------------------------------------------------------------- void audioPlayer_auxEffectUpdate(CAudioPlayer* ap) { if ((ap->mAudioTrack != 0) && (ap->mAuxEffect != 0)) { android_fxSend_attach(ap, true, ap->mAuxEffect, ap->mVolume.mLevel + ap->mAuxSendLevel); } } //----------------------------------------------------------------------------- /* * returns true if the given data sink is supported by AudioPlayer that doesn't * play to an OutputMix object, false otherwise * * pre-condition: the locator of the audio sink is not SL_DATALOCATOR_OUTPUTMIX */ bool audioPlayer_isSupportedNonOutputMixSink(const SLDataSink* pAudioSink) { bool result = true; const SLuint32 sinkLocatorType = *(SLuint32 *)pAudioSink->pLocator; const SLuint32 sinkFormatType = *(SLuint32 *)pAudioSink->pFormat; switch (sinkLocatorType) { case SL_DATALOCATOR_BUFFERQUEUE: case SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE: if (SL_DATAFORMAT_PCM != sinkFormatType) { SL_LOGE("Unsupported sink format 0x%x, expected SL_DATAFORMAT_PCM", (unsigned)sinkFormatType); result = false; } // it's no use checking the PCM format fields because additional characteristics // such as the number of channels, or sample size are unknown to the player at this stage break; default: SL_LOGE("Unsupported sink locator type 0x%x", (unsigned)sinkLocatorType); result = false; break; } return result; } //----------------------------------------------------------------------------- /* * returns the Android object type if the locator type combinations for the source and sinks * are supported by this implementation, INVALID_TYPE otherwise */ static AndroidObjectType audioPlayer_getAndroidObjectTypeForSourceSink(const CAudioPlayer *ap) { const SLDataSource *pAudioSrc = &ap->mDataSource.u.mSource; const SLDataSink *pAudioSnk = &ap->mDataSink.u.mSink; const SLuint32 sourceLocatorType = *(SLuint32 *)pAudioSrc->pLocator; const SLuint32 sinkLocatorType = *(SLuint32 *)pAudioSnk->pLocator; AndroidObjectType type = INVALID_TYPE; //-------------------------------------- // Sink / source matching check: // the following source / sink combinations are supported // SL_DATALOCATOR_BUFFERQUEUE / SL_DATALOCATOR_OUTPUTMIX // SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE / SL_DATALOCATOR_OUTPUTMIX // SL_DATALOCATOR_URI / SL_DATALOCATOR_OUTPUTMIX // SL_DATALOCATOR_ANDROIDFD / SL_DATALOCATOR_OUTPUTMIX // SL_DATALOCATOR_ANDROIDBUFFERQUEUE / SL_DATALOCATOR_OUTPUTMIX // SL_DATALOCATOR_ANDROIDBUFFERQUEUE / SL_DATALOCATOR_BUFFERQUEUE // SL_DATALOCATOR_URI / SL_DATALOCATOR_BUFFERQUEUE // SL_DATALOCATOR_ANDROIDFD / SL_DATALOCATOR_BUFFERQUEUE // SL_DATALOCATOR_URI / SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE // SL_DATALOCATOR_ANDROIDFD / SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE switch (sinkLocatorType) { case SL_DATALOCATOR_OUTPUTMIX: { switch (sourceLocatorType) { // Buffer Queue to AudioTrack case SL_DATALOCATOR_BUFFERQUEUE: case SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE: type = AUDIOPLAYER_FROM_PCM_BUFFERQUEUE; break; // URI or FD to MediaPlayer case SL_DATALOCATOR_URI: case SL_DATALOCATOR_ANDROIDFD: type = AUDIOPLAYER_FROM_URIFD; break; // Android BufferQueue to MediaPlayer (shared memory streaming) case SL_DATALOCATOR_ANDROIDBUFFERQUEUE: type = AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE; break; default: SL_LOGE("Source data locator 0x%x not supported with SL_DATALOCATOR_OUTPUTMIX sink", (unsigned)sourceLocatorType); break; } } break; case SL_DATALOCATOR_BUFFERQUEUE: case SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE: switch (sourceLocatorType) { // URI or FD decoded to PCM in a buffer queue case SL_DATALOCATOR_URI: case SL_DATALOCATOR_ANDROIDFD: type = AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE; break; // AAC ADTS Android buffer queue decoded to PCM in a buffer queue case SL_DATALOCATOR_ANDROIDBUFFERQUEUE: type = AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE; break; default: SL_LOGE("Source data locator 0x%x not supported with SL_DATALOCATOR_BUFFERQUEUE sink", (unsigned)sourceLocatorType); break; } break; default: SL_LOGE("Sink data locator 0x%x not supported", (unsigned)sinkLocatorType); break; } return type; } //----------------------------------------------------------------------------- /* * Callback associated with an SfPlayer of an SL ES AudioPlayer that gets its data * from a URI or FD, for prepare, prefetch, and play events */ static void sfplayer_handlePrefetchEvent(int event, int data1, int data2, void* user) { // FIXME see similar code and comment in player_handleMediaPlayerEventNotifications if (NULL == user) { return; } CAudioPlayer *ap = (CAudioPlayer *)user; if (!android::CallbackProtector::enterCbIfOk(ap->mCallbackProtector)) { // it is not safe to enter the callback (the track is about to go away) return; } union { char c[sizeof(int)]; int i; } u; u.i = event; SL_LOGV("sfplayer_handlePrefetchEvent(event='%c%c%c%c' (%d), data1=%d, data2=%d, user=%p) from " "SfAudioPlayer", u.c[3], u.c[2], u.c[1], u.c[0], event, data1, data2, user); switch(event) { case android::GenericPlayer::kEventPrepared: { SL_LOGV("Received GenericPlayer::kEventPrepared for CAudioPlayer %p", ap); // assume no callback slPrefetchCallback callback = NULL; void* callbackPContext; SLuint32 events; object_lock_exclusive(&ap->mObject); // mark object as prepared; same state is used for successful or unsuccessful prepare assert(ap->mAndroidObjState == ANDROID_PREPARING); ap->mAndroidObjState = ANDROID_READY; if (PLAYER_SUCCESS == data1) { // Most of successful prepare completion for ap->mAPlayer // is handled by GenericPlayer and its subclasses. } else { // SfPlayer prepare() failed prefetching, there is no event in SLPrefetchStatus to // indicate a prefetch error, so we signal it by sending simultaneously two events: // - SL_PREFETCHEVENT_FILLLEVELCHANGE with a level of 0 // - SL_PREFETCHEVENT_STATUSCHANGE with a status of SL_PREFETCHSTATUS_UNDERFLOW SL_LOGE(ERROR_PLAYER_PREFETCH_d, data1); if (IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) { ap->mPrefetchStatus.mLevel = 0; ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_UNDERFLOW; if (!(~ap->mPrefetchStatus.mCallbackEventsMask & (SL_PREFETCHEVENT_FILLLEVELCHANGE | SL_PREFETCHEVENT_STATUSCHANGE))) { callback = ap->mPrefetchStatus.mCallback; callbackPContext = ap->mPrefetchStatus.mContext; events = SL_PREFETCHEVENT_FILLLEVELCHANGE | SL_PREFETCHEVENT_STATUSCHANGE; } } } object_unlock_exclusive(&ap->mObject); // callback with no lock held if (NULL != callback) { (*callback)(&ap->mPrefetchStatus.mItf, callbackPContext, events); } } break; case android::GenericPlayer::kEventPrefetchFillLevelUpdate : { if (!IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) { break; } slPrefetchCallback callback = NULL; void* callbackPContext = NULL; // SLPrefetchStatusItf callback or no callback? interface_lock_exclusive(&ap->mPrefetchStatus); if (ap->mPrefetchStatus.mCallbackEventsMask & SL_PREFETCHEVENT_FILLLEVELCHANGE) { callback = ap->mPrefetchStatus.mCallback; callbackPContext = ap->mPrefetchStatus.mContext; } ap->mPrefetchStatus.mLevel = (SLpermille)data1; interface_unlock_exclusive(&ap->mPrefetchStatus); // callback with no lock held if (NULL != callback) { (*callback)(&ap->mPrefetchStatus.mItf, callbackPContext, SL_PREFETCHEVENT_FILLLEVELCHANGE); } } break; case android::GenericPlayer::kEventPrefetchStatusChange: { if (!IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) { break; } slPrefetchCallback callback = NULL; void* callbackPContext = NULL; // SLPrefetchStatusItf callback or no callback? object_lock_exclusive(&ap->mObject); if (ap->mPrefetchStatus.mCallbackEventsMask & SL_PREFETCHEVENT_STATUSCHANGE) { callback = ap->mPrefetchStatus.mCallback; callbackPContext = ap->mPrefetchStatus.mContext; } if (data1 >= android::kStatusIntermediate) { ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_SUFFICIENTDATA; } else if (data1 < android::kStatusIntermediate) { ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_UNDERFLOW; } object_unlock_exclusive(&ap->mObject); // callback with no lock held if (NULL != callback) { (*callback)(&ap->mPrefetchStatus.mItf, callbackPContext, SL_PREFETCHEVENT_STATUSCHANGE); } } break; case android::GenericPlayer::kEventEndOfStream: { audioPlayer_dispatch_headAtEnd_lockPlay(ap, true /*set state to paused?*/, true); if ((ap->mAudioTrack != 0) && (!ap->mSeek.mLoopEnabled)) { ap->mAudioTrack->stop(); } } break; case android::GenericPlayer::kEventChannelCount: { object_lock_exclusive(&ap->mObject); if (UNKNOWN_NUMCHANNELS == ap->mNumChannels && UNKNOWN_NUMCHANNELS != data1) { ap->mNumChannels = data1; android_audioPlayer_volumeUpdate(ap); } object_unlock_exclusive(&ap->mObject); } break; case android::GenericPlayer::kEventPlay: { slPlayCallback callback = NULL; void* callbackPContext = NULL; interface_lock_shared(&ap->mPlay); callback = ap->mPlay.mCallback; callbackPContext = ap->mPlay.mContext; interface_unlock_shared(&ap->mPlay); if (NULL != callback) { SLuint32 event = (SLuint32) data1; // SL_PLAYEVENT_HEAD* #ifndef USE_ASYNCHRONOUS_PLAY_CALLBACK // synchronous callback requires a synchronous GetPosition implementation (*callback)(&ap->mPlay.mItf, callbackPContext, event); #else // asynchronous callback works with any GetPosition implementation SLresult result = EnqueueAsyncCallback_ppi(ap, callback, &ap->mPlay.mItf, callbackPContext, event); if (SL_RESULT_SUCCESS != result) { ALOGW("Callback %p(%p, %p, 0x%x) dropped", callback, &ap->mPlay.mItf, callbackPContext, event); } #endif } } break; case android::GenericPlayer::kEventErrorAfterPrepare: { SL_LOGV("kEventErrorAfterPrepare"); // assume no callback slPrefetchCallback callback = NULL; void* callbackPContext = NULL; object_lock_exclusive(&ap->mObject); if (IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) { ap->mPrefetchStatus.mLevel = 0; ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_UNDERFLOW; if (!(~ap->mPrefetchStatus.mCallbackEventsMask & (SL_PREFETCHEVENT_FILLLEVELCHANGE | SL_PREFETCHEVENT_STATUSCHANGE))) { callback = ap->mPrefetchStatus.mCallback; callbackPContext = ap->mPrefetchStatus.mContext; } } object_unlock_exclusive(&ap->mObject); // FIXME there's interesting information in data1, but no API to convey it to client SL_LOGE("Error after prepare: %d", data1); // callback with no lock held if (NULL != callback) { (*callback)(&ap->mPrefetchStatus.mItf, callbackPContext, SL_PREFETCHEVENT_FILLLEVELCHANGE | SL_PREFETCHEVENT_STATUSCHANGE); } } break; case android::GenericPlayer::kEventHasVideoSize: //SL_LOGW("Unexpected kEventHasVideoSize"); break; default: break; } ap->mCallbackProtector->exitCb(); } // From EffectDownmix.h const uint32_t kSides = AUDIO_CHANNEL_OUT_SIDE_LEFT | AUDIO_CHANNEL_OUT_SIDE_RIGHT; const uint32_t kBacks = AUDIO_CHANNEL_OUT_BACK_LEFT | AUDIO_CHANNEL_OUT_BACK_RIGHT; const uint32_t kUnsupported = AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER | AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | AUDIO_CHANNEL_OUT_TOP_CENTER | AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT | AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER | AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT | AUDIO_CHANNEL_OUT_TOP_BACK_LEFT | AUDIO_CHANNEL_OUT_TOP_BACK_CENTER | AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT; //TODO(pmclean) This will need to be revisited when arbitrary N-channel support is added. SLresult android_audioPlayer_validateChannelMask(uint32_t mask, int numChans) { // Check that the number of channels falls within bounds. if (numChans < 0 || numChans > 8) { return SL_RESULT_CONTENT_UNSUPPORTED; } // Are there the right number of channels in the mask? if (audio_channel_count_from_out_mask(mask) != numChans) { return SL_RESULT_CONTENT_UNSUPPORTED; } // check against unsupported channels if (mask & kUnsupported) { ALOGE("Unsupported channels (top or front left/right of center)"); return SL_RESULT_CONTENT_UNSUPPORTED; } // verify has FL/FR if more than one channel if (numChans > 1 && (mask & AUDIO_CHANNEL_OUT_STEREO) != AUDIO_CHANNEL_OUT_STEREO) { ALOGE("Front channels must be present"); return SL_RESULT_CONTENT_UNSUPPORTED; } // verify uses SIDE as a pair (ok if not using SIDE at all) bool hasSides = false; if ((mask & kSides) != 0) { if ((mask & kSides) != kSides) { ALOGE("Side channels must be used as a pair"); return SL_RESULT_CONTENT_UNSUPPORTED; } hasSides = true; } // verify uses BACK as a pair (ok if not using BACK at all) bool hasBacks = false; if ((mask & kBacks) != 0) { if ((mask & kBacks) != kBacks) { ALOGE("Back channels must be used as a pair"); return SL_RESULT_CONTENT_UNSUPPORTED; } hasBacks = true; } return SL_RESULT_SUCCESS; } //----------------------------------------------------------------------------- SLresult android_audioPlayer_checkSourceSink(CAudioPlayer *pAudioPlayer) { // verify that the locator types for the source / sink combination is supported pAudioPlayer->mAndroidObjType = audioPlayer_getAndroidObjectTypeForSourceSink(pAudioPlayer); if (INVALID_TYPE == pAudioPlayer->mAndroidObjType) { return SL_RESULT_PARAMETER_INVALID; } const SLDataSource *pAudioSrc = &pAudioPlayer->mDataSource.u.mSource; const SLDataSink *pAudioSnk = &pAudioPlayer->mDataSink.u.mSink; // format check: const SLuint32 sourceLocatorType = *(SLuint32 *)pAudioSrc->pLocator; const SLuint32 sinkLocatorType = *(SLuint32 *)pAudioSnk->pLocator; const SLuint32 sourceFormatType = *(SLuint32 *)pAudioSrc->pFormat; const SLuint32 sinkFormatType = *(SLuint32 *)pAudioSnk->pFormat; const SLuint32 *df_representation = NULL; // pointer to representation field, if it exists switch (sourceLocatorType) { //------------------ // Buffer Queues case SL_DATALOCATOR_BUFFERQUEUE: case SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE: { SLDataLocator_BufferQueue *dl_bq = (SLDataLocator_BufferQueue *) pAudioSrc->pLocator; // Buffer format switch (sourceFormatType) { // currently only PCM buffer queues are supported, case SL_ANDROID_DATAFORMAT_PCM_EX: { const SLAndroidDataFormat_PCM_EX *df_pcm = (const SLAndroidDataFormat_PCM_EX *) pAudioSrc->pFormat; switch (df_pcm->representation) { case SL_ANDROID_PCM_REPRESENTATION_SIGNED_INT: case SL_ANDROID_PCM_REPRESENTATION_UNSIGNED_INT: case SL_ANDROID_PCM_REPRESENTATION_FLOAT: df_representation = &df_pcm->representation; break; default: SL_LOGE("Cannot create audio player: unsupported representation: %d", df_pcm->representation); return SL_RESULT_CONTENT_UNSUPPORTED; } } // SL_ANDROID_DATAFORMAT_PCM_EX - fall through to next test. case SL_DATAFORMAT_PCM: { const SLDataFormat_PCM *df_pcm = (const SLDataFormat_PCM *) pAudioSrc->pFormat; SLresult result = android_audioPlayer_validateChannelMask(df_pcm->channelMask, df_pcm->numChannels); if (result != SL_RESULT_SUCCESS) { SL_LOGE("Cannot create audio player: unsupported PCM data source with %u channels", (unsigned) df_pcm->numChannels); return result; } switch (df_pcm->samplesPerSec) { case SL_SAMPLINGRATE_8: case SL_SAMPLINGRATE_11_025: case SL_SAMPLINGRATE_12: case SL_SAMPLINGRATE_16: case SL_SAMPLINGRATE_22_05: case SL_SAMPLINGRATE_24: case SL_SAMPLINGRATE_32: case SL_SAMPLINGRATE_44_1: case SL_SAMPLINGRATE_48: break; case SL_SAMPLINGRATE_64: case SL_SAMPLINGRATE_88_2: case SL_SAMPLINGRATE_96: case SL_SAMPLINGRATE_192: default: SL_LOGE("Cannot create audio player: unsupported sample rate %u milliHz", (unsigned) df_pcm->samplesPerSec); return SL_RESULT_CONTENT_UNSUPPORTED; } switch (df_pcm->bitsPerSample) { case SL_PCMSAMPLEFORMAT_FIXED_8: if (df_representation != NULL && *df_representation != SL_ANDROID_PCM_REPRESENTATION_UNSIGNED_INT) { goto default_err; } break; case SL_PCMSAMPLEFORMAT_FIXED_16: case SL_PCMSAMPLEFORMAT_FIXED_24: if (df_representation != NULL && *df_representation != SL_ANDROID_PCM_REPRESENTATION_SIGNED_INT) { goto default_err; } break; case SL_PCMSAMPLEFORMAT_FIXED_32: if (df_representation != NULL && *df_representation != SL_ANDROID_PCM_REPRESENTATION_SIGNED_INT && *df_representation != SL_ANDROID_PCM_REPRESENTATION_FLOAT) { goto default_err; } break; // others default: default_err: // this should have already been rejected by checkDataFormat SL_LOGE("Cannot create audio player: unsupported sample bit depth %u", (SLuint32)df_pcm->bitsPerSample); return SL_RESULT_CONTENT_UNSUPPORTED; } switch (df_pcm->containerSize) { case 8: case 16: case 24: case 32: break; // others default: SL_LOGE("Cannot create audio player: unsupported container size %u", (unsigned) df_pcm->containerSize); return SL_RESULT_CONTENT_UNSUPPORTED; } // df_pcm->channelMask: the earlier platform-independent check and the // upcoming check by sles_to_android_channelMaskOut are sufficient switch (df_pcm->endianness) { case SL_BYTEORDER_LITTLEENDIAN: break; case SL_BYTEORDER_BIGENDIAN: SL_LOGE("Cannot create audio player: unsupported big-endian byte order"); return SL_RESULT_CONTENT_UNSUPPORTED; // native is proposed but not yet in spec default: SL_LOGE("Cannot create audio player: unsupported byte order %u", (unsigned) df_pcm->endianness); return SL_RESULT_CONTENT_UNSUPPORTED; } } //case SL_DATAFORMAT_PCM break; case SL_DATAFORMAT_MIME: case XA_DATAFORMAT_RAWIMAGE: SL_LOGE("Cannot create audio player with buffer queue data source " "without SL_DATAFORMAT_PCM format"); return SL_RESULT_CONTENT_UNSUPPORTED; default: // invalid data format is detected earlier assert(false); return SL_RESULT_INTERNAL_ERROR; } // switch (sourceFormatType) } // case SL_DATALOCATOR_BUFFERQUEUE or SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE break; //------------------ // URI case SL_DATALOCATOR_URI: { SLDataLocator_URI *dl_uri = (SLDataLocator_URI *) pAudioSrc->pLocator; if (NULL == dl_uri->URI) { return SL_RESULT_PARAMETER_INVALID; } // URI format switch (sourceFormatType) { case SL_DATAFORMAT_MIME: break; case SL_DATAFORMAT_PCM: case XA_DATAFORMAT_RAWIMAGE: SL_LOGE("Cannot create audio player with SL_DATALOCATOR_URI data source without " "SL_DATAFORMAT_MIME format"); return SL_RESULT_CONTENT_UNSUPPORTED; } // switch (sourceFormatType) // decoding format check if ((sinkLocatorType != SL_DATALOCATOR_OUTPUTMIX) && !audioPlayer_isSupportedNonOutputMixSink(pAudioSnk)) { return SL_RESULT_CONTENT_UNSUPPORTED; } } // case SL_DATALOCATOR_URI break; //------------------ // File Descriptor case SL_DATALOCATOR_ANDROIDFD: { // fd is already non null switch (sourceFormatType) { case SL_DATAFORMAT_MIME: break; case SL_DATAFORMAT_PCM: // FIXME implement SL_LOGD("[ FIXME implement PCM FD data sources ]"); break; case XA_DATAFORMAT_RAWIMAGE: SL_LOGE("Cannot create audio player with SL_DATALOCATOR_ANDROIDFD data source " "without SL_DATAFORMAT_MIME or SL_DATAFORMAT_PCM format"); return SL_RESULT_CONTENT_UNSUPPORTED; default: // invalid data format is detected earlier assert(false); return SL_RESULT_INTERNAL_ERROR; } // switch (sourceFormatType) if ((sinkLocatorType != SL_DATALOCATOR_OUTPUTMIX) && !audioPlayer_isSupportedNonOutputMixSink(pAudioSnk)) { return SL_RESULT_CONTENT_UNSUPPORTED; } } // case SL_DATALOCATOR_ANDROIDFD break; //------------------ // Stream case SL_DATALOCATOR_ANDROIDBUFFERQUEUE: { switch (sourceFormatType) { case SL_DATAFORMAT_MIME: { SLDataFormat_MIME *df_mime = (SLDataFormat_MIME *) pAudioSrc->pFormat; if (NULL == df_mime) { SL_LOGE("MIME type null invalid"); return SL_RESULT_CONTENT_UNSUPPORTED; } SL_LOGD("source MIME is %s", (char*)df_mime->mimeType); switch(df_mime->containerType) { case SL_CONTAINERTYPE_MPEG_TS: if (strcasecmp((char*)df_mime->mimeType, (const char *)XA_ANDROID_MIME_MP2TS)) { SL_LOGE("Invalid MIME (%s) for container SL_CONTAINERTYPE_MPEG_TS, expects %s", (char*)df_mime->mimeType, XA_ANDROID_MIME_MP2TS); return SL_RESULT_CONTENT_UNSUPPORTED; } if (pAudioPlayer->mAndroidObjType != AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE) { SL_LOGE("Invalid sink for container SL_CONTAINERTYPE_MPEG_TS"); return SL_RESULT_PARAMETER_INVALID; } break; case SL_CONTAINERTYPE_RAW: case SL_CONTAINERTYPE_AAC: if (strcasecmp((char*)df_mime->mimeType, (const char *)SL_ANDROID_MIME_AACADTS) && strcasecmp((char*)df_mime->mimeType, ANDROID_MIME_AACADTS_ANDROID_FRAMEWORK)) { SL_LOGE("Invalid MIME (%s) for container type %d, expects %s", (char*)df_mime->mimeType, df_mime->containerType, SL_ANDROID_MIME_AACADTS); return SL_RESULT_CONTENT_UNSUPPORTED; } if (pAudioPlayer->mAndroidObjType != AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE) { SL_LOGE("Invalid sink for container SL_CONTAINERTYPE_AAC"); return SL_RESULT_PARAMETER_INVALID; } break; default: SL_LOGE("Cannot create player with SL_DATALOCATOR_ANDROIDBUFFERQUEUE data source " "that is not fed MPEG-2 TS data or AAC ADTS data"); return SL_RESULT_CONTENT_UNSUPPORTED; } } break; default: SL_LOGE("Cannot create player with SL_DATALOCATOR_ANDROIDBUFFERQUEUE data source " "without SL_DATAFORMAT_MIME format"); return SL_RESULT_CONTENT_UNSUPPORTED; } } break; // case SL_DATALOCATOR_ANDROIDBUFFERQUEUE //------------------ // Address case SL_DATALOCATOR_ADDRESS: case SL_DATALOCATOR_IODEVICE: case SL_DATALOCATOR_OUTPUTMIX: case XA_DATALOCATOR_NATIVEDISPLAY: case SL_DATALOCATOR_MIDIBUFFERQUEUE: SL_LOGE("Cannot create audio player with data locator type 0x%x", (unsigned) sourceLocatorType); return SL_RESULT_CONTENT_UNSUPPORTED; default: SL_LOGE("Cannot create audio player with invalid data locator type 0x%x", (unsigned) sourceLocatorType); return SL_RESULT_PARAMETER_INVALID; }// switch (locatorType) return SL_RESULT_SUCCESS; } //----------------------------------------------------------------------------- // Callback associated with an AudioTrack of an SL ES AudioPlayer that gets its data // from a buffer queue. This will not be called once the AudioTrack has been destroyed. static void audioTrack_callBack_pullFromBuffQueue(int event, void* user, void *info) { CAudioPlayer *ap = (CAudioPlayer *)user; if (!android::CallbackProtector::enterCbIfOk(ap->mCallbackProtector)) { // it is not safe to enter the callback (the track is about to go away) return; } void * callbackPContext = NULL; switch(event) { case android::AudioTrack::EVENT_MORE_DATA: { //SL_LOGV("received event EVENT_MORE_DATA from AudioTrack TID=%d", gettid()); slPrefetchCallback prefetchCallback = NULL; void *prefetchContext = NULL; SLuint32 prefetchEvents = SL_PREFETCHEVENT_NONE; android::AudioTrack::Buffer* pBuff = (android::AudioTrack::Buffer*)info; // retrieve data from the buffer queue interface_lock_exclusive(&ap->mBufferQueue); if (ap->mBufferQueue.mCallbackPending) { // call callback with lock not held slBufferQueueCallback callback = ap->mBufferQueue.mCallback; if (NULL != callback) { callbackPContext = ap->mBufferQueue.mContext; interface_unlock_exclusive(&ap->mBufferQueue); (*callback)(&ap->mBufferQueue.mItf, callbackPContext); interface_lock_exclusive(&ap->mBufferQueue); ap->mBufferQueue.mCallbackPending = false; } } if (ap->mBufferQueue.mState.count != 0) { //SL_LOGV("nbBuffers in queue = %u",ap->mBufferQueue.mState.count); assert(ap->mBufferQueue.mFront != ap->mBufferQueue.mRear); BufferHeader *oldFront = ap->mBufferQueue.mFront; BufferHeader *newFront = &oldFront[1]; size_t availSource = oldFront->mSize - ap->mBufferQueue.mSizeConsumed; size_t availSink = pBuff->size; size_t bytesToCopy = availSource < availSink ? availSource : availSink; void *pSrc = (char *)oldFront->mBuffer + ap->mBufferQueue.mSizeConsumed; memcpy(pBuff->raw, pSrc, bytesToCopy); if (bytesToCopy < availSource) { ap->mBufferQueue.mSizeConsumed += bytesToCopy; // pBuff->size is already equal to bytesToCopy in this case } else { // consumed an entire buffer, dequeue pBuff->size = bytesToCopy; ap->mBufferQueue.mSizeConsumed = 0; if (newFront == &ap->mBufferQueue.mArray [ap->mBufferQueue.mNumBuffers + 1]) { newFront = ap->mBufferQueue.mArray; } ap->mBufferQueue.mFront = newFront; ap->mBufferQueue.mState.count--; ap->mBufferQueue.mState.playIndex++; ap->mBufferQueue.mCallbackPending = true; } } else { // empty queue // signal no data available pBuff->size = 0; // signal we're at the end of the content, but don't pause (see note in function) audioPlayer_dispatch_headAtEnd_lockPlay(ap, false /*set state to paused?*/, false); // signal underflow to prefetch status itf if (IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) { ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_UNDERFLOW; ap->mPrefetchStatus.mLevel = 0; // callback or no callback? prefetchEvents = ap->mPrefetchStatus.mCallbackEventsMask & (SL_PREFETCHEVENT_STATUSCHANGE | SL_PREFETCHEVENT_FILLLEVELCHANGE); if (SL_PREFETCHEVENT_NONE != prefetchEvents) { prefetchCallback = ap->mPrefetchStatus.mCallback; prefetchContext = ap->mPrefetchStatus.mContext; } } // stop the track so it restarts playing faster when new data is enqueued ap->mAudioTrack->stop(); } interface_unlock_exclusive(&ap->mBufferQueue); // notify client if (NULL != prefetchCallback) { assert(SL_PREFETCHEVENT_NONE != prefetchEvents); // spec requires separate callbacks for each event if (prefetchEvents & SL_PREFETCHEVENT_STATUSCHANGE) { (*prefetchCallback)(&ap->mPrefetchStatus.mItf, prefetchContext, SL_PREFETCHEVENT_STATUSCHANGE); } if (prefetchEvents & SL_PREFETCHEVENT_FILLLEVELCHANGE) { (*prefetchCallback)(&ap->mPrefetchStatus.mItf, prefetchContext, SL_PREFETCHEVENT_FILLLEVELCHANGE); } } } break; case android::AudioTrack::EVENT_MARKER: //SL_LOGI("received event EVENT_MARKER from AudioTrack"); audioTrack_handleMarker_lockPlay(ap); break; case android::AudioTrack::EVENT_NEW_POS: //SL_LOGI("received event EVENT_NEW_POS from AudioTrack"); audioTrack_handleNewPos_lockPlay(ap); break; case android::AudioTrack::EVENT_UNDERRUN: //SL_LOGI("received event EVENT_UNDERRUN from AudioTrack"); audioTrack_handleUnderrun_lockPlay(ap); break; case android::AudioTrack::EVENT_BUFFER_END: case android::AudioTrack::EVENT_LOOP_END: // These are unexpected so fall through default: // FIXME where does the notification of SL_PLAYEVENT_HEADMOVING fit? SL_LOGE("Encountered unknown AudioTrack event %d for CAudioPlayer %p", event, (CAudioPlayer *)user); break; } ap->mCallbackProtector->exitCb(); } //----------------------------------------------------------------------------- void android_audioPlayer_create(CAudioPlayer *pAudioPlayer) { // pAudioPlayer->mAndroidObjType has been set in android_audioPlayer_checkSourceSink() // and if it was == INVALID_TYPE, then IEngine_CreateAudioPlayer would never call us assert(INVALID_TYPE != pAudioPlayer->mAndroidObjType); // These initializations are in the same order as the field declarations in classes.h // FIXME Consolidate initializations (many of these already in IEngine_CreateAudioPlayer) // mAndroidObjType: see above comment pAudioPlayer->mAndroidObjState = ANDROID_UNINITIALIZED; pAudioPlayer->mSessionId = android::AudioSystem::newAudioUniqueId(); // placeholder: not necessary yet as session ID lifetime doesn't extend beyond player // android::AudioSystem::acquireAudioSessionId(pAudioPlayer->mSessionId); pAudioPlayer->mStreamType = ANDROID_DEFAULT_OUTPUT_STREAM_TYPE; // mAudioTrack pAudioPlayer->mCallbackProtector = new android::CallbackProtector(); // mAPLayer // mAuxEffect pAudioPlayer->mAuxSendLevel = 0; pAudioPlayer->mAmplFromDirectLevel = 1.0f; // matches initial mDirectLevel value pAudioPlayer->mDeferredStart = false; // This section re-initializes interface-specific fields that // can be set or used regardless of whether the interface is // exposed on the AudioPlayer or not switch (pAudioPlayer->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: pAudioPlayer->mPlaybackRate.mMinRate = AUDIOTRACK_MIN_PLAYBACKRATE_PERMILLE; pAudioPlayer->mPlaybackRate.mMaxRate = AUDIOTRACK_MAX_PLAYBACKRATE_PERMILLE; break; case AUDIOPLAYER_FROM_URIFD: pAudioPlayer->mPlaybackRate.mMinRate = MEDIAPLAYER_MIN_PLAYBACKRATE_PERMILLE; pAudioPlayer->mPlaybackRate.mMaxRate = MEDIAPLAYER_MAX_PLAYBACKRATE_PERMILLE; break; default: // use the default range break; } } //----------------------------------------------------------------------------- SLresult android_audioPlayer_setConfig(CAudioPlayer *ap, const SLchar *configKey, const void *pConfigValue, SLuint32 valueSize) { SLresult result; assert(NULL != ap && NULL != configKey && NULL != pConfigValue); if (strcmp((const char*)configKey, (const char*)SL_ANDROID_KEY_STREAM_TYPE) == 0) { // stream type if (KEY_STREAM_TYPE_PARAMSIZE > valueSize) { SL_LOGE(ERROR_CONFIG_VALUESIZE_TOO_LOW); result = SL_RESULT_BUFFER_INSUFFICIENT; } else { result = audioPlayer_setStreamType(ap, *(SLuint32*)pConfigValue); } } else { SL_LOGE(ERROR_CONFIG_UNKNOWN_KEY); result = SL_RESULT_PARAMETER_INVALID; } return result; } //----------------------------------------------------------------------------- SLresult android_audioPlayer_getConfig(CAudioPlayer* ap, const SLchar *configKey, SLuint32* pValueSize, void *pConfigValue) { SLresult result; assert(NULL != ap && NULL != configKey && NULL != pValueSize); if (strcmp((const char*)configKey, (const char*)SL_ANDROID_KEY_STREAM_TYPE) == 0) { // stream type if (NULL == pConfigValue) { result = SL_RESULT_SUCCESS; } else if (KEY_STREAM_TYPE_PARAMSIZE > *pValueSize) { SL_LOGE(ERROR_CONFIG_VALUESIZE_TOO_LOW); result = SL_RESULT_BUFFER_INSUFFICIENT; } else { result = audioPlayer_getStreamType(ap, (SLint32*)pConfigValue); } *pValueSize = KEY_STREAM_TYPE_PARAMSIZE; } else { SL_LOGE(ERROR_CONFIG_UNKNOWN_KEY); result = SL_RESULT_PARAMETER_INVALID; } return result; } // Called from android_audioPlayer_realize for a PCM buffer queue player // to determine if it can use a fast track. static bool canUseFastTrack(CAudioPlayer *pAudioPlayer) { assert(pAudioPlayer->mAndroidObjType == AUDIOPLAYER_FROM_PCM_BUFFERQUEUE); // no need to check the buffer queue size, application side // double-buffering (and more) is not a requirement for using fast tracks // Check a blacklist of interfaces that are incompatible with fast tracks. // The alternative, to check a whitelist of compatible interfaces, is // more maintainable but is too slow. As a compromise, in a debug build // we use both methods and warn if they produce different results. // In release builds, we only use the blacklist method. // If a blacklisted interface is added after realization using // DynamicInterfaceManagement::AddInterface, // then this won't be detected but the interface will be ineffective. bool blacklistResult = true; static const unsigned blacklist[] = { MPH_BASSBOOST, MPH_EFFECTSEND, MPH_ENVIRONMENTALREVERB, MPH_EQUALIZER, MPH_PLAYBACKRATE, MPH_PRESETREVERB, MPH_VIRTUALIZER, MPH_ANDROIDEFFECT, MPH_ANDROIDEFFECTSEND, // FIXME The problem with a blacklist is remembering to add new interfaces here }; for (unsigned i = 0; i < sizeof(blacklist)/sizeof(blacklist[0]); ++i) { if (IsInterfaceInitialized(&pAudioPlayer->mObject, blacklist[i])) { blacklistResult = false; break; } } #if LOG_NDEBUG == 0 bool whitelistResult = true; static const unsigned whitelist[] = { MPH_BUFFERQUEUE, MPH_DYNAMICINTERFACEMANAGEMENT, MPH_METADATAEXTRACTION, MPH_MUTESOLO, MPH_OBJECT, MPH_PLAY, MPH_PREFETCHSTATUS, MPH_VOLUME, MPH_ANDROIDCONFIGURATION, MPH_ANDROIDSIMPLEBUFFERQUEUE, MPH_ANDROIDBUFFERQUEUESOURCE, }; for (unsigned mph = MPH_MIN; mph < MPH_MAX; ++mph) { for (unsigned i = 0; i < sizeof(whitelist)/sizeof(whitelist[0]); ++i) { if (mph == whitelist[i]) { goto compatible; } } if (IsInterfaceInitialized(&pAudioPlayer->mObject, mph)) { whitelistResult = false; break; } compatible: ; } if (whitelistResult != blacklistResult) { ALOGW("whitelistResult != blacklistResult"); // and use blacklistResult below } #endif return blacklistResult; } //----------------------------------------------------------------------------- // FIXME abstract out the diff between CMediaPlayer and CAudioPlayer SLresult android_audioPlayer_realize(CAudioPlayer *pAudioPlayer, SLboolean async) { SLresult result = SL_RESULT_SUCCESS; SL_LOGV("Realize pAudioPlayer=%p", pAudioPlayer); AudioPlayback_Parameters app; app.sessionId = pAudioPlayer->mSessionId; app.streamType = pAudioPlayer->mStreamType; switch (pAudioPlayer->mAndroidObjType) { //----------------------------------- // AudioTrack case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: { // initialize platform-specific CAudioPlayer fields SLDataLocator_BufferQueue *dl_bq = (SLDataLocator_BufferQueue *) pAudioPlayer->mDynamicSource.mDataSource; SLDataFormat_PCM *df_pcm = (SLDataFormat_PCM *) pAudioPlayer->mDynamicSource.mDataSource->pFormat; uint32_t sampleRate = sles_to_android_sampleRate(df_pcm->samplesPerSec); audio_output_flags_t policy; if (canUseFastTrack(pAudioPlayer)) { policy = AUDIO_OUTPUT_FLAG_FAST; } else { policy = AUDIO_OUTPUT_FLAG_NONE; } pAudioPlayer->mAudioTrack = new android::AudioTrack( pAudioPlayer->mStreamType, // streamType sampleRate, // sampleRate sles_to_android_sampleFormat(df_pcm), // format sles_to_android_channelMaskOut(df_pcm->numChannels, df_pcm->channelMask), // channel mask 0, // frameCount policy, // flags audioTrack_callBack_pullFromBuffQueue, // callback (void *) pAudioPlayer, // user 0, // FIXME find appropriate frame count // notificationFrame pAudioPlayer->mSessionId); android::status_t status = pAudioPlayer->mAudioTrack->initCheck(); if (status != android::NO_ERROR) { SL_LOGE("AudioTrack::initCheck status %u", status); // FIXME should return a more specific result depending on status result = SL_RESULT_CONTENT_UNSUPPORTED; pAudioPlayer->mAudioTrack.clear(); return result; } // initialize platform-independent CAudioPlayer fields pAudioPlayer->mNumChannels = df_pcm->numChannels; pAudioPlayer->mSampleRateMilliHz = df_pcm->samplesPerSec; // Note: bad field name in SL ES // This use case does not have a separate "prepare" step pAudioPlayer->mAndroidObjState = ANDROID_READY; } break; //----------------------------------- // MediaPlayer case AUDIOPLAYER_FROM_URIFD: { pAudioPlayer->mAPlayer = new android::LocAVPlayer(&app, false /*hasVideo*/); pAudioPlayer->mAPlayer->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer /*notifUSer*/); switch (pAudioPlayer->mDataSource.mLocator.mLocatorType) { case SL_DATALOCATOR_URI: { // The legacy implementation ran Stagefright within the application process, and // so allowed local pathnames specified by URI that were openable by // the application but were not openable by mediaserver. // The current implementation runs Stagefright (mostly) within mediaserver, // which runs as a different UID and likely a different current working directory. // For backwards compatibility with any applications which may have relied on the // previous behavior, we convert an openable file URI into an FD. // Note that unlike SL_DATALOCATOR_ANDROIDFD, this FD is owned by us // and so we close it as soon as we've passed it (via Binder dup) to mediaserver. const char *uri = (const char *)pAudioPlayer->mDataSource.mLocator.mURI.URI; if (!isDistantProtocol(uri)) { // don't touch the original uri, we may need it later const char *pathname = uri; // skip over an optional leading file:// prefix if (!strncasecmp(pathname, "file://", 7)) { pathname += 7; } // attempt to open it as a file using the application's credentials int fd = ::open(pathname, O_RDONLY); if (fd >= 0) { // if open is successful, then check to see if it's a regular file struct stat statbuf; if (!::fstat(fd, &statbuf) && S_ISREG(statbuf.st_mode)) { // treat similarly to an FD data locator, but // let setDataSource take responsibility for closing fd pAudioPlayer->mAPlayer->setDataSource(fd, 0, statbuf.st_size, true); break; } // we were able to open it, but it's not a file, so let mediaserver try (void) ::close(fd); } } // if either the URI didn't look like a file, or open failed, or not a file pAudioPlayer->mAPlayer->setDataSource(uri); } break; case SL_DATALOCATOR_ANDROIDFD: { int64_t offset = (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.offset; pAudioPlayer->mAPlayer->setDataSource( (int)pAudioPlayer->mDataSource.mLocator.mFD.fd, offset == SL_DATALOCATOR_ANDROIDFD_USE_FILE_SIZE ? (int64_t)PLAYER_FD_FIND_FILE_SIZE : offset, (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.length); } break; default: SL_LOGE(ERROR_PLAYERREALIZE_UNKNOWN_DATASOURCE_LOCATOR); break; } } break; //----------------------------------- // StreamPlayer case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: { android::StreamPlayer* splr = new android::StreamPlayer(&app, false /*hasVideo*/, &pAudioPlayer->mAndroidBufferQueue, pAudioPlayer->mCallbackProtector); pAudioPlayer->mAPlayer = splr; splr->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer); } break; //----------------------------------- // AudioToCbRenderer case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: { android::AudioToCbRenderer* decoder = new android::AudioToCbRenderer(&app); pAudioPlayer->mAPlayer = decoder; // configures the callback for the sink buffer queue decoder->setDataPushListener(adecoder_writeToBufferQueue, pAudioPlayer); // configures the callback for the notifications coming from the SF code decoder->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer); switch (pAudioPlayer->mDataSource.mLocator.mLocatorType) { case SL_DATALOCATOR_URI: decoder->setDataSource( (const char*)pAudioPlayer->mDataSource.mLocator.mURI.URI); break; case SL_DATALOCATOR_ANDROIDFD: { int64_t offset = (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.offset; decoder->setDataSource( (int)pAudioPlayer->mDataSource.mLocator.mFD.fd, offset == SL_DATALOCATOR_ANDROIDFD_USE_FILE_SIZE ? (int64_t)PLAYER_FD_FIND_FILE_SIZE : offset, (int64_t)pAudioPlayer->mDataSource.mLocator.mFD.length); } break; default: SL_LOGE(ERROR_PLAYERREALIZE_UNKNOWN_DATASOURCE_LOCATOR); break; } } break; //----------------------------------- // AacBqToPcmCbRenderer case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { android::AacBqToPcmCbRenderer* bqtobq = new android::AacBqToPcmCbRenderer(&app, &pAudioPlayer->mAndroidBufferQueue); // configures the callback for the sink buffer queue bqtobq->setDataPushListener(adecoder_writeToBufferQueue, pAudioPlayer); pAudioPlayer->mAPlayer = bqtobq; // configures the callback for the notifications coming from the SF code, // but also implicitly configures the AndroidBufferQueue from which ADTS data is read pAudioPlayer->mAPlayer->init(sfplayer_handlePrefetchEvent, (void*)pAudioPlayer); } break; //----------------------------------- default: SL_LOGE(ERROR_PLAYERREALIZE_UNEXPECTED_OBJECT_TYPE_D, pAudioPlayer->mAndroidObjType); result = SL_RESULT_INTERNAL_ERROR; break; } // proceed with effect initialization // initialize EQ // FIXME use a table of effect descriptors when adding support for more effects if (memcmp(SL_IID_EQUALIZER, &pAudioPlayer->mEqualizer.mEqDescriptor.type, sizeof(effect_uuid_t)) == 0) { SL_LOGV("Need to initialize EQ for AudioPlayer=%p", pAudioPlayer); android_eq_init(pAudioPlayer->mSessionId, &pAudioPlayer->mEqualizer); } // initialize BassBoost if (memcmp(SL_IID_BASSBOOST, &pAudioPlayer->mBassBoost.mBassBoostDescriptor.type, sizeof(effect_uuid_t)) == 0) { SL_LOGV("Need to initialize BassBoost for AudioPlayer=%p", pAudioPlayer); android_bb_init(pAudioPlayer->mSessionId, &pAudioPlayer->mBassBoost); } // initialize Virtualizer if (memcmp(SL_IID_VIRTUALIZER, &pAudioPlayer->mVirtualizer.mVirtualizerDescriptor.type, sizeof(effect_uuid_t)) == 0) { SL_LOGV("Need to initialize Virtualizer for AudioPlayer=%p", pAudioPlayer); android_virt_init(pAudioPlayer->mSessionId, &pAudioPlayer->mVirtualizer); } // initialize EffectSend // FIXME initialize EffectSend return result; } //----------------------------------------------------------------------------- /** * Called with a lock on AudioPlayer, and blocks until safe to destroy */ SLresult android_audioPlayer_preDestroy(CAudioPlayer *pAudioPlayer) { SL_LOGD("android_audioPlayer_preDestroy(%p)", pAudioPlayer); SLresult result = SL_RESULT_SUCCESS; bool disableCallbacksBeforePreDestroy; switch (pAudioPlayer->mAndroidObjType) { // Not yet clear why this order is important, but it reduces detected deadlocks case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: disableCallbacksBeforePreDestroy = true; break; // Use the old behavior for all other use cases until proven // case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: default: disableCallbacksBeforePreDestroy = false; break; } if (disableCallbacksBeforePreDestroy) { object_unlock_exclusive(&pAudioPlayer->mObject); if (pAudioPlayer->mCallbackProtector != 0) { pAudioPlayer->mCallbackProtector->requestCbExitAndWait(); } object_lock_exclusive(&pAudioPlayer->mObject); } if (pAudioPlayer->mAPlayer != 0) { pAudioPlayer->mAPlayer->preDestroy(); } SL_LOGD("android_audioPlayer_preDestroy(%p) after mAPlayer->preDestroy()", pAudioPlayer); if (!disableCallbacksBeforePreDestroy) { object_unlock_exclusive(&pAudioPlayer->mObject); if (pAudioPlayer->mCallbackProtector != 0) { pAudioPlayer->mCallbackProtector->requestCbExitAndWait(); } object_lock_exclusive(&pAudioPlayer->mObject); } return result; } //----------------------------------------------------------------------------- SLresult android_audioPlayer_destroy(CAudioPlayer *pAudioPlayer) { SLresult result = SL_RESULT_SUCCESS; SL_LOGV("android_audioPlayer_destroy(%p)", pAudioPlayer); switch (pAudioPlayer->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: // We own the audio track for PCM buffer queue players if (pAudioPlayer->mAudioTrack != 0) { pAudioPlayer->mAudioTrack->stop(); // Note that there may still be another reference in post-unlock phase of SetPlayState pAudioPlayer->mAudioTrack.clear(); } break; case AUDIOPLAYER_FROM_URIFD: // intended fall-through case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: pAudioPlayer->mAPlayer.clear(); break; //----------------------------------- default: SL_LOGE(ERROR_PLAYERDESTROY_UNEXPECTED_OBJECT_TYPE_D, pAudioPlayer->mAndroidObjType); result = SL_RESULT_INTERNAL_ERROR; break; } // placeholder: not necessary yet as session ID lifetime doesn't extend beyond player // android::AudioSystem::releaseAudioSessionId(pAudioPlayer->mSessionId); pAudioPlayer->mCallbackProtector.clear(); // explicit destructor pAudioPlayer->mAudioTrack.~sp(); // note that SetPlayState(PLAYING) may still hold a reference pAudioPlayer->mCallbackProtector.~sp(); pAudioPlayer->mAuxEffect.~sp(); pAudioPlayer->mAPlayer.~sp(); return result; } //----------------------------------------------------------------------------- SLresult android_audioPlayer_setPlaybackRateAndConstraints(CAudioPlayer *ap, SLpermille rate, SLuint32 constraints) { SLresult result = SL_RESULT_SUCCESS; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: { // these asserts were already checked by the platform-independent layer assert((AUDIOTRACK_MIN_PLAYBACKRATE_PERMILLE <= rate) && (rate <= AUDIOTRACK_MAX_PLAYBACKRATE_PERMILLE)); assert(constraints & SL_RATEPROP_NOPITCHCORAUDIO); // get the content sample rate uint32_t contentRate = sles_to_android_sampleRate(ap->mSampleRateMilliHz); // apply the SL ES playback rate on the AudioTrack as a factor of its content sample rate if (ap->mAudioTrack != 0) { ap->mAudioTrack->setSampleRate(contentRate * (rate/1000.0f)); } } break; case AUDIOPLAYER_FROM_URIFD: { assert((MEDIAPLAYER_MIN_PLAYBACKRATE_PERMILLE <= rate) && (rate <= MEDIAPLAYER_MAX_PLAYBACKRATE_PERMILLE)); assert(constraints & SL_RATEPROP_NOPITCHCORAUDIO); // apply the SL ES playback rate on the GenericPlayer if (ap->mAPlayer != 0) { ap->mAPlayer->setPlaybackRate((int16_t)rate); } } break; default: SL_LOGE("Unexpected object type %d", ap->mAndroidObjType); result = SL_RESULT_FEATURE_UNSUPPORTED; break; } return result; } //----------------------------------------------------------------------------- // precondition // called with no lock held // ap != NULL // pItemCount != NULL SLresult android_audioPlayer_metadata_getItemCount(CAudioPlayer *ap, SLuint32 *pItemCount) { if (ap->mAPlayer == 0) { return SL_RESULT_PARAMETER_INVALID; } switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { android::AudioSfDecoder* decoder = static_cast(ap->mAPlayer.get()); *pItemCount = decoder->getPcmFormatKeyCount(); } break; default: *pItemCount = 0; break; } return SL_RESULT_SUCCESS; } //----------------------------------------------------------------------------- // precondition // called with no lock held // ap != NULL // pKeySize != NULL SLresult android_audioPlayer_metadata_getKeySize(CAudioPlayer *ap, SLuint32 index, SLuint32 *pKeySize) { if (ap->mAPlayer == 0) { return SL_RESULT_PARAMETER_INVALID; } SLresult res = SL_RESULT_SUCCESS; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { android::AudioSfDecoder* decoder = static_cast(ap->mAPlayer.get()); SLuint32 keyNameSize = 0; if (!decoder->getPcmFormatKeySize(index, &keyNameSize)) { res = SL_RESULT_PARAMETER_INVALID; } else { // *pKeySize is the size of the region used to store the key name AND // the information about the key (size, lang, encoding) *pKeySize = keyNameSize + sizeof(SLMetadataInfo); } } break; default: *pKeySize = 0; res = SL_RESULT_PARAMETER_INVALID; break; } return res; } //----------------------------------------------------------------------------- // precondition // called with no lock held // ap != NULL // pKey != NULL SLresult android_audioPlayer_metadata_getKey(CAudioPlayer *ap, SLuint32 index, SLuint32 size, SLMetadataInfo *pKey) { if (ap->mAPlayer == 0) { return SL_RESULT_PARAMETER_INVALID; } SLresult res = SL_RESULT_SUCCESS; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { android::AudioSfDecoder* decoder = static_cast(ap->mAPlayer.get()); if ((size < sizeof(SLMetadataInfo) || (!decoder->getPcmFormatKeyName(index, size - sizeof(SLMetadataInfo), (char*)pKey->data)))) { res = SL_RESULT_PARAMETER_INVALID; } else { // successfully retrieved the key value, update the other fields pKey->encoding = SL_CHARACTERENCODING_UTF8; memcpy((char *) pKey->langCountry, "en", 3); pKey->size = strlen((char*)pKey->data) + 1; } } break; default: res = SL_RESULT_PARAMETER_INVALID; break; } return res; } //----------------------------------------------------------------------------- // precondition // called with no lock held // ap != NULL // pValueSize != NULL SLresult android_audioPlayer_metadata_getValueSize(CAudioPlayer *ap, SLuint32 index, SLuint32 *pValueSize) { if (ap->mAPlayer == 0) { return SL_RESULT_PARAMETER_INVALID; } SLresult res = SL_RESULT_SUCCESS; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { android::AudioSfDecoder* decoder = static_cast(ap->mAPlayer.get()); SLuint32 valueSize = 0; if (!decoder->getPcmFormatValueSize(index, &valueSize)) { res = SL_RESULT_PARAMETER_INVALID; } else { // *pValueSize is the size of the region used to store the key value AND // the information about the value (size, lang, encoding) *pValueSize = valueSize + sizeof(SLMetadataInfo); } } break; default: *pValueSize = 0; res = SL_RESULT_PARAMETER_INVALID; break; } return res; } //----------------------------------------------------------------------------- // precondition // called with no lock held // ap != NULL // pValue != NULL SLresult android_audioPlayer_metadata_getValue(CAudioPlayer *ap, SLuint32 index, SLuint32 size, SLMetadataInfo *pValue) { if (ap->mAPlayer == 0) { return SL_RESULT_PARAMETER_INVALID; } SLresult res = SL_RESULT_SUCCESS; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { android::AudioSfDecoder* decoder = static_cast(ap->mAPlayer.get()); pValue->encoding = SL_CHARACTERENCODING_BINARY; memcpy((char *) pValue->langCountry, "en", 3); // applicable here? SLuint32 valueSize = 0; if ((size < sizeof(SLMetadataInfo) || (!decoder->getPcmFormatValueSize(index, &valueSize)) || (!decoder->getPcmFormatKeyValue(index, size - sizeof(SLMetadataInfo), (SLuint32*)pValue->data)))) { res = SL_RESULT_PARAMETER_INVALID; } else { pValue->size = valueSize; } } break; default: res = SL_RESULT_PARAMETER_INVALID; break; } return res; } //----------------------------------------------------------------------------- // preconditions // ap != NULL // mutex is locked // play state has changed void android_audioPlayer_setPlayState(CAudioPlayer *ap) { SLuint32 playState = ap->mPlay.mState; AndroidObjectState objState = ap->mAndroidObjState; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: switch (playState) { case SL_PLAYSTATE_STOPPED: SL_LOGV("setting AudioPlayer to SL_PLAYSTATE_STOPPED"); if (ap->mAudioTrack != 0) { ap->mAudioTrack->stop(); } break; case SL_PLAYSTATE_PAUSED: SL_LOGV("setting AudioPlayer to SL_PLAYSTATE_PAUSED"); if (ap->mAudioTrack != 0) { ap->mAudioTrack->pause(); } break; case SL_PLAYSTATE_PLAYING: SL_LOGV("setting AudioPlayer to SL_PLAYSTATE_PLAYING"); if (ap->mAudioTrack != 0) { // instead of ap->mAudioTrack->start(); ap->mDeferredStart = true; } break; default: // checked by caller, should not happen break; } break; case AUDIOPLAYER_FROM_URIFD: // intended fall-through case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: // FIXME report and use the return code to the lock mechanism, which is where play state // changes are updated (see object_unlock_exclusive_attributes()) aplayer_setPlayState(ap->mAPlayer, playState, &ap->mAndroidObjState); break; default: SL_LOGE(ERROR_PLAYERSETPLAYSTATE_UNEXPECTED_OBJECT_TYPE_D, ap->mAndroidObjType); break; } } //----------------------------------------------------------------------------- // call when either player event flags, marker position, or position update period changes void android_audioPlayer_usePlayEventMask(CAudioPlayer *ap) { IPlay *pPlayItf = &ap->mPlay; SLuint32 eventFlags = pPlayItf->mEventFlags; /*switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE:*/ if (ap->mAPlayer != 0) { assert(ap->mAudioTrack == 0); ap->mAPlayer->setPlayEvents((int32_t) eventFlags, (int32_t) pPlayItf->mMarkerPosition, (int32_t) pPlayItf->mPositionUpdatePeriod); return; } if (ap->mAudioTrack == 0) { return; } if (eventFlags & SL_PLAYEVENT_HEADATMARKER) { ap->mAudioTrack->setMarkerPosition((uint32_t)((((int64_t)pPlayItf->mMarkerPosition * sles_to_android_sampleRate(ap->mSampleRateMilliHz)))/1000)); } else { // clear marker ap->mAudioTrack->setMarkerPosition(0); } if (eventFlags & SL_PLAYEVENT_HEADATNEWPOS) { ap->mAudioTrack->setPositionUpdatePeriod( (uint32_t)((((int64_t)pPlayItf->mPositionUpdatePeriod * sles_to_android_sampleRate(ap->mSampleRateMilliHz)))/1000)); } else { // clear periodic update ap->mAudioTrack->setPositionUpdatePeriod(0); } if (eventFlags & SL_PLAYEVENT_HEADATEND) { // nothing to do for SL_PLAYEVENT_HEADATEND, callback event will be checked against mask } if (eventFlags & SL_PLAYEVENT_HEADMOVING) { // FIXME support SL_PLAYEVENT_HEADMOVING SL_LOGD("[ FIXME: IPlay_SetCallbackEventsMask(SL_PLAYEVENT_HEADMOVING) on an " "SL_OBJECTID_AUDIOPLAYER to be implemented ]"); } if (eventFlags & SL_PLAYEVENT_HEADSTALLED) { // nothing to do for SL_PLAYEVENT_HEADSTALLED, callback event will be checked against mask } } //----------------------------------------------------------------------------- SLresult android_audioPlayer_getDuration(IPlay *pPlayItf, SLmillisecond *pDurMsec) { CAudioPlayer *ap = (CAudioPlayer *)pPlayItf->mThis; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD: // intended fall-through case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: { int32_t durationMsec = ANDROID_UNKNOWN_TIME; if (ap->mAPlayer != 0) { ap->mAPlayer->getDurationMsec(&durationMsec); } *pDurMsec = durationMsec == ANDROID_UNKNOWN_TIME ? SL_TIME_UNKNOWN : durationMsec; break; } case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: default: { *pDurMsec = SL_TIME_UNKNOWN; } } return SL_RESULT_SUCCESS; } //----------------------------------------------------------------------------- void android_audioPlayer_getPosition(IPlay *pPlayItf, SLmillisecond *pPosMsec) { CAudioPlayer *ap = (CAudioPlayer *)pPlayItf->mThis; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: if ((ap->mSampleRateMilliHz == UNKNOWN_SAMPLERATE) || (ap->mAudioTrack == 0)) { *pPosMsec = 0; } else { uint32_t positionInFrames; ap->mAudioTrack->getPosition(&positionInFrames); *pPosMsec = ((int64_t)positionInFrames * 1000) / sles_to_android_sampleRate(ap->mSampleRateMilliHz); } break; case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_URIFD: case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: { int32_t posMsec = ANDROID_UNKNOWN_TIME; if (ap->mAPlayer != 0) { ap->mAPlayer->getPositionMsec(&posMsec); } *pPosMsec = posMsec == ANDROID_UNKNOWN_TIME ? 0 : posMsec; break; } default: *pPosMsec = 0; } } //----------------------------------------------------------------------------- SLresult android_audioPlayer_seek(CAudioPlayer *ap, SLmillisecond posMsec) { SLresult result = SL_RESULT_SUCCESS; switch(ap->mAndroidObjType) { case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: // intended fall-through case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: result = SL_RESULT_FEATURE_UNSUPPORTED; break; case AUDIOPLAYER_FROM_URIFD: // intended fall-through case AUDIOPLAYER_FROM_URIFD_TO_PCM_BUFFERQUEUE: if (ap->mAPlayer != 0) { ap->mAPlayer->seek(posMsec); } break; default: break; } return result; } //----------------------------------------------------------------------------- SLresult android_audioPlayer_loop(CAudioPlayer *ap, SLboolean loopEnable) { SLresult result = SL_RESULT_SUCCESS; switch (ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD: // case AUDIOPLAY_FROM_URIFD_TO_PCM_BUFFERQUEUE: // would actually work, but what's the point? if (ap->mAPlayer != 0) { ap->mAPlayer->loop((bool)loopEnable); } break; default: result = SL_RESULT_FEATURE_UNSUPPORTED; break; } return result; } //----------------------------------------------------------------------------- SLresult android_audioPlayer_setBufferingUpdateThresholdPerMille(CAudioPlayer *ap, SLpermille threshold) { SLresult result = SL_RESULT_SUCCESS; switch (ap->mAndroidObjType) { case AUDIOPLAYER_FROM_URIFD: if (ap->mAPlayer != 0) { ap->mAPlayer->setBufferingUpdateThreshold(threshold / 10); } break; default: {} } return result; } //----------------------------------------------------------------------------- void android_audioPlayer_bufferQueue_onRefilled_l(CAudioPlayer *ap) { // the AudioTrack associated with the AudioPlayer receiving audio from a PCM buffer // queue was stopped when the queue become empty, we restart as soon as a new buffer // has been enqueued since we're in playing state if (ap->mAudioTrack != 0) { // instead of ap->mAudioTrack->start(); ap->mDeferredStart = true; } // when the queue became empty, an underflow on the prefetch status itf was sent. Now the queue // has received new data, signal it has sufficient data if (IsInterfaceInitialized(&ap->mObject, MPH_PREFETCHSTATUS)) { // we wouldn't have been called unless we were previously in the underflow state assert(SL_PREFETCHSTATUS_UNDERFLOW == ap->mPrefetchStatus.mStatus); assert(0 == ap->mPrefetchStatus.mLevel); ap->mPrefetchStatus.mStatus = SL_PREFETCHSTATUS_SUFFICIENTDATA; ap->mPrefetchStatus.mLevel = 1000; // callback or no callback? SLuint32 prefetchEvents = ap->mPrefetchStatus.mCallbackEventsMask & (SL_PREFETCHEVENT_STATUSCHANGE | SL_PREFETCHEVENT_FILLLEVELCHANGE); if (SL_PREFETCHEVENT_NONE != prefetchEvents) { ap->mPrefetchStatus.mDeferredPrefetchCallback = ap->mPrefetchStatus.mCallback; ap->mPrefetchStatus.mDeferredPrefetchContext = ap->mPrefetchStatus.mContext; ap->mPrefetchStatus.mDeferredPrefetchEvents = prefetchEvents; } } } //----------------------------------------------------------------------------- /* * BufferQueue::Clear */ SLresult android_audioPlayer_bufferQueue_onClear(CAudioPlayer *ap) { SLresult result = SL_RESULT_SUCCESS; switch (ap->mAndroidObjType) { //----------------------------------- // AudioTrack case AUDIOPLAYER_FROM_PCM_BUFFERQUEUE: if (ap->mAudioTrack != 0) { ap->mAudioTrack->flush(); } break; default: result = SL_RESULT_INTERNAL_ERROR; break; } return result; } //----------------------------------------------------------------------------- void android_audioPlayer_androidBufferQueue_clear_l(CAudioPlayer *ap) { switch (ap->mAndroidObjType) { case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: if (ap->mAPlayer != 0) { android::StreamPlayer* splr = static_cast(ap->mAPlayer.get()); splr->appClear_l(); } break; case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: // nothing to do here, fall through default: break; } } void android_audioPlayer_androidBufferQueue_onRefilled_l(CAudioPlayer *ap) { switch (ap->mAndroidObjType) { case AUDIOPLAYER_FROM_TS_ANDROIDBUFFERQUEUE: if (ap->mAPlayer != 0) { android::StreamPlayer* splr = static_cast(ap->mAPlayer.get()); splr->queueRefilled(); } break; case AUDIOPLAYER_FROM_ADTS_ABQ_TO_PCM_BUFFERQUEUE: // FIXME this may require waking up the decoder if it is currently starved and isn't polling default: break; } }