| /* |
| ** |
| ** Copyright 2012, 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 INCLUDING_FROM_AUDIOFLINGER_H |
| #error This header file should only be included from AudioFlinger.h |
| #endif |
| |
| // base for record and playback |
| class TrackBase : public ExtendedAudioBufferProvider, public RefBase { |
| |
| public: |
| enum track_state { |
| IDLE, |
| TERMINATED, |
| FLUSHED, |
| STOPPED, |
| // next 2 states are currently used for fast tracks only |
| STOPPING_1, // waiting for first underrun |
| STOPPING_2, // waiting for presentation complete |
| RESUMING, |
| ACTIVE, |
| PAUSING, |
| PAUSED |
| }; |
| |
| TrackBase(ThreadBase *thread, |
| const sp<Client>& client, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| const sp<IMemory>& sharedBuffer, |
| int sessionId, |
| bool isOut); |
| virtual ~TrackBase(); |
| |
| virtual status_t start(AudioSystem::sync_event_t event, |
| int triggerSession) = 0; |
| virtual void stop() = 0; |
| sp<IMemory> getCblk() const { return mCblkMemory; } |
| audio_track_cblk_t* cblk() const { return mCblk; } |
| int sessionId() const { return mSessionId; } |
| virtual status_t setSyncEvent(const sp<SyncEvent>& event); |
| |
| protected: |
| TrackBase(const TrackBase&); |
| TrackBase& operator = (const TrackBase&); |
| |
| // AudioBufferProvider interface |
| virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) = 0; |
| virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); |
| |
| // ExtendedAudioBufferProvider interface is only needed for Track, |
| // but putting it in TrackBase avoids the complexity of virtual inheritance |
| virtual size_t framesReady() const { return SIZE_MAX; } |
| |
| audio_format_t format() const { return mFormat; } |
| |
| uint32_t channelCount() const { return mChannelCount; } |
| |
| audio_channel_mask_t channelMask() const { return mChannelMask; } |
| |
| uint32_t sampleRate() const; // FIXME inline after cblk sr moved |
| |
| // Return a pointer to the start of a contiguous slice of the track buffer. |
| // Parameter 'offset' is the requested start position, expressed in |
| // monotonically increasing frame units relative to the track epoch. |
| // Parameter 'frames' is the requested length, also in frame units. |
| // Always returns non-NULL. It is the caller's responsibility to |
| // verify that this will be successful; the result of calling this |
| // function with invalid 'offset' or 'frames' is undefined. |
| void* getBuffer(uint32_t offset, uint32_t frames) const; |
| |
| bool isStopped() const { |
| return (mState == STOPPED || mState == FLUSHED); |
| } |
| |
| // for fast tracks only |
| bool isStopping() const { |
| return mState == STOPPING_1 || mState == STOPPING_2; |
| } |
| bool isStopping_1() const { |
| return mState == STOPPING_1; |
| } |
| bool isStopping_2() const { |
| return mState == STOPPING_2; |
| } |
| |
| bool isTerminated() const { |
| return mState == TERMINATED; |
| } |
| |
| bool step(); // mStepCount is an implicit input |
| void reset(); |
| |
| bool isOut() const { return mIsOut; } |
| // true for Track and TimedTrack, false for RecordTrack, |
| // this could be a track type if needed later |
| |
| const wp<ThreadBase> mThread; |
| /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const |
| sp<IMemory> mCblkMemory; |
| audio_track_cblk_t* mCblk; |
| void* mBuffer; // start of track buffer, typically in shared memory |
| // except for OutputTrack when it is in local memory |
| void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize |
| // is based on mChannelCount and 16-bit samples |
| uint32_t mStepCount; // saves AudioBufferProvider::Buffer::frameCount as of |
| // time of releaseBuffer() for later use by step() |
| // we don't really need a lock for these |
| track_state mState; |
| const uint32_t mSampleRate; // initial sample rate only; for tracks which |
| // support dynamic rates, the current value is in control block |
| const audio_format_t mFormat; |
| const audio_channel_mask_t mChannelMask; |
| const uint8_t mChannelCount; |
| const size_t mFrameSize; // AudioFlinger's view of frame size in shared memory, |
| // where for AudioTrack (but not AudioRecord), |
| // 8-bit PCM samples are stored as 16-bit |
| const size_t mFrameCount;// size of track buffer given at createTrack() or |
| // openRecord(), and then adjusted as needed |
| |
| bool mStepServerFailed; |
| const int mSessionId; |
| Vector < sp<SyncEvent> >mSyncEvents; |
| const bool mIsOut; |
| ServerProxy* mServerProxy; |
| const int mId; |
| sp<NBAIO_Sink> mTeeSink; |
| sp<NBAIO_Source> mTeeSource; |
| }; |