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/*
**
** 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;
};