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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
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** See the License for the specific language governing permissions and
** limitations under the License.
#error This header file should only be included from AudioFlinger.h
// record track
class RecordTrack : public TrackBase {
RecordTrack(RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
audio_session_t sessionId,
uid_t uid,
audio_input_flags_t flags,
track_type type,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
virtual ~RecordTrack();
virtual status_t initCheck() const;
virtual status_t start(AudioSystem::sync_event_t event, audio_session_t triggerSession);
virtual void stop();
void destroy();
virtual void invalidate();
// clear the buffer overflow flag
void clearOverflow() { mOverflow = false; }
// set the buffer overflow flag and return previous value
bool setOverflow() { bool tmp = mOverflow; mOverflow = true;
return tmp; }
static void appendDumpHeader(String8& result);
void dump(char* buffer, size_t size, bool active);
void handleSyncStartEvent(const sp<SyncEvent>& event);
void clearSyncStartEvent();
void updateTrackFrameInfo(int64_t trackFramesReleased,
int64_t sourceFramesRead,
uint32_t halSampleRate,
const ExtendedTimestamp &timestamp);
virtual bool isFastTrack() const { return (mFlags & AUDIO_INPUT_FLAG_FAST) != 0; }
friend class AudioFlinger; // for mState
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
// releaseBuffer() not overridden
bool mOverflow; // overflow on most recent attempt to fill client buffer
AudioBufferProvider::Buffer mSink; // references client's buffer sink in shared memory
// sync event triggering actual audio capture. Frames read before this event will
// be dropped and therefore not read by the application.
sp<SyncEvent> mSyncStartEvent;
// number of captured frames to drop after the start sync event has been received.
// when < 0, maximum frames to drop before starting capture even if sync event is
// not received
ssize_t mFramesToDrop;
// used by resampler to find source frames
ResamplerBufferProvider *mResamplerBufferProvider;
// used by the record thread to convert frames to proper destination format
RecordBufferConverter *mRecordBufferConverter;
audio_input_flags_t mFlags;
// playback track, used by PatchPanel
class PatchRecord : virtual public RecordTrack, public PatchProxyBufferProvider {
PatchRecord(RecordThread *recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
audio_input_flags_t flags);
virtual ~PatchRecord();
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
// PatchProxyBufferProvider interface
virtual status_t obtainBuffer(Proxy::Buffer *buffer,
const struct timespec *timeOut = NULL);
virtual void releaseBuffer(Proxy::Buffer *buffer);
void setPeerProxy(PatchProxyBufferProvider *proxy) { mPeerProxy = proxy; }
sp<ClientProxy> mProxy;
PatchProxyBufferProvider* mPeerProxy;
struct timespec mPeerTimeout;
}; // end of PatchRecord