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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.
*/
#include <android/content/AttributionSourceState.h>
#ifndef INCLUDING_FROM_AUDIOFLINGER_H
#error This header file should only be included from AudioFlinger.h
#endif
// record track
class RecordTrack : public TrackBase {
public:
RecordTrack(RecordThread *thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_session_t sessionId,
pid_t creatorPid,
const AttributionSourceState& attributionSource,
audio_input_flags_t flags,
track_type type,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
int32_t startFrames = -1);
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; }
void appendDumpHeader(String8& result);
void appendDump(String8& result, 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; }
bool isDirect() const override
{ return (mFlags & AUDIO_INPUT_FLAG_DIRECT) != 0; }
void setSilenced(bool silenced) { if (!isPatchTrack()) mSilenced = silenced; }
bool isSilenced() const { return mSilenced; }
status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction);
status_t setPreferredMicrophoneFieldDimension(float zoom);
status_t shareAudioHistory(const std::string& sharedAudioPackageName,
int64_t sharedAudioStartMs);
int32_t startFrames() { return mStartFrames; }
static bool checkServerLatencySupported(
audio_format_t format, audio_input_flags_t flags) {
return audio_is_linear_pcm(format)
&& (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
}
private:
friend class AudioFlinger; // for mState
DISALLOW_COPY_AND_ASSIGN(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;
bool mSilenced;
std::string mSharedAudioPackageName = {};
int32_t mStartFrames = -1;
};
// playback track, used by PatchPanel
class PatchRecord : public RecordTrack, public PatchTrackBase {
public:
PatchRecord(RecordThread *recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
void *buffer,
size_t bufferSize,
audio_input_flags_t flags,
const Timeout& timeout = {});
virtual ~PatchRecord();
virtual Source* getSource() { return nullptr; }
// 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);
size_t writeFrames(const void* src, size_t frameCount, size_t frameSize) {
return writeFrames(this, src, frameCount, frameSize);
}
protected:
/** Write the source data into the buffer provider. @return written frame count. */
static size_t writeFrames(AudioBufferProvider* dest, const void* src,
size_t frameCount, size_t frameSize);
}; // end of PatchRecord
class PassthruPatchRecord : public PatchRecord, public Source {
public:
PassthruPatchRecord(RecordThread *recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
size_t frameCount,
audio_input_flags_t flags);
Source* getSource() override { return static_cast<Source*>(this); }
// Source interface
status_t read(void *buffer, size_t bytes, size_t *read) override;
status_t getCapturePosition(int64_t *frames, int64_t *time) override;
status_t standby() override;
// AudioBufferProvider interface
// This interface is used by RecordThread to pass the data obtained
// from HAL or other source to the client. PassthruPatchRecord receives
// the data in 'obtainBuffer' so these calls are stubbed out.
status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
// PatchProxyBufferProvider interface
// This interface is used from DirectOutputThread to acquire data from HAL.
bool producesBufferOnDemand() const override { return true; }
status_t obtainBuffer(Proxy::Buffer *buffer, const struct timespec *timeOut = nullptr) override;
void releaseBuffer(Proxy::Buffer *buffer) override;
private:
// This is to use with PatchRecord::writeFrames
struct PatchRecordAudioBufferProvider : public AudioBufferProvider {
explicit PatchRecordAudioBufferProvider(PassthruPatchRecord& passthru) :
mPassthru(passthru) {}
status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override {
return mPassthru.PatchRecord::getNextBuffer(buffer);
}
void releaseBuffer(AudioBufferProvider::Buffer* buffer) override {
return mPassthru.PatchRecord::releaseBuffer(buffer);
}
private:
PassthruPatchRecord& mPassthru;
};
sp<StreamInHalInterface> obtainStream(sp<ThreadBase>* thread);
PatchRecordAudioBufferProvider mPatchRecordAudioBufferProvider;
std::unique_ptr<void, decltype(free)*> mSinkBuffer; // frame size aligned continuous buffer
std::unique_ptr<void, decltype(free)*> mStubBuffer; // buffer used for AudioBufferProvider
size_t mUnconsumedFrames = 0;
std::mutex mReadLock;
std::condition_variable mReadCV;
size_t mReadBytes = 0; // GUARDED_BY(mReadLock)
status_t mReadError = NO_ERROR; // GUARDED_BY(mReadLock)
int64_t mLastReadFrames = 0; // accessed on RecordThread only
};