summaryrefslogtreecommitdiffstats
path: root/services/audioflinger/Threads.h
diff options
context:
space:
mode:
Diffstat (limited to 'services/audioflinger/Threads.h')
-rw-r--r--services/audioflinger/Threads.h176
1 files changed, 118 insertions, 58 deletions
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index a2fb874..5617c0c 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -36,6 +36,8 @@ public:
audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
virtual ~ThreadBase();
+ virtual status_t readyToRun();
+
void dumpBase(int fd, const Vector<String16>& args);
void dumpEffectChains(int fd, const Vector<String16>& args);
@@ -63,7 +65,7 @@ public:
class IoConfigEvent : public ConfigEvent {
public:
IoConfigEvent(int event, int param) :
- ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(event) {}
+ ConfigEvent(CFG_EVENT_IO), mEvent(event), mParam(param) {}
virtual ~IoConfigEvent() {}
int event() const { return mEvent; }
@@ -141,6 +143,7 @@ public:
void sendIoConfigEvent_l(int event, int param = 0);
void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
void processConfigEvents();
+ void processConfigEvents_l();
// see note at declaration of mStandby, mOutDevice and mInDevice
bool standby() const { return mStandby; }
@@ -156,7 +159,7 @@ public:
int sessionId,
effect_descriptor_t *desc,
int *enabled,
- status_t *status);
+ status_t *status /*non-NULL*/);
void disconnectEffect(const sp< EffectModule>& effect,
EffectHandle *handle,
bool unpinIfLast);
@@ -198,13 +201,13 @@ public:
// effect
void removeEffect_l(const sp< EffectModule>& effect);
// detach all tracks connected to an auxiliary effect
- virtual void detachAuxEffect_l(int effectId) {}
+ virtual void detachAuxEffect_l(int effectId __unused) {}
// returns either EFFECT_SESSION if effects on this audio session exist in one
// chain, or TRACK_SESSION if tracks on this audio session exist, or both
virtual uint32_t hasAudioSession(int sessionId) const = 0;
// the value returned by default implementation is not important as the
// strategy is only meaningful for PlaybackThread which implements this method
- virtual uint32_t getStrategyForSession_l(int sessionId) { return 0; }
+ virtual uint32_t getStrategyForSession_l(int sessionId __unused) { return 0; }
// suspend or restore effect according to the type of effect passed. a NULL
// type pointer means suspend all effects in the session
@@ -267,14 +270,15 @@ protected:
const sp<AudioFlinger> mAudioFlinger;
- // updated by PlaybackThread::readOutputParameters() or
- // RecordThread::readInputParameters()
+ // updated by PlaybackThread::readOutputParameters_l() or
+ // RecordThread::readInputParameters_l()
uint32_t mSampleRate;
size_t mFrameCount; // output HAL, direct output, record
audio_channel_mask_t mChannelMask;
uint32_t mChannelCount;
size_t mFrameSize;
audio_format_t mFormat;
+ size_t mBufferSize; // HAL buffer size for read() or write()
// Parameter sequence by client: binder thread calling setParameters():
// 1. Lock mLock
@@ -303,12 +307,12 @@ protected:
Vector<ConfigEvent *> mConfigEvents;
// These fields are written and read by thread itself without lock or barrier,
- // and read by other threads without lock or barrier via standby() , outDevice()
+ // and read by other threads without lock or barrier via standby(), outDevice()
// and inDevice().
// Because of the absence of a lock or barrier, any other thread that reads
// these fields must use the information in isolation, or be prepared to deal
// with possibility that it might be inconsistent with other information.
- bool mStandby; // Whether thread is currently in standby.
+ bool mStandby; // Whether thread is currently in standby.
audio_devices_t mOutDevice; // output device
audio_devices_t mInDevice; // input device
audio_source_t mAudioSource; // (see audio.h, audio_source_t)
@@ -358,7 +362,6 @@ public:
void dump(int fd, const Vector<String16>& args);
// Thread virtuals
- virtual status_t readyToRun();
virtual bool threadLoop();
// RefBase
@@ -391,7 +394,7 @@ protected:
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
-
+ virtual void onAddNewTrack_l();
// ThreadBase virtuals
virtual void preExit();
@@ -419,13 +422,13 @@ public:
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
int uid,
- status_t *status);
+ status_t *status /*non-NULL*/);
AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
@@ -447,7 +450,11 @@ public:
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
- int16_t *mixBuffer() const { return mMixBuffer; };
+ // FIXME rename mixBuffer() to sinkBuffer() and remove int16_t* dependency.
+ // Consider also removing and passing an explicit mMainBuffer initialization
+ // parameter to AF::PlaybackThread::Track::Track().
+ int16_t *mixBuffer() const {
+ return reinterpret_cast<int16_t *>(mSinkBuffer); };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
@@ -475,11 +482,68 @@ public:
status_t getTimestamp_l(AudioTimestamp& timestamp);
protected:
- // updated by readOutputParameters()
+ // updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
- int16_t* mMixBuffer; // frame size aligned mix buffer
- int8_t* mAllocMixBuffer; // mixer buffer allocation address
+ void* mSinkBuffer; // frame size aligned sink buffer
+
+ // TODO:
+ // Rearrange the buffer info into a struct/class with
+ // clear, copy, construction, destruction methods.
+ //
+ // mSinkBuffer also has associated with it:
+ //
+ // mSinkBufferSize: Sink Buffer Size
+ // mFormat: Sink Buffer Format
+
+ // Mixer Buffer (mMixerBuffer*)
+ //
+ // In the case of floating point or multichannel data, which is not in the
+ // sink format, it is required to accumulate in a higher precision or greater channel count
+ // buffer before downmixing or data conversion to the sink buffer.
+
+ // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer.
+ bool mMixerBufferEnabled;
+
+ // Storage, 32 byte aligned (may make this alignment a requirement later).
+ // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames.
+ void* mMixerBuffer;
+
+ // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize.
+ size_t mMixerBufferSize;
+
+ // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only.
+ audio_format_t mMixerBufferFormat;
+
+ // An internal flag set to true by MixerThread::prepareTracks_l()
+ // when mMixerBuffer contains valid data after mixing.
+ bool mMixerBufferValid;
+
+ // Effects Buffer (mEffectsBuffer*)
+ //
+ // In the case of effects data, which is not in the sink format,
+ // it is required to accumulate in a different buffer before data conversion
+ // to the sink buffer.
+
+ // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer.
+ bool mEffectBufferEnabled;
+
+ // Storage, 32 byte aligned (may make this alignment a requirement later).
+ // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames.
+ void* mEffectBuffer;
+
+ // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize.
+ size_t mEffectBufferSize;
+
+ // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only.
+ audio_format_t mEffectBufferFormat;
+
+ // An internal flag set to true by MixerThread::prepareTracks_l()
+ // when mEffectsBuffer contains valid data after mixing.
+ //
+ // When this is set, all mixer data is routed into the effects buffer
+ // for any processing (including output processing).
+ bool mEffectBufferValid;
// suspend count, > 0 means suspended. While suspended, the thread continues to pull from
// tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle
@@ -539,7 +603,7 @@ private:
void removeTrack_l(const sp<Track>& track);
void broadcast_l();
- void readOutputParameters();
+ void readOutputParameters_l();
virtual void dumpInternals(int fd, const Vector<String16>& args);
void dumpTracks(int fd, const Vector<String16>& args);
@@ -558,7 +622,7 @@ private:
// FIXME rename these former local variables of threadLoop to standard "m" names
nsecs_t standbyTime;
- size_t mixBufferSize;
+ size_t mSinkBufferSize;
// cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l()
uint32_t activeSleepTime;
@@ -623,13 +687,12 @@ private:
sp<NBLog::Writer> mFastMixerNBLogWriter;
public:
virtual bool hasFastMixer() const = 0;
- virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const
+ virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
{ FastTrackUnderruns dummy; return dummy; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
- virtual void flushOutput_l();
private:
// timestamp latch:
@@ -748,11 +811,11 @@ protected:
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_exit();
- virtual void flushOutput_l();
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
+ virtual void onAddNewTrack_l();
private:
void flushHw_l();
@@ -838,17 +901,28 @@ public:
// record thread
-class RecordThread : public ThreadBase, public AudioBufferProvider
- // derives from AudioBufferProvider interface for use by resampler
+class RecordThread : public ThreadBase
{
public:
+ class RecordTrack;
+ class ResamplerBufferProvider : public AudioBufferProvider
+ // derives from AudioBufferProvider interface for use by resampler
+ {
+ public:
+ ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack) { }
+ virtual ~ResamplerBufferProvider() { }
+ // AudioBufferProvider interface
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
+ virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+ private:
+ RecordTrack * const mRecordTrack;
+ };
+
#include "RecordTracks.h"
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
- uint32_t sampleRate,
- audio_channel_mask_t channelMask,
audio_io_handle_t id,
audio_devices_t outDevice,
audio_devices_t inDevice
@@ -867,23 +941,23 @@ public:
// Thread virtuals
virtual bool threadLoop();
- virtual status_t readyToRun();
// RefBase
virtual void onFirstRef();
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
+
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount,
+ size_t *pFrameCount,
int sessionId,
int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
- status_t *status);
+ status_t *status /*non-NULL*/);
status_t start(RecordTrack* recordTrack,
AudioSystem::sync_event_t event,
@@ -897,15 +971,12 @@ public:
AudioStreamIn* clearInput();
virtual audio_stream_t* stream() const;
- // AudioBufferProvider interface
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts);
- virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
- void readInputParameters();
- virtual unsigned int getInputFramesLost();
+ void readInputParameters_l();
+ virtual uint32_t getInputFramesLost();
virtual status_t addEffectChain_l(const sp<EffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<EffectChain>& chain);
@@ -920,44 +991,33 @@ public:
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const;
static void syncStartEventCallback(const wp<SyncEvent>& event);
- void handleSyncStartEvent(const sp<SyncEvent>& event);
virtual size_t frameCount() const { return mFrameCount; }
- bool hasFastRecorder() const { return false; }
+ bool hasFastCapture() const { return false; }
private:
- void clearSyncStartEvent();
-
// Enter standby if not already in standby, and set mStandby flag
- void standby();
+ void standbyIfNotAlreadyInStandby();
// Call the HAL standby method unconditionally, and don't change mStandby flag
- void inputStandBy();
+ void inputStandBy();
AudioStreamIn *mInput;
SortedVector < sp<RecordTrack> > mTracks;
- // mActiveTrack has dual roles: it indicates the current active track, and
+ // mActiveTracks has dual roles: it indicates the current active track(s), and
// is used together with mStartStopCond to indicate start()/stop() progress
- sp<RecordTrack> mActiveTrack;
+ SortedVector< sp<RecordTrack> > mActiveTracks;
+ // generation counter for mActiveTracks
+ int mActiveTracksGen;
Condition mStartStopCond;
- // updated by RecordThread::readInputParameters()
- AudioResampler *mResampler;
- // interleaved stereo pairs of fixed-point signed Q19.12
- int32_t *mRsmpOutBuffer;
- int16_t *mRsmpInBuffer; // [mFrameCount * mChannelCount]
- size_t mRsmpInIndex;
- size_t mBufferSize; // stream buffer size for read()
- const uint32_t mReqChannelCount;
- const uint32_t mReqSampleRate;
- ssize_t mBytesRead;
- // 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;
+ // resampler converts input at HAL Hz to output at AudioRecord client Hz
+ int16_t *mRsmpInBuffer; // see new[] for details on the size
+ size_t mRsmpInFrames; // size of resampler input in frames
+ size_t mRsmpInFramesP2;// size rounded up to a power-of-2
+
+ // rolling index that is never cleared
+ int32_t mRsmpInRear; // last filled frame + 1
// For dumpsys
const sp<NBAIO_Sink> mTeeSink;
generated by cgit v1.1 at 2025-01-08 17:50:26 +0000