Move members from ThreadBase to PlaybackThread
Move mNormalFrameCount and frameCountHAL(), since they're not used by
record threads. Also comment which fields are updated by readParameters().
Change-Id: I5fc0a8a89cc637976f22d49271a5a3e136dab4e1
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 1a513c4..6a224ac 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -266,10 +266,9 @@
audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
: Thread(false /*canCallJava*/),
mType(type),
- mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),
- // mChannelMask
- mChannelCount(0),
- mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+ mAudioFlinger(audioFlinger),
+ // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are
+ // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
mParamStatus(NO_ERROR),
mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
@@ -425,8 +424,6 @@
result.append(buffer);
snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
result.append(buffer);
- snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
- result.append(buffer);
snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
result.append(buffer);
snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
@@ -932,6 +929,7 @@
audio_devices_t device,
type_t type)
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
+ mNormalFrameCount(0), mMixBuffer(NULL),
mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
// mStreamTypes[] initialized in constructor body
mOutput(output),
@@ -1054,6 +1052,8 @@
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
+ snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);
+ result.append(buffer);
snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
@@ -4920,7 +4920,6 @@
mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mInputBytes / mFrameSize;
- mNormalFrameCount = mFrameCount; // not used by record, but used by input effects
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index a17c279..b1b33b0 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -126,10 +126,8 @@
audio_channel_mask_t channelMask() const { return mChannelMask; }
audio_format_t format() const { return mFormat; }
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
- // and returns the normal mix buffer's frame count.
- size_t frameCount() const { return mNormalFrameCount; }
- // Return's the HAL's frame count i.e. fast mixer buffer size.
- size_t frameCountHAL() const { return mFrameCount; }
+ // and returns the [normal mix] buffer's frame count.
+ virtual size_t frameCount() const = 0;
size_t frameSize() const { return mFrameSize; }
// Should be "virtual status_t requestExitAndWait()" and override same
@@ -263,9 +261,11 @@
Condition mWaitWorkCV;
const sp<AudioFlinger> mAudioFlinger;
+
+ // updated by PlaybackThread::readOutputParameters() or
+ // RecordThread::readInputParameters()
uint32_t mSampleRate;
size_t mFrameCount; // output HAL, direct output, record
- size_t mNormalFrameCount; // normal mixer and effects
audio_channel_mask_t mChannelMask;
uint32_t mChannelCount;
size_t mFrameSize;
@@ -461,8 +461,15 @@
// called with AudioFlinger lock held
void invalidateTracks(audio_stream_type_t streamType);
+ virtual size_t frameCount() const { return mNormalFrameCount; }
+
+ // Return's the HAL's frame count i.e. fast mixer buffer size.
+ size_t frameCountHAL() const { return mFrameCount; }
protected:
+ // updated by readOutputParameters()
+ size_t mNormalFrameCount; // normal mixer and effects
+
int16_t* mMixBuffer; // frame size aligned mix buffer
int8_t* mAllocMixBuffer; // mixer buffer allocation address
@@ -871,6 +878,8 @@
static void syncStartEventCallback(const wp<SyncEvent>& event);
void handleSyncStartEvent(const sp<SyncEvent>& event);
+ virtual size_t frameCount() const { return mFrameCount; }
+
private:
void clearSyncStartEvent();
@@ -886,6 +895,8 @@
// is used together with mStartStopCond to indicate start()/stop() progress
sp<RecordTrack> mActiveTrack;
Condition mStartStopCond;
+
+ // updated by RecordThread::readInputParameters()
AudioResampler *mResampler;
int32_t *mRsmpOutBuffer;
int16_t *mRsmpInBuffer;