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

2 files changed, 22 insertions, 12 deletions

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 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio
         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 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args)
     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 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
                                              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 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>&
 
     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 @@ void AudioFlinger::RecordThread::readInputParameters()
     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 @@ public:
                 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 @@ protected:
                 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 @@ public:
                 // 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 @@ public:
     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 @@ private:
             // 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;