Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 4c8b02a..e56efbb 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -185,7 +185,6 @@
     mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
     mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
     mCblk->out = 0;
-    mSampleRate = sampleRate;
     mFormat = format;
     // Update buffer size in case it has been limited by AudioFlinger during track creation
     mFrameCount = mCblk->frameCount;
@@ -196,7 +195,7 @@
     mRemainingFrames = notificationFrames;
     mUserData = user;
     // TODO: add audio hardware input latency here
-    mLatency = (1000*mFrameCount) / mSampleRate;
+    mLatency = (1000*mFrameCount) / sampleRate;
     mMarkerPosition = 0;
     mMarkerReached = false;
     mNewPosition = 0;
@@ -218,11 +217,6 @@
     return mLatency;
 }
 
-uint32_t AudioRecord::sampleRate() const
-{
-    return mSampleRate;
-}
-
 int AudioRecord::format() const
 {
     return mFormat;
@@ -321,6 +315,11 @@
     return !mActive;
 }
 
+uint32_t AudioRecord::getSampleRate()
+{
+    return mCblk->sampleRate;
+}
+
 status_t AudioRecord::setMarkerPosition(uint32_t marker)
 {
     if (mCbf == 0) return INVALID_OPERATION;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 289bd75..b2c067b 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -243,7 +243,6 @@
     mCblk->volume[0] = mCblk->volume[1] = 0x1000;
     mVolume[LEFT] = 1.0f;
     mVolume[RIGHT] = 1.0f;
-    mSampleRate = sampleRate;
     mStreamType = streamType;
     mFormat = format;
     mChannelCount = channelCount;
@@ -254,7 +253,7 @@
     mNotificationFrames = notificationFrames;
     mRemainingFrames = notificationFrames;
     mUserData = user;
-    mLatency = afLatency + (1000*mFrameCount) / mSampleRate;
+    mLatency = afLatency + (1000*mFrameCount) / sampleRate;
     mLoopCount = 0;
     mMarkerPosition = 0;
     mMarkerReached = false;
@@ -281,11 +280,6 @@
     return mStreamType;
 }
 
-uint32_t AudioTrack::sampleRate() const
-{
-    return mSampleRate;
-}
-
 int AudioTrack::format() const
 {
     return mFormat;
@@ -438,24 +432,23 @@
     *right = mVolume[RIGHT];
 }
 
-void AudioTrack::setSampleRate(int rate)
+status_t AudioTrack::setSampleRate(int rate)
 {
     int afSamplingRate;
 
     if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
-        return;
+        return NO_INIT;
     }
     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
-    if (rate <= 0) rate = 1;
-    if (rate > afSamplingRate*2) rate = afSamplingRate*2;
-    if (rate > MAX_SAMPLE_RATE) rate = MAX_SAMPLE_RATE;
+    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
 
-    mCblk->sampleRate = (uint16_t)rate;
+    mCblk->sampleRate = rate;
+    return NO_ERROR;
 }
 
 uint32_t AudioTrack::getSampleRate()
 {
-    return uint32_t(mCblk->sampleRate);
+    return mCblk->sampleRate;
 }
 
 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
@@ -866,7 +859,7 @@
     result.append(buffer);
     snprintf(buffer, 255, "  format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
     result.append(buffer);
-    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", mSampleRate, mStatus, mMuted);
+    snprintf(buffer, 255, "  sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
     result.append(buffer);
     snprintf(buffer, 255, "  active(%d), latency (%d)\n", mActive, mLatency);
     result.append(buffer);