am 88e209dc: Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

Merge commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c'

* commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c':
  Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

3 files changed, 21 insertions, 36 deletions

diff --git a/media/java/android/media/AudioTrack.java b/media/java/android/media/AudioTrack.java
index 3cd841d..5f1be9d 100644
--- a/media/java/android/media/AudioTrack.java
+++ b/media/java/android/media/AudioTrack.java
@@ -425,8 +425,7 @@ public class AudioTrack
     }
     
     /**
-     * Returns the current playback rate in Hz. Note that this rate may differ from the one set
-     * with {@link #setPlaybackRate(int)} as the value effectively used is implementation-dependent.
+     * Returns the current playback rate in Hz.
      */
     public int getPlaybackRate() {
         return native_get_playback_rate();
@@ -651,18 +650,13 @@ public class AudioTrack
      * Sets the playback sample rate for this track. This sets the sampling rate at which
      * the audio data will be consumed and played back, not the original sampling rate of the
      * content. Setting it to half the sample rate of the content will cause the playback to
-     * last twice as long, but will also result result in a negative pitch shift.
-     * The current implementation supports a maximum sample rate of 64kHz.
-     * Use {@link #getSampleRate()} to check the rate actually used in hardware after 
-     * potential clamping.
+     * last twice as long, but will also result in a negative pitch shift.
+     * The valid sample rate range if from 1Hz to twice the value returned by
+     * {@link #getNativeOutputSampleRate(int)}.
      * @param sampleRateInHz the sample rate expressed in Hz
      * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
      *    {@link #ERROR_INVALID_OPERATION}
      */
-    // FIXME: the implementation should support twice the hardware output sample rate
-    //   (see {@link #getNativeOutputSampleRate(int)}), but currently
-    //  due to the representation of the sample rate in the native layer, the sample rate
-    //  is limited to 65535Hz
     public int setPlaybackRate(int sampleRateInHz) {
         if (mState != STATE_INITIALIZED) {
             return ERROR_INVALID_OPERATION;
@@ -670,8 +664,7 @@ public class AudioTrack
         if (sampleRateInHz <= 0) {
             return ERROR_BAD_VALUE;
         }
-        native_set_playback_rate(sampleRateInHz);
-        return SUCCESS;
+        return native_set_playback_rate(sampleRateInHz);
     }
 
 
@@ -1031,8 +1024,8 @@ public class AudioTrack
 
     private native final void native_setVolume(float leftVolume, float rightVolume);
 
-    private native final void native_set_playback_rate(int sampleRateInHz);
-    private native final int  native_get_playback_rate();
+    private native final int native_set_playback_rate(int sampleRateInHz);
+    private native final int native_get_playback_rate();
 
     private native final int native_set_marker_pos(int marker);
     private native final int native_get_marker_pos();
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index cf0965e..0a6f4f7 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -185,7 +185,6 @@ status_t AudioRecord::set(
     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 @@ status_t AudioRecord::set(
     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 @@ uint32_t AudioRecord::latency() const
     return mLatency;
 }
 
-uint32_t AudioRecord::sampleRate() const
-{
-    return mSampleRate;
-}
-
 int AudioRecord::format() const
 {
     return mFormat;
@@ -321,6 +315,11 @@ bool AudioRecord::stopped() const
     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 4a1b69e..af7dae5 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -243,7 +243,6 @@ status_t AudioTrack::set(
     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 @@ status_t AudioTrack::set(
     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 @@ int AudioTrack::streamType() const
     return mStreamType;
 }
 
-uint32_t AudioTrack::sampleRate() const
-{
-    return mSampleRate;
-}
-
 int AudioTrack::format() const
 {
     return mFormat;
@@ -438,24 +432,23 @@ void AudioTrack::getVolume(float* left, float* right)
     *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 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
     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);