Merge "Add playback rate to AudioTrack"

3 files changed, 65 insertions, 2 deletions

diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 3c30148..07d946d 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -42,6 +42,8 @@
 #define AUDIO_TIMESTRETCH_PITCH_MAX    2.0f
 #define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
 
+// TODO: Consider putting these inlines into a class scope
+
 // Returns the source frames needed to resample to destination frames.  This is not a precise
 // value and depends on the resampler (and possibly how it handles rounding internally).
 // Nevertheless, this should be an upper bound on the requirements of the resampler.
@@ -66,4 +68,13 @@ static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSam
     return dstFrames > 2 ? dstFrames - 2 : 0;
 }
 
+static inline size_t sourceFramesNeededWithTimestretch(
+        uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate,
+        float speed) {
+    // required is the number of input frames the resampler needs
+    size_t required = sourceFramesNeeded(srcSampleRate, dstFramesRequired, dstSampleRate);
+    // to deliver this, the time stretcher requires:
+    return required * (double)speed + 1 + 1; // accounting for rounding dependencies
+}
+
 #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index e7e0703..a06197f 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -359,6 +359,21 @@ public:
     /* Return current source sample rate in Hz */
             uint32_t    getSampleRate() const;
 
+    /* Set source playback rate for timestretch
+     * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster
+     * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch
+     *
+     * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX
+     * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX
+     *
+     * Speed increases the playback rate of media, but does not alter pitch.
+     * Pitch increases the "tonal frequency" of media, but does not affect the playback rate.
+     */
+            status_t    setPlaybackRate(float speed, float pitch);
+
+    /* Return current playback rate */
+            void        getPlaybackRate(float *speed, float *pitch) const;
+
     /* Enables looping and sets the start and end points of looping.
      * Only supported for static buffer mode.
      *
@@ -719,6 +734,9 @@ protected:
             // increment mPosition by the delta of mServer, and return new value of mPosition
             uint32_t updateAndGetPosition_l();
 
+            // check sample rate and speed is compatible with AudioTrack
+            bool     isSampleRateSpeedAllowed_l(uint32_t sampleRate, float speed) const;
+
     // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
@@ -730,6 +748,8 @@ protected:
     float                   mVolume[2];
     float                   mSendLevel;
     mutable uint32_t        mSampleRate;            // mutable because getSampleRate() can update it
+    float                   mSpeed;                 // timestretch: 1.0f for normal speed.
+    float                   mPitch;                 // timestretch: 1.0f for normal pitch.
     size_t                  mFrameCount;            // corresponds to current IAudioTrack, value is
                                                     // reported back by AudioFlinger to the client
     size_t                  mReqFrameCount;         // frame count to request the first or next time
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 5644428..6cc2e2b 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -25,6 +25,7 @@
 #include <utils/Log.h>
 #include <utils/RefBase.h>
 #include <audio_utils/roundup.h>
+#include <media/AudioResamplerPublic.h>
 #include <media/SingleStateQueue.h>
 
 namespace android {
@@ -113,6 +114,14 @@ struct AudioTrackSharedStatic {
                     mPosLoopQueue;
 };
 
+
+struct AudioTrackPlaybackRate {
+    float mSpeed;
+    float mPitch;
+};
+
+typedef SingleStateQueue<AudioTrackPlaybackRate> AudioTrackPlaybackRateQueue;
+
 // ----------------------------------------------------------------------------
 
 // Important: do not add any virtual methods, including ~
@@ -159,6 +168,8 @@ private:
                 uint32_t    mSampleRate;    // AudioTrack only: client's requested sample rate in Hz
                                             // or 0 == default. Write-only client, read-only server.
 
+                AudioTrackPlaybackRateQueue::Shared mPlaybackRateQueue;
+
                 // client write-only, server read-only
                 uint16_t    mSendLevel;      // Fixed point U4.12 so 0x1000 means 1.0
 
@@ -313,7 +324,8 @@ public:
     AudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
             size_t frameSize, bool clientInServer = false)
         : ClientProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/,
-          clientInServer) { }
+          clientInServer),
+          mPlaybackRateMutator(&cblk->mPlaybackRateQueue) { }
     virtual ~AudioTrackClientProxy() { }
 
     // No barriers on the following operations, so the ordering of loads/stores
@@ -333,6 +345,13 @@ public:
         mCblk->mSampleRate = sampleRate;
     }
 
+    void        setPlaybackRate(float speed, float pitch) {
+        AudioTrackPlaybackRate playbackRate;
+        playbackRate.mSpeed = speed;
+        playbackRate.mPitch = pitch;
+        mPlaybackRateMutator.push(playbackRate);
+    }
+
     virtual void flush();
 
     virtual uint32_t    getUnderrunFrames() const {
@@ -344,6 +363,9 @@ public:
     bool        getStreamEndDone() const;
 
     status_t    waitStreamEndDone(const struct timespec *requested);
+
+private:
+    AudioTrackPlaybackRateQueue::Mutator   mPlaybackRateMutator;
 };
 
 class StaticAudioTrackClientProxy : public AudioTrackClientProxy {
@@ -458,8 +480,11 @@ class AudioTrackServerProxy : public ServerProxy {
 public:
     AudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
             size_t frameSize, bool clientInServer = false, uint32_t sampleRate = 0)
-        : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer) {
+        : ServerProxy(cblk, buffers, frameCount, frameSize, true /*isOut*/, clientInServer),
+          mPlaybackRateObserver(&cblk->mPlaybackRateQueue) {
         mCblk->mSampleRate = sampleRate;
+        mPlaybackRate.mSpeed = AUDIO_TIMESTRETCH_SPEED_NORMAL;
+        mPlaybackRate.mPitch = AUDIO_TIMESTRETCH_PITCH_NORMAL;
     }
 protected:
     virtual ~AudioTrackServerProxy() { }
@@ -493,6 +518,13 @@ public:
 
     // Return the total number of frames that AudioFlinger has obtained and released
     virtual size_t      framesReleased() const { return mCblk->mServer; }
+
+    // Return the playback speed and pitch read atomically. Not multi-thread safe on server side.
+    void                getPlaybackRate(float *speed, float *pitch);
+
+private:
+    AudioTrackPlaybackRate                  mPlaybackRate;  // last observed playback rate
+    AudioTrackPlaybackRateQueue::Observer   mPlaybackRateObserver;
 };
 
 class StaticAudioTrackServerProxy : public AudioTrackServerProxy {