Use AudioPlaybackRate to hold TimestretchBufferProvider parameters

Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.

Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2

3 files changed, 86 insertions, 26 deletions

diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 07d946d..53b8c13 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -18,6 +18,9 @@
 #define ANDROID_AUDIO_RESAMPLER_PUBLIC_H
 
 #include <stdint.h>
+#include <math.h>
+
+namespace android {
 
 // AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original
 // audio sample rate and the target rate when downsampling,
@@ -34,13 +37,76 @@
 // an int32_t of the phase increments, making the resulting sample rate inexact.
 #define AUDIO_RESAMPLER_UP_RATIO_MAX 65536
 
-#define AUDIO_TIMESTRETCH_SPEED_MIN    0.5f
-#define AUDIO_TIMESTRETCH_SPEED_MAX    2.0f
+// AUDIO_TIMESTRETCH_SPEED_MIN and AUDIO_TIMESTRETCH_SPEED_MAX define the min and max time stretch
+// speeds supported by the system. These are enforced by the system and values outside this range
+// will result in a runtime error.
+// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
+// the ones specified here
+// AUDIO_TIMESTRETCH_SPEED_MIN_DELTA is the minimum absolute speed difference that might trigger a
+// parameter update
+#define AUDIO_TIMESTRETCH_SPEED_MIN    0.01f
+#define AUDIO_TIMESTRETCH_SPEED_MAX    20.0f
 #define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f
+#define AUDIO_TIMESTRETCH_SPEED_MIN_DELTA 0.0001f
 
-#define AUDIO_TIMESTRETCH_PITCH_MIN    0.5f
-#define AUDIO_TIMESTRETCH_PITCH_MAX    2.0f
+// AUDIO_TIMESTRETCH_PITCH_MIN and AUDIO_TIMESTRETCH_PITCH_MAX define the min and max time stretch
+// pitch shifting supported by the system. These are not enforced by the system and values
+// outside this range might result in a pitch different than the one requested.
+// Depending on the AudioPlaybackRate::mStretchMode, the effective limits might be narrower than
+// the ones specified here.
+// AUDIO_TIMESTRETCH_PITCH_MIN_DELTA is the minimum absolute pitch difference that might trigger a
+// parameter update
+#define AUDIO_TIMESTRETCH_PITCH_MIN    0.25f
+#define AUDIO_TIMESTRETCH_PITCH_MAX    4.0f
 #define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
+#define AUDIO_TIMESTRETCH_PITCH_MIN_DELTA 0.0001f
+
+
+//Determines the current algorithm used for stretching
+enum AudioTimestretchStretchMode : int32_t {
+    AUDIO_TIMESTRETCH_STRETCH_DEFAULT            = 0,
+    AUDIO_TIMESTRETCH_STRETCH_SPEECH             = 1,
+    //TODO: add more stretch modes/algorithms
+};
+
+//Limits for AUDIO_TIMESTRETCH_STRETCH_SPEECH mode
+#define TIMESTRETCH_SONIC_SPEED_MIN 0.1f
+#define TIMESTRETCH_SONIC_SPEED_MAX 6.0f
+
+//Determines behavior of Timestretch if current algorithm can't perform
+//with current parameters.
+// FALLBACK_CUT_REPEAT: (internal only) for speed <1.0 will truncate frames
+//    for speed > 1.0 will repeat frames
+// FALLBACK_MUTE: will set all processed frames to zero
+// FALLBACK_FAIL:  will stop program execution and log a fatal error
+enum AudioTimestretchFallbackMode : int32_t {
+    AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT     = -1,
+    AUDIO_TIMESTRETCH_FALLBACK_DEFAULT        = 0,
+    AUDIO_TIMESTRETCH_FALLBACK_MUTE           = 1,
+    AUDIO_TIMESTRETCH_FALLBACK_FAIL           = 2,
+};
+
+struct AudioPlaybackRate {
+    float mSpeed;
+    float mPitch;
+    enum AudioTimestretchStretchMode  mStretchMode;
+    enum AudioTimestretchFallbackMode mFallbackMode;
+};
+
+static const AudioPlaybackRate AUDIO_PLAYBACK_RATE_DEFAULT = {
+        AUDIO_TIMESTRETCH_SPEED_NORMAL,
+        AUDIO_TIMESTRETCH_PITCH_NORMAL,
+        AUDIO_TIMESTRETCH_STRETCH_DEFAULT,
+        AUDIO_TIMESTRETCH_FALLBACK_DEFAULT
+};
+
+static inline bool isAudioPlaybackRateEqual(const AudioPlaybackRate &pr1,
+        const AudioPlaybackRate &pr2) {
+    return fabs(pr1.mSpeed - pr2.mSpeed) < AUDIO_TIMESTRETCH_SPEED_MIN_DELTA &&
+           fabs(pr1.mPitch - pr2.mPitch) < AUDIO_TIMESTRETCH_PITCH_MIN_DELTA &&
+           pr2.mStretchMode == pr2.mStretchMode &&
+           pr2.mFallbackMode == pr2.mFallbackMode;
+}
 
 // TODO: Consider putting these inlines into a class scope
 
@@ -77,4 +143,8 @@ static inline size_t sourceFramesNeededWithTimestretch(
     return required * (double)speed + 1 + 1; // accounting for rounding dependencies
 }
 
+} // namespace android
+
+// ---------------------------------------------------------------------------
+
 #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index a06197f..2d34c02 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -21,6 +21,7 @@
 #include <media/AudioSystem.h>
 #include <media/AudioTimestamp.h>
 #include <media/IAudioTrack.h>
+#include <media/AudioResamplerPublic.h>
 #include <utils/threads.h>
 
 namespace android {
@@ -369,10 +370,10 @@ public:
      * 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);
+            status_t    setPlaybackRate(const AudioPlaybackRate &playbackRate);
 
     /* Return current playback rate */
-            void        getPlaybackRate(float *speed, float *pitch) const;
+            const AudioPlaybackRate& getPlaybackRate() const;
 
     /* Enables looping and sets the start and end points of looping.
      * Only supported for static buffer mode.
@@ -748,8 +749,7 @@ 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.
+    AudioPlaybackRate       mPlaybackRate;
     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 6cc2e2b..1e5064f 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -114,13 +114,7 @@ struct AudioTrackSharedStatic {
                     mPosLoopQueue;
 };
 
-
-struct AudioTrackPlaybackRate {
-    float mSpeed;
-    float mPitch;
-};
-
-typedef SingleStateQueue<AudioTrackPlaybackRate> AudioTrackPlaybackRateQueue;
+typedef SingleStateQueue<AudioPlaybackRate> PlaybackRateQueue;
 
 // ----------------------------------------------------------------------------
 
@@ -168,7 +162,7 @@ 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;
+                PlaybackRateQueue::Shared mPlaybackRateQueue;
 
                 // client write-only, server read-only
                 uint16_t    mSendLevel;      // Fixed point U4.12 so 0x1000 means 1.0
@@ -345,10 +339,7 @@ public:
         mCblk->mSampleRate = sampleRate;
     }
 
-    void        setPlaybackRate(float speed, float pitch) {
-        AudioTrackPlaybackRate playbackRate;
-        playbackRate.mSpeed = speed;
-        playbackRate.mPitch = pitch;
+    void        setPlaybackRate(const AudioPlaybackRate& playbackRate) {
         mPlaybackRateMutator.push(playbackRate);
     }
 
@@ -365,7 +356,7 @@ public:
     status_t    waitStreamEndDone(const struct timespec *requested);
 
 private:
-    AudioTrackPlaybackRateQueue::Mutator   mPlaybackRateMutator;
+    PlaybackRateQueue::Mutator   mPlaybackRateMutator;
 };
 
 class StaticAudioTrackClientProxy : public AudioTrackClientProxy {
@@ -483,8 +474,7 @@ public:
         : 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;
+        mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
     }
 protected:
     virtual ~AudioTrackServerProxy() { }
@@ -520,11 +510,11 @@ public:
     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);
+    AudioPlaybackRate getPlaybackRate();
 
 private:
-    AudioTrackPlaybackRate                  mPlaybackRate;  // last observed playback rate
-    AudioTrackPlaybackRateQueue::Observer   mPlaybackRateObserver;
+    AudioPlaybackRate             mPlaybackRate;  // last observed playback rate
+    PlaybackRateQueue::Observer   mPlaybackRateObserver;
 };
 
 class StaticAudioTrackServerProxy : public AudioTrackServerProxy {