1 files changed, 37 insertions, 2 deletions
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 33e64ce..b84567e 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -41,6 +41,9 @@ public:
         MED_QUALITY=2,
         HIGH_QUALITY=3,
         VERY_HIGH_QUALITY=4,
+        DYN_LOW_QUALITY=5,
+        DYN_MED_QUALITY=6,
+        DYN_HIGH_QUALITY=7,
     };
 
     static AudioResampler* create(int bitDepth, int inChannelCount,
@@ -60,7 +63,7 @@ public:
     // A mono provider delivers a sequence of samples.
     // A stereo provider delivers a sequence of interleaved pairs of samples.
     // Multi-channel providers are not supported.
-    // In either case, 'out' holds interleaved pairs of fixed-point signed Q19.12.
+    // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
     // That is, for a mono provider, there is an implicit up-channeling.
     // Since this method accumulates, the caller is responsible for clearing 'out' initially.
     // FIXME assumes provider is always successful; it should return the actual frame count.
@@ -81,7 +84,7 @@ protected:
     static const uint32_t kPhaseMask = (1LU<<kNumPhaseBits)-1;
 
     // multiplier to calculate fixed point phase increment
-    static const double kPhaseMultiplier = 1L << kNumPhaseBits;
+    static const double kPhaseMultiplier;
 
     AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);
 
@@ -107,6 +110,38 @@ protected:
     uint64_t mLocalTimeFreq;
     int64_t mPTS;
 
+    // returns the inFrameCount required to generate outFrameCount frames.
+    //
+    // Placed here to be a consistent for all resamplers.
+    //
+    // Right now, we use the upper bound without regards to the current state of the
+    // input buffer using integer arithmetic, as follows:
+    //
+    // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
+    //
+    // The double precision equivalent (float may not be precise enough):
+    // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
+    //
+    // this relies on the fact that the mPhaseIncrement is rounded down from
+    // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
+    // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
+    //
+    // (so long as double precision is computed accurately enough to be considered
+    // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
+    // will not necessarily hold for floats).
+    //
+    // TODO:
+    // Greater accuracy and a tight bound is obtained by:
+    // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
+    // 2) using the exact integer formula where (ignoring 64b casting)
+    //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
+    //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
+    //
+    inline size_t getInFrameCountRequired(size_t outFrameCount) {
+        return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
+                + (mSampleRate - 1))/mSampleRate;
+    }
+
 private:
     const src_quality mQuality;