1 files changed, 32 insertions, 0 deletions
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h
index 97847a0..0634741 100644
--- a/include/media/AudioResamplerPublic.h
+++ b/include/media/AudioResamplerPublic.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_RESAMPLER_PUBLIC_H
 #define ANDROID_AUDIO_RESAMPLER_PUBLIC_H
 
+#include <stdint.h>
+
 // AUDIO_RESAMPLER_DOWN_RATIO_MAX is the maximum ratio between the original
 // audio sample rate and the target rate when downsampling,
 // as permitted in the audio framework, e.g. AudioTrack and AudioFlinger.
@@ -26,4 +28,34 @@
 // TODO: replace with an API
 #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256
 
+// AUDIO_RESAMPLER_UP_RATIO_MAX is the maximum suggested ratio between the original
+// audio sample rate and the target rate when upsampling.  It is loosely enforced by
+// the system. One issue with large upsampling ratios is the approximation by
+// an int32_t of the phase increments, making the resulting sample rate inexact.
+#define AUDIO_RESAMPLER_UP_RATIO_MAX 65536
+
+// 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.
+// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which
+// may not be true if the resampler is asynchronous.
+static inline size_t sourceFramesNeeded(
+        uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) {
+    // +1 for rounding - always do this even if matched ratio (resampler may use phases not ratio)
+    // +1 for additional sample needed for interpolation
+    return srcSampleRate == dstSampleRate ? dstFramesRequired :
+            size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
+}
+
+// An upper bound for the number of destination frames possible from srcFrames
+// after sample rate conversion.  This may be used for buffer sizing.
+static inline size_t destinationFramesPossible(size_t srcFrames, uint32_t srcSampleRate,
+        uint32_t dstSampleRate) {
+    if (srcSampleRate == dstSampleRate) {
+        return srcFrames;
+    }
+    uint64_t dstFrames = (uint64_t)srcFrames * dstSampleRate / srcSampleRate;
+    return dstFrames > 2 ? dstFrames - 2 : 0;
+}
+
 #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H