Merge "aaudio: optimize float to int conversion" (ae11d041) · Commits · e / os / android_frameworks_av

media/libaaudio/src/utility/AAudioUtilities.cpp

+7 −39

Original line number	Diff line number	Diff line
		@@ -25,6 +25,7 @@

		#include "aaudio/AAudio.h"
		#include <aaudio/AAudioTesting.h>
		#include <math.h>

		#include "utility/AAudioUtilities.h"

		@@ -50,44 +51,10 @@ int32_t AAudioConvert_formatToSizeInBytes(aaudio_format_t format) {
		return size;
		}


		// TODO expose and call clamp16_from_float function in primitives.h
		static inline int16_t clamp16_from_float(float f) {
		/* Offset is used to expand the valid range of [-1.0, 1.0) into the 16 lsbs of the
		* floating point significand. The normal shift is 3<<22, but the -15 offset
		* is used to multiply by 32768.
		*/
		static const float offset = (float)(3 << (22 - 15));
		/* zero = (0x10f << 22) = 0x43c00000 (not directly used) */
		static const int32_t limneg = (0x10f << 22) /zero/ - 32768; /* 0x43bf8000 */
		static const int32_t limpos = (0x10f << 22) /zero/ + 32767; /* 0x43c07fff */

		union {
		float f;
		int32_t i;
		} u;

		u.f = f + offset; /* recenter valid range */
		/* Now the valid range is represented as integers between [limneg, limpos].
		* Clamp using the fact that float representation (as an integer) is an ordered set.
		*/
		if (u.i < limneg)
		u.i = -32768;
		else if (u.i > limpos)
		u.i = 32767;
		return u.i; /* Return lower 16 bits, the part of interest in the significand. */
		}

		// Same but without clipping.
		// Convert -1.0f to +1.0f to -32768 to +32767
		static inline int16_t floatToInt16(float f) {
		static const float offset = (float)(3 << (22 - 15));
		union {
		float f;
		int32_t i;
		} u;
		u.f = f + offset; /* recenter valid range */
		return u.i; /* Return lower 16 bits, the part of interest in the significand. */
		static const float scale = 1 << 15;
		return (int16_t) roundf(fmaxf(fminf(f * scale, scale - 1.f), -scale));
		}

		static float clipAndClampFloatToPcm16(float sample, float scaler) {
		@@ -188,13 +155,14 @@ void AAudio_linearRamp(const int16_t *source,
		int32_t samplesPerFrame,
		float amplitude1,
		float amplitude2) {
		float scaler = amplitude1 / SHORT_SCALE;
		float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
		// Because we are converting from int16 to 1nt16, we do not have to scale by 1/32768.
		float scaler = amplitude1;
		float delta = (amplitude2 - amplitude1) / numFrames;
		for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
		for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
		// No need to clip because int16_t range is inherently limited.
		float sample = source++ scaler;
		*destination++ = floatToInt16(sample);
		*destination++ = (int16_t) roundf(sample);
		}
		scaler += delta;
		}