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/*
* Copyright (C) 2014 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#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.
// In practice, it is not recommended to downsample more than 6:1
// for best audio quality, even though the audio framework permits a larger
// downsampling ratio.
// 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
#define AUDIO_TIMESTRETCH_SPEED_MIN 0.5f
#define AUDIO_TIMESTRETCH_SPEED_MAX 2.0f
#define AUDIO_TIMESTRETCH_SPEED_NORMAL 1.0f
#define AUDIO_TIMESTRETCH_PITCH_MIN 0.5f
#define AUDIO_TIMESTRETCH_PITCH_MAX 2.0f
#define AUDIO_TIMESTRETCH_PITCH_NORMAL 1.0f
// 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
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