前些日子无聊实现的一个Audio PCM Resampler的代码，仅仅支持采样率为44.1khz的源数据的向下转换，可转换成8khz/11.025khz/16khz/22.050khz。
接口设计参考android-4.0.3_r1systemmediaaudio_utilsincludeaudio_utils
esampler.h，因此使用方法也和Android的那套接口保持一致。
// File: resampler.h
// Author: Loon
#ifndef __AUDIO_RESAMPLER_H
#define __AUDIO_RESAMPLER_H
#include
struct resampler_buffer {
union {
void* raw;
short* i16;
int8_t* i8;
};
size_t frame_count;
};
/* call back interface used by the resampler to get new data */
struct resampler_buffer_provider {
/**
* get a new buffer of data:
* as input: buffer->frame_count is the number of frames requested
* as output: buffer->frame_count is the number of frames returned
* buffer->raw points to data returned
*/
int (*get_next_buffer)(struct resampler_buffer_provider *provider,
struct resampler_buffer *buffer);
/**
* release a consumed buffer of data:
* as input: buffer->frame_count is the number of frames released
* buffer->raw points to data released
*/
void (*release_buffer)(struct resampler_buffer_provider *provider,
struct resampler_buffer *buffer);
};
struct resampler_itfe {
/**
* reset resampler state
*/
void (*reset)(struct resampler_itfe *resampler);
/**
* resample input from buffer provider and output at most *outFrameCount to out buffer.
* *outFrameCount is updated with the actual number of frames produced.
*/
int (*resample_from_provider)(struct resampler_itfe *resampler,
int16_t *out,
size_t *outFrameCount);
/**
* resample at most *inFrameCount frames from in buffer and output at most
* *outFrameCount to out buffer. *inFrameCount and *outFrameCount are updated respectively
* with the number of frames remaining in input and written to output.
*/
int (*resample_from_input)(struct resampler_itfe *resampler,
int16_t *in,
size_t *inFrameCount,
int16_t *out,
size_t *outFrameCount);
};
/**
* create a resampler according to input parameters passed.
* If resampler_buffer_provider is not NULL only resample_from_provider() can be called.
* If resampler_buffer_provider is NULL only resample_from_input() can be called.
*/
int create_down_resampler(uint32_t outSampleRate,
uint32_t channelCount,
uint32_t frame_count,
struct resampler_buffer_provider *provider,
struct resampler_itfe **);
/**
* release resampler resources.
*/
void release_down_resampler(struct resampler_itfe *);
#endif /** __AUDIO_RESAMPLER_H */
// File: resampler.c
// Author: Loon
#include
#include
#include
#include
#include
#include "resampler.h"
struct down_resampler {
struct resampler_itfe itfe;
struct resampler_buffer_provider *provider; // buffer provider installed by client
uint32_t out_sample_rate; // output sampling rate in Hz
uint32_t channel_count; // number of channels (interleaved)
uint32_t frame_count; // number of frames
int16_t *mInLeft;
int16_t *mInRight;
int16_t *mTmpLeft;
int16_t *mTmpRight;
int16_t *mTmp2Left;
int16_t *mTmp2Right;
int16_t *mOutLeft;
int16_t *mOutRight;
int mInInBuf;
int mInTmpBuf;
int mInTmp2Buf;
int mOutBufPos;
int mInOutBuf;
};
/*------------------------------------------------------------------------------*/
/*------------------------------------------------------------------------------*/
/* DownSampler Algorithm Implement */
/*
* 2.30 fixed point FIR filter coefficients for conversion 44100 -> 22050.
* (Works equivalently for 22010 -> 11025 or any other halving, of course.)
*
* Transition band from about 18 kHz, passband ripple < 0.1 dB,
* stopband ripple at about -55 dB, linear phase.
*
* Design and display in MATLAB or Octave using:
*
* filter = fir1(19, 0.5); filter = round(filter * 2**30); freqz(filter * 2**-30);
*/
static const int32_t filter_22khz_coeff[] = {
2089257, 2898328, -5820678, -10484531,
19038724, 30542725, -50469415, -81505260,
152544464, 478517512, 478517512, 152544464,
-81505260, -50469415, 30542725, 19038724,
-10484531, -5820678, 2898328, 2089257,
};
#define NUM_COEFF_22KHZ (sizeof(filter_22khz_coeff) / sizeof(filter_22khz_coeff[0]))
#define OVERLAP_22KHZ (NUM_COEFF_22KHZ - 2)
/*
* Convolution of signals A and reverse(B). (In our case, the filter response
* is symmetric, so the reversing doesn't matter.)
* A is taken to be in 0.16 fixed-point, and B is taken to be in 2.30 fixed-point.
* The answer will be in 16.16 fixed-point, unclipped.
*
* This function would probably be the prime candidate for SIMD conversion if
* you want more speed.
*/
int32_t fir_convolve(const int16_t* a, const int32_t* b, int num_samples)
{
int32_t sum = 1 << 13;
for (int i = 0; i < num_samples; ++i) {
sum += a[i] * (b[i] >> 16);
}
return sum >> 14;
}
/* Clip from 16.16 fixed-point to 0.16 fixed-point. */
int16_t clip(int32_t x)
{
if (x < -32768) {
return -32768;
} else if (x > 32767) {
return 32767;
} else {
return x;
}
}
/*
* Convert a chunk from 44 kHz to 22 kHz. Will update num_samples_in and num_samples_out
* accordingly, since it may leave input samples in the buffer due to overlap.
*
* Input and output are taken to be in 0.16 fixed-point.
*/
void resample_2_1(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
{
if (*num_samples_in < (int)NUM_COEFF_22KHZ) {
*num_samples_out = 0;
return;
}
int odd_smp = *num_samples_in & 0x1;
int num_samples = *num_samples_in - odd_smp - OVERLAP_22KHZ;
for (int i = 0; i < num_samples; i += 2) {
output[i / 2] = clip(fir_convolve(input + i, filter_22khz_coeff, NUM_COEFF_22KHZ));
}
memmove(input, input + num_samples, (OVERLAP_22KHZ + odd_smp) * sizeof(*input));
*num_samples_out = num_samples / 2;
*num_samples_in = OVERLAP_22KHZ + odd_smp;
}
/*
* 2.30 fixed point FIR filter coefficients for conversion 22050 -> 16000,
* or 11025 -> 8000.
*
* Transition band from about 14 kHz, passband ripple < 0.1 dB,
* stopband ripple at about -50 dB, linear phase.
*
* Design and display in MATLAB or Octave using:
*
* filter = fir1(23, 16000 / 22050); filter = round(filter * 2**30); freqz(filter * 2**-30);
*/
static const int32_t filter_16khz_coeff[] = {
2057290, -2973608, 1880478, 4362037,
-14639744, 18523609, -1609189, -38502470,
78073125, -68353935, -59103896, 617555440,
617555440, -59103896, -68353935, 78073125,
-38502470, -1609189, 18523609, -14639744,
4362037, 1880478, -2973608, 2057290,
};
#define NUM_COEFF_16KHZ (sizeof(filter_16khz_coeff) / sizeof(filter_16khz_coeff[0]))
#define OVERLAP_16KHZ (NUM_COEFF_16KHZ - 1)
/*
* Convert a chunk from 22 kHz to 16 kHz. Will update num_samples_in and
* num_samples_out accordingly, since it may leave input samples in the buffer
* due to overlap.
*
* This implementation is rather ad-hoc; it first low-pass filters the data
* into a temporary buffer, and then converts chunks of 441 input samples at a
* time into 320 output samples by simple linear interpolation. A better
* implementation would use a polyphase filter bank to do these two operations
* in one step.
*
* Input and output are taken to be in 0.16 fixed-point.
*/
#define RESAMPLE_16KHZ_SAMPLES_IN 441
#define RESAMPLE_16KHZ_SAMPLES_OUT 320
void resample_441_320(int16_t* input, int16_t* output, int* num_samples_in, int* num_samples_out)
{
const int num_blocks = (*num_samples_in - OVERLAP_16KHZ) / RESAMPLE_16KHZ_SAMPLES_IN;
if (num_blocks < 1) {
*num_samples_out = 0;
return;
}
for (int i = 0; i < num_blocks; ++i) {
uint32_t tmp[RESAMPLE_16KHZ_SAMPLES_IN];
for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_IN; ++j) {
tmp[j] = fir_convolve(input + i * RESAMPLE_16KHZ_SAMPLES_IN + j,
filter_16khz_coeff,
NUM_COEFF_16KHZ);
}
const float step_float = (float)RESAMPLE_16KHZ_SAMPLES_IN / (float)RESAMPLE_16KHZ_SAMPLES_OUT;
uint32_t in_sample_num = 0; // 16.16 fixed point
const uint32_t step = (uint32_t)(step_float * 65536.0f + 0.5f); // 16.16 fixed point
for (int j = 0; j < RESAMPLE_16KHZ_SAMPLES_OUT; ++j, in_sample_num += step) {
const uint32_t whole = in_sample_num >> 16;
const uint32_t frac = (in_sample_num & 0xffff); // 0.16 fixed point
const int32_t s1 = tmp[whole];
const int32_t s2 = tmp[whole + 1];
*output++ = clip(s1 + (((s2 - s1) * (int32_t)frac) >> 16));
}
}
const int samples_consumed = num_blocks * RESAMPLE_16KHZ_SAMPLES_IN;
memmove(input, input + samples_consumed, (*num_samples_in - samples_consumed) * sizeof(*input));
*num_samples_in -= samples_consumed;
*num_samples_out = RESAMPLE_16KHZ_SAMPLES_OUT * num_blocks;
}
/*------------------------------------------------------------------------------*/
/*------------------------------------------------------------------------------*/
/* DownSampler Callback Functions */
static void down_resampler_reset(struct resampler_itfe *resampler)
{
struct down_resampler *rsmp =
(struct down_resampler *)((char *)resampler - offsetof(struct down_resampler, itfe));
if (rsmp == NULL) {
return;
}
rsmp->mInInBuf = 0;
rsmp->mInTmpBuf = 0;
rsmp->mInTmp2Buf = 0;
rsmp->mOutBufPos = 0;
rsmp->mInOutBuf = 0;
}
// outputs a number of frames less or equal to *outFrameCount and updates *outFrameCount
// with the actual number of frames produced.
static int down_resampler_resample_from_provider(struct resampler_itfe *resampler,
int16_t *out,
size_t *outFrameCount)
{
struct down_resampler *rsmp =
(struct down_resampler *)((char *)resampler - offsetof(struct down_resampler, itfe));
if (rsmp == NULL || out == NULL || outFrameCount == NULL) {
return -EINVAL;
}
if (rsmp->provider == NULL) {
*outFrameCount = 0;
return -ENOSYS;
}
int16_t *outLeft = rsmp->mTmp2Left;
int16_t *outRight = rsmp->mTmp2Left;
if (rsmp->out_sample_rate == 22050) {
outLeft = rsmp->mTmpLeft;
outRight = rsmp->mTmpRight;
} else if (rsmp->out_sample_rate == 8000){
outLeft = rsmp->mOutLeft;
outRight = rsmp->mOutRight;
}
int outFrames = 0;
int remaingFrames = *outFrameCount;
if (rsmp->mInOutBuf) {
int frames = (remaingFrames > rsmp->mInOutBuf) ? rsmp->mInOutBuf : remaingFrames;
for (int i = 0; i < frames; ++i) {
out[i] = outLeft[rsmp->mOutBufPos + i];
}
if (rsmp->channel_count == 2) {
for (int i = 0; i < frames; ++i) {
out[i * 2] = outLeft[rsmp->mOutBufPos + i];
out[i * 2 + 1] = outRight[rsmp->mOutBufPos + i];
}
}
remaingFrames -= frames;
rsmp->mInOutBuf -= frames;
rsmp->mOutBufPos += frames;
outFrames += frames;
}
while (remaingFrames) {
struct resampler_buffer buf;
buf.frame_count = rsmp->frame_count - rsmp->mInInBuf;
int ret = rsmp->provider->get_next_buffer(rsmp->provider, &buf);
if (buf.raw == NULL) {
*outFrameCount = outFrames;
return ret;
}
for (size_t i = 0; i < buf.frame_count; ++i) {
rsmp->mInLeft[i + rsmp->mInInBuf] = buf.i16[i];
}
if (rsmp->channel_count == 2) {
for (size_t i = 0; i < buf.frame_count; ++i) {
rsmp->mInLeft[i + rsmp->mInInBuf] = buf.i16[i * 2];
rsmp->mInRight[i + rsmp->mInInBuf] = buf.i16[i * 2 + 1];
}
}
rsmp->mInInBuf += buf.frame_count;
rsmp->provider->release_buffer(rsmp->provider, &buf);
/* 44010 -> 22050 */
{
int samples_in_left = rsmp->mInInBuf;
int samples_out_left;
resample_2_1(rsmp->mInLeft, rsmp->mTmpLeft + rsmp->mInTmpBuf, &samples_in_left, &samples_out_left);
if (rsmp->channel_count == 2) {
int samples_in_right = rsmp->mInInBuf;
int samples_out_right;
resample_2_1(rsmp->mInRight, rsmp->mTmpRight + rsmp->mInTmpBuf, &samples_in_right, &samples_out_right);
}
rsmp->mInInBuf = samples_in_left;
rsmp->mInTmpBuf += samples_out_left;
rsmp->mInOutBuf = samples_out_left;
}
if (rsmp->out_sample_rate == 11025 || rsmp->out_sample_rate == 8000) {
/* 22050 - > 11025 */
int samples_in_left = rsmp->mInTmpBuf;
int samples_out_left;
resample_2_1(rsmp->mTmpLeft, rsmp->mTmp2Left + rsmp->mInTmp2Buf, &samples_in_left, &samples_out_left);
if (rsmp->channel_count == 2) {
int samples_in_right = rsmp->mInTmpBuf;
int samples_out_right;
resample_2_1(rsmp->mTmpRight, rsmp->mTmp2Right + rsmp->mInTmp2Buf, &samples_in_right, &samples_out_right);
}
rsmp->mInTmpBuf = samples_in_left;
rsmp->mInTmp2Buf += samples_out_left;
rsmp->mInOutBuf = samples_out_left;
if (rsmp->out_sample_rate == 8000) {
/* 11025 -> 8000*/
int samples_in_left = rsmp->mInTmp2Buf;
int samples_out_left;
resample_441_320(rsmp->mTmp2Left, rsmp->mOutLeft, &samples_in_left, &samples_out_left);
if (rsmp->channel_count == 2) {
int samples_in_right = rsmp->mInTmp2Buf;
int samples_out_right;
resample_441_320(rsmp->mTmp2Right, rsmp->mOutRight, &samples_in_right, &samples_out_right);
}
rsmp->mInTmp2Buf = samples_in_left;
rsmp->mInOutBuf = samples_out_left;
} else {
rsmp->mInTmp2Buf = 0;
}
} else if (rsmp->out_sample_rate == 16000) {
/* 22050 -> 16000*/
int samples_in_left = rsmp->mInTmpBuf;
int samples_out_left;
resample_441_320(rsmp->mTmpLeft, rsmp->mTmp2Left, &samples_in_left, &samples_out_left);
if (rsmp->channel_count == 2) {
int samples_in_right = rsmp->mInTmpBuf;
int samples_out_right;
resample_441_320(rsmp->mTmpRight, rsmp->mTmp2Right, &samples_in_right, &samples_out_right);
}
rsmp->mInTmpBuf = samples_in_left;
rsmp->mInOutBuf = samples_out_left;
} else {
rsmp->mInTmpBuf = 0;
}
int frames = (remaingFrames > rsmp->mInOutBuf) ? rsmp->mInOutBuf : remaingFrames;
for (int i = 0; i < frames; ++i) {
out[outFrames + i] = outLeft[i];
}
if (rsmp->channel_count == 2) {
for (int i = 0; i < frames; ++i) {
out[(outFrames + i) * 2] = outLeft[i];
out[(outFrames + i) * 2 + 1] = outRight[i];
}
}
remaingFrames -= frames;
outFrames += frames;
rsmp->mOutBufPos = frames;
rsmp->mInOutBuf -= frames;
}
return 0;
}
static int down_resampler_resample_from_input(struct resampler_itfe *resampler,
int16_t *in,
size_t *inFrameCount,
int16_t *out,
size_t *outFrameCount)
{
struct down_resampler *rsmp =
(struct down_resampler *)((char *)resampler - offsetof(struct down_resampler, itfe));
if (rsmp == NULL || in == NULL || inFrameCount == NULL ||
out == NULL || outFrameCount == NULL) {
return -EINVAL;
}
if (rsmp->provider != NULL) {
*outFrameCount = 0;
return -ENOSYS;
}
// Not implement yet!!!
return 0;
}
/*------------------------------------------------------------------------------*/
/*------------------------------------------------------------------------------*/
/* DownSampler Export Interfaces */
int create_down_resampler(uint32_t outSampleRate,
uint32_t channelCount,
uint32_t frame_count,
struct resampler_buffer_provider* provider,
struct resampler_itfe **resampler)
{
int error;
struct down_resampler *rsmp;
if (resampler == NULL) {
return -EINVAL;
}
if (outSampleRate != 8000 && outSampleRate != 11025 && outSampleRate != 16000 &&
outSampleRate != 22050) {
return -EINVAL;
}
*resampler = NULL;
rsmp = (struct down_resampler *)malloc(sizeof(struct down_resampler));
if (rsmp == NULL) return -ENOMEM;
rsmp->itfe.reset = down_resampler_reset;
rsmp->itfe.resample_from_provider = down_resampler_resample_from_provider;
rsmp->itfe.resample_from_input = down_resampler_resample_from_input;
rsmp->provider = provider;
rsmp->out_sample_rate = outSampleRate;
rsmp->channel_count = channelCount;
rsmp->frame_count = frame_count;
rsmp->mInLeft = malloc(sizeof(int16_t) * frame_count);
rsmp->mInRight = malloc(sizeof(int16_t) * frame_count);
rsmp->mTmpLeft = malloc(sizeof(int16_t) * frame_count);
rsmp->mTmpRight = malloc(sizeof(int16_t) * frame_count);
rsmp->mTmp2Left = malloc(sizeof(int16_t) * frame_count);
rsmp->mTmp2Right = malloc(sizeof(int16_t) * frame_count);
rsmp->mOutLeft = malloc(sizeof(int16_t) * frame_count);
rsmp->mOutRight = malloc(sizeof(int16_t) * frame_count);
if (rsmp->mInLeft == NULL || rsmp->mInRight == NULL ||
rsmp->mTmpLeft == NULL || rsmp->mTmpRight == NULL ||
rsmp->mTmp2Left == NULL || rsmp->mTmp2Right == NULL ||
rsmp->mOutLeft == NULL || rsmp->mOutRight == NULL)
return -ENOMEM;
down_resampler_reset(&rsmp->itfe);
*resampler = &rsmp->itfe;
return 0;
}
void release_down_resampler(struct resampler_itfe *resampler)
{
struct down_resampler *rsmp =
(struct down_resampler *)((char *)resampler - offsetof(struct down_resampler, itfe));
if (rsmp == NULL) {
return;
}
if (rsmp->mInLeft) free(rsmp->mInLeft);
if (rsmp->mInRight) free(rsmp->mInRight);
if (rsmp->mTmpLeft) free(rsmp->mTmpLeft);
if (rsmp->mTmpRight) free(rsmp->mTmpRight);
if (rsmp->mTmp2Left) free(rsmp->mTmp2Left);
if (rsmp->mTmp2Right) free(rsmp->mTmp2Right);
if (rsmp->mOutLeft) free(rsmp->mOutLeft);
if (rsmp->mOutRight) free(rsmp->mOutRight);
free(rsmp);
}
{var%20f='http://v.t.sina.com.cn/share/share.php?appkey=1515056452',u=z||d.location,p=['&url=',e(u),'&title=',e(t||d.title),'&source=',e(r),'&sourceUrl=',e(l),'&content=',c||'gb2312','&pic=',e(p||'')].join('');function%20a(){if(!window.open([f,p].join(''),'mb',['toolbar=0,status=0,resizable=1,width=440,height=430,left=',(s.width-440)/2,',top=',(s.height-430)/2].join('')))u.href=[f,p].join('');};if(/Firefox/.test(navigator.userAgent))setTimeout(a,0);else%20a();})(screen,document,encodeURIComponent,'','','https://www.xiaopingtou.cn//data/attach/topic/topicKPo7gB.jpg', '推荐 tao122188 的文章《Audio Resampler Implement》','https://www.xiaopingtou.net/article-81641.html','页面编码gb2312|utf-8默认gb2312'));){kind=link}