Commit a8c07773 authored by Michael Niedermayer's avatar Michael Niedermayer

Revert "Merge commit '0517c9e0'" bring the old...

Revert "Merge commit '0517c9e0'" bring the old audio resampling API back

This reverts commit d3edc65d, reversing
changes made to 150de78d.

Conflicts:

	libavcodec/version.h

It seems there are several applications still using it
Signed-off-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parent bbaa4432
......@@ -27,6 +27,8 @@ OBJS = allcodecs.o \
parser.o \
raw.o \
rawdec.o \
resample.o \
resample2.o \
utils.o \
# parts needed for many different codecs
......
......@@ -4096,6 +4096,103 @@ int avcodec_encode_subtitle(AVCodecContext *avctx, uint8_t *buf, int buf_size,
* @}
*/
#if FF_API_AVCODEC_RESAMPLE
/**
* @defgroup lavc_resample Audio resampling
* @ingroup libavc
* @deprecated use libswresample instead
*
* @{
*/
struct ReSampleContext;
struct AVResampleContext;
typedef struct ReSampleContext ReSampleContext;
/**
* Initialize audio resampling context.
*
* @param output_channels number of output channels
* @param input_channels number of input channels
* @param output_rate output sample rate
* @param input_rate input sample rate
* @param sample_fmt_out requested output sample format
* @param sample_fmt_in input sample format
* @param filter_length length of each FIR filter in the filterbank relative to the cutoff frequency
* @param log2_phase_count log2 of the number of entries in the polyphase filterbank
* @param linear if 1 then the used FIR filter will be linearly interpolated
between the 2 closest, if 0 the closest will be used
* @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
* @return allocated ReSampleContext, NULL if error occurred
*/
attribute_deprecated
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum AVSampleFormat sample_fmt_out,
enum AVSampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff);
attribute_deprecated
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
/**
* Free resample context.
*
* @param s a non-NULL pointer to a resample context previously
* created with av_audio_resample_init()
*/
attribute_deprecated
void audio_resample_close(ReSampleContext *s);
/**
* Initialize an audio resampler.
* Note, if either rate is not an integer then simply scale both rates up so they are.
* @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
* @param log2_phase_count log2 of the number of entries in the polyphase filterbank
* @param linear If 1 then the used FIR filter will be linearly interpolated
between the 2 closest, if 0 the closest will be used
* @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
*/
attribute_deprecated
struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);
/**
* Resample an array of samples using a previously configured context.
* @param src an array of unconsumed samples
* @param consumed the number of samples of src which have been consumed are returned here
* @param src_size the number of unconsumed samples available
* @param dst_size the amount of space in samples available in dst
* @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
* @return the number of samples written in dst or -1 if an error occurred
*/
attribute_deprecated
int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
/**
* Compensate samplerate/timestamp drift. The compensation is done by changing
* the resampler parameters, so no audible clicks or similar distortions occur
* @param compensation_distance distance in output samples over which the compensation should be performed
* @param sample_delta number of output samples which should be output less
*
* example: av_resample_compensate(c, 10, 500)
* here instead of 510 samples only 500 samples would be output
*
* note, due to rounding the actual compensation might be slightly different,
* especially if the compensation_distance is large and the in_rate used during init is small
*/
attribute_deprecated
void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
attribute_deprecated
void av_resample_close(struct AVResampleContext *c);
/**
* @}
*/
#endif
/**
* @addtogroup lavc_picture
* @{
......
/*
* samplerate conversion for both audio and video
* Copyright (c) 2000 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* samplerate conversion for both audio and video
*/
#include <string.h>
#include "avcodec.h"
#include "audioconvert.h"
#include "libavutil/opt.h"
#include "libavutil/mem.h"
#include "libavutil/samplefmt.h"
#if FF_API_AVCODEC_RESAMPLE
#define MAX_CHANNELS 8
struct AVResampleContext;
static const char *context_to_name(void *ptr)
{
return "audioresample";
}
static const AVOption options[] = {{NULL}};
static const AVClass audioresample_context_class = {
"ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
};
struct ReSampleContext {
struct AVResampleContext *resample_context;
short *temp[MAX_CHANNELS];
int temp_len;
float ratio;
/* channel convert */
int input_channels, output_channels, filter_channels;
AVAudioConvert *convert_ctx[2];
enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
unsigned sample_size[2]; ///< size of one sample in sample_fmt
short *buffer[2]; ///< buffers used for conversion to S16
unsigned buffer_size[2]; ///< sizes of allocated buffers
};
/* n1: number of samples */
static void stereo_to_mono(short *output, short *input, int n1)
{
short *p, *q;
int n = n1;
p = input;
q = output;
while (n >= 4) {
q[0] = (p[0] + p[1]) >> 1;
q[1] = (p[2] + p[3]) >> 1;
q[2] = (p[4] + p[5]) >> 1;
q[3] = (p[6] + p[7]) >> 1;
q += 4;
p += 8;
n -= 4;
}
while (n > 0) {
q[0] = (p[0] + p[1]) >> 1;
q++;
p += 2;
n--;
}
}
/* n1: number of samples */
static void mono_to_stereo(short *output, short *input, int n1)
{
short *p, *q;
int n = n1;
int v;
p = input;
q = output;
while (n >= 4) {
v = p[0]; q[0] = v; q[1] = v;
v = p[1]; q[2] = v; q[3] = v;
v = p[2]; q[4] = v; q[5] = v;
v = p[3]; q[6] = v; q[7] = v;
q += 8;
p += 4;
n -= 4;
}
while (n > 0) {
v = p[0]; q[0] = v; q[1] = v;
q += 2;
p += 1;
n--;
}
}
/*
5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
- Left = front_left + rear_gain * rear_left + center_gain * center
- Right = front_right + rear_gain * rear_right + center_gain * center
Where rear_gain is usually around 0.5-1.0 and
center_gain is almost always 0.7 (-3 dB)
*/
static void surround_to_stereo(short **output, short *input, int channels, int samples)
{
int i;
short l, r;
for (i = 0; i < samples; i++) {
int fl,fr,c,rl,rr;
fl = input[0];
fr = input[1];
c = input[2];
// lfe = input[3];
rl = input[4];
rr = input[5];
l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
/* output l & r. */
*output[0]++ = l;
*output[1]++ = r;
/* increment input. */
input += channels;
}
}
static void deinterleave(short **output, short *input, int channels, int samples)
{
int i, j;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*output[j]++ = *input++;
}
}
}
static void interleave(short *output, short **input, int channels, int samples)
{
int i, j;
for (i = 0; i < samples; i++) {
for (j = 0; j < channels; j++) {
*output++ = *input[j]++;
}
}
}
static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
{
int i;
short l, r;
for (i = 0; i < n; i++) {
l = *input1++;
r = *input2++;
*output++ = l; /* left */
*output++ = (l / 2) + (r / 2); /* center */
*output++ = r; /* right */
*output++ = 0; /* left surround */
*output++ = 0; /* right surroud */
*output++ = 0; /* low freq */
}
}
#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
static const uint8_t supported_resampling[MAX_CHANNELS] = {
// output ch: 1 2 3 4 5 6 7 8
SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
};
ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
int output_rate, int input_rate,
enum AVSampleFormat sample_fmt_out,
enum AVSampleFormat sample_fmt_in,
int filter_length, int log2_phase_count,
int linear, double cutoff)
{
ReSampleContext *s;
if (input_channels > MAX_CHANNELS) {
av_log(NULL, AV_LOG_ERROR,
"Resampling with input channels greater than %d is unsupported.\n",
MAX_CHANNELS);
return NULL;
}
if (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
int i;
av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
"output channels for %d input channel%s", input_channels,
input_channels > 1 ? "s:" : ":");
for (i = 0; i < MAX_CHANNELS; i++)
if (supported_resampling[input_channels-1] & (1<<i))
av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
av_log(NULL, AV_LOG_ERROR, "\n");
return NULL;
}
s = av_mallocz(sizeof(ReSampleContext));
if (!s) {
av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
return NULL;
}
s->ratio = (float)output_rate / (float)input_rate;
s->input_channels = input_channels;
s->output_channels = output_channels;
s->filter_channels = s->input_channels;
if (s->output_channels < s->filter_channels)
s->filter_channels = s->output_channels;
s->sample_fmt[0] = sample_fmt_in;
s->sample_fmt[1] = sample_fmt_out;
s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
s->sample_fmt[0], 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert %s sample format to s16 sample format\n",
av_get_sample_fmt_name(s->sample_fmt[0]));
av_free(s);
return NULL;
}
}
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
av_log(s, AV_LOG_ERROR,
"Cannot convert s16 sample format to %s sample format\n",
av_get_sample_fmt_name(s->sample_fmt[1]));
av_audio_convert_free(s->convert_ctx[0]);
av_free(s);
return NULL;
}
}
s->resample_context = av_resample_init(output_rate, input_rate,
filter_length, log2_phase_count,
linear, cutoff);
*(const AVClass**)s->resample_context = &audioresample_context_class;
return s;
}
/* resample audio. 'nb_samples' is the number of input samples */
/* XXX: optimize it ! */
int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
{
int i, nb_samples1;
short *bufin[MAX_CHANNELS];
short *bufout[MAX_CHANNELS];
short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
short *output_bak = NULL;
int lenout;
if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
/* nothing to do */
memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
return nb_samples;
}
if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
int istride[1] = { s->sample_size[0] };
int ostride[1] = { 2 };
const void *ibuf[1] = { input };
void *obuf[1];
unsigned input_size = nb_samples * s->input_channels * 2;
if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
av_free(s->buffer[0]);
s->buffer_size[0] = input_size;
s->buffer[0] = av_malloc(s->buffer_size[0]);
if (!s->buffer[0]) {
av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
return 0;
}
}
obuf[0] = s->buffer[0];
if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
ibuf, istride, nb_samples * s->input_channels) < 0) {
av_log(s->resample_context, AV_LOG_ERROR,
"Audio sample format conversion failed\n");
return 0;
}
input = s->buffer[0];
}
lenout= 2*s->output_channels*nb_samples * s->ratio + 16;
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
s->output_channels;
output_bak = output;
if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
av_free(s->buffer[1]);
s->buffer_size[1] = out_size;
s->buffer[1] = av_malloc(s->buffer_size[1]);
if (!s->buffer[1]) {
av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
return 0;
}
}
output = s->buffer[1];
}
/* XXX: move those malloc to resample init code */
for (i = 0; i < s->filter_channels; i++) {
bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
buftmp2[i] = bufin[i] + s->temp_len;
bufout[i] = av_malloc(lenout * sizeof(short));
}
if (s->input_channels == 2 && s->output_channels == 1) {
buftmp3[0] = output;
stereo_to_mono(buftmp2[0], input, nb_samples);
} else if (s->output_channels >= 2 && s->input_channels == 1) {
buftmp3[0] = bufout[0];
memcpy(buftmp2[0], input, nb_samples * sizeof(short));
} else if (s->input_channels == 6 && s->output_channels ==2) {
buftmp3[0] = bufout[0];
buftmp3[1] = bufout[1];
surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
} else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
for (i = 0; i < s->input_channels; i++) {
buftmp3[i] = bufout[i];
}
deinterleave(buftmp2, input, s->input_channels, nb_samples);
} else {
buftmp3[0] = output;
memcpy(buftmp2[0], input, nb_samples * sizeof(short));
}
nb_samples += s->temp_len;
/* resample each channel */
nb_samples1 = 0; /* avoid warning */
for (i = 0; i < s->filter_channels; i++) {
int consumed;
int is_last = i + 1 == s->filter_channels;
nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
&consumed, nb_samples, lenout, is_last);
s->temp_len = nb_samples - consumed;
s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
}
if (s->output_channels == 2 && s->input_channels == 1) {
mono_to_stereo(output, buftmp3[0], nb_samples1);
} else if (s->output_channels == 6 && s->input_channels == 2) {
ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
} else if ((s->output_channels == s->input_channels && s->input_channels >= 2) ||
(s->output_channels == 2 && s->input_channels == 6)) {
interleave(output, buftmp3, s->output_channels, nb_samples1);
}
if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
int istride[1] = { 2 };
int ostride[1] = { s->sample_size[1] };
const void *ibuf[1] = { output };
void *obuf[1] = { output_bak };
if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
ibuf, istride, nb_samples1 * s->output_channels) < 0) {
av_log(s->resample_context, AV_LOG_ERROR,
"Audio sample format conversion failed\n");
return 0;
}
}
for (i = 0; i < s->filter_channels; i++) {
av_free(bufin[i]);
av_free(bufout[i]);
}
return nb_samples1;
}
void audio_resample_close(ReSampleContext *s)
{
int i;
av_resample_close(s->resample_context);
for (i = 0; i < s->filter_channels; i++)
av_freep(&s->temp[i]);
av_freep(&s->buffer[0]);
av_freep(&s->buffer[1]);
av_audio_convert_free(s->convert_ctx[0]);
av_audio_convert_free(s->convert_ctx[1]);
av_free(s);
}
#endif
/*
* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "libavutil/avassert.h"
#include "avcodec.h"
#include "libavutil/common.h"
#if FF_API_AVCODEC_RESAMPLE
#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
#define FILTER_SHIFT 0
#define FELEM double
#define FELEM2 double
#define FELEML double
#define WINDOW_TYPE 24
#endif
typedef struct AVResampleContext{
const AVClass *av_class;
FELEM *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
static double bessel(double x){
double v=1;
double lastv=0;
double t=1;
int i;
x= x*x/4;
for(i=1; v != lastv; i++){
lastv=v;
t *= x/(i*i);
v += t;
}
return v;
}
/**
* Build a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
* @return 0 on success, negative on error
*/
static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i;
double x, y, w;
double *tab = av_malloc(tap_count * sizeof(*tab));
const int center= (tap_count-1)/2;
if (!tab)
return AVERROR(ENOMEM);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
for(ph=0;ph<phase_count;ph++) {
double norm = 0;
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch(type){
case 0:{
const float d= -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x);
else y= d*(-4 + 8*x - 5*x*x + x*x*x);
break;}
case 1:
w = 2.0*x / (factor*tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
break;
default:
w = 2.0*x / (factor*tap_count*M_PI);
y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
break;
}
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
#endif
}
}
#if 0
{
#define LEN 1024
int j,k;
double sine[LEN + tap_count];
double filtered[LEN];
double maxff=-2, minff=2, maxsf=-2, minsf=2;
for(i=0; i<LEN; i++){
double ss=0, sf=0, ff=0;
for(j=0; j<LEN+tap_count; j++)
sine[j]= cos(i*j*M_PI/LEN);
for(j=0; j<LEN; j++){
double sum=0;
ph=0;
for(k=0; k<tap_count; k++)
sum += filter[ph * tap_count + k] * sine[k+j];
filtered[j]= sum / (1<<FILTER_SHIFT);
ss+= sine[j + center] * sine[j + center];
ff+= filtered[j] * filtered[j];
sf+= sine[j + center] * filtered[j];
}
ss= sqrt(2*ss/LEN);
ff= sqrt(2*ff/LEN);
sf= 2*sf/LEN;
maxff= FFMAX(maxff, ff);
minff= FFMIN(minff, ff);
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
minff=minsf= 2;
maxff=maxsf= -2;
}
}
}
#endif
av_free(tab);
return 0;
}
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
if (!c)
return NULL;
c->phase_shift= phase_shift;
c->phase_mask= phase_count-1;
c->linear= linear;
c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
if (!c->filter_bank)
goto error;
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
goto error;
memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
goto error;
c->ideal_dst_incr= c->dst_incr;
c->index= -phase_count*((c->filter_length-1)/2);
return c;
error:
av_free(c->filter_bank);
av_free(c);
return NULL;
}
void av_resample_close(AVResampleContext *c){
av_freep(&c->filter_bank);
av_freep(&c);
}
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
c->compensation_distance= compensation_distance;
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
}
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
int frac= c->frac;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int compensation_distance= c->compensation_distance;
if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
int64_t index2= ((int64_t)index)<<32;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
for(dst_index=0; dst_index < dst_size; dst_index++){
dst[dst_index] = src[index2>>32];
index2 += incr;
}
index += dst_index * dst_incr;
index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
}else{
for(dst_index=0; dst_index < dst_size; dst_index++){
FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
int sample_index= index >> c->phase_shift;
FELEM2 val=0;
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
}else if(c->linear){
FELEM2 v2=0;
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
}
val+=(v2-val)*(FELEML)frac / c->src_incr;
}else{
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
}
}
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
#endif
frac += dst_incr_frac;
index += dst_incr;
if(frac >= c->src_incr){
frac -= c->src_incr;
index++;
}
if(dst_index + 1 == compensation_distance){
compensation_distance= 0;
dst_incr_frac= c->ideal_dst_incr % c->src_incr;
dst_incr= c->ideal_dst_incr / c->src_incr;
}
}
}
*consumed= FFMAX(index, 0) >> c->phase_shift;
if(index>=0) index &= c->phase_mask;
if(compensation_distance){
compensation_distance -= dst_index;
av_assert2(compensation_distance > 0);
}
if(update_ctx){
c->frac= frac;
c->index= index;
c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
c->compensation_distance= compensation_distance;
}
return dst_index;
}
#endif
......@@ -73,6 +73,9 @@
#ifndef FF_API_CODEC_ID
#define FF_API_CODEC_ID (LIBAVCODEC_VERSION_MAJOR < 56)
#endif
#ifndef FF_API_AVCODEC_RESAMPLE
#define FF_API_AVCODEC_RESAMPLE (LIBAVCODEC_VERSION_MAJOR < 56)
#endif
#ifndef FF_API_MMI
#define FF_API_MMI (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment