[FFmpeg-cvslog] avcodec: Drop deprecated audio resample API
James Almer
git at videolan.org
Sun Oct 22 05:25:09 EEST 2017
ffmpeg | branch: master | James Almer <jamrial at gmail.com> | Sat Oct 21 23:13:44 2017 -0300| [8f483108b503fa03ed5e956e25df4cb899171df5] | committer: James Almer
avcodec: Drop deprecated audio resample API
Deprecated in 03/2013.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8f483108b503fa03ed5e956e25df4cb899171df5
---
libavcodec/Makefile | 2 -
libavcodec/avcodec.h | 97 -----------
libavcodec/resample.c | 439 -------------------------------------------------
libavcodec/resample2.c | 319 -----------------------------------
libavcodec/version.h | 3 -
5 files changed, 860 deletions(-)
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index b52e5ada7a..651348972e 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -45,8 +45,6 @@ OBJS = allcodecs.o \
profiles.o \
qsv_api.o \
raw.o \
- resample.o \
- resample2.o \
utils.o \
vorbis_parser.o \
xiph.o \
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 42d230ea96..6922b5b6fc 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -5516,103 +5516,6 @@ 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
-
#if FF_API_AVPICTURE
/**
* @addtogroup lavc_picture
diff --git a/libavcodec/resample.c b/libavcodec/resample.c
deleted file mode 100644
index 4c5eb9f10e..0000000000
--- a/libavcodec/resample.c
+++ /dev/null
@@ -1,439 +0,0 @@
-/*
- * 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
-FF_DISABLE_DEPRECATION_WARNINGS
-
-#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->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_array((nb_samples + s->temp_len), sizeof(short));
- bufout[i] = av_malloc_array(lenout, sizeof(short));
-
- if (!bufin[i] || !bufout[i]) {
- av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
- nb_samples1 = 0;
- goto fail;
- }
-
- memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
- buftmp2[i] = bufin[i] + s->temp_len;
- }
-
- 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_array(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;
- }
- }
-
-fail:
- 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);
-}
-
-FF_ENABLE_DEPRECATION_WARNINGS
-#endif
diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
deleted file mode 100644
index 56ae9f7229..0000000000
--- a/libavcodec/resample2.c
+++ /dev/null
@@ -1,319 +0,0 @@
-/*
- * audio resampling
- * Copyright (c) 2004 Michael Niedermayer <michaelni at 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 at 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
-
-typedef int16_t FELEM;
-typedef int32_t FELEM2;
-typedef int64_t FELEML;
-#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
-
-typedef double FELEM;
-typedef double FELEM2;
-typedef double FELEML;
-#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_array(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_array(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
diff --git a/libavcodec/version.h b/libavcodec/version.h
index cd2ca5f1a2..acbc61d757 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -57,9 +57,6 @@
#ifndef FF_API_AUDIO_CONVERT
#define FF_API_AUDIO_CONVERT (LIBAVCODEC_VERSION_MAJOR < 58)
#endif
-#ifndef FF_API_AVCODEC_RESAMPLE
-#define FF_API_AVCODEC_RESAMPLE FF_API_AUDIO_CONVERT
-#endif
#ifndef FF_API_LOWRES
#define FF_API_LOWRES (LIBAVCODEC_VERSION_MAJOR < 59)
#endif
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