From 8f483108b503fa03ed5e956e25df4cb899171df5 Mon Sep 17 00:00:00 2001 From: James Almer Date: Sat, 21 Oct 2017 23:13:44 -0300 Subject: [PATCH] avcodec: Drop deprecated audio resample API Deprecated in 03/2013. --- libavcodec/Makefile | 2 - libavcodec/avcodec.h | 97 --------- libavcodec/resample.c | 439 ----------------------------------------- libavcodec/resample2.c | 319 ------------------------------ libavcodec/version.h | 3 - 5 files changed, 860 deletions(-) delete mode 100644 libavcodec/resample.c delete mode 100644 libavcodec/resample2.c 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 - -#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<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 - * - * 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 - */ - -#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;phphase_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_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; ifilter_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; ifilter_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; ifilter_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