libavcodec/resample.c

   1 /*
   2  * samplerate conversion for both audio and video
   3  * Copyright (c) 2000 Fabrice Bellard.
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 /**
  23  * @file resample.c
  24  * samplerate conversion for both audio and video
  25  */
  26
  27 #include "avcodec.h"
  28
  29 struct AVResampleContext;
  30
  31 struct ReSampleContext {
  32     struct AVResampleContext *resample_context;
  33     short *temp[2];
  34     int temp_len;
  35     float ratio;
  36     /* channel convert */
  37     int input_channels, output_channels, filter_channels;
  38 };
  39
  40 /* n1: number of samples */
  41 static void stereo_to_mono(short *output, short *input, int n1)
  42 {
  43     short *p, *q;
  44     int n = n1;
  45
  46     p = input;
  47     q = output;
  48     while (n >= 4) {
  49         q[0] = (p[0] + p[1]) >> 1;
  50         q[1] = (p[2] + p[3]) >> 1;
  51         q[2] = (p[4] + p[5]) >> 1;
  52         q[3] = (p[6] + p[7]) >> 1;
  53         q += 4;
  54         p += 8;
  55         n -= 4;
  56     }
  57     while (n > 0) {
  58         q[0] = (p[0] + p[1]) >> 1;
  59         q++;
  60         p += 2;
  61         n--;
  62     }
  63 }
  64
  65 /* n1: number of samples */
  66 static void mono_to_stereo(short *output, short *input, int n1)
  67 {
  68     short *p, *q;
  69     int n = n1;
  70     int v;
  71
  72     p = input;
  73     q = output;
  74     while (n >= 4) {
  75         v = p[0]; q[0] = v; q[1] = v;
  76         v = p[1]; q[2] = v; q[3] = v;
  77         v = p[2]; q[4] = v; q[5] = v;
  78         v = p[3]; q[6] = v; q[7] = v;
  79         q += 8;
  80         p += 4;
  81         n -= 4;
  82     }
  83     while (n > 0) {
  84         v = p[0]; q[0] = v; q[1] = v;
  85         q += 2;
  86         p += 1;
  87         n--;
  88     }
  89 }
  90
  91 /* XXX: should use more abstract 'N' channels system */
  92 static void stereo_split(short *output1, short *output2, short *input, int n)
  93 {
  94     int i;
  95
  96     for(i=0;i<n;i++) {
  97         *output1++ = *input++;
  98         *output2++ = *input++;
  99     }
 100 }
 101
 102 static void stereo_mux(short *output, short *input1, short *input2, int n)
 103 {
 104     int i;
 105
 106     for(i=0;i<n;i++) {
 107         *output++ = *input1++;
 108         *output++ = *input2++;
 109     }
 110 }
 111
 112 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 113 {
 114     int i;
 115     short l,r;
 116
 117     for(i=0;i<n;i++) {
 118       l=*input1++;
 119       r=*input2++;
 120       *output++ = l;           /* left */
 121       *output++ = (l/2)+(r/2); /* center */
 122       *output++ = r;           /* right */
 123       *output++ = 0;           /* left surround */
 124       *output++ = 0;           /* right surroud */
 125       *output++ = 0;           /* low freq */
 126     }
 127 }
 128
 129 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 130                                       int output_rate, int input_rate)
 131 {
 132     ReSampleContext *s;
 133
 134     if ( input_channels > 2)
 135       {
 136         av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
 137         return NULL;
 138       }
 139
 140     s = av_mallocz(sizeof(ReSampleContext));
 141     if (!s)
 142       {
 143         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
 144         return NULL;
 145       }
 146
 147     s->ratio = (float)output_rate / (float)input_rate;
 148
 149     s->input_channels = input_channels;
 150     s->output_channels = output_channels;
 151
 152     s->filter_channels = s->input_channels;
 153     if (s->output_channels < s->filter_channels)
 154         s->filter_channels = s->output_channels;
 155
 156 /*
 157  * AC-3 output is the only case where filter_channels could be greater than 2.
 158  * input channels can't be greater than 2, so resample the 2 channels and then
 159  * expand to 6 channels after the resampling.
 160  */
 161     if(s->filter_channels>2)
 162       s->filter_channels = 2;
 163
 164 #define TAPS 16
 165     s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8);
 166
 167     return s;
 168 }
 169
 170 /* resample audio. 'nb_samples' is the number of input samples */
 171 /* XXX: optimize it ! */
 172 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 173 {
 174     int i, nb_samples1;
 175     short *bufin[2];
 176     short *bufout[2];
 177     short *buftmp2[2], *buftmp3[2];
 178     int lenout;
 179
 180     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
 181         /* nothing to do */
 182         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 183         return nb_samples;
 184     }
 185
 186     /* XXX: move those malloc to resample init code */
 187     for(i=0; i<s->filter_channels; i++){
 188         bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
 189         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
 190         buftmp2[i] = bufin[i] + s->temp_len;
 191     }
 192
 193     /* make some zoom to avoid round pb */
 194     lenout= 4*nb_samples * s->ratio + 16;
 195     bufout[0]= av_malloc( lenout * sizeof(short) );
 196     bufout[1]= av_malloc( lenout * sizeof(short) );
 197
 198     if (s->input_channels == 2 &&
 199         s->output_channels == 1) {
 200         buftmp3[0] = output;
 201         stereo_to_mono(buftmp2[0], input, nb_samples);
 202     } else if (s->output_channels >= 2 && s->input_channels == 1) {
 203         buftmp3[0] = bufout[0];
 204         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
 205     } else if (s->output_channels >= 2) {
 206         buftmp3[0] = bufout[0];
 207         buftmp3[1] = bufout[1];
 208         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
 209     } else {
 210         buftmp3[0] = output;
 211         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
 212     }
 213
 214     nb_samples += s->temp_len;
 215
 216     /* resample each channel */
 217     nb_samples1 = 0; /* avoid warning */
 218     for(i=0;i<s->filter_channels;i++) {
 219         int consumed;
 220         int is_last= i+1 == s->filter_channels;
 221
 222         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
 223         s->temp_len= nb_samples - consumed;
 224         s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
 225         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
 226     }
 227
 228     if (s->output_channels == 2 && s->input_channels == 1) {
 229         mono_to_stereo(output, buftmp3[0], nb_samples1);
 230     } else if (s->output_channels == 2) {
 231         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 232     } else if (s->output_channels == 6) {
 233         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 234     }
 235
 236     for(i=0; i<s->filter_channels; i++)
 237         av_free(bufin[i]);
 238
 239     av_free(bufout[0]);
 240     av_free(bufout[1]);
 241     return nb_samples1;
 242 }
 243
 244 void audio_resample_close(ReSampleContext *s)
 245 {
 246     av_resample_close(s->resample_context);
 247     av_freep(&s->temp[0]);
 248     av_freep(&s->temp[1]);
 249     av_free(s);
 250 }