quicktime/ffmpeg/libavcodec/resample.c

   1 /*
   2  * Sample rate convertion for both audio and video
   3  * Copyright (c) 2000 Fabrice Bellard.
   4  *
   5  * This library is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2 of the License, or (at your option) any later version.
   9  *
  10  * This library is distributed in the hope that it will be useful,
  11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with this library; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  18  */
  19
  20 /**
  21  * @file resample.c
  22  * Sample rate convertion for both audio and video.
  23  */
  24
  25 #include "avcodec.h"
  26
  27 struct AVResampleContext;
  28
  29 struct ReSampleContext {
  30     struct AVResampleContext *resample_context;
  31     short *temp[2];
  32     int temp_len;
  33     float ratio;
  34     /* channel convert */
  35     int input_channels, output_channels, filter_channels;
  36 };
  37
  38 /* n1: number of samples */
  39 static void stereo_to_mono(short *output, short *input, int n1)
  40 {
  41     short *p, *q;
  42     int n = n1;
  43
  44     p = input;
  45     q = output;
  46     while (n >= 4) {
  47         q[0] = (p[0] + p[1]) >> 1;
  48         q[1] = (p[2] + p[3]) >> 1;
  49         q[2] = (p[4] + p[5]) >> 1;
  50         q[3] = (p[6] + p[7]) >> 1;
  51         q += 4;
  52         p += 8;
  53         n -= 4;
  54     }
  55     while (n > 0) {
  56         q[0] = (p[0] + p[1]) >> 1;
  57         q++;
  58         p += 2;
  59         n--;
  60     }
  61 }
  62
  63 /* n1: number of samples */
  64 static void mono_to_stereo(short *output, short *input, int n1)
  65 {
  66     short *p, *q;
  67     int n = n1;
  68     int v;
  69
  70     p = input;
  71     q = output;
  72     while (n >= 4) {
  73         v = p[0]; q[0] = v; q[1] = v;
  74         v = p[1]; q[2] = v; q[3] = v;
  75         v = p[2]; q[4] = v; q[5] = v;
  76         v = p[3]; q[6] = v; q[7] = v;
  77         q += 8;
  78         p += 4;
  79         n -= 4;
  80     }
  81     while (n > 0) {
  82         v = p[0]; q[0] = v; q[1] = v;
  83         q += 2;
  84         p += 1;
  85         n--;
  86     }
  87 }
  88
  89 /* XXX: should use more abstract 'N' channels system */
  90 static void stereo_split(short *output1, short *output2, short *input, int n)
  91 {
  92     int i;
  93
  94     for(i=0;i<n;i++) {
  95         *output1++ = *input++;
  96         *output2++ = *input++;
  97     }
  98 }
  99
 100 static void stereo_mux(short *output, short *input1, short *input2, int n)
 101 {
 102     int i;
 103
 104     for(i=0;i<n;i++) {
 105         *output++ = *input1++;
 106         *output++ = *input2++;
 107     }
 108 }
 109
 110 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 111 {
 112     int i;
 113     short l,r;
 114
 115     for(i=0;i<n;i++) {
 116       l=*input1++;
 117       r=*input2++;
 118       *output++ = l;           /* left */
 119       *output++ = (l/2)+(r/2); /* center */
 120       *output++ = r;           /* right */
 121       *output++ = 0;           /* left surround */
 122       *output++ = 0;           /* right surroud */
 123       *output++ = 0;           /* low freq */
 124     }
 125 }
 126
 127 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 128                                       int output_rate, int input_rate)
 129 {
 130     ReSampleContext *s;
 131
 132     if ( input_channels > 2)
 133       {
 134         av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.");
 135         return NULL;
 136       }
 137
 138     s = av_mallocz(sizeof(ReSampleContext));
 139     if (!s)
 140       {
 141         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.");
 142         return NULL;
 143       }
 144
 145     s->ratio = (float)output_rate / (float)input_rate;
 146
 147     s->input_channels = input_channels;
 148     s->output_channels = output_channels;
 149
 150     s->filter_channels = s->input_channels;
 151     if (s->output_channels < s->filter_channels)
 152         s->filter_channels = s->output_channels;
 153
 154 /*
 155  * ac3 output is the only case where filter_channels could be greater than 2.
 156  * input channels can't be greater than 2, so resample the 2 channels and then
 157  * expand to 6 channels after the resampling.
 158  */
 159     if(s->filter_channels>2)
 160       s->filter_channels = 2;
 161
 162     s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0);
 163
 164     return s;
 165 }
 166
 167 /* resample audio. 'nb_samples' is the number of input samples */
 168 /* XXX: optimize it ! */
 169 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 170 {
 171     int i, nb_samples1;
 172     short *bufin[2];
 173     short *bufout[2];
 174     short *buftmp2[2], *buftmp3[2];
 175     int lenout;
 176
 177     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
 178         /* nothing to do */
 179         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 180         return nb_samples;
 181     }
 182
 183     /* XXX: move those malloc to resample init code */
 184     for(i=0; i<s->filter_channels; i++){
 185         bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
 186         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
 187         buftmp2[i] = bufin[i] + s->temp_len;
 188     }
 189
 190     /* make some zoom to avoid round pb */
 191     lenout= (int)(nb_samples * s->ratio) + 16;
 192     bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
 193     bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
 194
 195     if (s->input_channels == 2 &&
 196         s->output_channels == 1) {
 197         buftmp3[0] = output;
 198         stereo_to_mono(buftmp2[0], input, nb_samples);
 199     } else if (s->output_channels >= 2 && s->input_channels == 1) {
 200         buftmp3[0] = bufout[0];
 201         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
 202     } else if (s->output_channels >= 2) {
 203         buftmp3[0] = bufout[0];
 204         buftmp3[1] = bufout[1];
 205         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
 206     } else {
 207         buftmp3[0] = output;
 208         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
 209     }
 210
 211     nb_samples += s->temp_len;
 212
 213     /* resample each channel */
 214     nb_samples1 = 0; /* avoid warning */
 215     for(i=0;i<s->filter_channels;i++) {
 216         int consumed;
 217         int is_last= i+1 == s->filter_channels;
 218
 219         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
 220         s->temp_len= nb_samples - consumed;
 221         s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
 222         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
 223     }
 224
 225     if (s->output_channels == 2 && s->input_channels == 1) {
 226         mono_to_stereo(output, buftmp3[0], nb_samples1);
 227     } else if (s->output_channels == 2) {
 228         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 229     } else if (s->output_channels == 6) {
 230         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 231     }
 232
 233     for(i=0; i<s->filter_channels; i++)
 234         av_free(bufin[i]);
 235
 236     av_free(bufout[0]);
 237     av_free(bufout[1]);
 238     return nb_samples1;
 239 }
 240
 241 void audio_resample_close(ReSampleContext *s)
 242 {
 243     av_resample_close(s->resample_context);
 244     av_freep(&s->temp[0]);
 245     av_freep(&s->temp[1]);
 246     av_free(s);
 247 }