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libavcodec/resample.c

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00001 /*
00002  * samplerate conversion for both audio and video
00003  * Copyright (c) 2000 Fabrice Bellard
00004  *
00005  * This file is part of FFmpeg.
00006  *
00007  * FFmpeg is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * FFmpeg is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with FFmpeg; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00027 #include "avcodec.h"
00028 #include "audioconvert.h"
00029 #include "opt.h"
00030 
00031 struct AVResampleContext;
00032 
00033 static const char *context_to_name(void *ptr)
00034 {
00035     return "audioresample";
00036 }
00037 
00038 static const AVOption options[] = {{NULL}};
00039 static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT };
00040 
00041 struct ReSampleContext {
00042     struct AVResampleContext *resample_context;
00043     short *temp[2];
00044     int temp_len;
00045     float ratio;
00046     /* channel convert */
00047     int input_channels, output_channels, filter_channels;
00048     AVAudioConvert *convert_ctx[2];
00049     enum SampleFormat sample_fmt[2]; 
00050     unsigned sample_size[2];         
00051     short *buffer[2];                
00052     unsigned buffer_size[2];         
00053 };
00054 
00055 /* n1: number of samples */
00056 static void stereo_to_mono(short *output, short *input, int n1)
00057 {
00058     short *p, *q;
00059     int n = n1;
00060 
00061     p = input;
00062     q = output;
00063     while (n >= 4) {
00064         q[0] = (p[0] + p[1]) >> 1;
00065         q[1] = (p[2] + p[3]) >> 1;
00066         q[2] = (p[4] + p[5]) >> 1;
00067         q[3] = (p[6] + p[7]) >> 1;
00068         q += 4;
00069         p += 8;
00070         n -= 4;
00071     }
00072     while (n > 0) {
00073         q[0] = (p[0] + p[1]) >> 1;
00074         q++;
00075         p += 2;
00076         n--;
00077     }
00078 }
00079 
00080 /* n1: number of samples */
00081 static void mono_to_stereo(short *output, short *input, int n1)
00082 {
00083     short *p, *q;
00084     int n = n1;
00085     int v;
00086 
00087     p = input;
00088     q = output;
00089     while (n >= 4) {
00090         v = p[0]; q[0] = v; q[1] = v;
00091         v = p[1]; q[2] = v; q[3] = v;
00092         v = p[2]; q[4] = v; q[5] = v;
00093         v = p[3]; q[6] = v; q[7] = v;
00094         q += 8;
00095         p += 4;
00096         n -= 4;
00097     }
00098     while (n > 0) {
00099         v = p[0]; q[0] = v; q[1] = v;
00100         q += 2;
00101         p += 1;
00102         n--;
00103     }
00104 }
00105 
00106 /* XXX: should use more abstract 'N' channels system */
00107 static void stereo_split(short *output1, short *output2, short *input, int n)
00108 {
00109     int i;
00110 
00111     for(i=0;i<n;i++) {
00112         *output1++ = *input++;
00113         *output2++ = *input++;
00114     }
00115 }
00116 
00117 static void stereo_mux(short *output, short *input1, short *input2, int n)
00118 {
00119     int i;
00120 
00121     for(i=0;i<n;i++) {
00122         *output++ = *input1++;
00123         *output++ = *input2++;
00124     }
00125 }
00126 
00127 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
00128 {
00129     int i;
00130     short l,r;
00131 
00132     for(i=0;i<n;i++) {
00133       l=*input1++;
00134       r=*input2++;
00135       *output++ = l;           /* left */
00136       *output++ = (l/2)+(r/2); /* center */
00137       *output++ = r;           /* right */
00138       *output++ = 0;           /* left surround */
00139       *output++ = 0;           /* right surroud */
00140       *output++ = 0;           /* low freq */
00141     }
00142 }
00143 
00144 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
00145                                         int output_rate, int input_rate,
00146                                         enum SampleFormat sample_fmt_out,
00147                                         enum SampleFormat sample_fmt_in,
00148                                         int filter_length, int log2_phase_count,
00149                                         int linear, double cutoff)
00150 {
00151     ReSampleContext *s;
00152 
00153     if ( input_channels > 2)
00154       {
00155         av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
00156         return NULL;
00157       }
00158 
00159     s = av_mallocz(sizeof(ReSampleContext));
00160     if (!s)
00161       {
00162         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
00163         return NULL;
00164       }
00165 
00166     s->ratio = (float)output_rate / (float)input_rate;
00167 
00168     s->input_channels = input_channels;
00169     s->output_channels = output_channels;
00170 
00171     s->filter_channels = s->input_channels;
00172     if (s->output_channels < s->filter_channels)
00173         s->filter_channels = s->output_channels;
00174 
00175     s->sample_fmt [0] = sample_fmt_in;
00176     s->sample_fmt [1] = sample_fmt_out;
00177     s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
00178     s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
00179 
00180     if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
00181         if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
00182                                                          s->sample_fmt[0], 1, NULL, 0))) {
00183             av_log(s, AV_LOG_ERROR,
00184                    "Cannot convert %s sample format to s16 sample format\n",
00185                    avcodec_get_sample_fmt_name(s->sample_fmt[0]));
00186             av_free(s);
00187             return NULL;
00188         }
00189     }
00190 
00191     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
00192         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
00193                                                          SAMPLE_FMT_S16, 1, NULL, 0))) {
00194             av_log(s, AV_LOG_ERROR,
00195                    "Cannot convert s16 sample format to %s sample format\n",
00196                    avcodec_get_sample_fmt_name(s->sample_fmt[1]));
00197             av_audio_convert_free(s->convert_ctx[0]);
00198             av_free(s);
00199             return NULL;
00200         }
00201     }
00202 
00203 /*
00204  * AC-3 output is the only case where filter_channels could be greater than 2.
00205  * input channels can't be greater than 2, so resample the 2 channels and then
00206  * expand to 6 channels after the resampling.
00207  */
00208     if(s->filter_channels>2)
00209       s->filter_channels = 2;
00210 
00211 #define TAPS 16
00212     s->resample_context= av_resample_init(output_rate, input_rate,
00213                          filter_length, log2_phase_count, linear, cutoff);
00214 
00215     *(const AVClass**)s->resample_context = &audioresample_context_class;
00216 
00217     return s;
00218 }
00219 
00220 #if LIBAVCODEC_VERSION_MAJOR < 53
00221 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
00222                                      int output_rate, int input_rate)
00223 {
00224     return av_audio_resample_init(output_channels, input_channels,
00225                                   output_rate, input_rate,
00226                                   SAMPLE_FMT_S16, SAMPLE_FMT_S16,
00227                                   TAPS, 10, 0, 0.8);
00228 }
00229 #endif
00230 
00231 /* resample audio. 'nb_samples' is the number of input samples */
00232 /* XXX: optimize it ! */
00233 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
00234 {
00235     int i, nb_samples1;
00236     short *bufin[2];
00237     short *bufout[2];
00238     short *buftmp2[2], *buftmp3[2];
00239     short *output_bak = NULL;
00240     int lenout;
00241 
00242     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
00243         /* nothing to do */
00244         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
00245         return nb_samples;
00246     }
00247 
00248     if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
00249         int istride[1] = { s->sample_size[0] };
00250         int ostride[1] = { 2 };
00251         const void *ibuf[1] = { input };
00252         void       *obuf[1];
00253         unsigned input_size = nb_samples*s->input_channels*2;
00254 
00255         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
00256             av_free(s->buffer[0]);
00257             s->buffer_size[0] = input_size;
00258             s->buffer[0] = av_malloc(s->buffer_size[0]);
00259             if (!s->buffer[0]) {
00260                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
00261                 return 0;
00262             }
00263         }
00264 
00265         obuf[0] = s->buffer[0];
00266 
00267         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
00268                              ibuf, istride, nb_samples*s->input_channels) < 0) {
00269             av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
00270             return 0;
00271         }
00272 
00273         input  = s->buffer[0];
00274     }
00275 
00276     lenout= 4*nb_samples * s->ratio + 16;
00277 
00278     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
00279         output_bak = output;
00280 
00281         if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
00282             av_free(s->buffer[1]);
00283             s->buffer_size[1] = lenout;
00284             s->buffer[1] = av_malloc(s->buffer_size[1]);
00285             if (!s->buffer[1]) {
00286                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
00287                 return 0;
00288             }
00289         }
00290 
00291         output = s->buffer[1];
00292     }
00293 
00294     /* XXX: move those malloc to resample init code */
00295     for(i=0; i<s->filter_channels; i++){
00296         bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
00297         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
00298         buftmp2[i] = bufin[i] + s->temp_len;
00299     }
00300 
00301     /* make some zoom to avoid round pb */
00302     bufout[0]= av_malloc( lenout * sizeof(short) );
00303     bufout[1]= av_malloc( lenout * sizeof(short) );
00304 
00305     if (s->input_channels == 2 &&
00306         s->output_channels == 1) {
00307         buftmp3[0] = output;
00308         stereo_to_mono(buftmp2[0], input, nb_samples);
00309     } else if (s->output_channels >= 2 && s->input_channels == 1) {
00310         buftmp3[0] = bufout[0];
00311         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
00312     } else if (s->output_channels >= 2) {
00313         buftmp3[0] = bufout[0];
00314         buftmp3[1] = bufout[1];
00315         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
00316     } else {
00317         buftmp3[0] = output;
00318         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
00319     }
00320 
00321     nb_samples += s->temp_len;
00322 
00323     /* resample each channel */
00324     nb_samples1 = 0; /* avoid warning */
00325     for(i=0;i<s->filter_channels;i++) {
00326         int consumed;
00327         int is_last= i+1 == s->filter_channels;
00328 
00329         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
00330         s->temp_len= nb_samples - consumed;
00331         s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
00332         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
00333     }
00334 
00335     if (s->output_channels == 2 && s->input_channels == 1) {
00336         mono_to_stereo(output, buftmp3[0], nb_samples1);
00337     } else if (s->output_channels == 2) {
00338         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
00339     } else if (s->output_channels == 6) {
00340         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
00341     }
00342 
00343     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
00344         int istride[1] = { 2 };
00345         int ostride[1] = { s->sample_size[1] };
00346         const void *ibuf[1] = { output };
00347         void       *obuf[1] = { output_bak };
00348 
00349         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
00350                              ibuf, istride, nb_samples1*s->output_channels) < 0) {
00351             av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
00352             return 0;
00353         }
00354     }
00355 
00356     for(i=0; i<s->filter_channels; i++)
00357         av_free(bufin[i]);
00358 
00359     av_free(bufout[0]);
00360     av_free(bufout[1]);
00361     return nb_samples1;
00362 }
00363 
00364 void audio_resample_close(ReSampleContext *s)
00365 {
00366     av_resample_close(s->resample_context);
00367     av_freep(&s->temp[0]);
00368     av_freep(&s->temp[1]);
00369     av_freep(&s->buffer[0]);
00370     av_freep(&s->buffer[1]);
00371     av_audio_convert_free(s->convert_ctx[0]);
00372     av_audio_convert_free(s->convert_ctx[1]);
00373     av_free(s);
00374 }

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