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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 };
00040 
00041 struct ReSampleContext {
00042     const AVClass *av_class;
00043     struct AVResampleContext *resample_context;
00044     short *temp[2];
00045     int temp_len;
00046     float ratio;
00047     /* channel convert */
00048     int input_channels, output_channels, filter_channels;
00049     AVAudioConvert *convert_ctx[2];
00050     enum SampleFormat sample_fmt[2]; 
00051     unsigned sample_size[2];         
00052     short *buffer[2];                
00053     unsigned buffer_size[2];         
00054 };
00055 
00056 /* n1: number of samples */
00057 static void stereo_to_mono(short *output, short *input, int n1)
00058 {
00059     short *p, *q;
00060     int n = n1;
00061 
00062     p = input;
00063     q = output;
00064     while (n >= 4) {
00065         q[0] = (p[0] + p[1]) >> 1;
00066         q[1] = (p[2] + p[3]) >> 1;
00067         q[2] = (p[4] + p[5]) >> 1;
00068         q[3] = (p[6] + p[7]) >> 1;
00069         q += 4;
00070         p += 8;
00071         n -= 4;
00072     }
00073     while (n > 0) {
00074         q[0] = (p[0] + p[1]) >> 1;
00075         q++;
00076         p += 2;
00077         n--;
00078     }
00079 }
00080 
00081 /* n1: number of samples */
00082 static void mono_to_stereo(short *output, short *input, int n1)
00083 {
00084     short *p, *q;
00085     int n = n1;
00086     int v;
00087 
00088     p = input;
00089     q = output;
00090     while (n >= 4) {
00091         v = p[0]; q[0] = v; q[1] = v;
00092         v = p[1]; q[2] = v; q[3] = v;
00093         v = p[2]; q[4] = v; q[5] = v;
00094         v = p[3]; q[6] = v; q[7] = v;
00095         q += 8;
00096         p += 4;
00097         n -= 4;
00098     }
00099     while (n > 0) {
00100         v = p[0]; q[0] = v; q[1] = v;
00101         q += 2;
00102         p += 1;
00103         n--;
00104     }
00105 }
00106 
00107 /* XXX: should use more abstract 'N' channels system */
00108 static void stereo_split(short *output1, short *output2, short *input, int n)
00109 {
00110     int i;
00111 
00112     for(i=0;i<n;i++) {
00113         *output1++ = *input++;
00114         *output2++ = *input++;
00115     }
00116 }
00117 
00118 static void stereo_mux(short *output, short *input1, short *input2, int n)
00119 {
00120     int i;
00121 
00122     for(i=0;i<n;i++) {
00123         *output++ = *input1++;
00124         *output++ = *input2++;
00125     }
00126 }
00127 
00128 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
00129 {
00130     int i;
00131     short l,r;
00132 
00133     for(i=0;i<n;i++) {
00134       l=*input1++;
00135       r=*input2++;
00136       *output++ = l;           /* left */
00137       *output++ = (l/2)+(r/2); /* center */
00138       *output++ = r;           /* right */
00139       *output++ = 0;           /* left surround */
00140       *output++ = 0;           /* right surroud */
00141       *output++ = 0;           /* low freq */
00142     }
00143 }
00144 
00145 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
00146                                         int output_rate, int input_rate,
00147                                         enum SampleFormat sample_fmt_out,
00148                                         enum SampleFormat sample_fmt_in,
00149                                         int filter_length, int log2_phase_count,
00150                                         int linear, double cutoff)
00151 {
00152     ReSampleContext *s;
00153 
00154     if ( input_channels > 2)
00155       {
00156         av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
00157         return NULL;
00158       }
00159 
00160     s = av_mallocz(sizeof(ReSampleContext));
00161     if (!s)
00162       {
00163         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
00164         return NULL;
00165       }
00166 
00167     s->ratio = (float)output_rate / (float)input_rate;
00168 
00169     s->input_channels = input_channels;
00170     s->output_channels = output_channels;
00171 
00172     s->filter_channels = s->input_channels;
00173     if (s->output_channels < s->filter_channels)
00174         s->filter_channels = s->output_channels;
00175 
00176     s->sample_fmt [0] = sample_fmt_in;
00177     s->sample_fmt [1] = sample_fmt_out;
00178     s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
00179     s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
00180 
00181     if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
00182         if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
00183                                                          s->sample_fmt[0], 1, NULL, 0))) {
00184             av_log(s, AV_LOG_ERROR,
00185                    "Cannot convert %s sample format to s16 sample format\n",
00186                    avcodec_get_sample_fmt_name(s->sample_fmt[0]));
00187             av_free(s);
00188             return NULL;
00189         }
00190     }
00191 
00192     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
00193         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
00194                                                          SAMPLE_FMT_S16, 1, NULL, 0))) {
00195             av_log(s, AV_LOG_ERROR,
00196                    "Cannot convert s16 sample format to %s sample format\n",
00197                    avcodec_get_sample_fmt_name(s->sample_fmt[1]));
00198             av_audio_convert_free(s->convert_ctx[0]);
00199             av_free(s);
00200             return NULL;
00201         }
00202     }
00203 
00204 /*
00205  * AC-3 output is the only case where filter_channels could be greater than 2.
00206  * input channels can't be greater than 2, so resample the 2 channels and then
00207  * expand to 6 channels after the resampling.
00208  */
00209     if(s->filter_channels>2)
00210       s->filter_channels = 2;
00211 
00212 #define TAPS 16
00213     s->resample_context= av_resample_init(output_rate, input_rate,
00214                          filter_length, log2_phase_count, linear, cutoff);
00215 
00216     s->av_class= &audioresample_context_class;
00217 
00218     return s;
00219 }
00220 
00221 #if LIBAVCODEC_VERSION_MAJOR < 53
00222 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
00223                                      int output_rate, int input_rate)
00224 {
00225     return av_audio_resample_init(output_channels, input_channels,
00226                                   output_rate, input_rate,
00227                                   SAMPLE_FMT_S16, SAMPLE_FMT_S16,
00228                                   TAPS, 10, 0, 0.8);
00229 }
00230 #endif
00231 
00232 /* resample audio. 'nb_samples' is the number of input samples */
00233 /* XXX: optimize it ! */
00234 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
00235 {
00236     int i, nb_samples1;
00237     short *bufin[2];
00238     short *bufout[2];
00239     short *buftmp2[2], *buftmp3[2];
00240     short *output_bak = NULL;
00241     int lenout;
00242 
00243     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
00244         /* nothing to do */
00245         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
00246         return nb_samples;
00247     }
00248 
00249     if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
00250         int istride[1] = { s->sample_size[0] };
00251         int ostride[1] = { 2 };
00252         const void *ibuf[1] = { input };
00253         void       *obuf[1];
00254         unsigned input_size = nb_samples*s->input_channels*2;
00255 
00256         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
00257             av_free(s->buffer[0]);
00258             s->buffer_size[0] = input_size;
00259             s->buffer[0] = av_malloc(s->buffer_size[0]);
00260             if (!s->buffer[0]) {
00261                 av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
00262                 return 0;
00263             }
00264         }
00265 
00266         obuf[0] = s->buffer[0];
00267 
00268         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
00269                              ibuf, istride, nb_samples*s->input_channels) < 0) {
00270             av_log(s, AV_LOG_ERROR, "Audio sample format conversion failed\n");
00271             return 0;
00272         }
00273 
00274         input  = s->buffer[0];
00275     }
00276 
00277     lenout= 4*nb_samples * s->ratio + 16;
00278 
00279     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
00280         output_bak = output;
00281 
00282         if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
00283             av_free(s->buffer[1]);
00284             s->buffer_size[1] = lenout;
00285             s->buffer[1] = av_malloc(s->buffer_size[1]);
00286             if (!s->buffer[1]) {
00287                 av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n");
00288                 return 0;
00289             }
00290         }
00291 
00292         output = s->buffer[1];
00293     }
00294 
00295     /* XXX: move those malloc to resample init code */
00296     for(i=0; i<s->filter_channels; i++){
00297         bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
00298         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
00299         buftmp2[i] = bufin[i] + s->temp_len;
00300     }
00301 
00302     /* make some zoom to avoid round pb */
00303     bufout[0]= av_malloc( lenout * sizeof(short) );
00304     bufout[1]= av_malloc( lenout * sizeof(short) );
00305 
00306     if (s->input_channels == 2 &&
00307         s->output_channels == 1) {
00308         buftmp3[0] = output;
00309         stereo_to_mono(buftmp2[0], input, nb_samples);
00310     } else if (s->output_channels >= 2 && s->input_channels == 1) {
00311         buftmp3[0] = bufout[0];
00312         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
00313     } else if (s->output_channels >= 2) {
00314         buftmp3[0] = bufout[0];
00315         buftmp3[1] = bufout[1];
00316         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
00317     } else {
00318         buftmp3[0] = output;
00319         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
00320     }
00321 
00322     nb_samples += s->temp_len;
00323 
00324     /* resample each channel */
00325     nb_samples1 = 0; /* avoid warning */
00326     for(i=0;i<s->filter_channels;i++) {
00327         int consumed;
00328         int is_last= i+1 == s->filter_channels;
00329 
00330         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
00331         s->temp_len= nb_samples - consumed;
00332         s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
00333         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
00334     }
00335 
00336     if (s->output_channels == 2 && s->input_channels == 1) {
00337         mono_to_stereo(output, buftmp3[0], nb_samples1);
00338     } else if (s->output_channels == 2) {
00339         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
00340     } else if (s->output_channels == 6) {
00341         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
00342     }
00343 
00344     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
00345         int istride[1] = { 2 };
00346         int ostride[1] = { s->sample_size[1] };
00347         const void *ibuf[1] = { output };
00348         void       *obuf[1] = { output_bak };
00349 
00350         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
00351                              ibuf, istride, nb_samples1*s->output_channels) < 0) {
00352             av_log(s, AV_LOG_ERROR, "Audio sample format convertion failed\n");
00353             return 0;
00354         }
00355     }
00356 
00357     for(i=0; i<s->filter_channels; i++)
00358         av_free(bufin[i]);
00359 
00360     av_free(bufout[0]);
00361     av_free(bufout[1]);
00362     return nb_samples1;
00363 }
00364 
00365 void audio_resample_close(ReSampleContext *s)
00366 {
00367     av_resample_close(s->resample_context);
00368     av_freep(&s->temp[0]);
00369     av_freep(&s->temp[1]);
00370     av_freep(&s->buffer[0]);
00371     av_freep(&s->buffer[1]);
00372     av_audio_convert_free(s->convert_ctx[0]);
00373     av_audio_convert_free(s->convert_ctx[1]);
00374     av_free(s);
00375 }

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