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  1. /*
  2.  * Copyright (c) 2012 Stefano Sabatini
  3.  *
  4.  * Permission is hereby granted, free of charge, to any person obtaining a copy
  5.  * of this software and associated documentation files (the "Software"), to deal
  6.  * in the Software without restriction, including without limitation the rights
  7.  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  8.  * copies of the Software, and to permit persons to whom the Software is
  9.  * furnished to do so, subject to the following conditions:
  10.  *
  11.  * The above copyright notice and this permission notice shall be included in
  12.  * all copies or substantial portions of the Software.
  13.  *
  14.  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  15.  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  16.  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  17.  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  18.  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  19.  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  20.  * THE SOFTWARE.
  21.  */
  22.  
  23. /**
  24.  * @example doc/examples/resampling_audio.c
  25.  * libswresample API use example.
  26.  */
  27.  
  28. #include <libavutil/opt.h>
  29. #include <libavutil/channel_layout.h>
  30. #include <libavutil/samplefmt.h>
  31. #include <libswresample/swresample.h>
  32.  
  33. static int get_format_from_sample_fmt(const char **fmt,
  34.                                       enum AVSampleFormat sample_fmt)
  35. {
  36.     int i;
  37.     struct sample_fmt_entry {
  38.         enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
  39.     } sample_fmt_entries[] = {
  40.         { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
  41.         { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
  42.         { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
  43.         { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
  44.         { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
  45.     };
  46.     *fmt = NULL;
  47.  
  48.     for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
  49.         struct sample_fmt_entry *entry = &sample_fmt_entries[i];
  50.         if (sample_fmt == entry->sample_fmt) {
  51.             *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
  52.             return 0;
  53.         }
  54.     }
  55.  
  56.     fprintf(stderr,
  57.             "Sample format %s not supported as output format\n",
  58.             av_get_sample_fmt_name(sample_fmt));
  59.     return AVERROR(EINVAL);
  60. }
  61.  
  62. /**
  63.  * Fill dst buffer with nb_samples, generated starting from t.
  64.  */
  65. void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
  66. {
  67.     int i, j;
  68.     double tincr = 1.0 / sample_rate, *dstp = dst;
  69.     const double c = 2 * M_PI * 440.0;
  70.  
  71.     /* generate sin tone with 440Hz frequency and duplicated channels */
  72.     for (i = 0; i < nb_samples; i++) {
  73.         *dstp = sin(c * *t);
  74.         for (j = 1; j < nb_channels; j++)
  75.             dstp[j] = dstp[0];
  76.         dstp += nb_channels;
  77.         *t += tincr;
  78.     }
  79. }
  80.  
  81. int main(int argc, char **argv)
  82. {
  83.     int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
  84.     int src_rate = 48000, dst_rate = 44100;
  85.     uint8_t **src_data = NULL, **dst_data = NULL;
  86.     int src_nb_channels = 0, dst_nb_channels = 0;
  87.     int src_linesize, dst_linesize;
  88.     int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
  89.     enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
  90.     const char *dst_filename = NULL;
  91.     FILE *dst_file;
  92.     int dst_bufsize;
  93.     const char *fmt;
  94.     struct SwrContext *swr_ctx;
  95.     double t;
  96.     int ret;
  97.  
  98.     if (argc != 2) {
  99.         fprintf(stderr, "Usage: %s output_file\n"
  100.                 "API example program to show how to resample an audio stream with libswresample.\n"
  101.                 "This program generates a series of audio frames, resamples them to a specified "
  102.                 "output format and rate and saves them to an output file named output_file.\n",
  103.             argv[0]);
  104.         exit(1);
  105.     }
  106.     dst_filename = argv[1];
  107.  
  108.     dst_file = fopen(dst_filename, "wb");
  109.     if (!dst_file) {
  110.         fprintf(stderr, "Could not open destination file %s\n", dst_filename);
  111.         exit(1);
  112.     }
  113.  
  114.     /* create resampler context */
  115.     swr_ctx = swr_alloc();
  116.     if (!swr_ctx) {
  117.         fprintf(stderr, "Could not allocate resampler context\n");
  118.         ret = AVERROR(ENOMEM);
  119.         goto end;
  120.     }
  121.  
  122.     /* set options */
  123.     av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
  124.     av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
  125.     av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
  126.  
  127.     av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
  128.     av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
  129.     av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
  130.  
  131.     /* initialize the resampling context */
  132.     if ((ret = swr_init(swr_ctx)) < 0) {
  133.         fprintf(stderr, "Failed to initialize the resampling context\n");
  134.         goto end;
  135.     }
  136.  
  137.     /* allocate source and destination samples buffers */
  138.  
  139.     src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
  140.     ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
  141.                                              src_nb_samples, src_sample_fmt, 0);
  142.     if (ret < 0) {
  143.         fprintf(stderr, "Could not allocate source samples\n");
  144.         goto end;
  145.     }
  146.  
  147.     /* compute the number of converted samples: buffering is avoided
  148.      * ensuring that the output buffer will contain at least all the
  149.      * converted input samples */
  150.     max_dst_nb_samples = dst_nb_samples =
  151.         av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
  152.  
  153.     /* buffer is going to be directly written to a rawaudio file, no alignment */
  154.     dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
  155.     ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
  156.                                              dst_nb_samples, dst_sample_fmt, 0);
  157.     if (ret < 0) {
  158.         fprintf(stderr, "Could not allocate destination samples\n");
  159.         goto end;
  160.     }
  161.  
  162.     t = 0;
  163.     do {
  164.         /* generate synthetic audio */
  165.         fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
  166.  
  167.         /* compute destination number of samples */
  168.         dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
  169.                                         src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
  170.         if (dst_nb_samples > max_dst_nb_samples) {
  171.             av_free(dst_data[0]);
  172.             ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
  173.                                    dst_nb_samples, dst_sample_fmt, 1);
  174.             if (ret < 0)
  175.                 break;
  176.             max_dst_nb_samples = dst_nb_samples;
  177.         }
  178.  
  179.         /* convert to destination format */
  180.         ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
  181.         if (ret < 0) {
  182.             fprintf(stderr, "Error while converting\n");
  183.             goto end;
  184.         }
  185.         dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
  186.                                                  ret, dst_sample_fmt, 1);
  187.         printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
  188.         fwrite(dst_data[0], 1, dst_bufsize, dst_file);
  189.     } while (t < 10);
  190.  
  191.     if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
  192.         goto end;
  193.     fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
  194.             "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
  195.             fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
  196.  
  197. end:
  198.     if (dst_file)
  199.         fclose(dst_file);
  200.  
  201.     if (src_data)
  202.         av_freep(&src_data[0]);
  203.     av_freep(&src_data);
  204.  
  205.     if (dst_data)
  206.         av_freep(&dst_data[0]);
  207.     av_freep(&dst_data);
  208.  
  209.     swr_free(&swr_ctx);
  210.     return ret < 0;
  211. }
  212.