yading@10: /*
yading@10:  * Copyright (c) 2012 Stefano Sabatini
yading@10:  *
yading@10:  * Permission is hereby granted, free of charge, to any person obtaining a copy
yading@10:  * of this software and associated documentation files (the "Software"), to deal
yading@10:  * in the Software without restriction, including without limitation the rights
yading@10:  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
yading@10:  * copies of the Software, and to permit persons to whom the Software is
yading@10:  * furnished to do so, subject to the following conditions:
yading@10:  *
yading@10:  * The above copyright notice and this permission notice shall be included in
yading@10:  * all copies or substantial portions of the Software.
yading@10:  *
yading@10:  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
yading@10:  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
yading@10:  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
yading@10:  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
yading@10:  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
yading@10:  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
yading@10:  * THE SOFTWARE.
yading@10:  */
yading@10: 
yading@10: /**
yading@10:  * @example doc/examples/resampling_audio.c
yading@10:  * libswresample API use example.
yading@10:  */
yading@10: 
yading@10: #include <libavutil/opt.h>
yading@10: #include <libavutil/channel_layout.h>
yading@10: #include <libavutil/samplefmt.h>
yading@10: #include <libswresample/swresample.h>
yading@10: 
yading@10: static int get_format_from_sample_fmt(const char **fmt,
yading@10:                                       enum AVSampleFormat sample_fmt)
yading@10: {
yading@10:     int i;
yading@10:     struct sample_fmt_entry {
yading@10:         enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
yading@10:     } sample_fmt_entries[] = {
yading@10:         { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
yading@10:         { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
yading@10:         { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
yading@10:         { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
yading@10:         { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
yading@10:     };
yading@10:     *fmt = NULL;
yading@10: 
yading@10:     for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
yading@10:         struct sample_fmt_entry *entry = &sample_fmt_entries[i];
yading@10:         if (sample_fmt == entry->sample_fmt) {
yading@10:             *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
yading@10:             return 0;
yading@10:         }
yading@10:     }
yading@10: 
yading@10:     fprintf(stderr,
yading@10:             "Sample format %s not supported as output format\n",
yading@10:             av_get_sample_fmt_name(sample_fmt));
yading@10:     return AVERROR(EINVAL);
yading@10: }
yading@10: 
yading@10: /**
yading@10:  * Fill dst buffer with nb_samples, generated starting from t.
yading@10:  */
yading@10: void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
yading@10: {
yading@10:     int i, j;
yading@10:     double tincr = 1.0 / sample_rate, *dstp = dst;
yading@10:     const double c = 2 * M_PI * 440.0;
yading@10: 
yading@10:     /* generate sin tone with 440Hz frequency and duplicated channels */
yading@10:     for (i = 0; i < nb_samples; i++) {
yading@10:         *dstp = sin(c * *t);
yading@10:         for (j = 1; j < nb_channels; j++)
yading@10:             dstp[j] = dstp[0];
yading@10:         dstp += nb_channels;
yading@10:         *t += tincr;
yading@10:     }
yading@10: }
yading@10: 
yading@10: int main(int argc, char **argv)
yading@10: {
yading@10:     int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
yading@10:     int src_rate = 48000, dst_rate = 44100;
yading@10:     uint8_t **src_data = NULL, **dst_data = NULL;
yading@10:     int src_nb_channels = 0, dst_nb_channels = 0;
yading@10:     int src_linesize, dst_linesize;
yading@10:     int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
yading@10:     enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
yading@10:     const char *dst_filename = NULL;
yading@10:     FILE *dst_file;
yading@10:     int dst_bufsize;
yading@10:     const char *fmt;
yading@10:     struct SwrContext *swr_ctx;
yading@10:     double t;
yading@10:     int ret;
yading@10: 
yading@10:     if (argc != 2) {
yading@10:         fprintf(stderr, "Usage: %s output_file\n"
yading@10:                 "API example program to show how to resample an audio stream with libswresample.\n"
yading@10:                 "This program generates a series of audio frames, resamples them to a specified "
yading@10:                 "output format and rate and saves them to an output file named output_file.\n",
yading@10:             argv[0]);
yading@10:         exit(1);
yading@10:     }
yading@10:     dst_filename = argv[1];
yading@10: 
yading@10:     dst_file = fopen(dst_filename, "wb");
yading@10:     if (!dst_file) {
yading@10:         fprintf(stderr, "Could not open destination file %s\n", dst_filename);
yading@10:         exit(1);
yading@10:     }
yading@10: 
yading@10:     /* create resampler context */
yading@10:     swr_ctx = swr_alloc();
yading@10:     if (!swr_ctx) {
yading@10:         fprintf(stderr, "Could not allocate resampler context\n");
yading@10:         ret = AVERROR(ENOMEM);
yading@10:         goto end;
yading@10:     }
yading@10: 
yading@10:     /* set options */
yading@10:     av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);
yading@10:     av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
yading@10:     av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
yading@10: 
yading@10:     av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);
yading@10:     av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
yading@10:     av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
yading@10: 
yading@10:     /* initialize the resampling context */
yading@10:     if ((ret = swr_init(swr_ctx)) < 0) {
yading@10:         fprintf(stderr, "Failed to initialize the resampling context\n");
yading@10:         goto end;
yading@10:     }
yading@10: 
yading@10:     /* allocate source and destination samples buffers */
yading@10: 
yading@10:     src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
yading@10:     ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
yading@10:                                              src_nb_samples, src_sample_fmt, 0);
yading@10:     if (ret < 0) {
yading@10:         fprintf(stderr, "Could not allocate source samples\n");
yading@10:         goto end;
yading@10:     }
yading@10: 
yading@10:     /* compute the number of converted samples: buffering is avoided
yading@10:      * ensuring that the output buffer will contain at least all the
yading@10:      * converted input samples */
yading@10:     max_dst_nb_samples = dst_nb_samples =
yading@10:         av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
yading@10: 
yading@10:     /* buffer is going to be directly written to a rawaudio file, no alignment */
yading@10:     dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
yading@10:     ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
yading@10:                                              dst_nb_samples, dst_sample_fmt, 0);
yading@10:     if (ret < 0) {
yading@10:         fprintf(stderr, "Could not allocate destination samples\n");
yading@10:         goto end;
yading@10:     }
yading@10: 
yading@10:     t = 0;
yading@10:     do {
yading@10:         /* generate synthetic audio */
yading@10:         fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
yading@10: 
yading@10:         /* compute destination number of samples */
yading@10:         dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
yading@10:                                         src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
yading@10:         if (dst_nb_samples > max_dst_nb_samples) {
yading@10:             av_free(dst_data[0]);
yading@10:             ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
yading@10:                                    dst_nb_samples, dst_sample_fmt, 1);
yading@10:             if (ret < 0)
yading@10:                 break;
yading@10:             max_dst_nb_samples = dst_nb_samples;
yading@10:         }
yading@10: 
yading@10:         /* convert to destination format */
yading@10:         ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
yading@10:         if (ret < 0) {
yading@10:             fprintf(stderr, "Error while converting\n");
yading@10:             goto end;
yading@10:         }
yading@10:         dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
yading@10:                                                  ret, dst_sample_fmt, 1);
yading@10:         printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
yading@10:         fwrite(dst_data[0], 1, dst_bufsize, dst_file);
yading@10:     } while (t < 10);
yading@10: 
yading@10:     if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
yading@10:         goto end;
yading@10:     fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
yading@10:             "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
yading@10:             fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
yading@10: 
yading@10: end:
yading@10:     if (dst_file)
yading@10:         fclose(dst_file);
yading@10: 
yading@10:     if (src_data)
yading@10:         av_freep(&src_data[0]);
yading@10:     av_freep(&src_data);
yading@10: 
yading@10:     if (dst_data)
yading@10:         av_freep(&dst_data[0]);
yading@10:     av_freep(&dst_data);
yading@10: 
yading@10:     swr_free(&swr_ctx);
yading@10:     return ret < 0;
yading@10: }