| 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 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 | static 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_freep(&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 | if (dst_bufsize < 0) { |
| 188 | fprintf(stderr, "Could not get sample buffer size\n"); |
| 189 | goto end; |
| 190 | } |
| 191 | printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret); |
| 192 | fwrite(dst_data[0], 1, dst_bufsize, dst_file); |
| 193 | } while (t < 10); |
| 194 | |
| 195 | if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0) |
| 196 | goto end; |
| 197 | fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n" |
| 198 | "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n", |
| 199 | fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename); |
| 200 | |
| 201 | end: |
| 202 | fclose(dst_file); |
| 203 | |
| 204 | if (src_data) |
| 205 | av_freep(&src_data[0]); |
| 206 | av_freep(&src_data); |
| 207 | |
| 208 | if (dst_data) |
| 209 | av_freep(&dst_data[0]); |
| 210 | av_freep(&dst_data); |
| 211 | |
| 212 | swr_free(&swr_ctx); |
| 213 | return ret < 0; |
| 214 | } |