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
33static 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 */
65static 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
81int main(int argc, char **argv)
82{
83    AVChannelLayout src_ch_layout = AV_CHANNEL_LAYOUT_STEREO, dst_ch_layout = AV_CHANNEL_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    char buf[64];
96    double t;
97    int ret;
98
99    if (argc != 2) {
100        fprintf(stderr, "Usage: %s output_file\n"
101                "API example program to show how to resample an audio stream with libswresample.\n"
102                "This program generates a series of audio frames, resamples them to a specified "
103                "output format and rate and saves them to an output file named output_file.\n",
104            argv[0]);
105        exit(1);
106    }
107    dst_filename = argv[1];
108
109    dst_file = fopen(dst_filename, "wb");
110    if (!dst_file) {
111        fprintf(stderr, "Could not open destination file %s\n", dst_filename);
112        exit(1);
113    }
114
115    /* create resampler context */
116    swr_ctx = swr_alloc();
117    if (!swr_ctx) {
118        fprintf(stderr, "Could not allocate resampler context\n");
119        ret = AVERROR(ENOMEM);
120        goto end;
121    }
122
123    /* set options */
124    av_opt_set_chlayout(swr_ctx, "in_chlayout",    &src_ch_layout, 0);
125    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);
126    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
127
128    av_opt_set_chlayout(swr_ctx, "out_chlayout",    &dst_ch_layout, 0);
129    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);
130    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
131
132    /* initialize the resampling context */
133    if ((ret = swr_init(swr_ctx)) < 0) {
134        fprintf(stderr, "Failed to initialize the resampling context\n");
135        goto end;
136    }
137
138    /* allocate source and destination samples buffers */
139
140    src_nb_channels = src_ch_layout.nb_channels;
141    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
142                                             src_nb_samples, src_sample_fmt, 0);
143    if (ret < 0) {
144        fprintf(stderr, "Could not allocate source samples\n");
145        goto end;
146    }
147
148    /* compute the number of converted samples: buffering is avoided
149     * ensuring that the output buffer will contain at least all the
150     * converted input samples */
151    max_dst_nb_samples = dst_nb_samples =
152        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
153
154    /* buffer is going to be directly written to a rawaudio file, no alignment */
155    dst_nb_channels = dst_ch_layout.nb_channels;
156    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
157                                             dst_nb_samples, dst_sample_fmt, 0);
158    if (ret < 0) {
159        fprintf(stderr, "Could not allocate destination samples\n");
160        goto end;
161    }
162
163    t = 0;
164    do {
165        /* generate synthetic audio */
166        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
167
168        /* compute destination number of samples */
169        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
170                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
171        if (dst_nb_samples > max_dst_nb_samples) {
172            av_freep(&dst_data[0]);
173            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
174                                   dst_nb_samples, dst_sample_fmt, 1);
175            if (ret < 0)
176                break;
177            max_dst_nb_samples = dst_nb_samples;
178        }
179
180        /* convert to destination format */
181        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
182        if (ret < 0) {
183            fprintf(stderr, "Error while converting\n");
184            goto end;
185        }
186        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
187                                                 ret, dst_sample_fmt, 1);
188        if (dst_bufsize < 0) {
189            fprintf(stderr, "Could not get sample buffer size\n");
190            goto end;
191        }
192        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
193        fwrite(dst_data[0], 1, dst_bufsize, dst_file);
194    } while (t < 10);
195
196    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
197        goto end;
198    av_channel_layout_describe(&dst_ch_layout, buf, sizeof(buf));
199    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
200            "ffplay -f %s -channel_layout %s -channels %d -ar %d %s\n",
201            fmt, buf, dst_nb_channels, dst_rate, dst_filename);
202
203end:
204    fclose(dst_file);
205
206    if (src_data)
207        av_freep(&src_data[0]);
208    av_freep(&src_data);
209
210    if (dst_data)
211        av_freep(&dst_data[0]);
212    av_freep(&dst_data);
213
214    swr_free(&swr_ctx);
215    return ret < 0;
216}
217