FFmpeg: doc/examples/resampling_audio.c Source File

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resampling_audio.c

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1 /*

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 */

23 /**

24 * @example doc/examples/resampling_audio.c

25 * libswresample API use example.

26 */

28 #include <libavutil/opt.h>

29 #include <libavutil/channel_layout.h>

30 #include <libavutil/samplefmt.h>

31 #include <libswresample/swresample.h>

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;

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 }

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 }

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;

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 }

81 int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,

82 int nb_samples, enum AVSampleFormat sample_fmt, int align)

83 {

84 int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;

86 *data = av_malloc(sizeof(*data) * nb_planes);

87 if (!*data)

88 return AVERROR(ENOMEM);

89 return av_samples_alloc(*data, linesize, nb_channels,

90 nb_samples, sample_fmt, align);

91 }

93 int main(int argc, char **argv)

94 {

95 int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;

96 int src_rate = 48000, dst_rate = 44100;

97 uint8_t **src_data = NULL, **dst_data = NULL;

98 int src_nb_channels = 0, dst_nb_channels = 0;

99 int src_linesize, dst_linesize;

100 int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;

101 enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;

102 const char *dst_filename = NULL;

103 FILE *dst_file;

104 int dst_bufsize;

105 const char *fmt;

106 struct SwrContext *swr_ctx;

107 double t;

108 int ret;

109

110 if (argc != 2) {

111 fprintf(stderr, "Usage: %s output_file\n"

112 "API example program to show how to resample an audio stream with libswresample.\n"

113 "This program generates a series of audio frames, resamples them to a specified "

114 "output format and rate and saves them to an output file named output_file.\n",

115 argv[0]);

116 exit(1);

117 }

118 dst_filename = argv[1];

119

120 dst_file = fopen(dst_filename, "wb");

121 if (!dst_file) {

122 fprintf(stderr, "Could not open destination file %s\n", dst_filename);

123 exit(1);

124 }

125

126 /* create resampler context */

127 swr_ctx = swr_alloc();

128 if (!swr_ctx) {

129 fprintf(stderr, "Could not allocate resampler context\n");

130 ret = AVERROR(ENOMEM);

131 goto end;

132 }

133

134 /* set options */

135 av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);

136 av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);

137 av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);

138

139 av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);

140 av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);

141 av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);

142

143 /* initialize the resampling context */

144 if ((ret = swr_init(swr_ctx)) < 0) {

145 fprintf(stderr, "Failed to initialize the resampling context\n");

146 goto end;

147 }

148

149 /* allocate source and destination samples buffers */

150

151 src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);

152 ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,

153 src_nb_samples, src_sample_fmt, 0);

154 if (ret < 0) {

155 fprintf(stderr, "Could not allocate source samples\n");

156 goto end;

157 }

158

159 /* compute the number of converted samples: buffering is avoided

160 * ensuring that the output buffer will contain at least all the

161 * converted input samples */

162 max_dst_nb_samples = dst_nb_samples =

163 av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

164

165 /* buffer is going to be directly written to a rawaudio file, no alignment */

166 dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);

167 ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,

168 dst_nb_samples, dst_sample_fmt, 0);

169 if (ret < 0) {

170 fprintf(stderr, "Could not allocate destination samples\n");

171 goto end;

172 }

173

174 t = 0;

175 do {

176 /* generate synthetic audio */

177 fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);

178

179 /* compute destination number of samples */

180 dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +

181 src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

182 if (dst_nb_samples > max_dst_nb_samples) {

183 av_free(dst_data[0]);

184 ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,

185 dst_nb_samples, dst_sample_fmt, 1);

186 if (ret < 0)

187 break;

188 max_dst_nb_samples = dst_nb_samples;

189 }

190

191 /* convert to destination format */

192 ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);

193 if (ret < 0) {

194 fprintf(stderr, "Error while converting\n");

195 goto end;

196 }

197 dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,

198 ret, dst_sample_fmt, 1);

199 printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);

200 fwrite(dst_data[0], 1, dst_bufsize, dst_file);

201 } while (t < 10);

202

203 if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)

204 goto end;

205 fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"

206 "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",

207 fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);

208

209 end:

210 if (dst_file)

211 fclose(dst_file);

212

213 if (src_data)

214 av_freep(&src_data[0]);

215 av_freep(&src_data);

216

217 if (dst_data)

218 av_freep(&dst_data[0]);

219 av_freep(&dst_data);

220

221 swr_free(&swr_ctx);

222 return ret < 0;

223 }

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