FFmpeg: libavfilter/af_aresample.c Source File

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libavfilter

af_aresample.c

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1 /*

4 *

5 * This file is part of FFmpeg.

6 *

7 * FFmpeg is free software; you can redistribute it and/or

8 * modify it under the terms of the GNU Lesser General Public

9 * License as published by the Free Software Foundation; either

10 * version 2.1 of the License, or (at your option) any later version.

11 *

12 * FFmpeg is distributed in the hope that it will be useful,

13 * but WITHOUT ANY WARRANTY; without even the implied warranty of

14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

15 * Lesser General Public License for more details.

16 *

17 * You should have received a copy of the GNU Lesser General Public

18 * License along with FFmpeg; if not, write to the Free Software

19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

20 */

22 /**

23 * @file

24 * resampling audio filter

25 */

27 #include "libavutil/avstring.h"

28 #include "libavutil/channel_layout.h"

29 #include "libavutil/opt.h"

30 #include "libavutil/samplefmt.h"

31 #include "libavutil/avassert.h"

32 #include "libswresample/swresample.h"

33 #include "avfilter.h"

34 #include "audio.h"

35 #include "internal.h"

37 typedef struct {

38 const AVClass *class;

39 int sample_rate_arg;

40 double ratio;

41 struct SwrContext *swr;

42 int64_t next_pts;

43 int req_fullfilled;

44 } AResampleContext;

46 static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)

47 {

48 AResampleContext *aresample = ctx->priv;

49 int ret = 0;

51 aresample->next_pts = AV_NOPTS_VALUE;

52 aresample->swr = swr_alloc();

53 if (!aresample->swr) {

54 ret = AVERROR(ENOMEM);

55 goto end;

56 }

58 if (opts) {

59 AVDictionaryEntry *e = NULL;

61 while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {

62 if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)

63 goto end;

64 }

65 av_dict_free(opts);

66 }

67 if (aresample->sample_rate_arg > 0)

68 av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);

69 end:

70 return ret;

71 }

73 static av_cold void uninit(AVFilterContext *ctx)

74 {

75 AResampleContext *aresample = ctx->priv;

76 swr_free(&aresample->swr);

77 }

79 static int query_formats(AVFilterContext *ctx)

80 {

81 AResampleContext *aresample = ctx->priv;

82 int out_rate = av_get_int(aresample->swr, "osr", NULL);

83 uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);

84 enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);

86 AVFilterLink *inlink = ctx->inputs[0];

87 AVFilterLink *outlink = ctx->outputs[0];

89 AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);

90 AVFilterFormats *out_formats;

91 AVFilterFormats *in_samplerates = ff_all_samplerates();

92 AVFilterFormats *out_samplerates;

93 AVFilterChannelLayouts *in_layouts = ff_all_channel_counts();

94 AVFilterChannelLayouts *out_layouts;

96 ff_formats_ref (in_formats, &inlink->out_formats);

97 ff_formats_ref (in_samplerates, &inlink->out_samplerates);

98 ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);

100 if(out_rate > 0) {

101 int ratelist[] = { out_rate, -1 };

102 out_samplerates = ff_make_format_list(ratelist);

103 } else {

104 out_samplerates = ff_all_samplerates();

105 }

106 ff_formats_ref(out_samplerates, &outlink->in_samplerates);

107

108 if(out_format != AV_SAMPLE_FMT_NONE) {

109 int formatlist[] = { out_format, -1 };

110 out_formats = ff_make_format_list(formatlist);

111 } else

112 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);

113 ff_formats_ref(out_formats, &outlink->in_formats);

114

115 if(out_layout) {

116 int64_t layout_list[] = { out_layout, -1 };

117 out_layouts = avfilter_make_format64_list(layout_list);

118 } else

119 out_layouts = ff_all_channel_counts();

120 ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

121

122 return 0;

123 }

124

125

126 static int config_output(AVFilterLink *outlink)

127 {

128 int ret;

129 AVFilterContext *ctx = outlink->src;

130 AVFilterLink *inlink = ctx->inputs[0];

131 AResampleContext *aresample = ctx->priv;

132 int out_rate;

133 uint64_t out_layout;

134 enum AVSampleFormat out_format;

135 char inchl_buf[128], outchl_buf[128];

136

137 aresample->swr = swr_alloc_set_opts(aresample->swr,

138 outlink->channel_layout, outlink->format, outlink->sample_rate,

139 inlink->channel_layout, inlink->format, inlink->sample_rate,

140 0, ctx);

141 if (!aresample->swr)

142 return AVERROR(ENOMEM);

143 if (!inlink->channel_layout)

144 av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);

145 if (!outlink->channel_layout)

146 av_opt_set_int(aresample->swr, "och", outlink->channels, 0);

147

148 ret = swr_init(aresample->swr);

149 if (ret < 0)

150 return ret;

151

152 out_rate = av_get_int(aresample->swr, "osr", NULL);

153 out_layout = av_get_int(aresample->swr, "ocl", NULL);

154 out_format = av_get_int(aresample->swr, "osf", NULL);

155 outlink->time_base = (AVRational) {1, out_rate};

156

157 av_assert0(outlink->sample_rate == out_rate);

158 av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);

159 av_assert0(outlink->format == out_format);

160

161 aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;

162

163 av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);

164 av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);

165

166 av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",

167 inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,

168 outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);

169 return 0;

170 }

171

172 static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)

173 {

174 AResampleContext *aresample = inlink->dst->priv;

175 const int n_in = insamplesref->nb_samples;

176 int64_t delay;

177 int n_out = n_in * aresample->ratio + 32;

178 AVFilterLink *const outlink = inlink->dst->outputs[0];

179 AVFrame *outsamplesref;

180 int ret;

181

182 delay = swr_get_delay(aresample->swr, outlink->sample_rate);

183 if (delay > 0)

184 n_out += delay;

185

186 outsamplesref = ff_get_audio_buffer(outlink, n_out);

187

188 if(!outsamplesref)

189 return AVERROR(ENOMEM);

190

191 av_frame_copy_props(outsamplesref, insamplesref);

192 outsamplesref->format = outlink->format;

193 av_frame_set_channels(outsamplesref, outlink->channels);

194 outsamplesref->channel_layout = outlink->channel_layout;

195 outsamplesref->sample_rate = outlink->sample_rate;

196

197 if(insamplesref->pts != AV_NOPTS_VALUE) {

198 int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);

199 int64_t outpts= swr_next_pts(aresample->swr, inpts);

200 aresample->next_pts =

201 outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);

202 } else {

203 outsamplesref->pts = AV_NOPTS_VALUE;

204 }

205 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,

206 (void *)insamplesref->extended_data, n_in);

207 if (n_out <= 0) {

208 av_frame_free(&outsamplesref);

209 av_frame_free(&insamplesref);

210 return 0;

211 }

212

213 outsamplesref->nb_samples = n_out;

214

215 ret = ff_filter_frame(outlink, outsamplesref);

216 aresample->req_fullfilled= 1;

217 av_frame_free(&insamplesref);

218 return ret;

219 }

220

221 static int request_frame(AVFilterLink *outlink)

222 {

223 AVFilterContext *ctx = outlink->src;

224 AResampleContext *aresample = ctx->priv;

225 AVFilterLink *const inlink = outlink->src->inputs[0];

226 int ret;

227

228 aresample->req_fullfilled = 0;

229 do{

230 ret = ff_request_frame(ctx->inputs[0]);

231 }while(!aresample->req_fullfilled && ret>=0);

232

233 if (ret == AVERROR_EOF) {

234 AVFrame *outsamplesref;

235 int n_out = 4096;

236 int64_t pts;

237

238 outsamplesref = ff_get_audio_buffer(outlink, n_out);

239 if (!outsamplesref)

240 return AVERROR(ENOMEM);

241

242 pts = swr_next_pts(aresample->swr, INT64_MIN);

243 pts = ROUNDED_DIV(pts, inlink->sample_rate);

244

245 n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);

246 if (n_out <= 0) {

247 av_frame_free(&outsamplesref);

248 return (n_out == 0) ? AVERROR_EOF : n_out;

249 }

250

251 outsamplesref->sample_rate = outlink->sample_rate;

252 outsamplesref->nb_samples = n_out;

253

254 outsamplesref->pts = pts;

255

256 return ff_filter_frame(outlink, outsamplesref);

257 }

258 return ret;

259 }

260

261 static const AVClass *resample_child_class_next(const AVClass *prev)

262 {

263 return prev ? NULL : swr_get_class();

264 }

265

266 static void *resample_child_next(void *obj, void *prev)

267 {

268 AResampleContext *s = obj;

269 return prev ? NULL : s->swr;

270 }

271

272 #define OFFSET(x) offsetof(AResampleContext, x)

273 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

274

275 static const AVOption options[] = {

276 {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },

277 {NULL}

278 };

279

280 static const AVClass aresample_class = {

281 .class_name = "aresample",

282 .item_name = av_default_item_name,

283 .option = options,

284 .version = LIBAVUTIL_VERSION_INT,

285 .child_class_next = resample_child_class_next,

286 .child_next = resample_child_next,

287 };

288

289 static const AVFilterPad aresample_inputs[] = {

290 {

291 .name = "default",

292 .type = AVMEDIA_TYPE_AUDIO,

293 .filter_frame = filter_frame,

294 },

295 { NULL }

296 };

297

298 static const AVFilterPad aresample_outputs[] = {

299 {

300 .name = "default",

301 .config_props = config_output,

302 .request_frame = request_frame,

303 .type = AVMEDIA_TYPE_AUDIO,

304 },

305 { NULL }

306 };

307

308 AVFilter ff_af_aresample = {

309 .name = "aresample",

310 .description = NULL_IF_CONFIG_SMALL("Resample audio data."),

311 .init_dict = init_dict,

312 .uninit = uninit,

313 .query_formats = query_formats,

314 .priv_size = sizeof(AResampleContext),

315 .priv_class = &aresample_class,

316 .inputs = aresample_inputs,

317 .outputs = aresample_outputs,

318 };

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