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 double ratio;

39 struct SwrContext *swr;

40 int64_t next_pts;

41 int req_fullfilled;

42 } AResampleContext;

44 static av_cold int init(AVFilterContext *ctx, const char *args)

45 {

46 AResampleContext *aresample = ctx->priv;

47 int ret = 0;

48 char *argd = av_strdup(args);

50 aresample->next_pts = AV_NOPTS_VALUE;

51 aresample->swr = swr_alloc();

52 if (!aresample->swr) {

53 ret = AVERROR(ENOMEM);

54 goto end;

55 }

57 if (args) {

58 char *ptr = argd, *token;

60 while (token = av_strtok(ptr, ":", &ptr)) {

61 char *value;

62 av_strtok(token, "=", &value);

64 if (value) {

65 if ((ret = av_opt_set(aresample->swr, token, value, 0)) < 0)

66 goto end;

67 } else {

68 int out_rate;

69 if ((ret = ff_parse_sample_rate(&out_rate, token, ctx)) < 0)

70 goto end;

71 if ((ret = av_opt_set_int(aresample->swr, "osr", out_rate, 0)) < 0)

72 goto end;

73 }

74 }

75 }

76 end:

77 av_free(argd);

78 return ret;

79 }

81 static av_cold void uninit(AVFilterContext *ctx)

82 {

83 AResampleContext *aresample = ctx->priv;

84 swr_free(&aresample->swr);

85 }

87 static int query_formats(AVFilterContext *ctx)

88 {

89 AResampleContext *aresample = ctx->priv;

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

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

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

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

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

97 AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);

98 AVFilterFormats *out_formats;

99 AVFilterFormats *in_samplerates = ff_all_samplerates();

100 AVFilterFormats *out_samplerates;

101 AVFilterChannelLayouts *in_layouts = ff_all_channel_counts();

102 AVFilterChannelLayouts *out_layouts;

103

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

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

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

107

108 if(out_rate > 0) {

109 out_samplerates = ff_make_format_list((int[]){ out_rate, -1 });

110 } else {

111 out_samplerates = ff_all_samplerates();

112 }

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

114

115 if(out_format != AV_SAMPLE_FMT_NONE) {

116 out_formats = ff_make_format_list((int[]){ out_format, -1 });

117 } else

118 out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);

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

120

121 if(out_layout) {

122 out_layouts = avfilter_make_format64_list((int64_t[]){ out_layout, -1 });

123 } else

124 out_layouts = ff_all_channel_counts();

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

126

127 return 0;

128 }

129

130

131 static int config_output(AVFilterLink *outlink)

132 {

133 int ret;

134 AVFilterContext *ctx = outlink->src;

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

136 AResampleContext *aresample = ctx->priv;

137 int out_rate;

138 uint64_t out_layout;

139 enum AVSampleFormat out_format;

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

141

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

143 outlink->channel_layout, outlink->format, outlink->sample_rate,

144 inlink->channel_layout, inlink->format, inlink->sample_rate,

145 0, ctx);

146 if (!aresample->swr)

147 return AVERROR(ENOMEM);

148 if (!inlink->channel_layout)

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

150 if (!outlink->channel_layout)

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

152

153 ret = swr_init(aresample->swr);

154 if (ret < 0)

155 return ret;

156

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

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

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

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

161

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

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

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

165

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

167

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

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

170

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

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

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

174 return 0;

175 }

176

177 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)

178 {

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

180 const int n_in = insamplesref->audio->nb_samples;

181 int n_out = n_in * aresample->ratio * 2 + 256;

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

183 AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);

184 int ret;

185

186 if(!outsamplesref)

187 return AVERROR(ENOMEM);

188

189 avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);

190 outsamplesref->format = outlink->format;

191 outsamplesref->audio->channels = outlink->channels;

192 outsamplesref->audio->channel_layout = outlink->channel_layout;

193 outsamplesref->audio->sample_rate = outlink->sample_rate;

194

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

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

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

198 aresample->next_pts =

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

200 } else {

201 outsamplesref->pts = AV_NOPTS_VALUE;

202 }

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

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

205 if (n_out <= 0) {

206 avfilter_unref_buffer(outsamplesref);

207 avfilter_unref_buffer(insamplesref);

208 return 0;

209 }

210

211 outsamplesref->audio->nb_samples = n_out;

212

213 ret = ff_filter_frame(outlink, outsamplesref);

214 aresample->req_fullfilled= 1;

215 avfilter_unref_buffer(insamplesref);

216 return ret;

217 }

218

219 static int request_frame(AVFilterLink *outlink)

220 {

221 AVFilterContext *ctx = outlink->src;

222 AResampleContext *aresample = ctx->priv;

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

224 int ret;

225

226 aresample->req_fullfilled = 0;

227 do{

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

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

230

231 if (ret == AVERROR_EOF) {

232 AVFilterBufferRef *outsamplesref;

233 int n_out = 4096;

234

235 outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);

236 if (!outsamplesref)

237 return AVERROR(ENOMEM);

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

239 if (n_out <= 0) {

240 avfilter_unref_buffer(outsamplesref);

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

242 }

243

244 outsamplesref->audio->sample_rate = outlink->sample_rate;

245 outsamplesref->audio->nb_samples = n_out;

246 #if 0

247 outsamplesref->pts = aresample->next_pts;

248 if(aresample->next_pts != AV_NOPTS_VALUE)

249 aresample->next_pts += av_rescale_q(n_out, (AVRational){1 ,outlink->sample_rate}, outlink->time_base);

250 #else

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

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

253 #endif

254

255 ff_filter_frame(outlink, outsamplesref);

256 return 0;

257 }

258 return ret;

259 }

260

261 static const AVFilterPad aresample_inputs[] = {

262 {

263 .name = "default",

264 .type = AVMEDIA_TYPE_AUDIO,

265 .filter_frame = filter_frame,

266 .min_perms = AV_PERM_READ,

267 },

268 { NULL },

269 };

270

271 static const AVFilterPad aresample_outputs[] = {

272 {

273 .name = "default",

274 .config_props = config_output,

275 .request_frame = request_frame,

276 .type = AVMEDIA_TYPE_AUDIO,

277 },

278 { NULL },

279 };

280

281 AVFilter avfilter_af_aresample = {

282 .name = "aresample",

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

284 .init = init,

285 .uninit = uninit,

286 .query_formats = query_formats,

287 .priv_size = sizeof(AResampleContext),

288 .inputs = aresample_inputs,

289 .outputs = aresample_outputs,

290 };

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