FFmpeg: libavfilter/af_astats.c Source File

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libavfilter

af_astats.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 #include <float.h>

24 #include "libavutil/opt.h"

25 #include "audio.h"

26 #include "avfilter.h"

27 #include "internal.h"

29 typedef struct ChannelStats {

30 double last;

31 double sigma_x, sigma_x2;

32 double avg_sigma_x2, min_sigma_x2, max_sigma_x2;

33 double min, max;

34 double min_run, max_run;

35 double min_runs, max_runs;

36 uint64_t min_count, max_count;

37 uint64_t nb_samples;

38 } ChannelStats;

40 typedef struct {

41 const AVClass *class;

42 ChannelStats *chstats;

43 int nb_channels;

44 uint64_t tc_samples;

45 double time_constant;

46 double mult;

47 } AudioStatsContext;

49 #define OFFSET(x) offsetof(AudioStatsContext, x)

50 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

52 static const AVOption astats_options[] = {

53 { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS },

54 { NULL }

55 };

57 AVFILTER_DEFINE_CLASS(astats);

59 static int query_formats(AVFilterContext *ctx)

60 {

61 AVFilterFormats *formats;

62 AVFilterChannelLayouts *layouts;

63 static const enum AVSampleFormat sample_fmts[] = {

64 AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,

65 AV_SAMPLE_FMT_NONE

66 };

68 layouts = ff_all_channel_layouts();

69 if (!layouts)

70 return AVERROR(ENOMEM);

71 ff_set_common_channel_layouts(ctx, layouts);

73 formats = ff_make_format_list(sample_fmts);

74 if (!formats)

75 return AVERROR(ENOMEM);

76 ff_set_common_formats(ctx, formats);

78 formats = ff_all_samplerates();

79 if (!formats)

80 return AVERROR(ENOMEM);

81 ff_set_common_samplerates(ctx, formats);

83 return 0;

84 }

86 static int config_output(AVFilterLink *outlink)

87 {

88 AudioStatsContext *s = outlink->src->priv;

89 int c;

91 s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels);

92 if (!s->chstats)

93 return AVERROR(ENOMEM);

94 s->nb_channels = outlink->channels;

95 s->mult = exp((-1 / s->time_constant / outlink->sample_rate));

96 s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;

98 for (c = 0; c < s->nb_channels; c++) {

99 ChannelStats *p = &s->chstats[c];

100

101 p->min = p->min_sigma_x2 = DBL_MAX;

102 p->max = p->max_sigma_x2 = DBL_MIN;

103 }

104

105 return 0;

106 }

107

108 static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)

109 {

110 if (d < p->min) {

111 p->min = d;

112 p->min_run = 1;

113 p->min_runs = 0;

114 p->min_count = 1;

115 } else if (d == p->min) {

116 p->min_count++;

117 p->min_run = d == p->last ? p->min_run + 1 : 1;

118 } else if (p->last == p->min) {

119 p->min_runs += p->min_run * p->min_run;

120 }

121

122 if (d > p->max) {

123 p->max = d;

124 p->max_run = 1;

125 p->max_runs = 0;

126 p->max_count = 1;

127 } else if (d == p->max) {

128 p->max_count++;

129 p->max_run = d == p->last ? p->max_run + 1 : 1;

130 } else if (p->last == p->max) {

131 p->max_runs += p->max_run * p->max_run;

132 }

133

134 p->sigma_x += d;

135 p->sigma_x2 += d * d;

136 p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;

137 p->last = d;

138

139 if (p->nb_samples >= s->tc_samples) {

140 p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);

141 p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);

142 }

143 p->nb_samples++;

144 }

145

146 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)

147 {

148 AudioStatsContext *s = inlink->dst->priv;

149 const int channels = s->nb_channels;

150 const double *src;

151 int i, c;

152

153 switch (inlink->format) {

154 case AV_SAMPLE_FMT_DBLP:

155 for (c = 0; c < channels; c++) {

156 ChannelStats *p = &s->chstats[c];

157 src = (const double *)buf->extended_data[c];

158

159 for (i = 0; i < buf->nb_samples; i++, src++)

160 update_stat(s, p, *src);

161 }

162 break;

163 case AV_SAMPLE_FMT_DBL:

164 src = (const double *)buf->extended_data[0];

165

166 for (i = 0; i < buf->nb_samples; i++) {

167 for (c = 0; c < channels; c++, src++)

168 update_stat(s, &s->chstats[c], *src);

169 }

170 break;

171 }

172

173 return ff_filter_frame(inlink->dst->outputs[0], buf);

174 }

175

176 #define LINEAR_TO_DB(x) (log10(x) * 20)

177

178 static void print_stats(AVFilterContext *ctx)

179 {

180 AudioStatsContext *s = ctx->priv;

181 uint64_t min_count = 0, max_count = 0, nb_samples = 0;

182 double min_runs = 0, max_runs = 0,

183 min = DBL_MAX, max = DBL_MIN,

184 max_sigma_x = 0,

185 sigma_x = 0,

186 sigma_x2 = 0,

187 min_sigma_x2 = DBL_MAX,

188 max_sigma_x2 = DBL_MIN;

189 int c;

190

191 for (c = 0; c < s->nb_channels; c++) {

192 ChannelStats *p = &s->chstats[c];

193

194 if (p->nb_samples < s->tc_samples)

195 p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples;

196

197 min = FFMIN(min, p->min);

198 max = FFMAX(max, p->max);

199 min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2);

200 max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2);

201 sigma_x += p->sigma_x;

202 sigma_x2 += p->sigma_x2;

203 min_count += p->min_count;

204 max_count += p->max_count;

205 min_runs += p->min_runs;

206 max_runs += p->max_runs;

207 nb_samples += p->nb_samples;

208 if (fabs(p->sigma_x) > fabs(max_sigma_x))

209 max_sigma_x = p->sigma_x;

210

211 av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1);

212 av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples);

213 av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min);

214 av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max);

215 av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));

216 av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));

217 av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));

218 if (p->min_sigma_x2 != 1)

219 av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));

220 av_log(ctx, AV_LOG_INFO, "Crest factor: %f\n", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);

221 av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));

222 av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);

223 }

224

225 av_log(ctx, AV_LOG_INFO, "Overall\n");

226 av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels));

227 av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min);

228 av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max);

229 av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));

230 av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));

231 av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));

232 if (min_sigma_x2 != 1)

233 av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));

234 av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));

235 av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);

236 av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);

237 }

238

239 static av_cold void uninit(AVFilterContext *ctx)

240 {

241 AudioStatsContext *s = ctx->priv;

242

243 print_stats(ctx);

244 av_freep(&s->chstats);

245 }

246

247 static const AVFilterPad astats_inputs[] = {

248 {

249 .name = "default",

250 .type = AVMEDIA_TYPE_AUDIO,

251 .filter_frame = filter_frame,

252 },

253 { NULL }

254 };

255

256 static const AVFilterPad astats_outputs[] = {

257 {

258 .name = "default",

259 .type = AVMEDIA_TYPE_AUDIO,

260 .config_props = config_output,

261 },

262 { NULL }

263 };

264

265 AVFilter avfilter_af_astats = {

266 .name = "astats",

267 .description = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."),

268 .query_formats = query_formats,

269 .priv_size = sizeof(AudioStatsContext),

270 .priv_class = &astats_class,

271 .uninit = uninit,

272 .inputs = astats_inputs,

273 .outputs = astats_outputs,

274 };

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