diff options
Diffstat (limited to 'libavfilter/af_astats.c')
-rw-r--r-- | libavfilter/af_astats.c | 531 |
1 files changed, 531 insertions, 0 deletions
diff --git a/libavfilter/af_astats.c b/libavfilter/af_astats.c new file mode 100644 index 0000000000..e7f9675c2e --- /dev/null +++ b/libavfilter/af_astats.c @@ -0,0 +1,531 @@ +/* + * Copyright (c) 2009 Rob Sykes <robs@users.sourceforge.net> + * Copyright (c) 2013 Paul B Mahol + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * FFmpeg is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with FFmpeg; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <float.h> + +#include "libavutil/opt.h" +#include "audio.h" +#include "avfilter.h" +#include "internal.h" + +typedef struct ChannelStats { + double last; + double sigma_x, sigma_x2; + double avg_sigma_x2, min_sigma_x2, max_sigma_x2; + double min, max; + double nmin, nmax; + double min_run, max_run; + double min_runs, max_runs; + double min_diff, max_diff; + double diff1_sum; + uint64_t mask, imask; + uint64_t min_count, max_count; + uint64_t nb_samples; +} ChannelStats; + +typedef struct { + const AVClass *class; + ChannelStats *chstats; + int nb_channels; + uint64_t tc_samples; + double time_constant; + double mult; + int metadata; + int reset_count; + int nb_frames; + int maxbitdepth; +} AudioStatsContext; + +#define OFFSET(x) offsetof(AudioStatsContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption astats_options[] = { + { "length", "set the window length", OFFSET(time_constant), AV_OPT_TYPE_DOUBLE, {.dbl=.05}, .01, 10, FLAGS }, + { "metadata", "inject metadata in the filtergraph", OFFSET(metadata), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS }, + { "reset", "recalculate stats after this many frames", OFFSET(reset_count), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(astats); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64P, + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + layouts = ff_all_channel_counts(); + if (!layouts) + return AVERROR(ENOMEM); + ret = ff_set_common_channel_layouts(ctx, layouts); + if (ret < 0) + return ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static void reset_stats(AudioStatsContext *s) +{ + int c; + + for (c = 0; c < s->nb_channels; c++) { + ChannelStats *p = &s->chstats[c]; + + p->min = p->nmin = p->min_sigma_x2 = DBL_MAX; + p->max = p->nmax = p->max_sigma_x2 = DBL_MIN; + p->min_diff = DBL_MAX; + p->max_diff = DBL_MIN; + p->sigma_x = 0; + p->sigma_x2 = 0; + p->avg_sigma_x2 = 0; + p->min_sigma_x2 = 0; + p->max_sigma_x2 = 0; + p->min_run = 0; + p->max_run = 0; + p->min_runs = 0; + p->max_runs = 0; + p->diff1_sum = 0; + p->mask = 0; + p->imask = 0xFFFFFFFFFFFFFFFF; + p->min_count = 0; + p->max_count = 0; + p->nb_samples = 0; + } +} + +static int config_output(AVFilterLink *outlink) +{ + AudioStatsContext *s = outlink->src->priv; + + s->chstats = av_calloc(sizeof(*s->chstats), outlink->channels); + if (!s->chstats) + return AVERROR(ENOMEM); + s->nb_channels = outlink->channels; + s->mult = exp((-1 / s->time_constant / outlink->sample_rate)); + s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5; + s->nb_frames = 0; + s->maxbitdepth = av_get_bytes_per_sample(outlink->format) * 8; + + reset_stats(s); + + return 0; +} + +static void bit_depth(AudioStatsContext *s, uint64_t mask, uint64_t imask, AVRational *depth) +{ + unsigned result = s->maxbitdepth; + + mask = mask & (~imask); + + for (; result && !(mask & 1); --result, mask >>= 1); + + depth->den = result; + depth->num = 0; + + for (; result; --result, mask >>= 1) + if (mask & 1) + depth->num++; +} + +static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d, double nd, int64_t i) +{ + if (d < p->min) { + p->min = d; + p->nmin = nd; + p->min_run = 1; + p->min_runs = 0; + p->min_count = 1; + } else if (d == p->min) { + p->min_count++; + p->min_run = d == p->last ? p->min_run + 1 : 1; + } else if (p->last == p->min) { + p->min_runs += p->min_run * p->min_run; + } + + if (d > p->max) { + p->max = d; + p->nmax = nd; + p->max_run = 1; + p->max_runs = 0; + p->max_count = 1; + } else if (d == p->max) { + p->max_count++; + p->max_run = d == p->last ? p->max_run + 1 : 1; + } else if (p->last == p->max) { + p->max_runs += p->max_run * p->max_run; + } + + p->sigma_x += nd; + p->sigma_x2 += nd * nd; + p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * nd * nd; + p->min_diff = FFMIN(p->min_diff, fabs(d - p->last)); + p->max_diff = FFMAX(p->max_diff, fabs(d - p->last)); + p->diff1_sum += fabs(d - p->last); + p->last = d; + p->mask |= i; + p->imask &= i; + + if (p->nb_samples >= s->tc_samples) { + p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2); + p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2); + } + p->nb_samples++; +} + +static void set_meta(AVDictionary **metadata, int chan, const char *key, + const char *fmt, double val) +{ + uint8_t value[128]; + uint8_t key2[128]; + + snprintf(value, sizeof(value), fmt, val); + if (chan) + snprintf(key2, sizeof(key2), "lavfi.astats.%d.%s", chan, key); + else + snprintf(key2, sizeof(key2), "lavfi.astats.%s", key); + av_dict_set(metadata, key2, value, 0); +} + +#define LINEAR_TO_DB(x) (log10(x) * 20) + +static void set_metadata(AudioStatsContext *s, AVDictionary **metadata) +{ + uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0; + double min_runs = 0, max_runs = 0, + min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0, + nmin = DBL_MAX, nmax = DBL_MIN, + max_sigma_x = 0, + diff1_sum = 0, + sigma_x = 0, + sigma_x2 = 0, + min_sigma_x2 = DBL_MAX, + max_sigma_x2 = DBL_MIN; + AVRational depth; + int c; + + for (c = 0; c < s->nb_channels; c++) { + ChannelStats *p = &s->chstats[c]; + + if (p->nb_samples < s->tc_samples) + p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples; + + min = FFMIN(min, p->min); + max = FFMAX(max, p->max); + nmin = FFMIN(nmin, p->nmin); + nmax = FFMAX(nmax, p->nmax); + min_diff = FFMIN(min_diff, p->min_diff); + max_diff = FFMAX(max_diff, p->max_diff); + diff1_sum += p->diff1_sum, + min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2); + max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2); + sigma_x += p->sigma_x; + sigma_x2 += p->sigma_x2; + min_count += p->min_count; + max_count += p->max_count; + min_runs += p->min_runs; + max_runs += p->max_runs; + mask |= p->mask; + imask &= p->imask; + nb_samples += p->nb_samples; + if (fabs(p->sigma_x) > fabs(max_sigma_x)) + max_sigma_x = p->sigma_x; + + set_meta(metadata, c + 1, "DC_offset", "%f", p->sigma_x / p->nb_samples); + set_meta(metadata, c + 1, "Min_level", "%f", p->min); + set_meta(metadata, c + 1, "Max_level", "%f", p->max); + set_meta(metadata, c + 1, "Min_difference", "%f", p->min_diff); + set_meta(metadata, c + 1, "Max_difference", "%f", p->max_diff); + set_meta(metadata, c + 1, "Mean_difference", "%f", p->diff1_sum / (p->nb_samples - 1)); + set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax))); + set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples))); + set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2))); + set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2))); + set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1); + set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count))); + set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count)); + bit_depth(s, p->mask, p->imask, &depth); + set_meta(metadata, c + 1, "Bit_depth", "%f", depth.num); + set_meta(metadata, c + 1, "Bit_depth2", "%f", depth.den); + } + + set_meta(metadata, 0, "Overall.DC_offset", "%f", max_sigma_x / (nb_samples / s->nb_channels)); + set_meta(metadata, 0, "Overall.Min_level", "%f", min); + set_meta(metadata, 0, "Overall.Max_level", "%f", max); + set_meta(metadata, 0, "Overall.Min_difference", "%f", min_diff); + set_meta(metadata, 0, "Overall.Max_difference", "%f", max_diff); + set_meta(metadata, 0, "Overall.Mean_difference", "%f", diff1_sum / (nb_samples - s->nb_channels)); + set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-nmin, nmax))); + set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples))); + set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2))); + set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2))); + set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count))); + set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels); + bit_depth(s, mask, imask, &depth); + set_meta(metadata, 0, "Overall.Bit_depth", "%f", depth.num); + set_meta(metadata, 0, "Overall.Bit_depth2", "%f", depth.den); + set_meta(metadata, 0, "Overall.Number_of_samples", "%f", nb_samples / s->nb_channels); +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *buf) +{ + AudioStatsContext *s = inlink->dst->priv; + AVDictionary **metadata = avpriv_frame_get_metadatap(buf); + const int channels = s->nb_channels; + int i, c; + + if (s->reset_count > 0) { + if (s->nb_frames >= s->reset_count) { + reset_stats(s); + s->nb_frames = 0; + } + s->nb_frames++; + } + + switch (inlink->format) { + case AV_SAMPLE_FMT_DBLP: + for (c = 0; c < channels; c++) { + ChannelStats *p = &s->chstats[c]; + const double *src = (const double *)buf->extended_data[c]; + + for (i = 0; i < buf->nb_samples; i++, src++) + update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 63))); + } + break; + case AV_SAMPLE_FMT_DBL: { + const double *src = (const double *)buf->extended_data[0]; + + for (i = 0; i < buf->nb_samples; i++) { + for (c = 0; c < channels; c++, src++) + update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 63))); + }} + break; + case AV_SAMPLE_FMT_FLTP: + for (c = 0; c < channels; c++) { + ChannelStats *p = &s->chstats[c]; + const float *src = (const float *)buf->extended_data[c]; + + for (i = 0; i < buf->nb_samples; i++, src++) + update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 31))); + } + break; + case AV_SAMPLE_FMT_FLT: { + const float *src = (const float *)buf->extended_data[0]; + + for (i = 0; i < buf->nb_samples; i++) { + for (c = 0; c < channels; c++, src++) + update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 31))); + }} + break; + case AV_SAMPLE_FMT_S64P: + for (c = 0; c < channels; c++) { + ChannelStats *p = &s->chstats[c]; + const int64_t *src = (const int64_t *)buf->extended_data[c]; + + for (i = 0; i < buf->nb_samples; i++, src++) + update_stat(s, p, *src, *src / (double)INT64_MAX, *src); + } + break; + case AV_SAMPLE_FMT_S64: { + const int64_t *src = (const int64_t *)buf->extended_data[0]; + + for (i = 0; i < buf->nb_samples; i++) { + for (c = 0; c < channels; c++, src++) + update_stat(s, &s->chstats[c], *src, *src / (double)INT64_MAX, *src); + }} + break; + case AV_SAMPLE_FMT_S32P: + for (c = 0; c < channels; c++) { + ChannelStats *p = &s->chstats[c]; + const int32_t *src = (const int32_t *)buf->extended_data[c]; + + for (i = 0; i < buf->nb_samples; i++, src++) + update_stat(s, p, *src, *src / (double)INT32_MAX, *src); + } + break; + case AV_SAMPLE_FMT_S32: { + const int32_t *src = (const int32_t *)buf->extended_data[0]; + + for (i = 0; i < buf->nb_samples; i++) { + for (c = 0; c < channels; c++, src++) + update_stat(s, &s->chstats[c], *src, *src / (double)INT32_MAX, *src); + }} + break; + case AV_SAMPLE_FMT_S16P: + for (c = 0; c < channels; c++) { + ChannelStats *p = &s->chstats[c]; + const int16_t *src = (const int16_t *)buf->extended_data[c]; + + for (i = 0; i < buf->nb_samples; i++, src++) + update_stat(s, p, *src, *src / (double)INT16_MAX, *src); + } + break; + case AV_SAMPLE_FMT_S16: { + const int16_t *src = (const int16_t *)buf->extended_data[0]; + + for (i = 0; i < buf->nb_samples; i++) { + for (c = 0; c < channels; c++, src++) + update_stat(s, &s->chstats[c], *src, *src / (double)INT16_MAX, *src); + }} + break; + } + + if (s->metadata) + set_metadata(s, metadata); + + return ff_filter_frame(inlink->dst->outputs[0], buf); +} + +static void print_stats(AVFilterContext *ctx) +{ + AudioStatsContext *s = ctx->priv; + uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0; + double min_runs = 0, max_runs = 0, + min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0, + nmin = DBL_MAX, nmax = DBL_MIN, + max_sigma_x = 0, + diff1_sum = 0, + sigma_x = 0, + sigma_x2 = 0, + min_sigma_x2 = DBL_MAX, + max_sigma_x2 = DBL_MIN; + AVRational depth; + int c; + + for (c = 0; c < s->nb_channels; c++) { + ChannelStats *p = &s->chstats[c]; + + if (p->nb_samples < s->tc_samples) + p->min_sigma_x2 = p->max_sigma_x2 = p->sigma_x2 / p->nb_samples; + + min = FFMIN(min, p->min); + max = FFMAX(max, p->max); + nmin = FFMIN(nmin, p->nmin); + nmax = FFMAX(nmax, p->nmax); + min_diff = FFMIN(min_diff, p->min_diff); + max_diff = FFMAX(max_diff, p->max_diff); + diff1_sum += p->diff1_sum, + min_sigma_x2 = FFMIN(min_sigma_x2, p->min_sigma_x2); + max_sigma_x2 = FFMAX(max_sigma_x2, p->max_sigma_x2); + sigma_x += p->sigma_x; + sigma_x2 += p->sigma_x2; + min_count += p->min_count; + max_count += p->max_count; + min_runs += p->min_runs; + max_runs += p->max_runs; + mask |= p->mask; + imask &= p->imask; + nb_samples += p->nb_samples; + if (fabs(p->sigma_x) > fabs(max_sigma_x)) + max_sigma_x = p->sigma_x; + + av_log(ctx, AV_LOG_INFO, "Channel: %d\n", c + 1); + av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", p->sigma_x / p->nb_samples); + av_log(ctx, AV_LOG_INFO, "Min level: %f\n", p->min); + av_log(ctx, AV_LOG_INFO, "Max level: %f\n", p->max); + av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", p->min_diff); + av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", p->max_diff); + av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", p->diff1_sum / (p->nb_samples - 1)); + av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax))); + av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples))); + av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2))); + if (p->min_sigma_x2 != 1) + av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2))); + av_log(ctx, AV_LOG_INFO, "Crest factor: %f\n", p->sigma_x2 ? FFMAX(-p->nmin, p->nmax) / sqrt(p->sigma_x2 / p->nb_samples) : 1); + 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))); + av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count); + bit_depth(s, p->mask, p->imask, &depth); + av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den); + } + + av_log(ctx, AV_LOG_INFO, "Overall\n"); + av_log(ctx, AV_LOG_INFO, "DC offset: %f\n", max_sigma_x / (nb_samples / s->nb_channels)); + av_log(ctx, AV_LOG_INFO, "Min level: %f\n", min); + av_log(ctx, AV_LOG_INFO, "Max level: %f\n", max); + av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", min_diff); + av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", max_diff); + av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", diff1_sum / (nb_samples - s->nb_channels)); + av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-nmin, nmax))); + av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples))); + av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2))); + if (min_sigma_x2 != 1) + av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2))); + av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count))); + av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels); + bit_depth(s, mask, imask, &depth); + av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den); + av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioStatsContext *s = ctx->priv; + + if (s->nb_channels) + print_stats(ctx); + av_freep(&s->chstats); +} + +static const AVFilterPad astats_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad astats_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +AVFilter ff_af_astats = { + .name = "astats", + .description = NULL_IF_CONFIG_SMALL("Show time domain statistics about audio frames."), + .query_formats = query_formats, + .priv_size = sizeof(AudioStatsContext), + .priv_class = &astats_class, + .uninit = uninit, + .inputs = astats_inputs, + .outputs = astats_outputs, +}; |