[PATCH v3 0/5] avfilter/af_volumedetect.c: Add 32bit float audio
changes since v1: - Defined callback and assigning to filter frame according to planar/packed and float/int. - Fixed rounding same value 2 times - Subnormal values are supported - Replaced square division with ldexp Yigithan Yigit (5): avfilter/af_volumedetect.c: Move logdb function avfilter/af_volumedetect.c: Add 32bit float audio support avfilter/af_volumedetect.c: Added functions for int/float and planar/packed avfilter/af_volumedetect.c: reindent after last commit Replace division with ldexp libavfilter/af_volumedetect.c | 255 ++++++++++++++++++++++++++-------- 1 file changed, 198 insertions(+), 57 deletions(-) -- 2.45.2
--- libavfilter/af_volumedetect.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index 8b001d1cf2..327801a7f9 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -24,6 +24,8 @@ #include "avfilter.h" #include "internal.h" +#define MAX_DB 91 + typedef struct VolDetectContext { /** * Number of samples at each PCM value. @@ -33,6 +35,14 @@ typedef struct VolDetectContext { uint64_t histogram[0x10001]; } VolDetectContext; +static inline double logdb(uint64_t v) +{ + double d = v / (double)(0x8000 * 0x8000); + if (!v) + return MAX_DB; + return -log10(d) * 10; +} + static int filter_frame(AVFilterLink *inlink, AVFrame *samples) { AVFilterContext *ctx = inlink->dst; @@ -56,16 +66,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *samples) return ff_filter_frame(inlink->dst->outputs[0], samples); } -#define MAX_DB 91 - -static inline double logdb(uint64_t v) -{ - double d = v / (double)(0x8000 * 0x8000); - if (!v) - return MAX_DB; - return -log10(d) * 10; -} - static void print_stats(AVFilterContext *ctx) { VolDetectContext *vd = ctx->priv; -- 2.45.2
--- libavfilter/af_volumedetect.c | 139 ++++++++++++++++++++++++++-------- 1 file changed, 107 insertions(+), 32 deletions(-) diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index 327801a7f9..edd2d56f7a 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -1,5 +1,6 @@ /* * Copyright (c) 2012 Nicolas George + * Copyright (c) 2024 Yigithan Yigit - 32 Bit Float Audio Support * * This file is part of FFmpeg. * @@ -20,48 +21,62 @@ #include "libavutil/channel_layout.h" #include "libavutil/avassert.h" +#include "libavutil/mem.h" #include "audio.h" #include "avfilter.h" #include "internal.h" +#define MAX_DB_FLT 1024 #define MAX_DB 91 +#define HISTOGRAM_SIZE 0x10000 +#define HISTOGRAM_SIZE_FLT (MAX_DB_FLT*2) + +typedef struct VolDetectContext VolDetectContext; typedef struct VolDetectContext { - /** - * Number of samples at each PCM value. - * histogram[0x8000 + i] is the number of samples at value i. - * The extra element is there for symmetry. - */ - uint64_t histogram[0x10001]; + uint64_t* histogram; ///< for integer number of samples at each PCM value, for float number of samples at each dB + uint64_t nb_samples; ///< number of samples + double sum2; ///< sum of the squares of the samples + double max; ///< maximum sample value + int is_float; ///< true if the input is in floating point + void (*process_samples)(VolDetectContext *vd, AVFrame *samples); } VolDetectContext; -static inline double logdb(uint64_t v) +static inline double logdb(double v, enum AVSampleFormat sample_fmt) { - double d = v / (double)(0x8000 * 0x8000); - if (!v) - return MAX_DB; - return -log10(d) * 10; + if (sample_fmt == AV_SAMPLE_FMT_FLT) { + if (!v) + return MAX_DB_FLT; + return -log10(v) * 10; + } else { + double d = v / (double)(0x8000 * 0x8000); + if (!v) + return MAX_DB; + return -log10(d) * 10; + } +} + +static void update_float_stats(VolDetectContext *vd, float *audio_data) +{ + double sample; + int idx; + if(!isfinite(*audio_data) || isnan(*audio_data)) + return; + sample = fabsf(*audio_data); + if (sample > vd->max) + vd->max = sample; + vd->sum2 += sample * sample; + idx = (int)floorf(logdb(sample * sample, AV_SAMPLE_FMT_FLT)) + MAX_DB_FLT; + vd->histogram[idx]++; + vd->nb_samples++; } static int filter_frame(AVFilterLink *inlink, AVFrame *samples) { AVFilterContext *ctx = inlink->dst; VolDetectContext *vd = ctx->priv; - int nb_samples = samples->nb_samples; - int nb_channels = samples->ch_layout.nb_channels; - int nb_planes = nb_channels; - int plane, i; - int16_t *pcm; - - if (!av_sample_fmt_is_planar(samples->format)) { - nb_samples *= nb_channels; - nb_planes = 1; - } - for (plane = 0; plane < nb_planes; plane++) { - pcm = (int16_t *)samples->extended_data[plane]; - for (i = 0; i < nb_samples; i++) - vd->histogram[pcm[i] + 0x8000]++; - } + + vd->process_samples(vd, samples); return ff_filter_frame(inlink->dst->outputs[0], samples); } @@ -73,6 +88,20 @@ static void print_stats(AVFilterContext *ctx) uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0; uint64_t histdb[MAX_DB + 1] = { 0 }; + if (!vd->nb_samples) + return; + if (vd->is_float) { + av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n", vd->nb_samples); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(vd->sum2 / vd->nb_samples, AV_SAMPLE_FMT_FLT)); + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -2.0*logdb(vd->max, AV_SAMPLE_FMT_FLT)); + for (i = 0; i < HISTOGRAM_SIZE_FLT && !vd->histogram[i]; i++); + for (; i >= 0 && sum < vd->nb_samples / 1000; i++) { + if (!vd->histogram[i]) + continue; + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %" PRId64 "\n", MAX_DB_FLT - i, vd->histogram[i]); + sum += vd->histogram[i]; + } + } else { for (i = 0; i < 0x10000; i++) nb_samples += vd->histogram[i]; av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples); @@ -92,26 +121,61 @@ static void print_stats(AVFilterContext *ctx) return; power = (power + nb_samples_shift / 2) / nb_samples_shift; av_assert0(power <= 0x8000 * 0x8000); - av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power)); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb((double)power, AV_SAMPLE_FMT_S16)); max_volume = 0x8000; while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] && !vd->histogram[0x8000 - max_volume]) max_volume--; - av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume)); + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16)); for (i = 0; i < 0x10000; i++) - histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i]; + histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000), AV_SAMPLE_FMT_S16)] += vd->histogram[i]; for (i = 0; i <= MAX_DB && !histdb[i]; i++); for (; i <= MAX_DB && sum < nb_samples / 1000; i++) { - av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]); + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i, histdb[i]); sum += histdb[i]; } + } +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + VolDetectContext *vd = ctx->priv; + size_t histogram_size; + + vd->is_float = outlink->format == AV_SAMPLE_FMT_FLT || + outlink->format == AV_SAMPLE_FMT_FLTP; + + if (!vd->is_float) { + /* + * Number of samples at each PCM value. + * Only used for integer formats. + * For 16 bit signed PCM there are 65536. + * histogram[0x8000 + i] is the number of samples at value i. + * The extra element is there for symmetry. + */ + histogram_size = HISTOGRAM_SIZE + 1; + } else { + /* + * The histogram is used to store the number of samples at each dB + * instead of the number of samples at each PCM value. + */ + histogram_size = HISTOGRAM_SIZE_FLT + 1; + } + vd->histogram = av_calloc(histogram_size, sizeof(uint64_t)); + if (!vd->histogram) + return AVERROR(ENOMEM); + return 0; } static av_cold void uninit(AVFilterContext *ctx) { + VolDetectContext *vd = ctx->priv; print_stats(ctx); + if (vd->histogram) + av_freep(&vd->histogram); } static const AVFilterPad volumedetect_inputs[] = { @@ -122,6 +186,14 @@ static const AVFilterPad volumedetect_inputs[] = { }, }; +static const AVFilterPad volumedetect_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, +}; + const AVFilter ff_af_volumedetect = { .name = "volumedetect", .description = NULL_IF_CONFIG_SMALL("Detect audio volume."), @@ -129,6 +201,9 @@ const AVFilter ff_af_volumedetect = { .uninit = uninit, .flags = AVFILTER_FLAG_METADATA_ONLY, FILTER_INPUTS(volumedetect_inputs), - FILTER_OUTPUTS(ff_audio_default_filterpad), - FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P), + FILTER_OUTPUTS(volumedetect_outputs), + FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_FLT, + AV_SAMPLE_FMT_FLTP), }; -- 2.45.2
Le 2 juillet 2024 04:33:51 GMT+03:00, Yigithan Yigit <yigithanyigitdevel@gmail.com> a écrit :
--- libavfilter/af_volumedetect.c | 139 ++++++++++++++++++++++++++-------- 1 file changed, 107 insertions(+), 32 deletions(-)
Did you try to compile this patch?
diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index 327801a7f9..edd2d56f7a 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -1,5 +1,6 @@ /* * Copyright (c) 2012 Nicolas George + * Copyright (c) 2024 Yigithan Yigit - 32 Bit Float Audio Support * * This file is part of FFmpeg. * @@ -20,48 +21,62 @@
#include "libavutil/channel_layout.h" #include "libavutil/avassert.h" +#include "libavutil/mem.h" #include "audio.h" #include "avfilter.h" #include "internal.h"
+#define MAX_DB_FLT 1024 #define MAX_DB 91 +#define HISTOGRAM_SIZE 0x10000 +#define HISTOGRAM_SIZE_FLT (MAX_DB_FLT*2) + +typedef struct VolDetectContext VolDetectContext;
typedef struct VolDetectContext { - /** - * Number of samples at each PCM value. - * histogram[0x8000 + i] is the number of samples at value i. - * The extra element is there for symmetry. - */ - uint64_t histogram[0x10001]; + uint64_t* histogram; ///< for integer number of samples at each PCM value, for float number of samples at each dB + uint64_t nb_samples; ///< number of samples + double sum2; ///< sum of the squares of the samples + double max; ///< maximum sample value + int is_float; ///< true if the input is in floating point + void (*process_samples)(VolDetectContext *vd, AVFrame *samples); } VolDetectContext;
-static inline double logdb(uint64_t v) +static inline double logdb(double v, enum AVSampleFormat sample_fmt) { - double d = v / (double)(0x8000 * 0x8000); - if (!v) - return MAX_DB; - return -log10(d) * 10; + if (sample_fmt == AV_SAMPLE_FMT_FLT) { + if (!v) + return MAX_DB_FLT; + return -log10(v) * 10; + } else { + double d = v / (double)(0x8000 * 0x8000); + if (!v) + return MAX_DB; + return -log10(d) * 10; + } +} + +static void update_float_stats(VolDetectContext *vd, float *audio_data) +{ + double sample; + int idx; + if(!isfinite(*audio_data) || isnan(*audio_data)) + return; + sample = fabsf(*audio_data); + if (sample > vd->max) + vd->max = sample; + vd->sum2 += sample * sample; + idx = (int)floorf(logdb(sample * sample, AV_SAMPLE_FMT_FLT)) + MAX_DB_FLT; + vd->histogram[idx]++; + vd->nb_samples++; }
static int filter_frame(AVFilterLink *inlink, AVFrame *samples) { AVFilterContext *ctx = inlink->dst; VolDetectContext *vd = ctx->priv; - int nb_samples = samples->nb_samples; - int nb_channels = samples->ch_layout.nb_channels; - int nb_planes = nb_channels; - int plane, i; - int16_t *pcm; - - if (!av_sample_fmt_is_planar(samples->format)) { - nb_samples *= nb_channels; - nb_planes = 1; - } - for (plane = 0; plane < nb_planes; plane++) { - pcm = (int16_t *)samples->extended_data[plane]; - for (i = 0; i < nb_samples; i++) - vd->histogram[pcm[i] + 0x8000]++; - } + + vd->process_samples(vd, samples);
return ff_filter_frame(inlink->dst->outputs[0], samples); } @@ -73,6 +88,20 @@ static void print_stats(AVFilterContext *ctx) uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0; uint64_t histdb[MAX_DB + 1] = { 0 };
+ if (!vd->nb_samples) + return; + if (vd->is_float) { + av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n", vd->nb_samples); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(vd->sum2 / vd->nb_samples, AV_SAMPLE_FMT_FLT)); + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -2.0*logdb(vd->max, AV_SAMPLE_FMT_FLT)); + for (i = 0; i < HISTOGRAM_SIZE_FLT && !vd->histogram[i]; i++); + for (; i >= 0 && sum < vd->nb_samples / 1000; i++) { + if (!vd->histogram[i]) + continue; + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %" PRId64 "\n", MAX_DB_FLT - i, vd->histogram[i]); + sum += vd->histogram[i]; + } + } else { for (i = 0; i < 0x10000; i++) nb_samples += vd->histogram[i]; av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples); @@ -92,26 +121,61 @@ static void print_stats(AVFilterContext *ctx) return; power = (power + nb_samples_shift / 2) / nb_samples_shift; av_assert0(power <= 0x8000 * 0x8000); - av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power)); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb((double)power, AV_SAMPLE_FMT_S16));
max_volume = 0x8000; while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] && !vd->histogram[0x8000 - max_volume]) max_volume--; - av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume)); + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16));
for (i = 0; i < 0x10000; i++) - histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i]; + histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000), AV_SAMPLE_FMT_S16)] += vd->histogram[i]; for (i = 0; i <= MAX_DB && !histdb[i]; i++); for (; i <= MAX_DB && sum < nb_samples / 1000; i++) { - av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]); + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i, histdb[i]); sum += histdb[i]; } + } +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + VolDetectContext *vd = ctx->priv; + size_t histogram_size; + + vd->is_float = outlink->format == AV_SAMPLE_FMT_FLT || + outlink->format == AV_SAMPLE_FMT_FLTP; + + if (!vd->is_float) { + /* + * Number of samples at each PCM value. + * Only used for integer formats. + * For 16 bit signed PCM there are 65536. + * histogram[0x8000 + i] is the number of samples at value i. + * The extra element is there for symmetry. + */ + histogram_size = HISTOGRAM_SIZE + 1; + } else { + /* + * The histogram is used to store the number of samples at each dB + * instead of the number of samples at each PCM value. + */ + histogram_size = HISTOGRAM_SIZE_FLT + 1; + } + vd->histogram = av_calloc(histogram_size, sizeof(uint64_t)); + if (!vd->histogram) + return AVERROR(ENOMEM); + return 0; }
static av_cold void uninit(AVFilterContext *ctx) { + VolDetectContext *vd = ctx->priv; print_stats(ctx); + if (vd->histogram) + av_freep(&vd->histogram); }
static const AVFilterPad volumedetect_inputs[] = { @@ -122,6 +186,14 @@ static const AVFilterPad volumedetect_inputs[] = { }, };
+static const AVFilterPad volumedetect_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, +}; + const AVFilter ff_af_volumedetect = { .name = "volumedetect", .description = NULL_IF_CONFIG_SMALL("Detect audio volume."), @@ -129,6 +201,9 @@ const AVFilter ff_af_volumedetect = { .uninit = uninit, .flags = AVFILTER_FLAG_METADATA_ONLY, FILTER_INPUTS(volumedetect_inputs), - FILTER_OUTPUTS(ff_audio_default_filterpad), - FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P), + FILTER_OUTPUTS(volumedetect_outputs), + FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_FLT, + AV_SAMPLE_FMT_FLTP), };
Le 2 juillet 2024 08:46:53 GMT+03:00, "Rémi Denis-Courmont" <remi@remlab.net> a écrit :
Le 2 juillet 2024 04:33:51 GMT+03:00, Yigithan Yigit <yigithanyigitdevel@gmail.com> a écrit :
--- libavfilter/af_volumedetect.c | 139 ++++++++++++++++++++++++++-------- 1 file changed, 107 insertions(+), 32 deletions(-)
Did you try to compile this patch?
Nvmd misread.
Le 2 juillet 2024 04:33:51 GMT+03:00, Yigithan Yigit <yigithanyigitdevel@gmail.com> a écrit :
--- libavfilter/af_volumedetect.c | 139 ++++++++++++++++++++++++++-------- 1 file changed, 107 insertions(+), 32 deletions(-)
diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index 327801a7f9..edd2d56f7a 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -1,5 +1,6 @@ /* * Copyright (c) 2012 Nicolas George + * Copyright (c) 2024 Yigithan Yigit - 32 Bit Float Audio Support * * This file is part of FFmpeg. * @@ -20,48 +21,62 @@
#include "libavutil/channel_layout.h" #include "libavutil/avassert.h" +#include "libavutil/mem.h" #include "audio.h" #include "avfilter.h" #include "internal.h"
+#define MAX_DB_FLT 1024 #define MAX_DB 91 +#define HISTOGRAM_SIZE 0x10000 +#define HISTOGRAM_SIZE_FLT (MAX_DB_FLT*2) + +typedef struct VolDetectContext VolDetectContext;
typedef struct VolDetectContext { - /** - * Number of samples at each PCM value. - * histogram[0x8000 + i] is the number of samples at value i. - * The extra element is there for symmetry. - */ - uint64_t histogram[0x10001]; + uint64_t* histogram; ///< for integer number of samples at each PCM value, for float number of samples at each dB + uint64_t nb_samples; ///< number of samples + double sum2; ///< sum of the squares of the samples + double max; ///< maximum sample value + int is_float; ///< true if the input is in floating point + void (*process_samples)(VolDetectContext *vd, AVFrame *samples); } VolDetectContext;
-static inline double logdb(uint64_t v) +static inline double logdb(double v, enum AVSampleFormat sample_fmt) { - double d = v / (double)(0x8000 * 0x8000); - if (!v) - return MAX_DB; - return -log10(d) * 10; + if (sample_fmt == AV_SAMPLE_FMT_FLT) {
There's no point in doing this. You've already up-converted to double precision and do all the calculations in double precision. Maybe that's fine or maybe not, but either way, this doesn't look sensible.
+ if (!v) + return MAX_DB_FLT; + return -log10(v) * 10; + } else { + double d = v / (double)(0x8000 * 0x8000); + if (!v) + return MAX_DB; + return -log10(d) * 10; + } +} + +static void update_float_stats(VolDetectContext *vd, float *audio_data) +{ + double sample; + int idx; + if(!isfinite(*audio_data) || isnan(*audio_data)) + return; + sample = fabsf(*audio_data); + if (sample > vd->max) + vd->max = sample; + vd->sum2 += sample * sample; + idx = (int)floorf(logdb(sample * sample, AV_SAMPLE_FMT_FLT)) + MAX_DB_FLT; + vd->histogram[idx]++; + vd->nb_samples++; }
static int filter_frame(AVFilterLink *inlink, AVFrame *samples) { AVFilterContext *ctx = inlink->dst; VolDetectContext *vd = ctx->priv; - int nb_samples = samples->nb_samples; - int nb_channels = samples->ch_layout.nb_channels; - int nb_planes = nb_channels; - int plane, i; - int16_t *pcm; - - if (!av_sample_fmt_is_planar(samples->format)) { - nb_samples *= nb_channels; - nb_planes = 1; - } - for (plane = 0; plane < nb_planes; plane++) { - pcm = (int16_t *)samples->extended_data[plane]; - for (i = 0; i < nb_samples; i++) - vd->histogram[pcm[i] + 0x8000]++; - } + + vd->process_samples(vd, samples);
return ff_filter_frame(inlink->dst->outputs[0], samples); } @@ -73,6 +88,20 @@ static void print_stats(AVFilterContext *ctx) uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0; uint64_t histdb[MAX_DB + 1] = { 0 };
+ if (!vd->nb_samples) + return; + if (vd->is_float) { + av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n", vd->nb_samples); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(vd->sum2 / vd->nb_samples, AV_SAMPLE_FMT_FLT)); + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -2.0*logdb(vd->max, AV_SAMPLE_FMT_FLT)); + for (i = 0; i < HISTOGRAM_SIZE_FLT && !vd->histogram[i]; i++); + for (; i >= 0 && sum < vd->nb_samples / 1000; i++) { + if (!vd->histogram[i]) + continue; + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %" PRId64 "\n", MAX_DB_FLT - i, vd->histogram[i]); + sum += vd->histogram[i]; + } + } else { for (i = 0; i < 0x10000; i++) nb_samples += vd->histogram[i]; av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples); @@ -92,26 +121,61 @@ static void print_stats(AVFilterContext *ctx) return; power = (power + nb_samples_shift / 2) / nb_samples_shift; av_assert0(power <= 0x8000 * 0x8000); - av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power)); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb((double)power, AV_SAMPLE_FMT_S16));
max_volume = 0x8000; while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] && !vd->histogram[0x8000 - max_volume]) max_volume--; - av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume)); + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16));
for (i = 0; i < 0x10000; i++) - histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i]; + histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000), AV_SAMPLE_FMT_S16)] += vd->histogram[i]; for (i = 0; i <= MAX_DB && !histdb[i]; i++); for (; i <= MAX_DB && sum < nb_samples / 1000; i++) { - av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]); + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i, histdb[i]); sum += histdb[i]; } + } +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + VolDetectContext *vd = ctx->priv; + size_t histogram_size; + + vd->is_float = outlink->format == AV_SAMPLE_FMT_FLT || + outlink->format == AV_SAMPLE_FMT_FLTP; + + if (!vd->is_float) { + /* + * Number of samples at each PCM value. + * Only used for integer formats. + * For 16 bit signed PCM there are 65536. + * histogram[0x8000 + i] is the number of samples at value i. + * The extra element is there for symmetry. + */ + histogram_size = HISTOGRAM_SIZE + 1; + } else { + /* + * The histogram is used to store the number of samples at each dB + * instead of the number of samples at each PCM value. + */ + histogram_size = HISTOGRAM_SIZE_FLT + 1; + } + vd->histogram = av_calloc(histogram_size, sizeof(uint64_t)); + if (!vd->histogram) + return AVERROR(ENOMEM); + return 0; }
static av_cold void uninit(AVFilterContext *ctx) { + VolDetectContext *vd = ctx->priv; print_stats(ctx); + if (vd->histogram) + av_freep(&vd->histogram); }
static const AVFilterPad volumedetect_inputs[] = { @@ -122,6 +186,14 @@ static const AVFilterPad volumedetect_inputs[] = { }, };
+static const AVFilterPad volumedetect_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, +}; + const AVFilter ff_af_volumedetect = { .name = "volumedetect", .description = NULL_IF_CONFIG_SMALL("Detect audio volume."), @@ -129,6 +201,9 @@ const AVFilter ff_af_volumedetect = { .uninit = uninit, .flags = AVFILTER_FLAG_METADATA_ONLY, FILTER_INPUTS(volumedetect_inputs), - FILTER_OUTPUTS(ff_audio_default_filterpad), - FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P), + FILTER_OUTPUTS(volumedetect_outputs), + FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_FLT, + AV_SAMPLE_FMT_FLTP), };
On Jul 2, 2024, at 8:51 AM, Rémi Denis-Courmont <remi@remlab.net> wrote:
Le 2 juillet 2024 04:33:51 GMT+03:00, Yigithan Yigit <yigithanyigitdevel@gmail.com <mailto:yigithanyigitdevel@gmail.com>> a écrit :
--- libavfilter/af_volumedetect.c | 139 ++++++++++++++++++++++++++-------- 1 file changed, 107 insertions(+), 32 deletions(-)
diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index 327801a7f9..edd2d56f7a 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -1,5 +1,6 @@ /* * Copyright (c) 2012 Nicolas George + * Copyright (c) 2024 Yigithan Yigit - 32 Bit Float Audio Support * * This file is part of FFmpeg. * @@ -20,48 +21,62 @@
#include "libavutil/channel_layout.h" #include "libavutil/avassert.h" +#include "libavutil/mem.h" #include "audio.h" #include "avfilter.h" #include "internal.h"
+#define MAX_DB_FLT 1024 #define MAX_DB 91 +#define HISTOGRAM_SIZE 0x10000 +#define HISTOGRAM_SIZE_FLT (MAX_DB_FLT*2) + +typedef struct VolDetectContext VolDetectContext;
typedef struct VolDetectContext { - /** - * Number of samples at each PCM value. - * histogram[0x8000 + i] is the number of samples at value i. - * The extra element is there for symmetry. - */ - uint64_t histogram[0x10001]; + uint64_t* histogram; ///< for integer number of samples at each PCM value, for float number of samples at each dB + uint64_t nb_samples; ///< number of samples + double sum2; ///< sum of the squares of the samples + double max; ///< maximum sample value + int is_float; ///< true if the input is in floating point + void (*process_samples)(VolDetectContext *vd, AVFrame *samples); } VolDetectContext;
-static inline double logdb(uint64_t v) +static inline double logdb(double v, enum AVSampleFormat sample_fmt) { - double d = v / (double)(0x8000 * 0x8000); - if (!v) - return MAX_DB; - return -log10(d) * 10; + if (sample_fmt == AV_SAMPLE_FMT_FLT) {
There's no point in doing this. You've already up-converted to double precision and do all the calculations in double precision. Maybe that's fine or maybe not, but either way, this doesn't look sensible.
+ if (!v) + return MAX_DB_FLT; + return -log10(v) * 10; + } else { + double d = v / (double)(0x8000 * 0x8000); + if (!v) + return MAX_DB; + return -log10(d) * 10; + } +} +
If I understand your concerns correctly, We should have function like this;
static inline double logdb(double v, enum AVSampleFormat sample_fmt) { if (!v) return sample_fmt == AV_SAMPLE_FMT_FLT ? MAX_DB_FLT : MAX_DB;
if (sample_fmt == AV_SAMPLE_FMT_S16) v = ldexp(v, -30);
return -log10(v) * 10; }
What do you think about that? Thanks for the feedback Yigithan
--- libavfilter/af_volumedetect.c | 66 +++++++++++++++++++++++++++++++++++ 1 file changed, 66 insertions(+) diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index edd2d56f7a..778f0cac6c 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -71,6 +71,64 @@ static void update_float_stats(VolDetectContext *vd, float *audio_data) vd->nb_samples++; } +static void process_float_planar_samples(VolDetectContext *vd, AVFrame *samples) +{ + int plane, i; + int nb_channels = samples->ch_layout.nb_channels; + int nb_samples = samples->nb_samples; + float *audio_data; + for (plane = 0; plane < nb_channels; plane++) { + audio_data = (float *)samples->extended_data[plane]; + for (i = 0; i < nb_samples; i++) { + update_float_stats(vd, &audio_data[i]); + } + } +} + +static void process_float_packed_samples(VolDetectContext *vd, AVFrame *samples) +{ + int i, j; + int nb_channels = samples->ch_layout.nb_channels; + int nb_samples = samples->nb_samples; + float *audio_data; + for (i = 0; i < nb_samples; i++) { + audio_data = (float *)samples->extended_data[0]; + for (j = 0; j < nb_channels; j++) { + update_float_stats(vd, &audio_data[i * nb_channels + j]); + } + } +} + +static void process_int_planar_samples(VolDetectContext *vd, AVFrame *samples) +{ + int plane, i; + int nb_channels = samples->ch_layout.nb_channels; + int nb_samples = samples->nb_samples; + int16_t *pcm; + for (plane = 0; plane < nb_channels; plane++) { + pcm = (int16_t *)samples->extended_data[plane]; + for (i = 0; i < nb_samples; i++) { + vd->histogram[pcm[i] + 0x8000]++; + vd->nb_samples++; + } + } +} + +static void process_int_packed_samples(VolDetectContext *vd, AVFrame *samples) +{ + int i, j; + int nb_channels = samples->ch_layout.nb_channels; + int nb_samples = samples->nb_samples; + int16_t *pcm; + for (i = 0; i < nb_samples; i++) { + pcm = (int16_t *)samples->extended_data[0]; + for (j = 0; j < nb_channels; j++) { + vd->histogram[pcm[i * nb_channels + j] + 0x8000]++; + vd->nb_samples++; + } + } +} + static int filter_frame(AVFilterLink *inlink, AVFrame *samples) { AVFilterContext *ctx = inlink->dst; @@ -157,12 +215,20 @@ static int config_output(AVFilterLink *outlink) * The extra element is there for symmetry. */ histogram_size = HISTOGRAM_SIZE + 1; + if (av_sample_fmt_is_planar(outlink->format)) + vd->process_samples = process_int_planar_samples; + else + vd->process_samples = process_int_packed_samples; } else { /* * The histogram is used to store the number of samples at each dB * instead of the number of samples at each PCM value. */ histogram_size = HISTOGRAM_SIZE_FLT + 1; + if (av_sample_fmt_is_planar(outlink->format)) + vd->process_samples = process_float_planar_samples; + else + vd->process_samples = process_float_packed_samples; } vd->histogram = av_calloc(histogram_size, sizeof(uint64_t)); if (!vd->histogram) -- 2.45.2
Why did you ignore my comment from the last iteration? -- Anton Khirnov
--- libavfilter/af_volumedetect.c | 62 +++++++++++++++++------------------ 1 file changed, 31 insertions(+), 31 deletions(-) diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index 778f0cac6c..a53212015d 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -160,40 +160,40 @@ static void print_stats(AVFilterContext *ctx) sum += vd->histogram[i]; } } else { - for (i = 0; i < 0x10000; i++) - nb_samples += vd->histogram[i]; - av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples); - if (!nb_samples) - return; + for (i = 0; i < 0x10000; i++) + nb_samples += vd->histogram[i]; + av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples); + if (!nb_samples) + return; - /* If nb_samples > 1<<34, there is a risk of overflow in the - multiplication or the sum: shift all histogram values to avoid that. - The total number of samples must be recomputed to avoid rounding - errors. */ - shift = av_log2(nb_samples >> 33); - for (i = 0; i < 0x10000; i++) { - nb_samples_shift += vd->histogram[i] >> shift; - power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift); - } - if (!nb_samples_shift) - return; - power = (power + nb_samples_shift / 2) / nb_samples_shift; - av_assert0(power <= 0x8000 * 0x8000); - av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb((double)power, AV_SAMPLE_FMT_S16)); + /* If nb_samples > 1<<34, there is a risk of overflow in the + multiplication or the sum: shift all histogram values to avoid that. + The total number of samples must be recomputed to avoid rounding + errors. */ + shift = av_log2(nb_samples >> 33); + for (i = 0; i < 0x10000; i++) { + nb_samples_shift += vd->histogram[i] >> shift; + power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift); + } + if (!nb_samples_shift) + return; + power = (power + nb_samples_shift / 2) / nb_samples_shift; + av_assert0(power <= 0x8000 * 0x8000); + av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb((double)power, AV_SAMPLE_FMT_S16)); - max_volume = 0x8000; - while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] && - !vd->histogram[0x8000 - max_volume]) - max_volume--; - av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16)); + max_volume = 0x8000; + while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] && + !vd->histogram[0x8000 - max_volume]) + max_volume--; + av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16)); - for (i = 0; i < 0x10000; i++) - histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000), AV_SAMPLE_FMT_S16)] += vd->histogram[i]; - for (i = 0; i <= MAX_DB && !histdb[i]; i++); - for (; i <= MAX_DB && sum < nb_samples / 1000; i++) { - av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i, histdb[i]); - sum += histdb[i]; - } + for (i = 0; i < 0x10000; i++) + histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000), AV_SAMPLE_FMT_S16)] += vd->histogram[i]; + for (i = 0; i <= MAX_DB && !histdb[i]; i++); + for (; i <= MAX_DB && sum < nb_samples / 1000; i++) { + av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i, histdb[i]); + sum += histdb[i]; + } } } -- 2.45.2
--- libavfilter/af_volumedetect.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c index a53212015d..856d5a295d 100644 --- a/libavfilter/af_volumedetect.c +++ b/libavfilter/af_volumedetect.c @@ -49,7 +49,7 @@ static inline double logdb(double v, enum AVSampleFormat sample_fmt) return MAX_DB_FLT; return -log10(v) * 10; } else { - double d = v / (double)(0x8000 * 0x8000); + double d = ldexp(v, -30); if (!v) return MAX_DB; return -log10(d) * 10; -- 2.45.2
participants (3)
-
Anton Khirnov -
Rémi Denis-Courmont -
Yigithan Yigit