avfilter: add apsnr filter

Paul B Mahol committed 3 years ago

Commit 951def81 parent a1928df4 files · +73 −3Browse files

Unified · Split

4 files

doc/filters.texi +7 −0
libavfilter/Makefile +1 −0
libavfilter/af_asdr.c +64 −3
libavfilter/allfilters.c +1 −0

@@ -2836,6 +2836,13 @@ Default value is 8. · 2835 unmodified lines
2836	2836
2837	2837	This filter supports the all above options as @ref{commands}.
2838	2838
	2839	+@section apsnr
	2840	+Measure Audio Peak Signal-to-Noise Ratio.
	2841	+
	2842	+This filter takes two audio streams for input, and outputs first
	2843	+audio stream.
	2844	+Results are in dB per channel at end of either input.
	2845	+
2839	2846	@section apsyclip
2840	2847	Apply Psychoacoustic clipper to input audio stream.
2841	2848

libavfilter/Makefile+1−0

@@ -84,6 +84,7 @@ OBJS-$(CONFIG_APAD_FILTER) += af_apad.o · 83 unmodified lines
84	84	OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
85	85	OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o
86	86	OBJS-$(CONFIG_APHASESHIFT_FILTER) += af_afreqshift.o
	87	+OBJS-$(CONFIG_APSNR_FILTER) += af_asdr.o
87	88	OBJS-$(CONFIG_APSYCLIP_FILTER) += af_apsyclip.o
88	89	OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
89	90	OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o

libavfilter/af_asdr.c+64−3

@@ -18,6 +18,8 @@ · 17 unmodified lines
18	18	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19	19	*/
20	20
	21	+#include <float.h>
	22	+
21	23	#include "libavutil/channel_layout.h"
22	24	#include "libavutil/common.h"
23	25
@@ -27,6 +29,8 @@ · 3 unmodified lines
27	29
28	30	typedef struct AudioSDRContext {
29	31	int channels;
	32	+ uint64_t nb_samples;
	33	+ double max;
30	34	double *sum_u;
31	35	double *sum_uv;
32	36
@@ -67,6 +71,34 @@ static int sdr_##name(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)\ · 34 unmodified lines
67	71	SDR_FILTER(fltp, float)
68	72	SDR_FILTER(dblp, double)
69	73
	74	+#define PSNR_FILTER(name, type) \
	75	+static int psnr_##name(AVFilterContext ctx, void arg, int jobnr,int nb_jobs)\
	76	+{ \
	77	+ AudioSDRContext *s = ctx->priv; \
	78	+ AVFrame *u = s->cache[0]; \
	79	+ AVFrame *v = s->cache[1]; \
	80	+ const int channels = u->ch_layout.nb_channels; \
	81	+ const int start = (channels * jobnr) / nb_jobs; \
	82	+ const int end = (channels * (jobnr+1)) / nb_jobs; \
	83	+ const int nb_samples = u->nb_samples; \
	84	+ \
	85	+ for (int ch = start; ch < end; ch++) { \
	86	+ const type const us = (type )u->extended_data[ch]; \
	87	+ const type const vs = (type )v->extended_data[ch]; \
	88	+ double sum_uv = 0.; \
	89	+ \
	90	+ for (int n = 0; n < nb_samples; n++) \
	91	+ sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]); \
	92	+ \
	93	+ s->sum_uv[ch] += sum_uv; \
	94	+ } \
	95	+ \
	96	+ return 0; \
	97	+}
	98	+
	99	+PSNR_FILTER(fltp, float)
	100	+PSNR_FILTER(dblp, double)
	101	+
70	102	static int activate(AVFilterContext *ctx)
71	103	{
72	104	AudioSDRContext *s = ctx->priv;
@@ -97,6 +129,7 @@ static int activate(AVFilterContext *ctx) · 24 unmodified lines
97	129	out = s->cache[0];
98	130	s->cache[0] = NULL;
99	131
	132	+ s->nb_samples += available;
100	133	return ff_filter_frame(outlink, out);
101	134	}
102	135
@@ -126,7 +159,12 @@ static int config_output(AVFilterLink *outlink) · 23 unmodified lines
126	159	AudioSDRContext *s = ctx->priv;
127	160
128	161	s->channels = inlink->ch_layout.nb_channels;
129		− s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
	162	+
	163	+ if (!strcmp(ctx->filter->name, "asdr"))
	164	+ s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? sdr_fltp : sdr_dblp;
	165	+ else
	166	+ s->filter = inlink->format == AV_SAMPLE_FMT_FLTP ? psnr_fltp : psnr_dblp;
	167	+ s->max = inlink->format == AV_SAMPLE_FMT_FLTP ? FLT_MAX : DBL_MAX;
130	168
131	169	s->sum_u = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->sum_u));
132	170	s->sum_uv = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->sum_uv));
@@ -140,8 +178,16 @@ static av_cold void uninit(AVFilterContext *ctx) · 7 unmodified lines
140	178	{
141	179	AudioSDRContext *s = ctx->priv;
142	180
143		− for (int ch = 0; ch < s->channels; ch++)
144		− av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 20. * log10(s->sum_u[ch] / s->sum_uv[ch]));
	181	+ if (!strcmp(ctx->filter->name, "asdr")) {
	182	+ for (int ch = 0; ch < s->channels; ch++)
	183	+ av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 20. * log10(s->sum_u[ch] / s->sum_uv[ch]));
	184	+ } else {
	185	+ for (int ch = 0; ch < s->channels; ch++) {
	186	+ double psnr = s->sum_uv[ch] > 0.0 ? 2.0 * log(s->max) - log(s->nb_samples / s->sum_uv[ch]) : INFINITY;
	187	+
	188	+ av_log(ctx, AV_LOG_INFO, "PSNR ch%d: %g dB\n", ch, psnr);
	189	+ }
	190	+ }
145	191
146	192	av_frame_free(&s->cache[0]);
147	193	av_frame_free(&s->cache[1]);
@@ -183,3 +229,18 @@ const AVFilter ff_af_asdr = { · 35 unmodified lines
183	229	FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
184	230	AV_SAMPLE_FMT_DBLP),
185	231	};
	232	+
	233	+const AVFilter ff_af_apsnr = {
	234	+ .name = "apsnr",
	235	+ .description = NULL_IF_CONFIG_SMALL("Measure Audio Peak Signal-to-Noise Ratio."),
	236	+ .priv_size = sizeof(AudioSDRContext),
	237	+ .activate = activate,
	238	+ .uninit = uninit,
	239	+ .flags = AVFILTER_FLAG_METADATA_ONLY \|
	240	+ AVFILTER_FLAG_SLICE_THREADS \|
	241	+ AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
	242	+ FILTER_INPUTS(inputs),
	243	+ FILTER_OUTPUTS(outputs),
	244	+ FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP,
	245	+ AV_SAMPLE_FMT_DBLP),
	246	+};

libavfilter/allfilters.c+1−0

@@ -70,6 +70,7 @@ extern const AVFilter ff_af_apad; · 69 unmodified lines
70	70	extern const AVFilter ff_af_aperms;
71	71	extern const AVFilter ff_af_aphaser;
72	72	extern const AVFilter ff_af_aphaseshift;
	73	+extern const AVFilter ff_af_apsnr;
73	74	extern const AVFilter ff_af_apsyclip;
74	75	extern const AVFilter ff_af_apulsator;
75	76	extern const AVFilter ff_af_arealtime;

avfilter: add apsnr filter

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