avfilter/af_anlms: add double sample format support

Paul B Mahol committed 2 years ago

Commit 42e45ea1 parent ea6817d3 files · +194 −80Browse files

Showing the changes to libavfilter (2 of 3 files). Show all

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2 files

libavfilter/af_anlms.c +39 −80
libavfilter/anlms_template.c +141 −0

@@ -26,6 +26,7 @@ · 25 unmodified lines
26	26	#include "audio.h"
27	27	#include "avfilter.h"
28	28	#include "filters.h"
	29	+#include "formats.h"
29	30	#include "internal.h"
30	31
31	32	enum OutModes {
@@ -45,6 +46,7 @@ typedef struct AudioNLMSContext { · 13 unmodified lines
45	46	float eps;
46	47	float leakage;
47	48	int output_mode;
	49	+ int precision;
48	50
49	51	int kernel_size;
50	52	AVFrame *offset;
@@ -56,6 +58,8 @@ typedef struct AudioNLMSContext { · 5 unmodified lines
56	58
57	59	int anlmf;
58	60
	61	+ int (filter_channels)(AVFilterContext ctx, void *arg, int jobnr, int nb_jobs);
	62	+
59	63	AVFloatDSPContext *fdsp;
60	64	} AudioNLMSContext;
61	65
@@ -74,93 +78,32 @@ static const AVOption anlms_options[] = { · 12 unmodified lines
74	78	{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" },
75	79	{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" },
76	80	{ "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, "mode" },
	81	+ { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "precision" },
	82	+ { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "precision" },
	83	+ { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "precision" },
	84	+ { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "precision" },
77	85	{ NULL }
78	86	};
79	87
80	88	AVFILTER_DEFINE_CLASS_EXT(anlms, "anlm(f\|s)", anlms_options);
81	89
82		−static float fir_sample(AudioNLMSContext s, float sample, float delay,
83		− float coeffs, float tmp, int *offset)
	90	+static int query_formats(AVFilterContext *ctx)
84	91	{
85		− const int order = s->order;
86		− float output;
87		−
88		− delay[*offset] = sample;
89		−
90		− memcpy(tmp, coeffs + order - offset, order sizeof(float));
91		−
92		− output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
93		−
94		− if (--(*offset) < 0)
95		− *offset = order - 1;
96		−
97		− return output;
98		−}
99		−
100		−static float process_sample(AudioNLMSContext *s, float input, float desired,
101		− float delay, float coeffs, float tmp, int offsetp)
102		−{
103		− const int order = s->order;
104		− const float leakage = s->leakage;
105		− const float mu = s->mu;
106		− const float a = 1.f - leakage;
107		− float sum, output, e, norm, b;
108		− int offset = *offsetp;
109		−
110		− delay[offset + order] = input;
111		−
112		− output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
113		− e = desired - output;
114		−
115		− sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
116		−
117		− norm = s->eps + sum;
118		− b = mu * e / norm;
119		− if (s->anlmf)
120		− b = e e;
121		−
122		− memcpy(tmp, delay + offset, order * sizeof(float));
123		−
124		− s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
	92	+ AudioNLMSContext *s = ctx->priv;
	93	+ static const enum AVSampleFormat sample_fmts[3][3] = {
	94	+ { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
	95	+ { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
	96	+ { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
	97	+ };
	98	+ int ret;
125	99
126		− s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
	100	+ if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
	101	+ return ret;
127	102
128		− memcpy(coeffs + order, coeffs, order * sizeof(float));
	103	+ if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
	104	+ return ret;
129	105
130		− switch (s->output_mode) {
131		− case IN_MODE: output = input; break;
132		− case DESIRED_MODE: output = desired; break;
133		− case OUT_MODE: output = desired - output; break;
134		− case NOISE_MODE: output = input - output; break;
135		− case ERROR_MODE: break;
136		− }
137		− return output;
138		−}
139		−
140		−static int process_channels(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
141		−{
142		− AudioNLMSContext *s = ctx->priv;
143		− AVFrame *out = arg;
144		− const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
145		− const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
146		−
147		− for (int c = start; c < end; c++) {
148		− const float input = (const float )s->frame[0]->extended_data[c];
149		− const float desired = (const float )s->frame[1]->extended_data[c];
150		− float delay = (float )s->delay->extended_data[c];
151		− float coeffs = (float )s->coeffs->extended_data[c];
152		− float tmp = (float )s->tmp->extended_data[c];
153		− int offset = (int )s->offset->extended_data[c];
154		− float output = (float )out->extended_data[c];
155		−
156		− for (int n = 0; n < out->nb_samples; n++) {
157		− output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
158		− if (ctx->is_disabled)
159		− output[n] = input[n];
160		− }
161		− }
162		−
163		− return 0;
	106	+ return ff_set_common_all_samplerates(ctx);
164	107	}
165	108
166	109	static int activate(AVFilterContext *ctx)
@@ -195,7 +138,7 @@ static int activate(AVFilterContext *ctx) · 28 unmodified lines
195	138	return AVERROR(ENOMEM);
196	139	}
197	140
198		− ff_filter_execute(ctx, process_channels, out, NULL,
	141	+ ff_filter_execute(ctx, s->filter_channels, out, NULL,
199	142	FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
200	143
201	144	out->pts = s->frame[0]->pts;
@@ -228,6 +171,13 @@ static int activate(AVFilterContext *ctx) · 26 unmodified lines
228	171	return 0;
229	172	}
230	173
	174	+#define DEPTH 32
	175	+#include "anlms_template.c"
	176	+
	177	+#undef DEPTH
	178	+#define DEPTH 64
	179	+#include "anlms_template.c"
	180	+
231	181	static int config_output(AVFilterLink *outlink)
232	182	{
233	183	AVFilterContext *ctx = outlink->src;
@@ -247,6 +197,15 @@ static int config_output(AVFilterLink *outlink) · 13 unmodified lines
247	197	if (!s->delay \|\| !s->coeffs \|\| !s->offset \|\| !s->tmp)
248	198	return AVERROR(ENOMEM);
249	199
	200	+ switch (outlink->format) {
	201	+ case AV_SAMPLE_FMT_DBLP:
	202	+ s->filter_channels = filter_channels_double;
	203	+ break;
	204	+ case AV_SAMPLE_FMT_FLTP:
	205	+ s->filter_channels = filter_channels_float;
	206	+ break;
	207	+ }
	208	+
250	209	return 0;
251	210	}
252	211
@@ -317,7 +276,7 @@ const AVFilter ff_af_anlmf = { · 64 unmodified lines
317	276	.activate = activate,
318	277	FILTER_INPUTS(inputs),
319	278	FILTER_OUTPUTS(outputs),
320		− FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
	279	+ FILTER_QUERY_FUNC(query_formats),
321	280	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL \|
322	281	AVFILTER_FLAG_SLICE_THREADS,
323	282	.process_command = ff_filter_process_command,

libavfilter/anlms_template.c+141−0

@@ -0,0 +1,141 @@
	1	+/*
	2	+ * This file is part of FFmpeg.
	3	+ *
	4	+ * FFmpeg is free software; you can redistribute it and/or
	5	+ * modify it under the terms of the GNU Lesser General Public
	6	+ * License as published by the Free Software Foundation; either
	7	+ * version 2.1 of the License, or (at your option) any later version.
	8	+ *
	9	+ * FFmpeg is distributed in the hope that it will be useful,
	10	+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
	11	+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	12	+ * Lesser General Public License for more details.
	13	+ *
	14	+ * You should have received a copy of the GNU Lesser General Public
	15	+ * License along with FFmpeg; if not, write to the Free Software
	16	+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	17	+ */
	18	+
	19	+#undef ONE
	20	+#undef ftype
	21	+#undef SAMPLE_FORMAT
	22	+#if DEPTH == 32
	23	+#define SAMPLE_FORMAT float
	24	+#define ftype float
	25	+#define ONE 1.f
	26	+#else
	27	+#define SAMPLE_FORMAT double
	28	+#define ftype double
	29	+#define ONE 1.0
	30	+#endif
	31	+
	32	+#define fn3(a,b) a##_##b
	33	+#define fn2(a,b) fn3(a,b)
	34	+#define fn(a) fn2(a, SAMPLE_FORMAT)
	35	+
	36	+#if DEPTH == 64
	37	+static double scalarproduct_double(const double v1, const double v2, int len)
	38	+{
	39	+ double p = 0.0;
	40	+
	41	+ for (int i = 0; i < len; i++)
	42	+ p += v1[i] * v2[i];
	43	+
	44	+ return p;
	45	+}
	46	+#endif
	47	+
	48	+static ftype fn(fir_sample)(AudioNLMSContext s, ftype sample, ftype delay,
	49	+ ftype coeffs, ftype tmp, int *offset)
	50	+{
	51	+ const int order = s->order;
	52	+ ftype output;
	53	+
	54	+ delay[*offset] = sample;
	55	+
	56	+ memcpy(tmp, coeffs + order - offset, order sizeof(ftype));
	57	+
	58	+#if DEPTH == 32
	59	+ output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
	60	+#else
	61	+ output = scalarproduct_double(delay, tmp, s->kernel_size);
	62	+#endif
	63	+
	64	+ if (--(*offset) < 0)
	65	+ *offset = order - 1;
	66	+
	67	+ return output;
	68	+}
	69	+
	70	+static ftype fn(process_sample)(AudioNLMSContext *s, ftype input, ftype desired,
	71	+ ftype delay, ftype coeffs, ftype tmp, int offsetp)
	72	+{
	73	+ const int order = s->order;
	74	+ const ftype leakage = s->leakage;
	75	+ const ftype mu = s->mu;
	76	+ const ftype a = ONE - leakage;
	77	+ ftype sum, output, e, norm, b;
	78	+ int offset = *offsetp;
	79	+
	80	+ delay[offset + order] = input;
	81	+
	82	+ output = fn(fir_sample)(s, input, delay, coeffs, tmp, offsetp);
	83	+ e = desired - output;
	84	+
	85	+#if DEPTH == 32
	86	+ sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
	87	+#else
	88	+ sum = scalarproduct_double(delay, delay, s->kernel_size);
	89	+#endif
	90	+ norm = s->eps + sum;
	91	+ b = mu * e / norm;
	92	+ if (s->anlmf)
	93	+ b = e e;
	94	+
	95	+ memcpy(tmp, delay + offset, order * sizeof(ftype));
	96	+
	97	+#if DEPTH == 32
	98	+ s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
	99	+ s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
	100	+#else
	101	+ s->fdsp->vector_dmul_scalar(coeffs, coeffs, a, s->kernel_size);
	102	+ s->fdsp->vector_dmac_scalar(coeffs, tmp, b, s->kernel_size);
	103	+#endif
	104	+
	105	+ memcpy(coeffs + order, coeffs, order * sizeof(ftype));
	106	+
	107	+ switch (s->output_mode) {
	108	+ case IN_MODE: output = input; break;
	109	+ case DESIRED_MODE: output = desired; break;
	110	+ case OUT_MODE: output = desired - output; break;
	111	+ case NOISE_MODE: output = input - output; break;
	112	+ case ERROR_MODE: break;
	113	+ }
	114	+ return output;
	115	+}
	116	+
	117	+static int fn(filter_channels)(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
	118	+{
	119	+ AudioNLMSContext *s = ctx->priv;
	120	+ AVFrame *out = arg;
	121	+ const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
	122	+ const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
	123	+
	124	+ for (int c = start; c < end; c++) {
	125	+ const ftype input = (const ftype )s->frame[0]->extended_data[c];
	126	+ const ftype desired = (const ftype )s->frame[1]->extended_data[c];
	127	+ ftype delay = (ftype )s->delay->extended_data[c];
	128	+ ftype coeffs = (ftype )s->coeffs->extended_data[c];
	129	+ ftype tmp = (ftype )s->tmp->extended_data[c];
	130	+ int offset = (int )s->offset->extended_data[c];
	131	+ ftype output = (ftype )out->extended_data[c];
	132	+
	133	+ for (int n = 0; n < out->nb_samples; n++) {
	134	+ output[n] = fn(process_sample)(s, input[n], desired[n], delay, coeffs, tmp, offset);
	135	+ if (ctx->is_disabled)
	136	+ output[n] = input[n];
	137	+ }
	138	+ }
	139	+
	140	+ return 0;
	141	+}

avfilter/af_anlms: add double sample format support

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