avfilter/af_afir: add support for double sample format

doc/filters.texi

@@ -1655,6 +1655,22 @@ Allowed range is from @var{1} to @var{32}. Default is @var{1}.
 Set IR stream which will be used for convolution, starting from @var{0}, should always be
 lower than supplied value by @code{nbirs} option. Default is @var{0}.
 This option can be changed at runtime via @ref{commands}.
+
+@item precision
+Set which precision to use when processing samples.
+
+@table @option
+@item auto
+Auto pick internal sample format depending on other filters.
+
+@item float
+Always use single-floating point precision sample format.
+
+@item double
+Always use double-floating point precision sample format.
+@end table
+
+Default value is auto.
 @end table
 
 @subsection Examples

libavfilter/af_afir.h

+/*
+ * Copyright (c) 2017 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
+ */
+
+#ifndef AVFILTER_AFIR_H
+#define AVFILTER_AFIR_H
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/frame.h"
+#include "libavutil/rational.h"
+#include "libavutil/tx.h"
+#include "avfilter.h"
+#include "af_afirdsp.h"
+
+typedef struct AudioFIRSegment {
+    int nb_partitions;
+    int part_size;
+    int block_size;
+    int fft_length;
+    int coeff_size;
+    int input_size;
+    int input_offset;
+
+    int *output_offset;
+    int *part_index;
+
+    AVFrame *sumin;
+    AVFrame *sumout;
+    AVFrame *blockin;
+    AVFrame *blockout;
+    AVFrame *buffer;
+    AVFrame *coeff;
+    AVFrame *input;
+    AVFrame *output;
+
+    AVTXContext **tx, **itx;
+    av_tx_fn tx_fn, itx_fn;
+} AudioFIRSegment;
+
+typedef struct AudioFIRContext {
+    const AVClass *class;
+
+    float wet_gain;
+    float dry_gain;
+    float length;
+    int gtype;
+    float ir_gain;
+    int ir_format;
+    float max_ir_len;
+    int response;
+    int w, h;
+    AVRational frame_rate;
+    int ir_channel;
+    int minp;
+    int maxp;
+    int nb_irs;
+    int selir;
+    int precision;
+    int format;
+
+    double gain;
+
+    int eof_coeffs[32];
+    int have_coeffs;
+    int nb_taps;
+    int nb_channels;
+    int nb_coef_channels;
+    int one2many;
+
+    AudioFIRSegment seg[1024];
+    int nb_segments;
+
+    AVFrame *in;
+    AVFrame *ir[32];
+    AVFrame *video;
+    int min_part_size;
+    int64_t pts;
+
+    AudioFIRDSPContext afirdsp;
+    AVFloatDSPContext *fdsp;
+} AudioFIRContext;
+
+#endif /* AVFILTER_AFIR_H */

libavfilter/af_afirdsp.h

@@ -29,6 +29,8 @@
 typedef struct AudioFIRDSPContext {
     void (*fcmul_add)(float *sum, const float *t, const float *c,
                       ptrdiff_t len);
+    void (*dcmul_add)(double *sum, const double *t, const double *c,
+                      ptrdiff_t len);
 } AudioFIRDSPContext;
 
 void ff_afir_init_x86(AudioFIRDSPContext *s);
@@ -50,9 +52,27 @@ static void fcmul_add_c(float *sum, const float *t, const float *c, ptrdiff_t le
     sum[2 * n] += t[2 * n] * c[2 * n];
 }
 
+static void dcmul_add_c(double *sum, const double *t, const double *c, ptrdiff_t len)
+{
+    int n;
+
+    for (n = 0; n < len; n++) {
+        const double cre = c[2 * n    ];
+        const double cim = c[2 * n + 1];
+        const double tre = t[2 * n    ];
+        const double tim = t[2 * n + 1];
+
+        sum[2 * n    ] += tre * cre - tim * cim;
+        sum[2 * n + 1] += tre * cim + tim * cre;
+    }
+
+    sum[2 * n] += t[2 * n] * c[2 * n];
+}
+
 static av_unused void ff_afir_init(AudioFIRDSPContext *dsp)
 {
     dsp->fcmul_add = fcmul_add_c;
+    dsp->dcmul_add = dcmul_add_c;
 
     if (ARCH_X86)
         ff_afir_init_x86(dsp);