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authorPaul B Mahol <onemda@gmail.com>2017-01-26 17:03:08 +0100
committerPaul B Mahol <onemda@gmail.com>2017-05-09 20:47:52 +0200
commit49bbfb9d13936ee8bb7fee9983ca3710dc683a2e (patch)
treef132a0d6a8f1dc1b06e76725eca90fbfb248bc06 /libavfilter/af_afir.h
parentf1a4dd5e480932ee580fb686988599d46bb71637 (diff)
avfilter: add arbitrary audio FIR filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Diffstat (limited to 'libavfilter/af_afir.h')
-rw-r--r--libavfilter/af_afir.h83
1 files changed, 83 insertions, 0 deletions
diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
new file mode 100644
index 0000000000..7414f5438e
--- /dev/null
+++ b/libavfilter/af_afir.h
@@ -0,0 +1,83 @@
+/*
+ * 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/audio_fifo.h"
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "libavcodec/avfft.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+#define MAX_IR_DURATION 30
+
+typedef struct AudioFIRContext {
+ const AVClass *class;
+
+ float wet_gain;
+ float dry_gain;
+ float length;
+ int again;
+
+ float gain;
+
+ int eof_coeffs;
+ int have_coeffs;
+ int nb_coeffs;
+ int nb_taps;
+ int part_size;
+ int part_index;
+ int coeff_size;
+ int block_size;
+ int nb_partitions;
+ int nb_channels;
+ int ir_length;
+ int fft_length;
+ int nb_coef_channels;
+ int one2many;
+ int nb_samples;
+ int want_skip;
+ int need_padding;
+
+ RDFTContext **rdft, **irdft;
+ float **sum;
+ float **block;
+ FFTComplex **coeff;
+
+ AVAudioFifo *fifo[2];
+ AVFrame *in[2];
+ AVFrame *buffer;
+ int64_t pts;
+ int index;
+
+ AVFloatDSPContext *fdsp;
+ void (*fcmul_add)(float *sum, const float *t, const float *c,
+ ptrdiff_t len);
+} AudioFIRContext;
+
+void ff_afir_init_x86(AudioFIRContext *s);
+
+#endif /* AVFILTER_AFIR_H */