[FFmpeg-cvslog] avfilter: add firequalizer filter
Muhammad Faiz
git at videolan.org
Mon Feb 22 18:49:27 CET 2016
ffmpeg | branch: master | Muhammad Faiz <mfcc64 at gmail.com> | Wed Feb 17 01:02:22 2016 +0700| [bfc61b0fcc77701921a1a026d308db518396fed1] | committer: Muhammad Faiz
avfilter: add firequalizer filter
Reviewed-by: Paul B Mahol <onemda at gmail.com>
Signed-off-by: Muhammad Faiz <mfcc64 at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bfc61b0fcc77701921a1a026d308db518396fed1
---
Changelog | 1 +
MAINTAINERS | 1 +
configure | 2 +
doc/filters.texi | 109 ++++++++
libavfilter/Makefile | 1 +
libavfilter/af_firequalizer.c | 592 +++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
8 files changed, 708 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 7d672fb..fa4400c 100644
--- a/Changelog
+++ b/Changelog
@@ -6,6 +6,7 @@ version <next>:
- fieldhint filter
- loop video filter and aloop audio filter
- Bob Weaver deinterlacing filter
+- firequalizer filter
version 3.0:
diff --git a/MAINTAINERS b/MAINTAINERS
index 0705a69..f518aed 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -353,6 +353,7 @@ Filters:
af_biquads.c Paul B Mahol
af_chorus.c Paul B Mahol
af_compand.c Paul B Mahol
+ af_firequalizer.c Muhammad Faiz
af_ladspa.c Paul B Mahol
af_pan.c Nicolas George
af_sidechaincompress.c Paul B Mahol
diff --git a/configure b/configure
index 6b3ee5f..3d1ee49 100755
--- a/configure
+++ b/configure
@@ -2861,6 +2861,8 @@ eq_filter_deps="gpl"
fftfilt_filter_deps="avcodec"
fftfilt_filter_select="rdft"
find_rect_filter_deps="avcodec avformat gpl"
+firequalizer_filter_deps="avcodec"
+firequalizer_filter_select="rdft"
flite_filter_deps="libflite"
frei0r_filter_deps="frei0r dlopen"
frei0r_src_filter_deps="frei0r dlopen"
diff --git a/doc/filters.texi b/doc/filters.texi
index 250367e..aca7663 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2366,6 +2366,115 @@ Sets the difference coefficient (default: 2.5). 0.0 means mono sound
Enable clipping. By default is enabled.
@end table
+ at section firequalizer
+Apply FIR Equalization using arbitrary frequency response.
+
+The filter accepts the following option:
+
+ at table @option
+ at item gain
+Set gain curve equation (in dB). The expression can contain variables:
+ at table @option
+ at item f
+the evaluated frequency
+ at item sr
+sample rate
+ at item ch
+channel number, set to 0 when multichannels evaluation is disabled
+ at item chid
+channel id, see libavutil/channel_layout.h, set to the first channel id when
+multichannels evaluation is disabled
+ at item chs
+number of channels
+ at item chlayout
+channel_layout, see libavutil/channel_layout.h
+
+ at end table
+and functions:
+ at table @option
+ at item gain_interpolate(f)
+interpolate gain on frequency f based on gain_entry
+ at end table
+This option is also available as command. Default is @code{gain_interpolate(f)}.
+
+ at item gain_entry
+Set gain entry for gain_interpolate function. The expression can
+contain functions:
+ at table @option
+ at item entry(f, g)
+store gain entry at frequency f with value g
+ at end table
+This option is also available as command.
+
+ at item delay
+Set filter delay in seconds. Higher value means more accurate.
+Default is @code{0.01}.
+
+ at item accuracy
+Set filter accuracy in Hz. Lower value means more accurate.
+Default is @code{5}.
+
+ at item wfunc
+Set window function. Acceptable values are:
+ at table @option
+ at item rectangular
+rectangular window, useful when gain curve is already smooth
+ at item hann
+hann window (default)
+ at item hamming
+hamming window
+ at item blackman
+blackman window
+ at item nuttall3
+3-terms continuous 1st derivative nuttall window
+ at item mnuttall3
+minimum 3-terms discontinuous nuttall window
+ at item nuttall
+4-terms continuous 1st derivative nuttall window
+ at item bnuttall
+minimum 4-terms discontinuous nuttall (blackman-nuttall) window
+ at item bharris
+blackman-harris window
+ at end table
+
+ at item fixed
+If enabled, use fixed number of audio samples. This improves speed when
+filtering with large delay. Default is disabled.
+
+ at item multi
+Enable multichannels evaluation on gain. Default is disabled.
+ at end table
+
+ at subsection Examples
+ at itemize
+ at item
+lowpass at 1000 Hz:
+ at example
+firequalizer=gain='if(lt(f,1000), 0, -INF)'
+ at end example
+ at item
+lowpass at 1000 Hz with gain_entry:
+ at example
+firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
+ at end example
+ at item
+custom equalization:
+ at example
+firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
+ at end example
+ at item
+higher delay:
+ at example
+firequalizer=delay=0.1:fixed=on
+ at end example
+ at item
+lowpass on left channel, highpass on right channel:
+ at example
+firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
+:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
+ at end example
+ at end itemize
+
@section flanger
Apply a flanging effect to the audio.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 10c2e0b..a8469fd 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -80,6 +80,7 @@ OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o
OBJS-$(CONFIG_EXTRASTEREO_FILTER) += af_extrastereo.o
+OBJS-$(CONFIG_FIREQUALIZER_FILTER) += af_firequalizer.o
OBJS-$(CONFIG_FLANGER_FILTER) += af_flanger.o generate_wave_table.o
OBJS-$(CONFIG_HIGHPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_JOIN_FILTER) += af_join.o
diff --git a/libavfilter/af_firequalizer.c b/libavfilter/af_firequalizer.c
new file mode 100644
index 0000000..4c9d95e
--- /dev/null
+++ b/libavfilter/af_firequalizer.c
@@ -0,0 +1,592 @@
+/*
+ * Copyright (c) 2016 Muhammad Faiz <mfcc64 at gmail.com>
+ *
+ * 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
+ */
+
+#include "libavutil/opt.h"
+#include "libavutil/eval.h"
+#include "libavutil/avassert.h"
+#include "libavcodec/avfft.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+#define RDFT_BITS_MIN 4
+#define RDFT_BITS_MAX 16
+
+enum WindowFunc {
+ WFUNC_MIN,
+ WFUNC_RECTANGULAR = WFUNC_MIN,
+ WFUNC_HANN,
+ WFUNC_HAMMING,
+ WFUNC_BLACKMAN,
+ WFUNC_NUTTALL3,
+ WFUNC_MNUTTALL3,
+ WFUNC_NUTTALL,
+ WFUNC_BNUTTALL,
+ WFUNC_BHARRIS,
+ WFUNC_MAX = WFUNC_BHARRIS
+};
+
+#define NB_GAIN_ENTRY_MAX 4096
+typedef struct {
+ double freq;
+ double gain;
+} GainEntry;
+
+typedef struct {
+ int buf_idx;
+ int overlap_idx;
+} OverlapIndex;
+
+typedef struct {
+ const AVClass *class;
+
+ RDFTContext *analysis_irdft;
+ RDFTContext *rdft;
+ RDFTContext *irdft;
+ int analysis_rdft_len;
+ int rdft_len;
+
+ float *analysis_buf;
+ float *kernel_tmp_buf;
+ float *kernel_buf;
+ float *conv_buf;
+ OverlapIndex *conv_idx;
+ int fir_len;
+ int nsamples_max;
+ int64_t next_pts;
+ int frame_nsamples_max;
+ int remaining;
+
+ char *gain_cmd;
+ char *gain_entry_cmd;
+ const char *gain;
+ const char *gain_entry;
+ double delay;
+ double accuracy;
+ int wfunc;
+ int fixed;
+ int multi;
+
+ int nb_gain_entry;
+ int gain_entry_err;
+ GainEntry gain_entry_tbl[NB_GAIN_ENTRY_MAX];
+} FIREqualizerContext;
+
+#define OFFSET(x) offsetof(FIREqualizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption firequalizer_options[] = {
+ { "gain", "set gain curve", OFFSET(gain), AV_OPT_TYPE_STRING, { .str = "gain_interpolate(f)" }, 0, 0, FLAGS },
+ { "gain_entry", "set gain entry", OFFSET(gain_entry), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, FLAGS },
+ { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.0, 1e10, FLAGS },
+ { "accuracy", "set accuracy", OFFSET(accuracy), AV_OPT_TYPE_DOUBLE, { .dbl = 5.0 }, 0.0, 1e10, FLAGS },
+ { "wfunc", "set window function", OFFSET(wfunc), AV_OPT_TYPE_INT, { .i64 = WFUNC_HANN }, WFUNC_MIN, WFUNC_MAX, FLAGS, "wfunc" },
+ { "rectangular", "rectangular window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_RECTANGULAR }, 0, 0, FLAGS, "wfunc" },
+ { "hann", "hann window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_HANN }, 0, 0, FLAGS, "wfunc" },
+ { "hamming", "hamming window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_HAMMING }, 0, 0, FLAGS, "wfunc" },
+ { "blackman", "blackman window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BLACKMAN }, 0, 0, FLAGS, "wfunc" },
+ { "nuttall3", "3-term nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_NUTTALL3 }, 0, 0, FLAGS, "wfunc" },
+ { "mnuttall3", "minimum 3-term nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_MNUTTALL3 }, 0, 0, FLAGS, "wfunc" },
+ { "nuttall", "nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_NUTTALL }, 0, 0, FLAGS, "wfunc" },
+ { "bnuttall", "blackman-nuttall window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BNUTTALL }, 0, 0, FLAGS, "wfunc" },
+ { "bharris", "blackman-harris window", 0, AV_OPT_TYPE_CONST, { .i64 = WFUNC_BHARRIS }, 0, 0, FLAGS, "wfunc" },
+ { "fixed", "set fixed frame samples", OFFSET(fixed), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
+ { "multi", "set multi channels mode", OFFSET(multi), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(firequalizer);
+
+static void common_uninit(FIREqualizerContext *s)
+{
+ av_rdft_end(s->analysis_irdft);
+ av_rdft_end(s->rdft);
+ av_rdft_end(s->irdft);
+ s->analysis_irdft = s->rdft = s->irdft = NULL;
+
+ av_freep(&s->analysis_buf);
+ av_freep(&s->kernel_tmp_buf);
+ av_freep(&s->kernel_buf);
+ av_freep(&s->conv_buf);
+ av_freep(&s->conv_idx);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ FIREqualizerContext *s = ctx->priv;
+
+ common_uninit(s);
+ av_freep(&s->gain_cmd);
+ av_freep(&s->gain_entry_cmd);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts;
+ AVFilterFormats *formats;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static void fast_convolute(FIREqualizerContext *s, const float *kernel_buf, float *conv_buf,
+ OverlapIndex *idx, float *data, int nsamples)
+{
+ if (nsamples <= s->nsamples_max) {
+ float *buf = conv_buf + idx->buf_idx * s->rdft_len;
+ float *obuf = conv_buf + !idx->buf_idx * s->rdft_len + idx->overlap_idx;
+ int k;
+
+ memcpy(buf, data, nsamples * sizeof(*data));
+ memset(buf + nsamples, 0, (s->rdft_len - nsamples) * sizeof(*data));
+ av_rdft_calc(s->rdft, buf);
+
+ buf[0] *= kernel_buf[0];
+ buf[1] *= kernel_buf[1];
+ for (k = 2; k < s->rdft_len; k += 2) {
+ float re, im;
+ re = buf[k] * kernel_buf[k] - buf[k+1] * kernel_buf[k+1];
+ im = buf[k] * kernel_buf[k+1] + buf[k+1] * kernel_buf[k];
+ buf[k] = re;
+ buf[k+1] = im;
+ }
+
+ av_rdft_calc(s->irdft, buf);
+ for (k = 0; k < s->rdft_len - idx->overlap_idx; k++)
+ buf[k] += obuf[k];
+ memcpy(data, buf, nsamples * sizeof(*data));
+ idx->buf_idx = !idx->buf_idx;
+ idx->overlap_idx = nsamples;
+ } else {
+ while (nsamples > s->nsamples_max * 2) {
+ fast_convolute(s, kernel_buf, conv_buf, idx, data, s->nsamples_max);
+ data += s->nsamples_max;
+ nsamples -= s->nsamples_max;
+ }
+ fast_convolute(s, kernel_buf, conv_buf, idx, data, nsamples/2);
+ fast_convolute(s, kernel_buf, conv_buf, idx, data + nsamples/2, nsamples - nsamples/2);
+ }
+}
+
+static double entry_func(void *p, double freq, double gain)
+{
+ AVFilterContext *ctx = p;
+ FIREqualizerContext *s = ctx->priv;
+
+ if (s->nb_gain_entry >= NB_GAIN_ENTRY_MAX) {
+ av_log(ctx, AV_LOG_ERROR, "entry table overflow.\n");
+ s->gain_entry_err = AVERROR(EINVAL);
+ return 0;
+ }
+
+ if (isnan(freq)) {
+ av_log(ctx, AV_LOG_ERROR, "nan frequency (%g, %g).\n", freq, gain);
+ s->gain_entry_err = AVERROR(EINVAL);
+ return 0;
+ }
+
+ if (s->nb_gain_entry > 0 && freq <= s->gain_entry_tbl[s->nb_gain_entry - 1].freq) {
+ av_log(ctx, AV_LOG_ERROR, "unsorted frequency (%g, %g).\n", freq, gain);
+ s->gain_entry_err = AVERROR(EINVAL);
+ return 0;
+ }
+
+ s->gain_entry_tbl[s->nb_gain_entry].freq = freq;
+ s->gain_entry_tbl[s->nb_gain_entry].gain = gain;
+ s->nb_gain_entry++;
+ return 0;
+}
+
+static int gain_entry_compare(const void *key, const void *memb)
+{
+ const double *freq = key;
+ const GainEntry *entry = memb;
+
+ if (*freq < entry[0].freq)
+ return -1;
+ if (*freq > entry[1].freq)
+ return 1;
+ return 0;
+}
+
+static double gain_interpolate_func(void *p, double freq)
+{
+ AVFilterContext *ctx = p;
+ FIREqualizerContext *s = ctx->priv;
+ GainEntry *res;
+ double d0, d1, d;
+
+ if (isnan(freq))
+ return freq;
+
+ if (!s->nb_gain_entry)
+ return 0;
+
+ if (freq <= s->gain_entry_tbl[0].freq)
+ return s->gain_entry_tbl[0].gain;
+
+ if (freq >= s->gain_entry_tbl[s->nb_gain_entry-1].freq)
+ return s->gain_entry_tbl[s->nb_gain_entry-1].gain;
+
+ res = bsearch(&freq, &s->gain_entry_tbl, s->nb_gain_entry - 1, sizeof(*res), gain_entry_compare);
+ av_assert0(res);
+
+ d = res[1].freq - res[0].freq;
+ d0 = freq - res[0].freq;
+ d1 = res[1].freq - freq;
+
+ if (d0 && d1)
+ return (d0 * res[1].gain + d1 * res[0].gain) / d;
+
+ if (d0)
+ return res[1].gain;
+
+ return res[0].gain;
+}
+
+static const char *const var_names[] = {
+ "f",
+ "sr",
+ "ch",
+ "chid",
+ "chs",
+ "chlayout",
+ NULL
+};
+
+enum VarOffset {
+ VAR_F,
+ VAR_SR,
+ VAR_CH,
+ VAR_CHID,
+ VAR_CHS,
+ VAR_CHLAYOUT,
+ VAR_NB
+};
+
+static int generate_kernel(AVFilterContext *ctx, const char *gain, const char *gain_entry)
+{
+ FIREqualizerContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
+ const char *gain_entry_func_names[] = { "entry", NULL };
+ const char *gain_func_names[] = { "gain_interpolate", NULL };
+ double (*gain_entry_funcs[])(void *, double, double) = { entry_func, NULL };
+ double (*gain_funcs[])(void *, double) = { gain_interpolate_func, NULL };
+ double vars[VAR_NB];
+ AVExpr *gain_expr;
+ int ret, k, center, ch;
+
+ s->nb_gain_entry = 0;
+ s->gain_entry_err = 0;
+ if (gain_entry) {
+ double result = 0.0;
+ ret = av_expr_parse_and_eval(&result, gain_entry, NULL, NULL, NULL, NULL,
+ gain_entry_func_names, gain_entry_funcs, ctx, 0, ctx);
+ if (ret < 0)
+ return ret;
+ if (s->gain_entry_err < 0)
+ return s->gain_entry_err;
+ }
+
+ av_log(ctx, AV_LOG_DEBUG, "nb_gain_entry = %d.\n", s->nb_gain_entry);
+
+ ret = av_expr_parse(&gain_expr, gain, var_names,
+ gain_func_names, gain_funcs, NULL, NULL, 0, ctx);
+ if (ret < 0)
+ return ret;
+
+ vars[VAR_CHS] = inlink->channels;
+ vars[VAR_CHLAYOUT] = inlink->channel_layout;
+ vars[VAR_SR] = inlink->sample_rate;
+ for (ch = 0; ch < inlink->channels; ch++) {
+ vars[VAR_CH] = ch;
+ vars[VAR_CHID] = av_channel_layout_extract_channel(inlink->channel_layout, ch);
+ vars[VAR_F] = 0.0;
+ s->analysis_buf[0] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
+ vars[VAR_F] = 0.5 * inlink->sample_rate;
+ s->analysis_buf[1] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
+
+ for (k = 1; k < s->analysis_rdft_len/2; k++) {
+ vars[VAR_F] = k * ((double)inlink->sample_rate /(double)s->analysis_rdft_len);
+ s->analysis_buf[2*k] = pow(10.0, 0.05 * av_expr_eval(gain_expr, vars, ctx));
+ s->analysis_buf[2*k+1] = 0.0;
+ }
+
+ av_rdft_calc(s->analysis_irdft, s->analysis_buf);
+ center = s->fir_len / 2;
+
+ for (k = 0; k <= center; k++) {
+ double u = k * (M_PI/center);
+ double win;
+ switch (s->wfunc) {
+ case WFUNC_RECTANGULAR:
+ win = 1.0;
+ break;
+ case WFUNC_HANN:
+ win = 0.5 + 0.5 * cos(u);
+ break;
+ case WFUNC_HAMMING:
+ win = 0.53836 + 0.46164 * cos(u);
+ break;
+ case WFUNC_BLACKMAN:
+ win = 0.48 + 0.5 * cos(u) + 0.02 * cos(2*u);
+ break;
+ case WFUNC_NUTTALL3:
+ win = 0.40897 + 0.5 * cos(u) + 0.09103 * cos(2*u);
+ break;
+ case WFUNC_MNUTTALL3:
+ win = 0.4243801 + 0.4973406 * cos(u) + 0.0782793 * cos(2*u);
+ break;
+ case WFUNC_NUTTALL:
+ win = 0.355768 + 0.487396 * cos(u) + 0.144232 * cos(2*u) + 0.012604 * cos(3*u);
+ break;
+ case WFUNC_BNUTTALL:
+ win = 0.3635819 + 0.4891775 * cos(u) + 0.1365995 * cos(2*u) + 0.0106411 * cos(3*u);
+ break;
+ case WFUNC_BHARRIS:
+ win = 0.35875 + 0.48829 * cos(u) + 0.14128 * cos(2*u) + 0.01168 * cos(3*u);
+ break;
+ default:
+ av_assert0(0);
+ }
+ s->analysis_buf[k] *= (2.0/s->analysis_rdft_len) * (2.0/s->rdft_len) * win;
+ }
+
+ for (k = 0; k < center - k; k++) {
+ float tmp = s->analysis_buf[k];
+ s->analysis_buf[k] = s->analysis_buf[center - k];
+ s->analysis_buf[center - k] = tmp;
+ }
+
+ for (k = 1; k <= center; k++)
+ s->analysis_buf[center + k] = s->analysis_buf[center - k];
+
+ memset(s->analysis_buf + s->fir_len, 0, (s->rdft_len - s->fir_len) * sizeof(*s->analysis_buf));
+ av_rdft_calc(s->rdft, s->analysis_buf);
+
+ for (k = 0; k < s->rdft_len; k++) {
+ if (isnan(s->analysis_buf[k]) || isinf(s->analysis_buf[k])) {
+ av_log(ctx, AV_LOG_ERROR, "filter kernel contains nan or infinity.\n");
+ av_expr_free(gain_expr);
+ return AVERROR(EINVAL);
+ }
+ }
+
+ memcpy(s->kernel_tmp_buf + ch * s->rdft_len, s->analysis_buf, s->rdft_len * sizeof(*s->analysis_buf));
+ if (!s->multi)
+ break;
+ }
+
+ memcpy(s->kernel_buf, s->kernel_tmp_buf, (s->multi ? inlink->channels : 1) * s->rdft_len * sizeof(*s->kernel_buf));
+ av_expr_free(gain_expr);
+ return 0;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ FIREqualizerContext *s = ctx->priv;
+ int rdft_bits;
+
+ common_uninit(s);
+
+ s->next_pts = 0;
+ s->frame_nsamples_max = 0;
+
+ s->fir_len = FFMAX(2 * (int)(inlink->sample_rate * s->delay) + 1, 3);
+ s->remaining = s->fir_len - 1;
+
+ for (rdft_bits = RDFT_BITS_MIN; rdft_bits <= RDFT_BITS_MAX; rdft_bits++) {
+ s->rdft_len = 1 << rdft_bits;
+ s->nsamples_max = s->rdft_len - s->fir_len + 1;
+ if (s->nsamples_max * 2 >= s->fir_len)
+ break;
+ }
+
+ if (rdft_bits > RDFT_BITS_MAX) {
+ av_log(ctx, AV_LOG_ERROR, "too large delay, please decrease it.\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (!(s->rdft = av_rdft_init(rdft_bits, DFT_R2C)) || !(s->irdft = av_rdft_init(rdft_bits, IDFT_C2R)))
+ return AVERROR(ENOMEM);
+
+ for ( ; rdft_bits <= RDFT_BITS_MAX; rdft_bits++) {
+ s->analysis_rdft_len = 1 << rdft_bits;
+ if (inlink->sample_rate <= s->accuracy * s->analysis_rdft_len)
+ break;
+ }
+
+ if (rdft_bits > RDFT_BITS_MAX) {
+ av_log(ctx, AV_LOG_ERROR, "too small accuracy, please increase it.\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (!(s->analysis_irdft = av_rdft_init(rdft_bits, IDFT_C2R)))
+ return AVERROR(ENOMEM);
+
+ s->analysis_buf = av_malloc_array(s->analysis_rdft_len, sizeof(*s->analysis_buf));
+ s->kernel_tmp_buf = av_malloc_array(s->rdft_len * (s->multi ? inlink->channels : 1), sizeof(*s->kernel_tmp_buf));
+ s->kernel_buf = av_malloc_array(s->rdft_len * (s->multi ? inlink->channels : 1), sizeof(*s->kernel_buf));
+ s->conv_buf = av_calloc(2 * s->rdft_len * inlink->channels, sizeof(*s->conv_buf));
+ s->conv_idx = av_calloc(inlink->channels, sizeof(*s->conv_idx));
+ if (!s->analysis_buf || !s->kernel_tmp_buf || !s->kernel_buf || !s->conv_buf || !s->conv_idx)
+ return AVERROR(ENOMEM);
+
+ av_log(ctx, AV_LOG_DEBUG, "sample_rate = %d, channels = %d, analysis_rdft_len = %d, rdft_len = %d, fir_len = %d, nsamples_max = %d.\n",
+ inlink->sample_rate, inlink->channels, s->analysis_rdft_len, s->rdft_len, s->fir_len, s->nsamples_max);
+
+ if (s->fixed)
+ inlink->min_samples = inlink->max_samples = inlink->partial_buf_size = s->nsamples_max;
+
+ return generate_kernel(ctx, s->gain_cmd ? s->gain_cmd : s->gain,
+ s->gain_entry_cmd ? s->gain_entry_cmd : s->gain_entry);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ FIREqualizerContext *s = ctx->priv;
+ int ch;
+
+ for (ch = 0; ch < inlink->channels; ch++) {
+ fast_convolute(s, s->kernel_buf + (s->multi ? ch * s->rdft_len : 0),
+ s->conv_buf + 2 * ch * s->rdft_len, s->conv_idx + ch,
+ (float *) frame->extended_data[ch], frame->nb_samples);
+ }
+
+ s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, av_make_q(1, inlink->sample_rate), inlink->time_base);
+ s->frame_nsamples_max = FFMAX(s->frame_nsamples_max, frame->nb_samples);
+ return ff_filter_frame(ctx->outputs[0], frame);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ FIREqualizerContext *s= ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+ if (ret == AVERROR_EOF && s->remaining > 0 && s->frame_nsamples_max > 0) {
+ AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(s->remaining, s->frame_nsamples_max));
+
+ if (!frame)
+ return AVERROR(ENOMEM);
+
+ av_samples_set_silence(frame->extended_data, 0, frame->nb_samples, outlink->channels, frame->format);
+ frame->pts = s->next_pts;
+ s->remaining -= frame->nb_samples;
+ ret = filter_frame(ctx->inputs[0], frame);
+ }
+
+ return ret;
+}
+
+static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
+ char *res, int res_len, int flags)
+{
+ FIREqualizerContext *s = ctx->priv;
+ int ret = AVERROR(ENOSYS);
+
+ if (!strcmp(cmd, "gain")) {
+ char *gain_cmd;
+
+ gain_cmd = av_strdup(args);
+ if (!gain_cmd)
+ return AVERROR(ENOMEM);
+
+ ret = generate_kernel(ctx, gain_cmd, s->gain_entry_cmd ? s->gain_entry_cmd : s->gain_entry);
+ if (ret >= 0) {
+ av_freep(&s->gain_cmd);
+ s->gain_cmd = gain_cmd;
+ } else {
+ av_freep(&gain_cmd);
+ }
+ } else if (!strcmp(cmd, "gain_entry")) {
+ char *gain_entry_cmd;
+
+ gain_entry_cmd = av_strdup(args);
+ if (!gain_entry_cmd)
+ return AVERROR(ENOMEM);
+
+ ret = generate_kernel(ctx, s->gain_cmd ? s->gain_cmd : s->gain, gain_entry_cmd);
+ if (ret >= 0) {
+ av_freep(&s->gain_entry_cmd);
+ s->gain_entry_cmd = gain_entry_cmd;
+ } else {
+ av_freep(&gain_entry_cmd);
+ }
+ }
+
+ return ret;
+}
+
+static const AVFilterPad firequalizer_inputs[] = {
+ {
+ .name = "default",
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ .type = AVMEDIA_TYPE_AUDIO,
+ .needs_writable = 1,
+ },
+ { NULL }
+};
+
+static const AVFilterPad firequalizer_outputs[] = {
+ {
+ .name = "default",
+ .request_frame = request_frame,
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_firequalizer = {
+ .name = "firequalizer",
+ .description = NULL_IF_CONFIG_SMALL("Finite Impulse Response Equalizer"),
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .process_command = process_command,
+ .priv_size = sizeof(FIREqualizerContext),
+ .inputs = firequalizer_inputs,
+ .outputs = firequalizer_outputs,
+ .priv_class = &firequalizer_class,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ed52649..3163831 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -101,6 +101,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);
REGISTER_FILTER(EXTRASTEREO, extrastereo, af);
+ REGISTER_FILTER(FIREQUALIZER, firequalizer, af);
REGISTER_FILTER(FLANGER, flanger, af);
REGISTER_FILTER(HIGHPASS, highpass, af);
REGISTER_FILTER(JOIN, join, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 4a462e7..480c464 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
-#define LIBAVFILTER_VERSION_MINOR 34
+#define LIBAVFILTER_VERSION_MINOR 35
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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