[FFmpeg-cvslog] avfilter: add afirsrc filter
Paul B Mahol
git at videolan.org
Fri Feb 7 18:10:17 EET 2020
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Fri Jan 31 18:07:35 2020 +0100| [ae5a43530036a90229df0d4578b6d2c306918b54] | committer: Paul B Mahol
avfilter: add afirsrc filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ae5a43530036a90229df0d4578b6d2c306918b54
---
Changelog | 1 +
doc/filters.texi | 38 ++++++
libavfilter/Makefile | 1 +
libavfilter/allfilters.c | 1 +
libavfilter/asrc_afirsrc.c | 330 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/version.h | 2 +-
6 files changed, 372 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 179ca71c7b..68b12a455f 100644
--- a/Changelog
+++ b/Changelog
@@ -34,6 +34,7 @@ version <next>:
- Argonaut Games ASF demuxer
- xfade video filter
- xfade_opencl filter
+- afirsrc audio filter source
version 4.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index f96ba638b2..99ea34cd16 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -5857,6 +5857,44 @@ aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
@end itemize
+ at section afirsrc
+
+Generate a FIR coefficients using frequency sampling method.
+
+The resulting stream can be used with @ref{afir} filter for filtering the audio signal.
+
+The filter accepts the following options:
+
+ at table @option
+ at item taps, t
+Set number of filter coefficents in output audio stream.
+Default value is 1025.
+
+ at item frequency, f
+Set frequency points from where magnitude and phase are set.
+This must be in non decreasing order, and first element must be 0, while last element
+must be 1. Elements are separated by white spaces.
+
+ at item magnitude, m
+Set magnitude value for every frequency point set by @option{frequency}.
+Number of values must be same as number of frequency points.
+Values are separated by white spaces.
+
+ at item phase, p
+Set phase value for every frequency point set by @option{frequency}.
+Number of values must be same as number of frequency points.
+Values are separated by white spaces.
+
+ at item sample_rate, r
+Set sample rate, default is 44100.
+
+ at item nb_samples, n
+Set number of samples per each frame. Default is 1024.
+
+ at item win_func, w
+Set window function. Default is blackman.
+ at end table
+
@section anullsrc
The null audio source, return unprocessed audio frames. It is mainly useful
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 3053740dd3..cc00e2c4ac 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -144,6 +144,7 @@ OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o
OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o
+OBJS-$(CONFIG_AFIRSRC_FILTER) += asrc_afirsrc.o
OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o
OBJS-$(CONFIG_FLITE_FILTER) += asrc_flite.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 1052978cd4..01a7a8bf9f 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -138,6 +138,7 @@ extern AVFilter ff_af_volume;
extern AVFilter ff_af_volumedetect;
extern AVFilter ff_asrc_aevalsrc;
+extern AVFilter ff_asrc_afirsrc;
extern AVFilter ff_asrc_anoisesrc;
extern AVFilter ff_asrc_anullsrc;
extern AVFilter ff_asrc_flite;
diff --git a/libavfilter/asrc_afirsrc.c b/libavfilter/asrc_afirsrc.c
new file mode 100644
index 0000000000..b90ffad57f
--- /dev/null
+++ b/libavfilter/asrc_afirsrc.c
@@ -0,0 +1,330 @@
+/*
+ * Copyright (c) 2020 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
+ */
+
+#include "libavutil/eval.h"
+#include "libavutil/opt.h"
+#include "libavutil/tx.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "window_func.h"
+
+typedef struct AudioFIRSourceContext {
+ const AVClass *class;
+
+ char *freq_points_str;
+ char *magnitude_str;
+ char *phase_str;
+ int nb_taps;
+ int sample_rate;
+ int nb_samples;
+ int win_func;
+
+ AVComplexFloat *complexf;
+ float *freq;
+ float *magnitude;
+ float *phase;
+ int freq_size;
+ int magnitude_size;
+ int phase_size;
+ int nb_freq;
+ int nb_magnitude;
+ int nb_phase;
+
+ float *taps;
+ float *win;
+ int64_t pts;
+
+ AVTXContext *tx_ctx;
+ av_tx_fn tx_fn;
+} AudioFIRSourceContext;
+
+#define OFFSET(x) offsetof(AudioFIRSourceContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption afirsrc_options[] = {
+ { "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
+ { "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=1025}, 9, UINT16_MAX, FLAGS },
+ { "frequency", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
+ { "f", "set frequency points", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str="0 1"}, 0, 0, FLAGS },
+ { "magnitude", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
+ { "m", "set magnitude values", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, FLAGS },
+ { "phase", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
+ { "p", "set phase values", OFFSET(phase_str), AV_OPT_TYPE_STRING, {.str="0 0"}, 0, 0, FLAGS },
+ { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
+ { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
+ { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
+ { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
+ { "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64=WFUNC_BLACKMAN}, 0, NB_WFUNC-1, FLAGS, "win_func" },
+ { "w", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64=WFUNC_BLACKMAN}, 0, NB_WFUNC-1, FLAGS, "win_func" },
+ { "rect", "Rectangular", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT}, 0, 0, FLAGS, "win_func" },
+ { "bartlett", "Bartlett", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, FLAGS, "win_func" },
+ { "hanning", "Hanning", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING}, 0, 0, FLAGS, "win_func" },
+ { "hamming", "Hamming", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING}, 0, 0, FLAGS, "win_func" },
+ { "blackman", "Blackman", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BLACKMAN}, 0, 0, FLAGS, "win_func" },
+ { "welch", "Welch", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_WELCH}, 0, 0, FLAGS, "win_func" },
+ { "flattop", "Flat-top", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_FLATTOP}, 0, 0, FLAGS, "win_func" },
+ { "bharris", "Blackman-Harris", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHARRIS}, 0, 0, FLAGS, "win_func" },
+ { "bnuttall", "Blackman-Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BNUTTALL}, 0, 0, FLAGS, "win_func" },
+ { "bhann", "Bartlett-Hann", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHANN}, 0, 0, FLAGS, "win_func" },
+ { "sine", "Sine", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE}, 0, 0, FLAGS, "win_func" },
+ { "nuttall", "Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_NUTTALL}, 0, 0, FLAGS, "win_func" },
+ { "lanczos", "Lanczos", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_LANCZOS}, 0, 0, FLAGS, "win_func" },
+ { "gauss", "Gauss", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_GAUSS}, 0, 0, FLAGS, "win_func" },
+ { "tukey", "Tukey", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_TUKEY}, 0, 0, FLAGS, "win_func" },
+ { "dolph", "Dolph-Chebyshev", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_DOLPH}, 0, 0, FLAGS, "win_func" },
+ { "cauchy", "Cauchy", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_CAUCHY}, 0, 0, FLAGS, "win_func" },
+ { "parzen", "Parzen", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_PARZEN}, 0, 0, FLAGS, "win_func" },
+ { "poisson", "Poisson", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_POISSON}, 0, 0, FLAGS, "win_func" },
+ { "bohman" , "Bohman", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BOHMAN}, 0, 0, FLAGS, "win_func" },
+ {NULL}
+};
+
+AVFILTER_DEFINE_CLASS(afirsrc);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioFIRSourceContext *s = ctx->priv;
+
+ if (!(s->nb_taps & 1)) {
+ av_log(s, AV_LOG_WARNING, "Number of taps %d must be odd length.\n", s->nb_taps);
+ s->nb_taps |= 1;
+ }
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioFIRSourceContext *s = ctx->priv;
+
+ av_freep(&s->win);
+ av_freep(&s->taps);
+ av_freep(&s->freq);
+ av_freep(&s->magnitude);
+ av_freep(&s->phase);
+ av_freep(&s->complexf);
+ av_tx_uninit(&s->tx_ctx);
+}
+
+static av_cold int query_formats(AVFilterContext *ctx)
+{
+ AudioFIRSourceContext *s = ctx->priv;
+ static const int64_t chlayouts[] = { AV_CH_LAYOUT_MONO, -1 };
+ int sample_rates[] = { s->sample_rate, -1 };
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLT,
+ AV_SAMPLE_FMT_NONE
+ };
+
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ int 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;
+
+ layouts = avfilter_make_format64_list(chlayouts);
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_rates);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static int parse_string(char *str, float **items, int *nb_items, int *items_size)
+{
+ float *new_items;
+ char *tail;
+
+ new_items = av_fast_realloc(NULL, items_size, 1 * sizeof(float));
+ if (!new_items)
+ return AVERROR(ENOMEM);
+ *items = new_items;
+
+ tail = str;
+ if (!tail)
+ return AVERROR(EINVAL);
+
+ do {
+ (*items)[(*nb_items)++] = av_strtod(tail, &tail);
+ new_items = av_fast_realloc(*items, items_size, (*nb_items + 1) * sizeof(float));
+ if (!new_items)
+ return AVERROR(ENOMEM);
+ *items = new_items;
+ if (tail && *tail)
+ tail++;
+ } while (tail && *tail);
+
+ return 0;
+}
+
+static void lininterp(AVComplexFloat *complexf,
+ const float *freq,
+ const float *magnitude,
+ const float *phase,
+ int m, int minterp)
+{
+ for (int i = 0; i < minterp; i++) {
+ for (int j = 1; j < m; j++) {
+ const float x = i / (float)minterp;
+
+ if (x <= freq[j]) {
+ const float mg = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (magnitude[j] - magnitude[j-1]) + magnitude[j-1];
+ const float ph = (x - freq[j-1]) / (freq[j] - freq[j-1]) * (phase[j] - phase[j-1]) + phase[j-1];
+
+ complexf[i].re = mg * cosf(ph);
+ complexf[i].im = mg * sinf(ph);
+ break;
+ }
+ }
+ }
+}
+
+static av_cold int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFIRSourceContext *s = ctx->priv;
+ float overlap, scale = 1.f, compensation;
+ int fft_size, middle, ret;
+
+ s->nb_freq = s->nb_magnitude = s->nb_phase = 0;
+
+ ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
+ if (ret < 0)
+ return ret;
+
+ ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
+ if (ret < 0)
+ return ret;
+
+ ret = parse_string(s->phase_str, &s->phase, &s->nb_phase, &s->phase_size);
+ if (ret < 0)
+ return ret;
+
+ if (s->nb_freq != s->nb_magnitude && s->nb_freq != s->nb_phase && s->nb_freq >= 2) {
+ av_log(ctx, AV_LOG_ERROR, "Number of frequencies, magnitudes and phases must be same and >= 2.\n");
+ return AVERROR(EINVAL);
+ }
+
+ for (int i = 0; i < s->nb_freq; i++) {
+ if (i == 0 && s->freq[i] != 0.f) {
+ av_log(ctx, AV_LOG_ERROR, "First frequency must be 0.\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (i == s->nb_freq - 1 && s->freq[i] != 1.f) {
+ av_log(ctx, AV_LOG_ERROR, "Last frequency must be 1.\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (i && s->freq[i] < s->freq[i-1]) {
+ av_log(ctx, AV_LOG_ERROR, "Frequencies must be in increasing order.\n");
+ return AVERROR(EINVAL);
+ }
+ }
+
+ fft_size = 1 << (av_log2(s->nb_taps) + 1);
+ s->complexf = av_calloc(fft_size * 2, sizeof(*s->complexf));
+ if (!s->complexf)
+ return AVERROR(ENOMEM);
+
+ ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
+ if (ret < 0)
+ return ret;
+
+ s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
+ if (!s->taps)
+ return AVERROR(ENOMEM);
+
+ s->win = av_calloc(s->nb_taps, sizeof(*s->win));
+ if (!s->win)
+ return AVERROR(ENOMEM);
+
+ generate_window_func(s->win, s->nb_taps, s->win_func, &overlap);
+
+ lininterp(s->complexf, s->freq, s->magnitude, s->phase, s->nb_freq, fft_size / 2);
+
+ s->tx_fn(s->tx_ctx, s->complexf + fft_size, s->complexf, sizeof(float));
+
+ compensation = 2.f / fft_size;
+ middle = s->nb_taps / 2;
+
+ for (int i = 0; i <= middle; i++) {
+ s->taps[ i] = s->complexf[fft_size + middle - i].re * compensation * s->win[i];
+ s->taps[middle + i] = s->complexf[fft_size + i].re * compensation * s->win[middle + i];
+ }
+
+ s->pts = 0;
+
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFIRSourceContext *s = ctx->priv;
+ AVFrame *frame;
+ int nb_samples;
+
+ nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts);
+ if (!nb_samples)
+ return AVERROR_EOF;
+
+ if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
+ return AVERROR(ENOMEM);
+
+ memcpy(frame->data[0], s->taps + s->pts, nb_samples * sizeof(float));
+
+ frame->pts = s->pts;
+ s->pts += nb_samples;
+ return ff_filter_frame(outlink, frame);
+}
+
+static const AVFilterPad afirsrc_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_asrc_afirsrc = {
+ .name = "afirsrc",
+ .description = NULL_IF_CONFIG_SMALL("Generate a FIR coefficients audio stream."),
+ .query_formats = query_formats,
+ .init = init,
+ .uninit = uninit,
+ .priv_size = sizeof(AudioFIRSourceContext),
+ .inputs = NULL,
+ .outputs = afirsrc_outputs,
+ .priv_class = &afirsrc_class,
+};
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 4f1e7b1bf9..9e8c82cbd3 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
-#define LIBAVFILTER_VERSION_MINOR 74
+#define LIBAVFILTER_VERSION_MINOR 75
#define LIBAVFILTER_VERSION_MICRO 100
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