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[FFmpeg-devel] avfilter: add afirsrc filter

Message ID 20200201125002.11306-1-onemda@gmail.com
State Accepted
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Series [FFmpeg-devel] avfilter: add afirsrc filter
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Commit Message

Paul B Mahol Feb. 1, 2020, 12:50 p.m. UTC
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi           |  38 +++++
 libavfilter/Makefile       |   1 +
 libavfilter/allfilters.c   |   1 +
 libavfilter/asrc_afirsrc.c | 330 +++++++++++++++++++++++++++++++++++++
 4 files changed, 370 insertions(+)
 create mode 100644 libavfilter/asrc_afirsrc.c

Comments

Paul B Mahol Feb. 7, 2020, 9:15 a.m. UTC | #1
On 2/1/20, Paul B Mahol <onemda@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda@gmail.com>
> ---
>  doc/filters.texi           |  38 +++++
>  libavfilter/Makefile       |   1 +
>  libavfilter/allfilters.c   |   1 +
>  libavfilter/asrc_afirsrc.c | 330 +++++++++++++++++++++++++++++++++++++
>  4 files changed, 370 insertions(+)
>  create mode 100644 libavfilter/asrc_afirsrc.c
>


Will apply.
diff mbox series

Patch

diff --git a/doc/filters.texi b/doc/filters.texi
index a3cc92200d..7081e1c0dc 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
 
+@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:
+
+@table @option
+@item taps, t
+Set number of filter coefficents in output audio stream.
+Default value is 1025.
+
+@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.
+
+@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.
+
+@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.
+
+@item sample_rate, r
+Set sample rate, default is 44100.
+
+@item nb_samples, n
+Set number of samples per each frame. Default is 1024.
+
+@item win_func, w
+Set window function. Default is blackman.
+@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 72804323d5..55898e3b5d 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 f7ab2def92..e31a958f40 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -137,6 +137,7 @@  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,
+};