[FFmpeg-cvslog] avfilter: add chorus filter
Paul B Mahol
git at videolan.org
Sun Apr 12 16:28:25 CEST 2015
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Tue Jun 24 08:35:37 2014 +0000| [d2fc702ace340bb22290bc575a0194082dd00627] | committer: Paul B Mahol
avfilter: add chorus filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d2fc702ace340bb22290bc575a0194082dd00627
---
Changelog | 1 +
doc/filters.texi | 55 +++++++
libavfilter/Makefile | 1 +
libavfilter/af_chorus.c | 379 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 4 +-
6 files changed, 439 insertions(+), 2 deletions(-)
diff --git a/Changelog b/Changelog
index e731cd7..2e30b24 100644
--- a/Changelog
+++ b/Changelog
@@ -16,6 +16,7 @@ version <next>:
- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
- WebM Live Chunk Muxer
- nvenc level and tier options
+- chorus filter
version 2.6:
diff --git a/doc/filters.texi b/doc/filters.texi
index b33379d..0f2e720 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1320,6 +1320,61 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav
@end example
+ at section chorus
+Add a chorus effect to the audio.
+
+Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
+
+Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
+constant, with chorus, it is varied using using sinusoidal or triangular modulation.
+The modulation depth defines the range the modulated delay is played before or after
+the delay. Hence the delayed sound will sound slower or faster, that is the delayed
+sound tuned around the original one, like in a chorus where some vocals are slightly
+off key.
+
+It accepts the following parameters:
+ at table @option
+ at item in_gain
+Set input gain. Default is 0.4.
+
+ at item out_gain
+Set output gain. Default is 0.4.
+
+ at item delays
+Set delays. A typical delay is around 40ms to 60ms.
+
+ at item decays
+Set decays.
+
+ at item speeds
+Set speeds.
+
+ at item depths
+Set depths.
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+A single delay:
+ at example
+chorus=0.7:0.9:55:0.4:0.25:2
+ at end example
+
+ at item
+Two delays:
+ at example
+chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
+ at end example
+
+ at item
+Fuller sounding chorus with three delays:
+ at example
+chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
+ at end example
+ at end itemize
+
@section compand
Compress or expand the audio's dynamic range.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 73e7adf..48cee50 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -64,6 +64,7 @@ OBJS-$(CONFIG_BIQUAD_FILTER) += af_biquads.o
OBJS-$(CONFIG_BS2B_FILTER) += af_bs2b.o
OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
+OBJS-$(CONFIG_CHORUS_FILTER) += af_chorus.o generate_wave_table.o
OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
diff --git a/libavfilter/af_chorus.c b/libavfilter/af_chorus.c
new file mode 100644
index 0000000..93fb36b
--- /dev/null
+++ b/libavfilter/af_chorus.c
@@ -0,0 +1,379 @@
+/*
+ * Copyright (c) 1998 Juergen Mueller And Sundry Contributors
+ * This source code is freely redistributable and may be used for
+ * any purpose. This copyright notice must be maintained.
+ * Juergen Mueller And Sundry Contributors are not responsible for
+ * the consequences of using this software.
+ *
+ * Copyright (c) 2015 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
+ */
+
+/**
+ * @file
+ * chorus audio filter
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "generate_wave_table.h"
+
+typedef struct ChorusContext {
+ const AVClass *class;
+ float in_gain, out_gain;
+ char *delays_str;
+ char *decays_str;
+ char *speeds_str;
+ char *depths_str;
+ float *delays;
+ float *decays;
+ float *speeds;
+ float *depths;
+ uint8_t **chorusbuf;
+ int **phase;
+ int *length;
+ int32_t **lookup_table;
+ int *counter;
+ int num_chorus;
+ int max_samples;
+ int channels;
+ int modulation;
+ int fade_out;
+ int64_t next_pts;
+} ChorusContext;
+
+#define OFFSET(x) offsetof(ChorusContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption chorus_options[] = {
+ { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
+ { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
+ { "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+ { "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+ { "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+ { "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(chorus);
+
+static void count_items(char *item_str, int *nb_items)
+{
+ char *p;
+
+ *nb_items = 1;
+ for (p = item_str; *p; p++) {
+ if (*p == '|')
+ (*nb_items)++;
+ }
+
+}
+
+static void fill_items(char *item_str, int *nb_items, float *items)
+{
+ char *p, *saveptr = NULL;
+ int i, new_nb_items = 0;
+
+ p = item_str;
+ for (i = 0; i < *nb_items; i++) {
+ char *tstr = av_strtok(p, "|", &saveptr);
+ p = NULL;
+ new_nb_items += sscanf(tstr, "%f", &items[i]) == 1;
+ }
+
+ *nb_items = new_nb_items;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ ChorusContext *s = ctx->priv;
+ int nb_delays, nb_decays, nb_speeds, nb_depths;
+
+ if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
+ av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
+ return AVERROR(EINVAL);
+ }
+
+ count_items(s->delays_str, &nb_delays);
+ count_items(s->decays_str, &nb_decays);
+ count_items(s->speeds_str, &nb_speeds);
+ count_items(s->depths_str, &nb_depths);
+
+ s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
+ s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
+ s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
+ s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
+
+ if (!s->delays || !s->decays || !s->speeds || !s->depths)
+ return AVERROR(ENOMEM);
+
+ fill_items(s->delays_str, &nb_delays, s->delays);
+ fill_items(s->decays_str, &nb_decays, s->decays);
+ fill_items(s->speeds_str, &nb_speeds, s->speeds);
+ fill_items(s->depths_str, &nb_depths, s->depths);
+
+ if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
+ av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
+ return AVERROR(EINVAL);
+ }
+
+ s->num_chorus = nb_delays;
+
+ if (s->num_chorus < 1) {
+ av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
+ return AVERROR(EINVAL);
+ }
+
+ s->length = av_calloc(s->num_chorus, sizeof(*s->length));
+ s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
+
+ if (!s->length || !s->lookup_table)
+ return AVERROR(ENOMEM);
+
+ s->next_pts = AV_NOPTS_VALUE;
+
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_layouts();
+ 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 int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ ChorusContext *s = ctx->priv;
+ float sum_in_volume = 1.0;
+ int n;
+
+ s->channels = outlink->channels;
+
+ for (n = 0; n < s->num_chorus; n++) {
+ int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
+ int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
+
+ s->length[n] = outlink->sample_rate / s->speeds[n];
+
+ s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
+ if (!s->lookup_table[n])
+ return AVERROR(ENOMEM);
+
+ ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
+ s->length[n], 0., depth_samples, 0);
+ s->max_samples = FFMAX(s->max_samples, samples);
+ }
+
+ for (n = 0; n < s->num_chorus; n++)
+ sum_in_volume += s->decays[n];
+
+ if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
+ av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
+
+ s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
+ if (!s->counter)
+ return AVERROR(ENOMEM);
+
+ s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
+ if (!s->phase)
+ return AVERROR(ENOMEM);
+
+ for (n = 0; n < outlink->channels; n++) {
+ s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
+ if (!s->phase[n])
+ return AVERROR(ENOMEM);
+ }
+
+ s->fade_out = s->max_samples;
+
+ return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
+ outlink->channels,
+ s->max_samples,
+ outlink->format, 0);
+}
+
+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ChorusContext *s = ctx->priv;
+ AVFrame *out_frame;
+ int c, i, n;
+
+ if (av_frame_is_writable(frame)) {
+ out_frame = frame;
+ } else {
+ out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+ if (!out_frame)
+ return AVERROR(ENOMEM);
+ av_frame_copy_props(out_frame, frame);
+ }
+
+ for (c = 0; c < inlink->channels; c++) {
+ const float *src = (const float *)frame->extended_data[c];
+ float *dst = (float *)out_frame->extended_data[c];
+ float *chorusbuf = (float *)s->chorusbuf[c];
+ int *phase = s->phase[c];
+
+ for (i = 0; i < frame->nb_samples; i++) {
+ float out, in = src[i];
+
+ out = in * s->in_gain;
+
+ for (n = 0; n < s->num_chorus; n++) {
+ out += chorusbuf[MOD(s->max_samples + s->counter[c] -
+ s->lookup_table[n][phase[n]],
+ s->max_samples)] * s->decays[n];
+ phase[n] = MOD(phase[n] + 1, s->length[n]);
+ }
+
+ out *= s->out_gain;
+
+ dst[i] = out;
+
+ chorusbuf[s->counter[c]] = in;
+ s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
+ }
+ }
+
+ s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
+
+ if (frame != out_frame)
+ av_frame_free(&frame);
+
+ return ff_filter_frame(ctx->outputs[0], out_frame);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ ChorusContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+
+ if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
+ int nb_samples = FFMIN(s->fade_out, 2048);
+ AVFrame *frame;
+
+ frame = ff_get_audio_buffer(outlink, nb_samples);
+ if (!frame)
+ return AVERROR(ENOMEM);
+ s->fade_out -= nb_samples;
+
+ av_samples_set_silence(frame->extended_data, 0,
+ frame->nb_samples,
+ outlink->channels,
+ frame->format);
+
+ frame->pts = s->next_pts;
+ if (s->next_pts != AV_NOPTS_VALUE)
+ s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+
+ ret = filter_frame(ctx->inputs[0], frame);
+ }
+
+ return ret;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ ChorusContext *s = ctx->priv;
+ int n;
+
+ av_freep(&s->delays);
+ av_freep(&s->decays);
+ av_freep(&s->speeds);
+ av_freep(&s->depths);
+
+ if (s->chorusbuf)
+ av_freep(&s->chorusbuf[0]);
+ av_freep(&s->chorusbuf);
+
+ if (s->phase)
+ for (n = 0; n < s->channels; n++)
+ av_freep(&s->phase[n]);
+ av_freep(&s->phase);
+
+ av_freep(&s->counter);
+ av_freep(&s->length);
+
+ if (s->lookup_table)
+ for (n = 0; n < s->num_chorus; n++)
+ av_freep(&s->lookup_table[n]);
+ av_freep(&s->lookup_table);
+}
+
+static const AVFilterPad chorus_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad chorus_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = request_frame,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_chorus = {
+ .name = "chorus",
+ .description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(ChorusContext),
+ .priv_class = &chorus_class,
+ .init = init,
+ .uninit = uninit,
+ .inputs = chorus_inputs,
+ .outputs = chorus_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 6bc01c5..7961dca 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -80,6 +80,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(BS2B, bs2b, af);
REGISTER_FILTER(CHANNELMAP, channelmap, af);
REGISTER_FILTER(CHANNELSPLIT, channelsplit, af);
+ REGISTER_FILTER(CHORUS, chorus, af);
REGISTER_FILTER(COMPAND, compand, af);
REGISTER_FILTER(DCSHIFT, dcshift, af);
REGISTER_FILTER(EARWAX, earwax, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 43b7ab9..a4a9e63 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,8 +30,8 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 5
-#define LIBAVFILTER_VERSION_MINOR 13
-#define LIBAVFILTER_VERSION_MICRO 101
+#define LIBAVFILTER_VERSION_MINOR 14
+#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
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