[FFmpeg-cvslog] aphaser filter
Paul B Mahol
git at videolan.org
Wed Apr 3 10:37:47 CEST 2013
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Mar 30 04:03:39 2013 +0000| [bf65752848ec5799c72e1c38c574fd88c917fbe1] | committer: Paul B Mahol
aphaser filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bf65752848ec5799c72e1c38c574fd88c917fbe1
---
Changelog | 1 +
doc/filters.texi | 37 +++++
libavfilter/Makefile | 1 +
libavfilter/af_aphaser.c | 360 ++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 4 +-
6 files changed, 402 insertions(+), 2 deletions(-)
diff --git a/Changelog b/Changelog
index 9f57061..5faa414 100644
--- a/Changelog
+++ b/Changelog
@@ -15,6 +15,7 @@ version <next>:
- new ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
- OpenCL support
+- audio phaser filter
version 1.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index 401125b..08c1945 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -6266,6 +6266,43 @@ following one, the permission might not be received as expected in that
following filter. Inserting a @ref{format} or @ref{aformat} filter before the
perms/aperms filter can avoid this problem.
+ at section aphaser
+Add a phasing effect to the input audio.
+
+A phaser filter creates series of peaks and troughs in the frequency spectrum.
+The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.
+
+The filter accepts parameters as a list of @var{key}=@var{value}
+pairs, separated by ":".
+
+A description of the accepted parameters follows.
+
+ at table @option
+ at item in_gain
+Set input gain. Default is 0.4.
+
+ at item out_gain
+Set output gain. Default is 0.74
+
+ at item delay
+Set delay in milliseconds. Default is 3.0.
+
+ at item decay
+Set decay. Default is 0.4.
+
+ at item speed
+Set modulation speed in Hz. Default is 0.5.
+
+ at item type
+Set modulation type. Default is triangular.
+
+It accepts the following values:
+ at table @samp
+ at item triangular, t
+ at item sinusoidal, s
+ at end table
+ at end table
+
@section aselect, select
Select frames to pass in output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index e865aef..9a12273 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -58,6 +58,7 @@ OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
+OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o
diff --git a/libavfilter/af_aphaser.c b/libavfilter/af_aphaser.c
new file mode 100644
index 0000000..141278f
--- /dev/null
+++ b/libavfilter/af_aphaser.c
@@ -0,0 +1,360 @@
+/*
+ * Copyright (c) 2013 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
+ * phaser audio filter
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+enum WaveType {
+ WAVE_SIN,
+ WAVE_TRI,
+ WAVE_NB,
+};
+
+typedef struct AudioPhaserContext {
+ const AVClass *class;
+ double in_gain, out_gain;
+ double delay;
+ double decay;
+ double speed;
+
+ enum WaveType type;
+
+ int delay_buffer_length;
+ double *delay_buffer;
+
+ int modulation_buffer_length;
+ int32_t *modulation_buffer;
+
+ int delay_pos, modulation_pos;
+
+ void (*phaser)(struct AudioPhaserContext *p,
+ uint8_t * const *src, uint8_t **dst,
+ int nb_samples, int channels);
+} AudioPhaserContext;
+
+#define OFFSET(x) offsetof(AudioPhaserContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aphaser_options[] = {
+ { "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
+ { "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
+ { "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
+ { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
+ { "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
+ { "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
+ { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
+ { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
+ { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
+ { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
+ { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(aphaser);
+
+static av_cold int init(AVFilterContext *ctx, const char *args)
+{
+ AudioPhaserContext *p = ctx->priv;
+
+ if (p->in_gain > (1 - p->decay * p->decay))
+ av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
+ if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
+ av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
+
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_NONE
+ };
+
+ layouts = ff_all_channel_layouts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ff_set_common_channel_layouts(ctx, layouts);
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_set_common_formats(ctx, formats);
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_set_common_samplerates(ctx, formats);
+
+ return 0;
+}
+
+static void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt,
+ void *table, int table_size,
+ double min, double max, double phase)
+{
+ uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
+
+ for (i = 0; i < table_size; i++) {
+ uint32_t point = (i + phase_offset) % table_size;
+ double d;
+
+ switch (wave_type) {
+ case WAVE_SIN:
+ d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
+ break;
+ case WAVE_TRI:
+ d = (double)point * 2 / table_size;
+ switch (4 * point / table_size) {
+ case 0: d = d + 0.5; break;
+ case 1:
+ case 2: d = 1.5 - d; break;
+ case 3: d = d - 1.5; break;
+ }
+ break;
+ default:
+ av_assert0(0);
+ }
+
+ d = d * (max - min) + min;
+ switch (sample_fmt) {
+ case AV_SAMPLE_FMT_FLT: {
+ float *fp = (float *)table;
+ *fp++ = (float)d;
+ table = fp;
+ continue; }
+ case AV_SAMPLE_FMT_DBL: {
+ double *dp = (double *)table;
+ *dp++ = d;
+ table = dp;
+ continue; }
+ }
+
+ d += d < 0 ? -0.5 : 0.5;
+ switch (sample_fmt) {
+ case AV_SAMPLE_FMT_S16: {
+ int16_t *sp = table;
+ *sp++ = (int16_t)d;
+ table = sp;
+ continue; }
+ case AV_SAMPLE_FMT_S32: {
+ int32_t *ip = table;
+ *ip++ = (int32_t)d;
+ table = ip;
+ continue; }
+ default:
+ av_assert0(0);
+ }
+ }
+}
+
+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
+
+#define PHASER_PLANAR(name, type) \
+static void phaser_## name ##p(AudioPhaserContext *p, \
+ uint8_t * const *src, uint8_t **dst, \
+ int nb_samples, int channels) \
+{ \
+ int i, c, delay_pos, modulation_pos; \
+ \
+ for (c = 0; c < channels; c++) { \
+ type *s = (type *)src[c]; \
+ type *d = (type *)dst[c]; \
+ double *buffer = p->delay_buffer + \
+ c * p->delay_buffer_length; \
+ \
+ delay_pos = p->delay_pos; \
+ modulation_pos = p->modulation_pos; \
+ \
+ for (i = 0; i < nb_samples; i++, s++, d++) { \
+ double v = *s * p->in_gain + buffer[ \
+ MOD(delay_pos + p->modulation_buffer[ \
+ modulation_pos], \
+ p->delay_buffer_length)] * p->decay; \
+ \
+ modulation_pos = MOD(modulation_pos + 1, \
+ p->modulation_buffer_length); \
+ delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
+ buffer[delay_pos] = v; \
+ \
+ *d = v * p->out_gain; \
+ } \
+ } \
+ \
+ p->delay_pos = delay_pos; \
+ p->modulation_pos = modulation_pos; \
+}
+
+#define PHASER(name, type) \
+static void phaser_## name (AudioPhaserContext *p, \
+ uint8_t * const *src, uint8_t **dst, \
+ int nb_samples, int channels) \
+{ \
+ int i, c, delay_pos, modulation_pos; \
+ type *s = (type *)src[0]; \
+ type *d = (type *)dst[0]; \
+ double *buffer = p->delay_buffer; \
+ \
+ delay_pos = p->delay_pos; \
+ modulation_pos = p->modulation_pos; \
+ \
+ for (i = 0; i < nb_samples; i++) { \
+ int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
+ p->delay_buffer_length) * channels; \
+ int npos; \
+ \
+ delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
+ npos = delay_pos * channels; \
+ for (c = 0; c < channels; c++, s++, d++) { \
+ double v = *s * p->in_gain + buffer[pos + c] * p->decay; \
+ \
+ buffer[npos + c] = v; \
+ \
+ *d = v * p->out_gain; \
+ } \
+ \
+ modulation_pos = MOD(modulation_pos + 1, \
+ p->modulation_buffer_length); \
+ } \
+ \
+ p->delay_pos = delay_pos; \
+ p->modulation_pos = modulation_pos; \
+}
+
+PHASER_PLANAR(dbl, double)
+PHASER_PLANAR(flt, float)
+PHASER_PLANAR(s16, int16_t)
+PHASER_PLANAR(s32, int32_t)
+
+PHASER(dbl, double)
+PHASER(flt, float)
+PHASER(s16, int16_t)
+PHASER(s32, int32_t)
+
+static int config_output(AVFilterLink *outlink)
+{
+ AudioPhaserContext *p = outlink->src->priv;
+ AVFilterLink *inlink = outlink->src->inputs[0];
+
+ p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
+ p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
+ p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
+ p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));
+
+ if (!p->modulation_buffer || !p->delay_buffer)
+ return AVERROR(ENOMEM);
+
+ generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
+ p->modulation_buffer, p->modulation_buffer_length,
+ 1., p->delay_buffer_length, M_PI / 2.0);
+
+ p->delay_pos = p->modulation_pos = 0;
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
+ case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
+ case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
+ case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
+ case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
+ case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
+ case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
+ case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
+ default: av_assert0(0);
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
+{
+ AudioPhaserContext *p = inlink->dst->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ AVFrame *outbuf;
+
+ if (av_frame_is_writable(inbuf)) {
+ outbuf = inbuf;
+ } else {
+ outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
+ if (!outbuf)
+ return AVERROR(ENOMEM);
+ av_frame_copy_props(outbuf, inbuf);
+ }
+
+ p->phaser(p, inbuf->extended_data, outbuf->extended_data,
+ outbuf->nb_samples, av_frame_get_channels(outbuf));
+
+ if (inbuf != outbuf)
+ av_frame_free(&inbuf);
+
+ return ff_filter_frame(outlink, outbuf);
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioPhaserContext *p = ctx->priv;
+
+ av_freep(&p->delay_buffer);
+ av_freep(&p->modulation_buffer);
+}
+
+static const AVFilterPad aphaser_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad aphaser_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+static const char *const shorthand[] = { "in_gain", "out_gain", "delay", "decay", "speed", "type", NULL };
+
+AVFilter avfilter_af_aphaser = {
+ .name = "aphaser",
+ .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioPhaserContext),
+ .init = init,
+ .uninit = uninit,
+ .inputs = aphaser_inputs,
+ .outputs = aphaser_outputs,
+ .priv_class = &aphaser_class,
+ .shorthand = shorthand,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 4ca180a..4972322 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -54,6 +54,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(ANULL, anull, af);
REGISTER_FILTER(APAD, apad, af);
REGISTER_FILTER(APERMS, aperms, af);
+ REGISTER_FILTER(APHASER, aphaser, af);
REGISTER_FILTER(ARESAMPLE, aresample, af);
REGISTER_FILTER(ASELECT, aselect, af);
REGISTER_FILTER(ASENDCMD, asendcmd, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index e623896..379b9ab 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -29,8 +29,8 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
-#define LIBAVFILTER_VERSION_MINOR 48
-#define LIBAVFILTER_VERSION_MICRO 105
+#define LIBAVFILTER_VERSION_MINOR 49
+#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \
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