[FFmpeg-devel] [PATCH] lavfi: add aecho filter
Paul B Mahol
onemda at gmail.com
Tue Jul 9 16:53:14 CEST 2013
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 29 +++++
libavfilter/Makefile | 1 +
libavfilter/af_aecho.c | 297 +++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 328 insertions(+)
create mode 100644 libavfilter/af_aecho.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 234ff2e..9472675 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -347,6 +347,35 @@ aconvert=u8:auto
@end example
@end itemize
+ at section aecho
+
+Apply echoing to the input audio.
+
+Echoes are reflected sound and can occur naturally amongst mountains
+(and sometimes large buildings) when talking or shouting; digital echo
+effects emulate this behaviour and are often used to help fill out the
+sound of a single instrument or vocal. The time difference between the
+original signal and the reflection is the @code{delay}, and the
+loudness of the reflected signal is the @code{decay}.
+Multiple echoes can have different delays and decays.
+
+A description of the accepted parameters follows.
+
+ at table @option
+ at item in_gain
+Set input volume of reflected signal.
+
+ at item out_gain
+Set output volume of reflected signal.
+
+ at item delay1..7
+Set time interval in miliseconds between original signal and reflection.
+
+ at item decay1..7
+Loudness of reflected signal.
+
+ at end table
+
@section afade
Apply fade-in/out effect to input audio.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index cf76ee1..306b24c 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -52,6 +52,7 @@ OBJS-$(CONFIG_AVFORMAT) += lavfutils.o
OBJS-$(CONFIG_SWSCALE) += lswsutils.o
OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
+OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
diff --git a/libavfilter/af_aecho.c b/libavfilter/af_aecho.c
new file mode 100644
index 0000000..9d4010d
--- /dev/null
+++ b/libavfilter/af_aecho.c
@@ -0,0 +1,297 @@
+/*
+ * 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
+ *
+ */
+
+#include "libavutil/avstring.h"
+#include "libavutil/opt.h"
+#include "libavutil/samplefmt.h"
+#include "libavutil/avassert.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+#define MAX_ECHOS 7
+
+typedef struct AudioEchoContext {
+ const AVClass *class;
+ float in_gain, out_gain;
+ float delay[MAX_ECHOS], decay[MAX_ECHOS];
+ int nb_echos;
+ int delay_index;
+ uint8_t **delayptrs;
+ int max_samples, fade_out;
+ int samples[MAX_ECHOS];
+ int64_t next_pts;
+
+ void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
+ uint8_t * const *src, uint8_t **dst,
+ int nb_samples, int channels);
+} AudioEchoContext;
+
+#define OFFSET(x) offsetof(AudioEchoContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aecho_options[] = {
+ { "in_gain", "", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "out_gain", "", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay1", "", OFFSET(delay[0]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay1", "", OFFSET(decay[0]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay2", "", OFFSET(delay[1]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay2", "", OFFSET(decay[1]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay3", "", OFFSET(delay[2]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay3", "", OFFSET(decay[2]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay4", "", OFFSET(delay[3]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay4", "", OFFSET(decay[3]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay5", "", OFFSET(delay[4]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay5", "", OFFSET(decay[4]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay6", "", OFFSET(delay[5]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay6", "", OFFSET(decay[5]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { "delay7", "", OFFSET(delay[6]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 90000, A },
+ { "decay7", "", OFFSET(decay[6]), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { NULL },
+};
+
+AVFILTER_DEFINE_CLASS(aecho);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioEchoContext *s = ctx->priv;
+ int i;
+
+ for (i = 0; i < MAX_ECHOS; i++) {
+ if (!s->delay[i] || !s->decay[i])
+ break;
+ }
+ s->nb_echos = i;
+ if (!s->nb_echos) {
+ av_log(ctx, AV_LOG_ERROR, "at least one decay & delay must be set");
+ return AVERROR(EINVAL);
+ }
+
+ s->next_pts = AV_NOPTS_VALUE;
+
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts;
+ AVFilterFormats *formats;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
+ 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 av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioEchoContext *s = ctx->priv;
+
+ if (s->delayptrs)
+ av_freep(s->delayptrs[0]);
+ av_freep(&s->delayptrs);
+}
+
+#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
+
+#define ECHO(name, type, min, max) \
+static void echo_samples_## name ##p(AudioEchoContext *ctx, \
+ uint8_t **delayptrs, \
+ uint8_t * const *src, uint8_t **dst, \
+ int nb_samples, int channels) \
+{ \
+ const double out_gain = ctx->out_gain; \
+ const double in_gain = ctx->in_gain; \
+ const int nb_echos = ctx->nb_echos; \
+ const int max_samples = ctx->max_samples; \
+ int i, j, chan, index; \
+ \
+ for (chan = 0; chan < channels; chan++) { \
+ const type *s = (type *)src[chan]; \
+ type *d = (type *)dst[chan]; \
+ type *dbuf = (type *)delayptrs[chan]; \
+ \
+ index = ctx->delay_index; \
+ for (i = 0; i < nb_samples; i++, s++, d++) { \
+ double out, in; \
+ \
+ in = *s; \
+ out = in * in_gain; \
+ for (j = 0; j < nb_echos; j++) { \
+ int ix = index + max_samples - ctx->samples[j]; \
+ ix = MOD(ix, max_samples); \
+ out += dbuf[ix] * ctx->decay[j]; \
+ } \
+ out *= out_gain; \
+ \
+ *d = av_clipd(out, min, max); \
+ dbuf[index] = in; \
+ \
+ index = MOD(index + 1, max_samples); \
+ } \
+ } \
+ ctx->delay_index = index; \
+}
+
+ECHO(dbl, double, -1.0, 1.0 )
+ECHO(flt, float, -1.0, 1.0 )
+ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
+ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioEchoContext *s = ctx->priv;
+ float volume = 1.0;
+ int i;
+
+ for (i = 0; i < s->nb_echos; i++) {
+ s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
+ s->max_samples = FFMAX(s->max_samples, s->samples[i]);
+ volume += s->decay[i];
+ }
+ s->fade_out = s->max_samples;
+
+ if (volume * s->in_gain * s->out_gain > 1.0)
+ av_log(ctx, AV_LOG_WARNING,
+ "out_gain %f can cause saturation of output\n", s->out_gain);
+
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
+ case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
+ case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
+ case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
+ }
+
+ uninit(ctx);
+ return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
+ outlink->channels,
+ s->max_samples,
+ outlink->format, 0);
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioEchoContext *s = ctx->priv;
+ AVFrame *out_frame;
+
+ 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);
+ }
+
+ s->echo_samples(s, s->delayptrs, frame->data, out_frame->data,
+ frame->nb_samples, inlink->channels);
+
+ if (frame != out_frame)
+ av_frame_free(&frame);
+
+ s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
+ return ff_filter_frame(ctx->outputs[0], out_frame);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioEchoContext *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);
+
+ s->echo_samples(s, s->delayptrs, frame->data, frame->data,
+ frame->nb_samples, outlink->channels);
+
+ 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);
+
+ return ff_filter_frame(outlink, frame);
+ }
+
+ return ret;
+}
+
+static const AVFilterPad aecho_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL },
+};
+
+static const AVFilterPad aecho_outputs[] = {
+ {
+ .name = "default",
+ .request_frame = request_frame,
+ .config_props = config_output,
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL },
+};
+
+AVFilter avfilter_af_aecho = {
+ .name = "aecho",
+ .description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioEchoContext),
+ .priv_class = &aecho_class,
+ .init = init,
+ .uninit = uninit,
+ .inputs = aecho_inputs,
+ .outputs = aecho_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 9a11feb..26472f8 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -48,6 +48,7 @@ void avfilter_register_all(void)
#if FF_API_ACONVERT_FILTER
REGISTER_FILTER(ACONVERT, aconvert, af);
#endif
+ REGISTER_FILTER(AECHO, aecho, af);
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
--
1.7.11.2
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