[FFmpeg-devel] [PATCH] avfilter: add adelay filter
Paul B Mahol
onemda at gmail.com
Fri Sep 13 19:42:08 CEST 2013
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 15 +++
libavfilter/Makefile | 1 +
libavfilter/af_adelay.c | 296 +++++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 313 insertions(+)
create mode 100644 libavfilter/af_adelay.c
diff --git a/doc/filters.texi b/doc/filters.texi
index 7f8d1b2..d4cec8a 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -347,6 +347,21 @@ aconvert=u8:auto
@end example
@end itemize
+ at section adelay
+
+Delay one or more audio channels.
+
+The filter accepts the following option:
+
+ at table @option
+ at item delays
+Set list of delays in milliseconds for each channel.
+At least one delay greater than 0 should be provided.
+Unused delays will be silently ignored. If number
+of given delays is smaller than numer of channels all
+remaining channels will be un-delayed.
+ at end table
+
@section aecho
Apply echoing to the input audio.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index b57d4c9..5a82c84 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_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
diff --git a/libavfilter/af_adelay.c b/libavfilter/af_adelay.c
new file mode 100644
index 0000000..b74ddaf
--- /dev/null
+++ b/libavfilter/af_adelay.c
@@ -0,0 +1,296 @@
+/*
+ * 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 "avfilter.h"
+#include "audio.h"
+#include "internal.h"
+
+typedef struct ChanDelay {
+ int delay;
+ unsigned delay_index;
+ unsigned index;
+ uint8_t *samples;
+} ChanDelay;
+
+typedef struct AudioDelayContext {
+ const AVClass *class;
+ char *delays;
+ ChanDelay *chandelay;
+ int nb_delays;
+ int block_align;
+ unsigned max_delay;
+ int64_t next_pts;
+
+ void (*delay_channel)(ChanDelay *d, int nb_samples,
+ const uint8_t *src, uint8_t *dst);
+} AudioDelayContext;
+
+#define OFFSET(x) offsetof(AudioDelayContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption adelay_options[] = {
+ { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adelay);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioDelayContext *s = ctx->priv;
+
+ if (!s->delays) {
+ av_log(ctx, AV_LOG_ERROR, "Missing delays.\n");
+ return AVERROR(EINVAL);
+ }
+
+ return 0;
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterChannelLayouts *layouts;
+ AVFilterFormats *formats;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_U8P, 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;
+}
+
+#define DELAY(name, type, fill) \
+static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
+ const uint8_t *ssrc, uint8_t *ddst) \
+{ \
+ const type *src = (type *)ssrc; \
+ type *dst = (type *)ddst; \
+ type *samples = (type *)d->samples; \
+ \
+ while (nb_samples) { \
+ if (d->delay_index < d->delay) { \
+ const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
+ \
+ memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
+ memset(dst, fill, len * sizeof(type)); \
+ d->delay_index += len; \
+ src += len; \
+ dst += len; \
+ nb_samples -= len; \
+ } else { \
+ *dst = samples[d->index]; \
+ samples[d->index] = *src; \
+ nb_samples--; \
+ d->index++; \
+ src++, dst++; \
+ d->index %= d->delay; \
+ } \
+ } \
+}
+
+DELAY(u8, uint8_t, 0x80)
+DELAY(s16, int16_t, 0)
+DELAY(s32, int32_t, 0)
+DELAY(flt, float, 0)
+DELAY(dbl, double, 0)
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioDelayContext *s = ctx->priv;
+ char *p, *arg, *saveptr = NULL;
+ int i;
+
+ s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
+ if (!s->chandelay)
+ return AVERROR(ENOMEM);
+ s->nb_delays = inlink->channels;
+ s->block_align = av_get_bytes_per_sample(inlink->format);
+
+ p = s->delays;
+ for (i = 0; i < s->nb_delays; i++) {
+ ChanDelay *d = &s->chandelay[i];
+ float delay;
+
+ if (!(arg = av_strtok(p, "|", &saveptr)))
+ break;
+
+ p = NULL;
+ sscanf(arg, "%f", &delay);
+
+ d->delay = delay * inlink->sample_rate / 1000.0;
+ if (d->delay < 0) {
+ av_log(ctx, AV_LOG_ERROR, "Delay must be non-negative number.\n");
+ return AVERROR(EINVAL);
+ }
+ }
+
+ for (i = 0; i < s->nb_delays; i++) {
+ ChanDelay *d = &s->chandelay[i];
+
+ if (!d->delay)
+ continue;
+
+ d->samples = av_malloc_array(d->delay, s->block_align);
+ if (!d->samples)
+ return AVERROR(ENOMEM);
+
+ s->max_delay = FFMAX(s->max_delay, d->delay);
+ }
+
+ if (!s->max_delay) {
+ av_log(ctx, AV_LOG_ERROR, "At least one delay >0 needed.\n");
+ return AVERROR(EINVAL);
+ }
+
+ switch (inlink->format) {
+ case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
+ case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
+ case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
+ case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
+ case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
+ }
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioDelayContext *s = ctx->priv;
+ AVFrame *out_frame;
+ int i;
+
+ if (ctx->is_disabled)
+ return ff_filter_frame(ctx->outputs[0], frame);
+
+ out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
+ if (!out_frame)
+ return AVERROR(ENOMEM);
+ av_frame_copy_props(out_frame, frame);
+
+ for (i = 0; i < s->nb_delays; i++) {
+ ChanDelay *d = &s->chandelay[i];
+ const uint8_t *src = frame->extended_data[i];
+ uint8_t *dst = out_frame->extended_data[i];
+
+ if (!d->delay)
+ memcpy(dst, src, frame->nb_samples * s->block_align);
+ else
+ s->delay_channel(d, frame->nb_samples, src, dst);
+ }
+
+ s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
+ av_frame_free(&frame);
+ return ff_filter_frame(ctx->outputs[0], out_frame);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioDelayContext *s = ctx->priv;
+ int ret;
+
+ ret = ff_request_frame(ctx->inputs[0]);
+ if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
+ int nb_samples = FFMIN(s->max_delay, 2048);
+ AVFrame *frame;
+
+ frame = ff_get_audio_buffer(outlink, nb_samples);
+ if (!frame)
+ return AVERROR(ENOMEM);
+ s->max_delay -= 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)
+{
+ AudioDelayContext *s = ctx->priv;
+ int i;
+
+ for (i = 0; i < s->nb_delays; i++)
+ av_free(s->chandelay[i].samples);
+ av_freep(&s->chandelay);
+}
+
+static const AVFilterPad adelay_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad adelay_outputs[] = {
+ {
+ .name = "default",
+ .request_frame = request_frame,
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter avfilter_af_adelay = {
+ .name = "adelay",
+ .description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioDelayContext),
+ .priv_class = &adelay_class,
+ .init = init,
+ .uninit = uninit,
+ .inputs = adelay_inputs,
+ .outputs = adelay_outputs,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 7eea4bf..f7e4342 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(ADELAY, adelay, af);
REGISTER_FILTER(AECHO, aecho, af);
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);
--
1.7.11.2
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