avfilter: add adelay filter
authorPaul B Mahol <onemda@gmail.com>
Fri, 13 Sep 2013 11:36:52 +0000 (11:36 +0000)
committerPaul B Mahol <onemda@gmail.com>
Mon, 16 Sep 2013 14:33:07 +0000 (14:33 +0000)
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Changelog
doc/filters.texi
libavfilter/Makefile
libavfilter/af_adelay.c [new file with mode: 0644]
libavfilter/allfilters.c
libavfilter/version.h

index b5d49c0..904c36d 100644 (file)
--- a/Changelog
+++ b/Changelog
@@ -23,6 +23,8 @@ version <next>
 - FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
 - changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
   more consistent with other muxers.
+- adelay filter
+
 
 version 2.0:
 
index 7f8d1b2..3404f8b 100644 (file)
@@ -347,6 +347,33 @@ aconvert=u8:auto
 @end example
 @end itemize
 
+@section adelay
+
+Delay one or more audio channels.
+
+Samples in delayed channel are filled with silence.
+
+The filter accepts the following option:
+
+@table @option
+@item delays
+Set list of delays in milliseconds for each channel separated by '|'.
+At least one delay greater than 0 should be provided.
+Unused delays will be silently ignored. If number of given delays is
+smaller than number of channels all remaining channels will not be delayed.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
+the second channel (and any other channels that may be present) unchanged.
+@example
+adelay=1500:0:500
+@end example
+@end itemize
+
 @section aecho
 
 Apply echoing to the input audio.
index b57d4c9..5a82c84 100644 (file)
@@ -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 (file)
index 0000000..d51264f
--- /dev/null
@@ -0,0 +1,283 @@
+/*
+ * 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 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->index >= d->delay ? 0 : d->index;               \
+        }                                                                 \
+    }                                                                     \
+}
+
+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 must be specified.\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 || !s->delays)
+        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,
+    .uninit        = uninit,
+    .inputs        = adelay_inputs,
+    .outputs       = adelay_outputs,
+    .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+};
index 7eea4bf..f7e4342 100644 (file)
@@ -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);
index f48d4ed..f0d4952 100644 (file)
@@ -30,7 +30,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  3
-#define LIBAVFILTER_VERSION_MINOR  84
+#define LIBAVFILTER_VERSION_MINOR  85
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \