@@ -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:

@@ -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.

@@ -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

new file mode 100644

@@ -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,

+};

@@ -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);

@@ -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, \