[FFmpeg-devel] [PATCH] avfilter: add adelay filter

Paul B Mahol onemda at gmail.com
Mon Sep 16 10:49:39 CEST 2013


On 9/15/13, Paul B Mahol <onemda at gmail.com> wrote:
> On 9/15/13, Stefano Sabatini <stefasab at gmail.com> wrote:
>> On date Friday 2013-09-13 17:42:08 +0000, Paul B Mahol encoded:
>>> 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:
>>> +
>>
>> Please specify what happens when an audio channels is delayed (I
>> suppose it is filled with silence).
>
> Added.
>
>>
>>> + 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.
>>
>> Missing separator declaration. Also I wonder if it makes sense to
>> specify time durations instead.
>
>
> Well, usage is to use reasonably small delays.
> Minutes/hours needs memory.
>
>>
>>> + 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");
>>
>> Nit: no need for final point (no complete sentence)
>>
>>> +        return AVERROR(EINVAL);
>>
>> or maybe it could work a as a no-op (simplify scripting sometimes).
>
> It was trivial change, thus changed that way (also less lines).
>
>>
>>> +    }
>>> +
>>> +    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");
>>
>> Nit: a non negative number
>>
>> Same remark about milliseconds vs. time duration specification.
>>
>> Also: would it make sense to specify a negative delay?
>
> That would make code more complicated. Feel free to send patch.
>
>>
>>> +            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");
>>
>> Nit: is needed / must be specified.
>
> Done.

I gonna push if there will be no more comments.


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