[FFmpeg-devel] [PATCH] aphaser filter
Paul B Mahol
onemda at gmail.com
Mon Apr 1 19:14:16 CEST 2013
On 3/31/13, Stefano Sabatini <stefasab at gmail.com> wrote:
> On date Saturday 2013-03-30 21:55:29 +0000, Paul B Mahol encoded:
>> Signed-off-by: Paul B Mahol <onemda at gmail.com>
>> ---
>> doc/filters.texi | 25 +++++
>> libavfilter/Makefile | 1 +
>> libavfilter/af_aphaser.c | 271
>> +++++++++++++++++++++++++++++++++++++++++++++++
>> libavfilter/allfilters.c | 1 +
>> 4 files changed, 298 insertions(+)
>> create mode 100644 libavfilter/af_aphaser.c
>>
>> diff --git a/doc/filters.texi b/doc/filters.texi
>> index 4190cca..2b8e58b 100644
>> --- a/doc/filters.texi
>> +++ b/doc/filters.texi
>> @@ -6262,6 +6262,31 @@ following one, the permission might not be received
>> as expected in that
>> following filter. Inserting a @ref{format} or @ref{aformat} filter before
>> the
>> perms/aperms filter can avoid this problem.
>>
>> + at section aphaser
>> +Add a phasing effect to the input audio.
>> +
>
> Maybe add a note about what a phaser is and what is used for.
>
>> +The filter accepts parameters as a list of @var{key}=@var{value}
>> +pairs, separated by ":".
>> +
>> +A description of the accepted parameters follows.
>> +
>> + at table @option
>> + at item in_gain
>> +Set input gain. Default is 0.4.
>> +
>> + at item out_gain
>> +Set output gain. Default is 0.74
>> +
>> + at item delay
>
>> +Set delay in miliseconds. Default is 3.0.
>
> typo
>
>> +
>> + at item decay
>> +Set decay. Default is 0.4.
>> +
>> + at item speed
>> +Set modulation speed in Hz. Default is 0.5.
>
> You could mention the ranges for the various options.
>
>> + at end table
>> +
>> @section aselect, select
>> Select frames to pass in output.
>>
>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
>> index 690b1cb..a4bdf2e 100644
>> --- a/libavfilter/Makefile
>> +++ b/libavfilter/Makefile
>> @@ -57,6 +57,7 @@ OBJS-$(CONFIG_AMIX_FILTER) +=
>> af_amix.o
>> OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
>> OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
>> OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
>> +OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o
>> OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
>> OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
>> OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o
>> diff --git a/libavfilter/af_aphaser.c b/libavfilter/af_aphaser.c
>> new file mode 100644
>> index 0000000..dd5fd4c
>> --- /dev/null
>> +++ b/libavfilter/af_aphaser.c
>> @@ -0,0 +1,271 @@
>> +/*
>> + * 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
>> + */
>> +
>> +/**
>> + * @file
>> + * phaser audio filter
>> + */
>> +
>> +#include "libavutil/avassert.h"
>> +#include "libavutil/opt.h"
>> +#include "audio.h"
>> +#include "avfilter.h"
>> +#include "internal.h"
>> +
>> +enum WaveType {
>> + WAVE_SINE,
>> + WAVE_TRIANGLE,
>> +};
>> +
>> +typedef struct {
>> + const AVClass *class;
>> + double in_gain, out_gain;
>> + double delay;
>> + double decay;
>> + double speed;
>> +
>> + int delay_buffer_length;
>> + double *delay_buffer;
>> +
>> + int modulation_buffer_length;
>> + int32_t *modulation_buffer;
>> +
>> + int delay_pos, modulation_pos;
>> +} AudioPhaserContext;
>> +
>> +#define OFFSET(x) offsetof(AudioPhaserContext, x)
>> +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
>> +
>> +static const AVOption aphaser_options[] = {
>> + { "in_gain", "set input gain", OFFSET(in_gain),
>> AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
>> + { "out_gain", "set output gain", OFFSET(out_gain),
>> AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
>
>> + { "delay", "set delay in miliseconds", OFFSET(delay),
>> AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
>
> mili -> milli typo
>
>> + { "decay", "set decay", OFFSET(decay),
>> AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
>> + { "speed", "set modulation speed", OFFSET(speed),
>> AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
>> + { NULL },
>> +};
>> +
>> +AVFILTER_DEFINE_CLASS(aphaser);
>> +
>> +static av_cold int init(AVFilterContext *ctx, const char *args)
>> +{
>> + AudioPhaserContext *p = ctx->priv;
>> +
>> + if (p->in_gain > (1 - p->decay * p->decay))
>> + av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
>> + if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
>> + av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
>> +
>> + return 0;
>> +}
>> +
>> +static int query_formats(AVFilterContext *ctx)
>> +{
>> + AVFilterFormats *formats;
>> + AVFilterChannelLayouts *layouts;
>> + static const enum AVSampleFormat sample_fmts[] = {
>> + 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 void generate_wave_table(int wave_type, enum AVSampleFormat
>> sample_fmt,
>
> enum WaveType wave_type
>
>> + void *table, int table_size,
>> + double min, double max, double phase)
>> +{
>> + uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
>> +
>> + for (i = 0; i < table_size; i++) {
>> + uint32_t point = (i + phase_offset) % table_size;
>> + double d;
>> +
>> + switch (wave_type) {
>> + case WAVE_SINE:
>> + d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
>> + break;
>> + case WAVE_TRIANGLE:
>> + d = (double)point * 2 / table_size;
>> + switch (4 * point / table_size) {
>> + case 0: d = d + 0.5; break;
>> + case 1:
>> + case 2: d = 1.5 - d; break;
>> + case 3: d = d - 1.5; break;
>> + }
>> + break;
>> + default:
>> + av_assert0(0);
>> + }
>> +
>> + d = d * (max - min) + min;
>> + switch (sample_fmt) {
>> + case AV_SAMPLE_FMT_FLT: {
>> + float *fp = (float *)table;
>> + *fp++ = (float)d;
>> + table = fp;
>> + continue; }
>> + case AV_SAMPLE_FMT_DBL: {
>> + double *dp = (double *)table;
>> + *dp++ = d;
>> + table = dp;
>> + continue; }
>> + }
>> +
>> + d += d < 0 ? -0.5 : +0.5;
>> + switch (sample_fmt) {
>> + case AV_SAMPLE_FMT_S16: {
>> + int16_t *sp = table;
>> + *sp++ = (int16_t)d;
>> + table = sp;
>> + continue; }
>> + case AV_SAMPLE_FMT_S32: {
>> + int32_t *ip = table;
>> + *ip++ = (int32_t)d;
>> + table = ip;
>> + continue; }
>> + default:
>> + av_assert0(0);
>> + }
>> + }
>> +}
>> +
>> +static int config_input(AVFilterLink *inlink)
>> +{
>> + AudioPhaserContext *p = inlink->dst->priv;
>> +
>> + p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate +
>> 0.5;
>> + p->delay_buffer = av_calloc(p->delay_buffer_length,
>> sizeof(*p->delay_buffer) * inlink->channels);
>> + p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
>> + p->modulation_buffer = av_malloc(p->modulation_buffer_length *
>> sizeof(*p->modulation_buffer));
>> +
>> + if (!p->modulation_buffer || !p->delay_buffer)
>> + return AVERROR(ENOMEM);
>> +
>> + generate_wave_table(WAVE_TRIANGLE, AV_SAMPLE_FMT_S32,
>> + p->modulation_buffer,
>> p->modulation_buffer_length,
>> + 1., (double)p->delay_buffer_length, M_PI / 2.0);
>> +
>> + p->delay_pos = p->modulation_pos = 0;
>> +
>> + return 0;
>> +}
>> +
>> +static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
>> +{
>> + AudioPhaserContext *p = inlink->dst->priv;
>> + AVFilterLink *outlink = inlink->dst->outputs[0];
>> + AVFrame *out_buf;
>> + int i, c, delay_pos, modulation_pos;
>> +
>> + if (av_frame_is_writable(buf)) {
>> + out_buf = buf;
>> + } else {
>> + out_buf = ff_get_audio_buffer(inlink, buf->nb_samples);
>> + if (!out_buf)
>> + return AVERROR(ENOMEM);
>> + out_buf->pts = buf->pts;
>
> copy props
>
>> + }
>> +
>> + for (c = 0; c < av_frame_get_channels(buf); c++) {
>> + double *src = (double *)buf->extended_data[c];
>> + double *dst = (double *)out_buf->extended_data[c];
>> + double *buffer = p->delay_buffer + c * p->delay_buffer_length;
>> +
>> + delay_pos = p->delay_pos;
>> + modulation_pos = p->modulation_pos;
>> +
>> + for (i = 0; i < buf->nb_samples; i++, src++, dst++) {
>> + double d = *src * p->in_gain + buffer[
>> + (delay_pos + p->modulation_buffer[modulation_pos])
>> %
>> + p->delay_buffer_length] * p->decay;
>> +
>> + modulation_pos = (modulation_pos + 1) %
>> p->modulation_buffer_length;
>> + delay_pos = (delay_pos + 1) % p->delay_buffer_length;
>> + buffer[delay_pos] = d;
>> +
>> + *dst = d * p->out_gain;
>> + }
>> + }
>> +
>> + p->delay_pos = delay_pos;
>> + p->modulation_pos = modulation_pos;
>> +
>> + if (buf != out_buf)
>> + av_frame_free(&buf);
>> +
>> + return ff_filter_frame(outlink, out_buf);
>> +}
>> +
>> +static av_cold void uninit(AVFilterContext *ctx)
>> +{
>> + AudioPhaserContext *p = ctx->priv;
>> +
>> + av_freep(&p->delay_buffer);
>> + av_freep(&p->modulation_buffer);
>> +}
>> +
>> +static const AVFilterPad aphaser_inputs[] = {
>> + {
>> + .name = "default",
>> + .type = AVMEDIA_TYPE_AUDIO,
>> + .filter_frame = filter_frame,
>> + .config_props = config_input,
>> + },
>> + { NULL }
>> +};
>> +
>> +static const AVFilterPad aphaser_outputs[] = {
>> + {
>> + .name = "default",
>> + .type = AVMEDIA_TYPE_AUDIO,
>> + },
>> + { NULL }
>> +};
>> +
>> +static const char *const shorthand[] = { "in_gain", "out_gain", "delay",
>> "decay", "speed", NULL };
>> +
>> +AVFilter avfilter_af_aphaser = {
>> + .name = "aphaser",
>> + .description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the
>> audio."),
>> + .query_formats = query_formats,
>> + .priv_size = sizeof(AudioPhaserContext),
>> + .init = init,
>> + .uninit = uninit,
>> + .inputs = aphaser_inputs,
>> + .outputs = aphaser_outputs,
>> + .priv_class = &aphaser_class,
>> + .shorthand = shorthand,
>> +};
>> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
>> index 45a67e5..287d459 100644
>> --- a/libavfilter/allfilters.c
>> +++ b/libavfilter/allfilters.c
>> @@ -53,6 +53,7 @@ void avfilter_register_all(void)
>> REGISTER_FILTER(ANULL, anull, af);
>> REGISTER_FILTER(APAD, apad, af);
>> REGISTER_FILTER(APERMS, aperms, af);
>> + REGISTER_FILTER(APHASER, aphaser, af);
>> REGISTER_FILTER(ARESAMPLE, aresample, af);
>> REGISTER_FILTER(ASELECT, aselect, af);
>> REGISTER_FILTER(ASENDCMD, asendcmd, af);
>
> No more comments from me, can't comment on the algo itself but I can't
> spot evident mistakes.
>
> It could be useful to compare the output with the one from the sox
> wrapper.
I compared it with sox itself and it's bitexact.
Sox wrapper is missing multichannel support and this effect in sox
runs on single channel (whatever such seperation is useful or not).
>
> Thanks.
> --
> FFmpeg = Fascinating Fascinating Multimedia Plastic Earthshaking God
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> ffmpeg-devel mailing list
> ffmpeg-devel at ffmpeg.org
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>
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