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avfilter: remove af_compand_fork

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All relevant changes except the float/double one have been merged

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
pull/56/merge
Michael Niedermayer 11 years ago
parent c62d83a5e7
commit 8f946ac8ea
3 changed files with 0 additions and 588 deletions
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  1. 1
      libavfilter/Makefile
  2. 586
      libavfilter/af_compand_fork.c
  3. 1
      libavfilter/allfilters.c

1
libavfilter/Makefile
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@ -88,7 +88,6 @@ OBJS-$(CONFIG_BIQUAD_FILTER) += af_biquads.o
OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_COMPAND_FORK_FILTER) += af_compand_fork.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o

586
libavfilter/af_compand_fork.c
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@ -1,586 +0,0 @@
/*
* Copyright (c) 1999 Chris Bagwell
* Copyright (c) 1999 Nick Bailey
* Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
* Copyright (c) 2013 Paul B Mahol
* Copyright (c) 2014 Andrew Kelley
*
* 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
* audio compand filter
*/
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/mathematics.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
typedef struct ChanParam {
float attack;
float decay;
float volume;
} ChanParam;
typedef struct CompandSegment {
float x, y;
float a, b;
} CompandSegment;
typedef struct CompandContext {
const AVClass *class;
int nb_channels;
int nb_segments;
char *attacks, *decays, *points;
CompandSegment *segments;
ChanParam *channels;
float in_min_lin;
float out_min_lin;
double curve_dB;
double gain_dB;
double initial_volume;
double delay;
AVFrame *delay_frame;
int delay_samples;
int delay_count;
int delay_index;
int64_t pts;
int (*compand)(AVFilterContext *ctx, AVFrame *frame);
} CompandContext;
#define OFFSET(x) offsetof(CompandContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
static const AVOption compand_options[] = {
{ "attacks", "set time over which increase of volume is determined", OFFSET(attacks), AV_OPT_TYPE_STRING, { .str = "0.3" }, 0, 0, A },
{ "decays", "set time over which decrease of volume is determined", OFFSET(decays), AV_OPT_TYPE_STRING, { .str = "0.8" }, 0, 0, A },
{ "points", "set points of transfer function", OFFSET(points), AV_OPT_TYPE_STRING, { .str = "-70/-70|-60/-20" }, 0, 0, A },
{ "soft-knee", "set soft-knee", OFFSET(curve_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0.01 }, 0.01, 900, A },
{ "gain", "set output gain", OFFSET(gain_dB), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 900, A },
{ "volume", "set initial volume", OFFSET(initial_volume), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, -900, 0, A },
{ "delay", "set delay for samples before sending them to volume adjuster", OFFSET(delay), AV_OPT_TYPE_DOUBLE, { .dbl = 0 }, 0, 20, A },
{ NULL }
};
static const AVClass compand_class = {
.class_name = "compand filter",
.item_name = av_default_item_name,
.option = compand_options,
.version = LIBAVUTIL_VERSION_INT,
};
static av_cold int init(AVFilterContext *ctx)
{
CompandContext *s = ctx->priv;
s->pts = AV_NOPTS_VALUE;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
CompandContext *s = ctx->priv;
av_freep(&s->channels);
av_freep(&s->segments);
av_frame_free(&s->delay_frame);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts;
AVFilterFormats *formats;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
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 count_items(char *item_str, int *nb_items)
{
char *p;
*nb_items = 1;
for (p = item_str; *p; p++) {
if (*p == '|')
(*nb_items)++;
}
}
static void update_volume(ChanParam *cp, float in)
{
float delta = in - cp->volume;
if (delta > 0.0)
cp->volume += delta * cp->attack;
else
cp->volume += delta * cp->decay;
}
static float get_volume(CompandContext *s, float in_lin)
{
CompandSegment *cs;
float in_log, out_log;
int i;
if (in_lin < s->in_min_lin)
return s->out_min_lin;
in_log = logf(in_lin);
for (i = 1; i < s->nb_segments; i++)
if (in_log <= s->segments[i].x)
break;
cs = &s->segments[i - 1];
in_log -= cs->x;
out_log = cs->y + in_log * (cs->a * in_log + cs->b);
return expf(out_log);
}
static int compand_nodelay(AVFilterContext *ctx, AVFrame *frame)
{
CompandContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
const int channels = s->nb_channels;
const int nb_samples = frame->nb_samples;
AVFrame *out_frame;
int chan, i;
int err;
if (av_frame_is_writable(frame)) {
out_frame = frame;
} else {
out_frame = ff_get_audio_buffer(inlink, nb_samples);
if (!out_frame) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
err = av_frame_copy_props(out_frame, frame);
if (err < 0) {
av_frame_free(&out_frame);
av_frame_free(&frame);
return err;
}
}
for (chan = 0; chan < channels; chan++) {
const float *src = (float *)frame->extended_data[chan];
float *dst = (float *)out_frame->extended_data[chan];
ChanParam *cp = &s->channels[chan];
for (i = 0; i < nb_samples; i++) {
update_volume(cp, fabs(src[i]));
dst[i] = av_clipf(src[i] * get_volume(s, cp->volume), -1.0f, 1.0f);
}
}
if (frame != out_frame)
av_frame_free(&frame);
return ff_filter_frame(ctx->outputs[0], out_frame);
}
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
static int compand_delay(AVFilterContext *ctx, AVFrame *frame)
{
CompandContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
const int channels = s->nb_channels;
const int nb_samples = frame->nb_samples;
int chan, i, dindex = 0, oindex, count = 0;
AVFrame *out_frame = NULL;
int err;
if (s->pts == AV_NOPTS_VALUE) {
s->pts = (frame->pts == AV_NOPTS_VALUE) ? 0 : frame->pts;
}
for (chan = 0; chan < channels; chan++) {
AVFrame *delay_frame = s->delay_frame;
const float *src = (float *)frame->extended_data[chan];
float *dbuf = (float *)delay_frame->extended_data[chan];
ChanParam *cp = &s->channels[chan];
float *dst;
count = s->delay_count;
dindex = s->delay_index;
for (i = 0, oindex = 0; i < nb_samples; i++) {
const float in = src[i];
update_volume(cp, fabs(in));
if (count >= s->delay_samples) {
if (!out_frame) {
out_frame = ff_get_audio_buffer(inlink, nb_samples - i);
if (!out_frame) {
av_frame_free(&frame);
return AVERROR(ENOMEM);
}
err = av_frame_copy_props(out_frame, frame);
if (err < 0) {
av_frame_free(&out_frame);
av_frame_free(&frame);
return err;
}
out_frame->pts = s->pts;
s->pts += av_rescale_q(nb_samples - i,
(AVRational){ 1, inlink->sample_rate },
inlink->time_base);
}
dst = (float *)out_frame->extended_data[chan];
dst[oindex++] = av_clipf(dbuf[dindex] *
get_volume(s, cp->volume), -1.0f, 1.0f);
} else {
count++;
}
dbuf[dindex] = in;
dindex = MOD(dindex + 1, s->delay_samples);
}
}
s->delay_count = count;
s->delay_index = dindex;
av_frame_free(&frame);
return out_frame ? ff_filter_frame(ctx->outputs[0], out_frame) : 0;
}
static int compand_drain(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
CompandContext *s = ctx->priv;
const int channels = s->nb_channels;
AVFrame *frame = NULL;
int chan, i, dindex;
/* 2048 is to limit output frame size during drain */
frame = ff_get_audio_buffer(outlink, FFMIN(2048, s->delay_count));
if (!frame)
return AVERROR(ENOMEM);
frame->pts = s->pts;
s->pts += av_rescale_q(frame->nb_samples,
(AVRational){ 1, outlink->sample_rate }, outlink->time_base);
for (chan = 0; chan < channels; chan++) {
AVFrame *delay_frame = s->delay_frame;
float *dbuf = (float *)delay_frame->extended_data[chan];
float *dst = (float *)frame->extended_data[chan];
ChanParam *cp = &s->channels[chan];
dindex = s->delay_index;
for (i = 0; i < frame->nb_samples; i++) {
dst[i] = av_clipf(dbuf[dindex] * get_volume(s, cp->volume),
-1.0f, 1.0f);
dindex = MOD(dindex + 1, s->delay_samples);
}
}
s->delay_count -= frame->nb_samples;
s->delay_index = dindex;
return ff_filter_frame(outlink, frame);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
CompandContext *s = ctx->priv;
const int sample_rate = outlink->sample_rate;
double radius = s->curve_dB * M_LN10 / 20.0;
char *p, *saveptr = NULL;
const int channels =
av_get_channel_layout_nb_channels(outlink->channel_layout);
int nb_attacks, nb_decays, nb_points;
int new_nb_items, num;
int i;
int err;
count_items(s->attacks, &nb_attacks);
count_items(s->decays, &nb_decays);
count_items(s->points, &nb_points);
if (channels <= 0) {
av_log(ctx, AV_LOG_ERROR, "Invalid number of channels: %d\n", channels);
return AVERROR(EINVAL);
}
if (nb_attacks > channels || nb_decays > channels) {
av_log(ctx, AV_LOG_ERROR,
"Number of attacks/decays bigger than number of channels.\n");
return AVERROR(EINVAL);
}
uninit(ctx);
s->nb_channels = channels;
s->channels = av_mallocz_array(channels, sizeof(*s->channels));
s->nb_segments = (nb_points + 4) * 2;
s->segments = av_mallocz_array(s->nb_segments, sizeof(*s->segments));
if (!s->channels || !s->segments) {
uninit(ctx);
return AVERROR(ENOMEM);
}
p = s->attacks;
for (i = 0, new_nb_items = 0; i < nb_attacks; i++) {
char *tstr = av_strtok(p, "|", &saveptr);
p = NULL;
new_nb_items += sscanf(tstr, "%f", &s->channels[i].attack) == 1;
if (s->channels[i].attack < 0) {
uninit(ctx);
return AVERROR(EINVAL);
}
}
nb_attacks = new_nb_items;
p = s->decays;
for (i = 0, new_nb_items = 0; i < nb_decays; i++) {
char *tstr = av_strtok(p, "|", &saveptr);
p = NULL;
new_nb_items += sscanf(tstr, "%f", &s->channels[i].decay) == 1;
if (s->channels[i].decay < 0) {
uninit(ctx);
return AVERROR(EINVAL);
}
}
nb_decays = new_nb_items;
if (nb_attacks != nb_decays) {
av_log(ctx, AV_LOG_ERROR,
"Number of attacks %d differs from number of decays %d.\n",
nb_attacks, nb_decays);
uninit(ctx);
return AVERROR(EINVAL);
}
#define S(x) s->segments[2 * ((x) + 1)]
p = s->points;
for (i = 0, new_nb_items = 0; i < nb_points; i++) {
char *tstr = av_strtok(p, "|", &saveptr);
p = NULL;
if (sscanf(tstr, "%f/%f", &S(i).x, &S(i).y) != 2) {
av_log(ctx, AV_LOG_ERROR,
"Invalid and/or missing input/output value.\n");
uninit(ctx);
return AVERROR(EINVAL);
}
if (i && S(i - 1).x > S(i).x) {
av_log(ctx, AV_LOG_ERROR,
"Transfer function input values must be increasing.\n");
uninit(ctx);
return AVERROR(EINVAL);
}
S(i).y -= S(i).x;
av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
new_nb_items++;
}
num = new_nb_items;
/* Add 0,0 if necessary */
if (num == 0 || S(num - 1).x)
num++;
#undef S
#define S(x) s->segments[2 * (x)]
/* Add a tail off segment at the start */
S(0).x = S(1).x - 2 * s->curve_dB;
S(0).y = S(1).y;
num++;
/* Join adjacent colinear segments */
for (i = 2; i < num; i++) {
double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
int j;
/* here we purposefully lose precision so that we can compare floats */
if (fabs(g1 - g2))
continue;
num--;
for (j = --i; j < num; j++)
S(j) = S(j + 1);
}
for (i = 0; !i || s->segments[i - 2].x; i += 2) {
s->segments[i].y += s->gain_dB;
s->segments[i].x *= M_LN10 / 20;
s->segments[i].y *= M_LN10 / 20;
}
#define L(x) s->segments[i - (x)]
for (i = 4; s->segments[i - 2].x; i += 2) {
double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
L(4).a = 0;
L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
L(2).a = 0;
L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
len = sqrt(pow(L(2).x - L(4).x, 2.) + pow(L(2).y - L(4).y, 2.));
r = FFMIN(radius, len);
L(3).x = L(2).x - r * cos(theta);
L(3).y = L(2).y - r * sin(theta);
theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
len = sqrt(pow(L(0).x - L(2).x, 2.) + pow(L(0).y - L(2).y, 2.));
r = FFMIN(radius, len / 2);
x = L(2).x + r * cos(theta);
y = L(2).y + r * sin(theta);
cx = (L(3).x + L(2).x + x) / 3;
cy = (L(3).y + L(2).y + y) / 3;
L(2).x = x;
L(2).y = y;
in1 = cx - L(3).x;
out1 = cy - L(3).y;
in2 = L(2).x - L(3).x;
out2 = L(2).y - L(3).y;
L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
L(3).b = out1 / in1 - L(3).a * in1;
}
L(3).x = 0;
L(3).y = L(2).y;
s->in_min_lin = exp(s->segments[1].x);
s->out_min_lin = exp(s->segments[1].y);
for (i = 0; i < channels; i++) {
ChanParam *cp = &s->channels[i];
if (cp->attack > 1.0 / sample_rate)
cp->attack = 1.0 - exp(-1.0 / (sample_rate * cp->attack));
else
cp->attack = 1.0;
if (cp->decay > 1.0 / sample_rate)
cp->decay = 1.0 - exp(-1.0 / (sample_rate * cp->decay));
else
cp->decay = 1.0;
cp->volume = pow(10.0, s->initial_volume / 20);
}
s->delay_samples = s->delay * sample_rate;
if (s->delay_samples <= 0) {
s->compand = compand_nodelay;
return 0;
}
s->delay_frame = av_frame_alloc();
if (!s->delay_frame) {
uninit(ctx);
return AVERROR(ENOMEM);
}
s->delay_frame->format = outlink->format;
s->delay_frame->nb_samples = s->delay_samples;
s->delay_frame->channel_layout = outlink->channel_layout;
err = av_frame_get_buffer(s->delay_frame, 32);
if (err)
return err;
s->compand = compand_delay;
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
CompandContext *s = ctx->priv;
return s->compand(ctx, frame);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
CompandContext *s = ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && s->delay_count)
ret = compand_drain(outlink);
return ret;
}
static const AVFilterPad compand_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad compand_outputs[] = {
{
.name = "default",
.request_frame = request_frame,
.config_props = config_output,
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_compand_fork = {
.name = "compand_fork",
.description = NULL_IF_CONFIG_SMALL(
"Compress or expand audio dynamic range."),
.query_formats = query_formats,
.priv_size = sizeof(CompandContext),
.priv_class = &compand_class,
.init = init,
.uninit = uninit,
.inputs = compand_inputs,
.outputs = compand_outputs,
};

1
libavfilter/allfilters.c
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@ -83,7 +83,6 @@ void avfilter_register_all(void)
REGISTER_FILTER(CHANNELMAP, channelmap, af);
REGISTER_FILTER(CHANNELSPLIT, channelsplit, af);
REGISTER_FILTER(COMPAND, compand, af);
REGISTER_FILTER(COMPAND_FORK, compand_fork, af);
REGISTER_FILTER(EARWAX, earwax, af);
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);

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