[FFmpeg-cvslog] avfilter: remove af_compand_fork
Michael Niedermayer
git at videolan.org
Wed Feb 26 13:22:06 CET 2014
ffmpeg | branch: master | Michael Niedermayer <michaelni at gmx.at> | Wed Feb 26 13:13:50 2014 +0100| [8f946ac8ea108b8865cc5bd16af3ab063f8003ba] | committer: Michael Niedermayer
avfilter: remove af_compand_fork
All relevant changes except the float/double one have been merged
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=8f946ac8ea108b8865cc5bd16af3ab063f8003ba
---
libavfilter/Makefile | 1 -
libavfilter/af_compand_fork.c | 586 -----------------------------------------
libavfilter/allfilters.c | 1 -
3 files changed, 588 deletions(-)
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index b3bc27c..c36b67e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/af_compand_fork.c b/libavfilter/af_compand_fork.c
deleted file mode 100644
index ae366e6..0000000
--- a/libavfilter/af_compand_fork.c
+++ /dev/null
@@ -1,586 +0,0 @@
-/*
- * Copyright (c) 1999 Chris Bagwell
- * Copyright (c) 1999 Nick Bailey
- * Copyright (c) 2007 Rob Sykes <robs at 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,
-};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index d9a9975..d042b64 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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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