mirror of
https://github.com/xenia-project/FFmpeg.git
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307 lines
10 KiB
C
307 lines
10 KiB
C
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/*
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* Copyright (c) 2002 Anders Johansson <ajh@atri.curtin.edu.au>
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* Copyright (c) 2011 Clément Bœsch <ubitux@gmail.com>
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* Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Audio panning filter (channels mixing)
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* Original code written by Anders Johansson for MPlayer,
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* reimplemented for FFmpeg.
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*/
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#include <stdio.h>
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#include "libavutil/audioconvert.h"
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#include "libavutil/avstring.h"
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#include "avfilter.h"
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#include "internal.h"
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#define MAX_CHANNELS 63
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typedef struct {
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int64_t out_channel_layout;
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union {
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double d[MAX_CHANNELS][MAX_CHANNELS];
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// i is 1:7:8 fixed-point, i.e. in [-128*256; +128*256[
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int i[MAX_CHANNELS][MAX_CHANNELS];
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} gain;
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int64_t need_renorm;
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int need_renumber;
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int nb_input_channels;
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int nb_output_channels;
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} PanContext;
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static int parse_channel_name(char **arg, int *rchannel, int *rnamed)
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{
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char buf[8];
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int len, i, channel_id;
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int64_t layout, layout0;
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if (sscanf(*arg, " %7[A-Z] %n", buf, &len)) {
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layout0 = layout = av_get_channel_layout(buf);
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for (i = 32; i > 0; i >>= 1) {
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if (layout >= (int64_t)1 << i) {
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channel_id += i;
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layout >>= i;
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}
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}
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if (channel_id >= MAX_CHANNELS || layout0 != (int64_t)1 << channel_id)
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return AVERROR(EINVAL);
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*rchannel = channel_id;
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*rnamed = 1;
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*arg += len;
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return 0;
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}
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if (sscanf(*arg, " c%d %n", &channel_id, &len) &&
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channel_id >= 0 && channel_id < MAX_CHANNELS) {
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*rchannel = channel_id;
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*rnamed = 0;
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*arg += len;
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return 0;
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}
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return AVERROR(EINVAL);
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}
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static void skip_spaces(char **arg)
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{
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int len = 0;
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sscanf(*arg, " %n", &len);
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*arg += len;
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}
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static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
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{
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PanContext *const pan = ctx->priv;
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char *arg, *arg0, *tokenizer, *args = av_strdup(args0);
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int out_ch_id, in_ch_id, len, named;
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int nb_in_channels[2] = { 0, 0 }; // number of unnamed and named input channels
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double gain;
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if (!args)
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return AVERROR(ENOMEM);
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arg = av_strtok(args, ":", &tokenizer);
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pan->out_channel_layout = av_get_channel_layout(arg);
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if (!pan->out_channel_layout) {
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av_log(ctx, AV_LOG_ERROR, "Unknown channel layout \"%s\"\n", arg);
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return AVERROR(EINVAL);
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}
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pan->nb_output_channels = av_get_channel_layout_nb_channels(pan->out_channel_layout);
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/* parse channel specifications */
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while ((arg = arg0 = av_strtok(NULL, ":", &tokenizer))) {
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/* channel name */
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if (parse_channel_name(&arg, &out_ch_id, &named)) {
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av_log(ctx, AV_LOG_ERROR,
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"Expected out channel name, got \"%.8s\"\n", arg);
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return AVERROR(EINVAL);
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}
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if (named) {
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if (!((pan->out_channel_layout >> out_ch_id) & 1)) {
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av_log(ctx, AV_LOG_ERROR,
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"Channel \"%.8s\" does not exist in the chosen layout\n", arg0);
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return AVERROR(EINVAL);
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}
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/* get the channel number in the output channel layout:
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* out_channel_layout & ((1 << out_ch_id) - 1) are all the
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* channels that come before out_ch_id,
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* so their count is the index of out_ch_id */
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out_ch_id = av_get_channel_layout_nb_channels(pan->out_channel_layout & (((int64_t)1 << out_ch_id) - 1));
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}
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if (out_ch_id < 0 || out_ch_id >= pan->nb_output_channels) {
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av_log(ctx, AV_LOG_ERROR,
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"Invalid out channel name \"%.8s\"\n", arg0);
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return AVERROR(EINVAL);
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}
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if (*arg == '=') {
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arg++;
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} else if (*arg == '<') {
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pan->need_renorm |= (int64_t)1 << out_ch_id;
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arg++;
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} else {
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av_log(ctx, AV_LOG_ERROR,
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"Syntax error after channel name in \"%.8s\"\n", arg0);
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return AVERROR(EINVAL);
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}
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/* gains */
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while (1) {
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gain = 1;
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if (sscanf(arg, " %lf %n* %n", &gain, &len, &len))
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arg += len;
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if (parse_channel_name(&arg, &in_ch_id, &named)){
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av_log(ctx, AV_LOG_ERROR,
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"Expected in channel name, got \"%.8s\"\n", arg);
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return AVERROR(EINVAL);
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}
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nb_in_channels[named]++;
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if (nb_in_channels[!named]) {
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av_log(ctx, AV_LOG_ERROR,
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"Can not mix named and numbered channels\n");
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return AVERROR(EINVAL);
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}
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pan->gain.d[out_ch_id][in_ch_id] = gain;
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if (!*arg)
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break;
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if (*arg != '+') {
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av_log(ctx, AV_LOG_ERROR, "Syntax error near \"%.8s\"\n", arg);
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return AVERROR(EINVAL);
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}
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arg++;
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skip_spaces(&arg);
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}
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}
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pan->need_renumber = !!nb_in_channels[1];
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av_free(args);
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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PanContext *pan = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AVFilterFormats *formats;
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const enum AVSampleFormat sample_fmts[] = {AV_SAMPLE_FMT_S16, -1};
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const int packing_fmts[] = {AVFILTER_PACKED, -1};
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avfilter_set_common_sample_formats (ctx, avfilter_make_format_list(sample_fmts));
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avfilter_set_common_packing_formats(ctx, avfilter_make_format_list(packing_fmts));
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// inlink supports any channel layout
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formats = avfilter_make_all_channel_layouts();
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avfilter_formats_ref(formats, &inlink->out_chlayouts);
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// outlink supports only requested output channel layout
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formats = NULL;
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avfilter_add_format(&formats, pan->out_channel_layout);
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avfilter_formats_ref(formats, &outlink->in_chlayouts);
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return 0;
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}
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static int config_props(AVFilterLink *link)
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{
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AVFilterContext *ctx = link->dst;
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PanContext *pan = ctx->priv;
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char buf[1024], *cur;
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int i, j, k, r;
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double t;
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pan->nb_input_channels = av_get_channel_layout_nb_channels(link->channel_layout);
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if (pan->need_renumber) {
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// input channels were given by their name: renumber them
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for (i = j = 0; i < MAX_CHANNELS; i++) {
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if ((link->channel_layout >> i) & 1) {
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for (k = 0; k < pan->nb_output_channels; k++)
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pan->gain.d[k][j] = pan->gain.d[k][i];
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j++;
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}
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}
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}
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// renormalize
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for (i = 0; i < pan->nb_output_channels; i++) {
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if (!((pan->need_renorm >> i) & 1))
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continue;
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t = 0;
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for (j = 0; j < pan->nb_input_channels; j++)
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t += pan->gain.d[i][j];
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if (t > -1E-5 && t < 1E-5) {
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// t is almost 0 but not exactly, this is probably a mistake
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if (t)
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av_log(ctx, AV_LOG_WARNING,
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"Degenerate coefficients while renormalizing\n");
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continue;
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}
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for (j = 0; j < pan->nb_input_channels; j++)
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pan->gain.d[i][j] /= t;
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}
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// summary
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for (i = 0; i < pan->nb_output_channels; i++) {
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cur = buf;
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for (j = 0; j < pan->nb_input_channels; j++) {
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r = snprintf(cur, buf + sizeof(buf) - cur, "%s%.3g i%d",
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j ? " + " : "", pan->gain.d[i][j], j);
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cur += FFMIN(buf + sizeof(buf) - cur, r);
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}
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av_log(ctx, AV_LOG_INFO, "o%d = %s\n", i, buf);
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}
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// convert to integer
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for (i = 0; i < pan->nb_output_channels; i++) {
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for (j = 0; j < pan->nb_input_channels; j++) {
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if (pan->gain.d[i][j] < -128 || pan->gain.d[i][j] > 128)
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av_log(ctx, AV_LOG_WARNING,
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"Gain #%d->#%d too large, clamped\n", j, i);
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pan->gain.i[i][j] = av_clipf(pan->gain.d[i][j], -128, 128) * 256.0;
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}
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}
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return 0;
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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PanContext *const pan = inlink->dst->priv;
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int i, o, n = insamples->audio->nb_samples;
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/* input */
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const int16_t *in = (int16_t *)insamples->data[0];
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const int16_t *in_end = in + n * pan->nb_input_channels;
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/* output */
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AVFilterLink *const outlink = inlink->dst->outputs[0];
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AVFilterBufferRef *outsamples = avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, n);
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int16_t *out = (int16_t *)outsamples->data[0];
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for (; in < in_end; in += pan->nb_input_channels) {
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for (o = 0; o < pan->nb_output_channels; o++) {
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int v = 0;
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for (i = 0; i < pan->nb_input_channels; i++)
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v += pan->gain.i[o][i] * in[i];
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*(out++) = v >> 8;
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}
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}
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avfilter_filter_samples(outlink, outsamples);
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avfilter_unref_buffer(insamples);
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}
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AVFilter avfilter_af_pan = {
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.name = "pan",
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.description = NULL_IF_CONFIG_SMALL("Remix channels with coefficients (panning)"),
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.priv_size = sizeof(PanContext),
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.init = init,
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.query_formats = query_formats,
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.inputs = (const AVFilterPad[]) {
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{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_props,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}
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},
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.outputs = (const AVFilterPad[]) {
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{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO, },
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{ .name = NULL}
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},
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};
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