ffmpeg/libavfilter/af_pan.c
Michael Niedermayer 1c60088885 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  x86: Only use optimizations with cmov if the CPU supports the instruction
  x86: Add CPU flag for the i686 cmov instruction
  x86: remove unused inline asm macros from dsputil_mmx.h
  x86: move some inline asm macros to the only places they are used
  lavfi: Add the af_channelmap audio channel mapping filter.
  lavfi: add join audio filter.
  lavfi: allow audio filters to request a given number of samples.
  lavfi: support automatically inserting the fifo filter when needed.
  lavfi/audio: eliminate ff_default_filter_samples().

Conflicts:
	Changelog
	libavcodec/x86/h264dsp_mmx.c
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/avfilter.h
	libavfilter/avfiltergraph.c
	libavfilter/version.h
	libavutil/x86/cpu.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-24 02:09:53 +02:00

388 lines
13 KiB
C

/*
* Copyright (c) 2002 Anders Johansson <ajh@atri.curtin.edu.au>
* Copyright (c) 2011 Clément Bœsch <ubitux@gmail.com>
* Copyright (c) 2011 Nicolas George <nicolas.george@normalesup.org>
*
* 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 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 panning filter (channels mixing)
* Original code written by Anders Johansson for MPlayer,
* reimplemented for FFmpeg.
*/
#include <stdio.h>
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#define MAX_CHANNELS 63
typedef struct PanContext {
int64_t out_channel_layout;
double gain[MAX_CHANNELS][MAX_CHANNELS];
int64_t need_renorm;
int need_renumber;
int nb_input_channels;
int nb_output_channels;
int pure_gains;
/* channel mapping specific */
int channel_map[SWR_CH_MAX];
struct SwrContext *swr;
} PanContext;
static int parse_channel_name(char **arg, int *rchannel, int *rnamed)
{
char buf[8];
int len, i, channel_id = 0;
int64_t layout, layout0;
/* try to parse a channel name, e.g. "FL" */
if (sscanf(*arg, " %7[A-Z] %n", buf, &len)) {
layout0 = layout = av_get_channel_layout(buf);
/* channel_id <- first set bit in layout */
for (i = 32; i > 0; i >>= 1) {
if (layout >= (int64_t)1 << i) {
channel_id += i;
layout >>= i;
}
}
/* reject layouts that are not a single channel */
if (channel_id >= MAX_CHANNELS || layout0 != (int64_t)1 << channel_id)
return AVERROR(EINVAL);
*rchannel = channel_id;
*rnamed = 1;
*arg += len;
return 0;
}
/* try to parse a channel number, e.g. "c2" */
if (sscanf(*arg, " c%d %n", &channel_id, &len) &&
channel_id >= 0 && channel_id < MAX_CHANNELS) {
*rchannel = channel_id;
*rnamed = 0;
*arg += len;
return 0;
}
return AVERROR(EINVAL);
}
static void skip_spaces(char **arg)
{
int len = 0;
sscanf(*arg, " %n", &len);
*arg += len;
}
static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
{
PanContext *const pan = ctx->priv;
char *arg, *arg0, *tokenizer, *args = av_strdup(args0);
int out_ch_id, in_ch_id, len, named, ret;
int nb_in_channels[2] = { 0, 0 }; // number of unnamed and named input channels
double gain;
if (!args0) {
av_log(ctx, AV_LOG_ERROR,
"pan filter needs a channel layout and a set "
"of channels definitions as parameter\n");
return AVERROR(EINVAL);
}
if (!args)
return AVERROR(ENOMEM);
arg = av_strtok(args, ":", &tokenizer);
ret = ff_parse_channel_layout(&pan->out_channel_layout, arg, ctx);
if (ret < 0)
return ret;
pan->nb_output_channels = av_get_channel_layout_nb_channels(pan->out_channel_layout);
/* parse channel specifications */
while ((arg = arg0 = av_strtok(NULL, ":", &tokenizer))) {
/* channel name */
if (parse_channel_name(&arg, &out_ch_id, &named)) {
av_log(ctx, AV_LOG_ERROR,
"Expected out channel name, got \"%.8s\"\n", arg);
return AVERROR(EINVAL);
}
if (named) {
if (!((pan->out_channel_layout >> out_ch_id) & 1)) {
av_log(ctx, AV_LOG_ERROR,
"Channel \"%.8s\" does not exist in the chosen layout\n", arg0);
return AVERROR(EINVAL);
}
/* get the channel number in the output channel layout:
* out_channel_layout & ((1 << out_ch_id) - 1) are all the
* channels that come before out_ch_id,
* so their count is the index of out_ch_id */
out_ch_id = av_get_channel_layout_nb_channels(pan->out_channel_layout & (((int64_t)1 << out_ch_id) - 1));
}
if (out_ch_id < 0 || out_ch_id >= pan->nb_output_channels) {
av_log(ctx, AV_LOG_ERROR,
"Invalid out channel name \"%.8s\"\n", arg0);
return AVERROR(EINVAL);
}
if (*arg == '=') {
arg++;
} else if (*arg == '<') {
pan->need_renorm |= (int64_t)1 << out_ch_id;
arg++;
} else {
av_log(ctx, AV_LOG_ERROR,
"Syntax error after channel name in \"%.8s\"\n", arg0);
return AVERROR(EINVAL);
}
/* gains */
while (1) {
gain = 1;
if (sscanf(arg, " %lf %n* %n", &gain, &len, &len))
arg += len;
if (parse_channel_name(&arg, &in_ch_id, &named)){
av_log(ctx, AV_LOG_ERROR,
"Expected in channel name, got \"%.8s\"\n", arg);
return AVERROR(EINVAL);
}
nb_in_channels[named]++;
if (nb_in_channels[!named]) {
av_log(ctx, AV_LOG_ERROR,
"Can not mix named and numbered channels\n");
return AVERROR(EINVAL);
}
pan->gain[out_ch_id][in_ch_id] = gain;
if (!*arg)
break;
if (*arg != '+') {
av_log(ctx, AV_LOG_ERROR, "Syntax error near \"%.8s\"\n", arg);
return AVERROR(EINVAL);
}
arg++;
skip_spaces(&arg);
}
}
pan->need_renumber = !!nb_in_channels[1];
av_free(args);
return 0;
}
static int are_gains_pure(const PanContext *pan)
{
int i, j;
for (i = 0; i < MAX_CHANNELS; i++) {
int nb_gain = 0;
for (j = 0; j < MAX_CHANNELS; j++) {
double gain = pan->gain[i][j];
/* channel mapping is effective only if 0% or 100% of a channel is
* selected... */
if (gain != 0. && gain != 1.)
return 0;
/* ...and if the output channel is only composed of one input */
if (gain && nb_gain++)
return 0;
}
}
return 1;
}
static int query_formats(AVFilterContext *ctx)
{
PanContext *pan = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts;
pan->pure_gains = are_gains_pure(pan);
/* libswr supports any sample and packing formats */
ff_set_common_formats(ctx, ff_all_formats(AVMEDIA_TYPE_AUDIO));
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_samplerates(ctx, formats);
// inlink supports any channel layout
layouts = ff_all_channel_layouts();
ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts);
// outlink supports only requested output channel layout
layouts = NULL;
ff_add_channel_layout(&layouts, pan->out_channel_layout);
ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts);
return 0;
}
static int config_props(AVFilterLink *link)
{
AVFilterContext *ctx = link->dst;
PanContext *pan = ctx->priv;
char buf[1024], *cur;
int i, j, k, r;
double t;
pan->nb_input_channels = av_get_channel_layout_nb_channels(link->channel_layout);
if (pan->need_renumber) {
// input channels were given by their name: renumber them
for (i = j = 0; i < MAX_CHANNELS; i++) {
if ((link->channel_layout >> i) & 1) {
for (k = 0; k < pan->nb_output_channels; k++)
pan->gain[k][j] = pan->gain[k][i];
j++;
}
}
}
// sanity check; can't be done in query_formats since the inlink
// channel layout is unknown at that time
if (pan->nb_input_channels > SWR_CH_MAX ||
pan->nb_output_channels > SWR_CH_MAX) {
av_log(ctx, AV_LOG_ERROR,
"libswresample support a maximum of %d channels. "
"Feel free to ask for a higher limit.\n", SWR_CH_MAX);
return AVERROR_PATCHWELCOME;
}
// init libswresample context
pan->swr = swr_alloc_set_opts(pan->swr,
pan->out_channel_layout, link->format, link->sample_rate,
link->channel_layout, link->format, link->sample_rate,
0, ctx);
if (!pan->swr)
return AVERROR(ENOMEM);
// gains are pure, init the channel mapping
if (pan->pure_gains) {
// get channel map from the pure gains
for (i = 0; i < pan->nb_output_channels; i++) {
int ch_id = -1;
for (j = 0; j < pan->nb_input_channels; j++) {
if (pan->gain[i][j]) {
ch_id = j;
break;
}
}
pan->channel_map[i] = ch_id;
}
av_opt_set_int(pan->swr, "icl", pan->out_channel_layout, 0);
av_opt_set_int(pan->swr, "uch", pan->nb_output_channels, 0);
swr_set_channel_mapping(pan->swr, pan->channel_map);
} else {
// renormalize
for (i = 0; i < pan->nb_output_channels; i++) {
if (!((pan->need_renorm >> i) & 1))
continue;
t = 0;
for (j = 0; j < pan->nb_input_channels; j++)
t += pan->gain[i][j];
if (t > -1E-5 && t < 1E-5) {
// t is almost 0 but not exactly, this is probably a mistake
if (t)
av_log(ctx, AV_LOG_WARNING,
"Degenerate coefficients while renormalizing\n");
continue;
}
for (j = 0; j < pan->nb_input_channels; j++)
pan->gain[i][j] /= t;
}
av_opt_set_int(pan->swr, "icl", link->channel_layout, 0);
av_opt_set_int(pan->swr, "ocl", pan->out_channel_layout, 0);
swr_set_matrix(pan->swr, pan->gain[0], pan->gain[1] - pan->gain[0]);
}
r = swr_init(pan->swr);
if (r < 0)
return r;
// summary
for (i = 0; i < pan->nb_output_channels; i++) {
cur = buf;
for (j = 0; j < pan->nb_input_channels; j++) {
r = snprintf(cur, buf + sizeof(buf) - cur, "%s%.3g i%d",
j ? " + " : "", pan->gain[i][j], j);
cur += FFMIN(buf + sizeof(buf) - cur, r);
}
av_log(ctx, AV_LOG_INFO, "o%d = %s\n", i, buf);
}
// add channel mapping summary if possible
if (pan->pure_gains) {
av_log(ctx, AV_LOG_INFO, "Pure channel mapping detected:");
for (i = 0; i < pan->nb_output_channels; i++)
if (pan->channel_map[i] < 0)
av_log(ctx, AV_LOG_INFO, " M");
else
av_log(ctx, AV_LOG_INFO, " %d", pan->channel_map[i]);
av_log(ctx, AV_LOG_INFO, "\n");
return 0;
}
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int n = insamples->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
PanContext *pan = inlink->dst->priv;
swr_convert(pan->swr, outsamples->data, n, (void *)insamples->data, n);
avfilter_copy_buffer_ref_props(outsamples, insamples);
outsamples->audio->channel_layout = outlink->channel_layout;
ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
}
static av_cold void uninit(AVFilterContext *ctx)
{
PanContext *pan = ctx->priv;
swr_free(&pan->swr);
}
AVFilter avfilter_af_pan = {
.name = "pan",
.description = NULL_IF_CONFIG_SMALL("Remix channels with coefficients (panning)."),
.priv_size = sizeof(PanContext),
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL}
},
.outputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}
},
};