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/*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* HDCD decoding filter, using libhdcd
*/
#include <hdcd/hdcd_simple.h>
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
typedef struct HDCDContext {
const AVClass *class;
hdcd_simple *shdcd;
/* AVOption members */
/** analyze mode replaces the audio with a solid tone and adjusts
* the amplitude to signal some specific aspect of the decoding
* process. See docs or HDCD_ANA_* defines. */
int analyze_mode;
/* end AVOption members */
} HDCDContext;
#define OFFSET(x) offsetof(HDCDContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define HDCD_ANA_MAX 6
static const AVOption hdcd_options[] = {
{ "analyze_mode", "Replace audio with solid tone and signal some processing aspect in the amplitude.",
OFFSET(analyze_mode), AV_OPT_TYPE_INT, { .i64=HDCD_ANA_OFF }, 0, HDCD_ANA_MAX, A, "analyze_mode"},
{ "off", HDCD_ANA_OFF_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_OFF}, 0, 0, A, "analyze_mode" },
{ "lle", HDCD_ANA_LLE_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_LLE}, 0, 0, A, "analyze_mode" },
{ "pe", HDCD_ANA_PE_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_PE}, 0, 0, A, "analyze_mode" },
{ "cdt", HDCD_ANA_CDT_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_CDT}, 0, 0, A, "analyze_mode" },
{ "tgm", HDCD_ANA_TGM_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_TGM}, 0, 0, A, "analyze_mode" },
{ "pel", HDCD_ANA_PEL_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_PEL}, 0, 0, A, "analyze_mode" },
{ "ltgm", HDCD_ANA_LTGM_DESC, 0, AV_OPT_TYPE_CONST, { .i64 = HDCD_ANA_LTGM}, 0, 0, A, "analyze_mode" },
{ NULL }
};
static const AVClass hdcd_class = {
.class_name = "HDCD filter",
.item_name = av_default_item_name,
.option = hdcd_options,
.version = LIBAVFILTER_VERSION_INT,
};
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
HDCDContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
const int16_t *in_data;
int32_t *out_data;
int n, result;
int channel_count = av_get_channel_layout_nb_channels(in->channel_layout);
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
result = av_frame_copy_props(out, in);
if (result) {
av_frame_free(&out);
av_frame_free(&in);
return result;
}
in_data = (int16_t *)in->data[0];
out_data = (int32_t *)out->data[0];
for (n = 0; n < in->nb_samples * channel_count; n++)
out_data[n] = in_data[n];
hdcd_process(s->shdcd, out_data, in->nb_samples);
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *in_formats, *out_formats, *sample_rates = NULL;
AVFilterChannelLayouts *layouts = NULL;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
static const enum AVSampleFormat sample_fmts_in[] = {
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE
};
static const enum AVSampleFormat sample_fmts_out[] = {
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_NONE
};
ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
ff_set_common_channel_layouts(ctx, layouts);
in_formats = ff_make_format_list(sample_fmts_in);
out_formats = ff_make_format_list(sample_fmts_out);
if (!in_formats || !out_formats)
return AVERROR(ENOMEM);
ff_formats_ref(in_formats, &inlink->out_formats);
ff_formats_ref(out_formats, &outlink->in_formats);
ff_add_format(&sample_rates, 44100);
ff_set_common_samplerates(ctx, sample_rates);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
HDCDContext *s = ctx->priv;
char detect_str[256] = "";
/* log the HDCD decode information */
hdcd_detect_str(s->shdcd, detect_str, sizeof(detect_str));
av_log(ctx, AV_LOG_INFO, "%s\n", detect_str);
hdcd_free(s->shdcd);
}
/** callback for error logging */
static void af_hdcd_log(const void *priv, const char *fmt, va_list args)
{
av_vlog((AVFilterContext *)priv, AV_LOG_VERBOSE, fmt, args);
}
static av_cold int init(AVFilterContext *ctx)
{
HDCDContext *s = ctx->priv;
s->shdcd = hdcd_new();
hdcd_logger_attach(s->shdcd, af_hdcd_log, ctx);
if (s->analyze_mode)
hdcd_analyze_mode(s->shdcd, s->analyze_mode);
av_log(ctx, AV_LOG_VERBOSE, "Analyze mode: [%d] %s\n",
s->analyze_mode, hdcd_str_analyze_mode_desc(s->analyze_mode));
return 0;
}
static const AVFilterPad avfilter_af_hdcd_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad avfilter_af_hdcd_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_hdcd = {
.name = "hdcd",
.description = NULL_IF_CONFIG_SMALL("Apply High Definition Compatible Digital (HDCD) decoding."),
.priv_size = sizeof(HDCDContext),
.priv_class = &hdcd_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = avfilter_af_hdcd_inputs,
.outputs = avfilter_af_hdcd_outputs,
};
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