/*
  * Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
  * Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
  *
  * 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
  */
 
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"
 
 typedef struct DCShiftContext {
     const AVClass *class;
     double dcshift;

     double limiterthreshold;

     double limitergain;
 } DCShiftContext;
 
 #define OFFSET(x) offsetof(DCShiftContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption dcshift_options[] = {
     { "shift",       "set DC shift",     OFFSET(dcshift),       AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
     { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
     { NULL }
 };
 
 AVFILTER_DEFINE_CLASS(dcshift);
 
 static av_cold int init(AVFilterContext *ctx)
 {
     DCShiftContext *s = ctx->priv;
 

     s->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));

 
     return 0;
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layouts;
     AVFilterFormats *formats;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
     };

     int ret;

     layouts = ff_all_channel_counts();

     if (!layouts)
         return AVERROR(ENOMEM);

     ret = ff_set_common_channel_layouts(ctx, layouts);
     if (ret < 0)
         return ret;

 
     formats = ff_make_format_list(sample_fmts);
     if (!formats)
         return AVERROR(ENOMEM);

     ret = ff_set_common_formats(ctx, formats);
     if (ret < 0)
         return ret;

 
     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);

     return ff_set_common_samplerates(ctx, formats);

 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];

     AVFrame *out;

     DCShiftContext *s = ctx->priv;
     int i, j;
     double dcshift = s->dcshift;
 

     if (av_frame_is_writable(in)) {
         out = in;
     } else {

         out = ff_get_audio_buffer(outlink, in->nb_samples);

         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out, in);

     }
 
     if (s->limitergain > 0) {
         for (i = 0; i < inlink->channels; i++) {
             const int32_t *src = (int32_t *)in->extended_data[i];
             int32_t *dst = (int32_t *)out->extended_data[i];
 
             for (j = 0; j < in->nb_samples; j++) {
                 double d;
 
                 d = src[j];
 

                 if (d > s->limiterthreshold && dcshift > 0) {
                     d = (d - s->limiterthreshold) * s->limitergain /
                              (INT32_MAX - s->limiterthreshold) +
                              s->limiterthreshold + dcshift;
                 } else if (d < -s->limiterthreshold && dcshift < 0) {
                     d = (d + s->limiterthreshold) * s->limitergain /
                              (INT32_MAX - s->limiterthreshold) -
                              s->limiterthreshold + dcshift;

                 } else {
                     d = dcshift * INT32_MAX + d;
                 }
 
                 dst[j] = av_clipl_int32(d);
             }
         }
     } else {
         for (i = 0; i < inlink->channels; i++) {
             const int32_t *src = (int32_t *)in->extended_data[i];
             int32_t *dst = (int32_t *)out->extended_data[i];
 
             for (j = 0; j < in->nb_samples; j++) {
                 double d = dcshift * (INT32_MAX + 1.) + src[j];
 
                 dst[j] = av_clipl_int32(d);
             }
         }
     }
 

     if (out != in)
         av_frame_free(&in);

     return ff_filter_frame(outlink, out);
 }
 static const AVFilterPad dcshift_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad dcshift_outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };
 
 AVFilter ff_af_dcshift = {
     .name           = "dcshift",
     .description    = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
     .query_formats  = query_formats,
     .priv_size      = sizeof(DCShiftContext),
     .priv_class     = &dcshift_class,
     .init           = init,
     .inputs         = dcshift_inputs,
     .outputs        = dcshift_outputs,
     .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
 };