/*
  * Copyright (c) 2013 Nicolas George
  *
  * 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 "avfilter.h"
 #include "internal.h"
 
 typedef struct {
     const AVClass *class;
     int sample_rate;
     int rescale_pts;
 } ASetRateContext;
 
 #define CONTEXT ASetRateContext
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 #define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
     { name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type,          \
       { .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
 
 #define OPT_INT(name, field, def, min, max, descr, ...) \
     OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
 
 static const AVOption asetrate_options[] = {

     OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
     OPT_INT("r",           sample_rate, 44100, 1, INT_MAX, "set the sample rate",),

     {NULL},
 };
 
 AVFILTER_DEFINE_CLASS(asetrate);
 
 static av_cold int query_formats(AVFilterContext *ctx)
 {
     ASetRateContext *sr = ctx->priv;
     int sample_rates[] = { sr->sample_rate, -1 };
 

     return ff_formats_ref(ff_make_format_list(sample_rates),

                    &ctx->outputs[0]->in_samplerates);
 }
 
 static av_cold int config_props(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     ASetRateContext *sr = ctx->priv;
     AVFilterLink *inlink = ctx->inputs[0];
     AVRational intb = ctx->inputs[0]->time_base;
     int inrate = inlink->sample_rate;
 
     if (intb.num == 1 && intb.den == inrate) {
         outlink->time_base.num = 1;
         outlink->time_base.den = outlink->sample_rate;
     } else {
         outlink->time_base = intb;
         sr->rescale_pts = 1;
         if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
             av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
     }
     return 0;
 }
 
 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     ASetRateContext *sr = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
 
     frame->sample_rate = outlink->sample_rate;
     if (sr->rescale_pts)
         frame->pts = av_rescale(frame->pts, inlink->sample_rate,
                                            outlink->sample_rate);
     return ff_filter_frame(outlink, frame);
 }
 
 static const AVFilterPad asetrate_inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .filter_frame = filter_frame,
     },
     { NULL }
 };
 
 static const AVFilterPad asetrate_outputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_props,
     },
     { NULL }
 };
 

 AVFilter ff_af_asetrate = {

     .name          = "asetrate",
     .description   = NULL_IF_CONFIG_SMALL("Change the sample rate without "
                                           "altering the data."),
     .query_formats = query_formats,
     .priv_size     = sizeof(ASetRateContext),
     .inputs        = asetrate_inputs,
     .outputs       = asetrate_outputs,
     .priv_class    = &asetrate_class,
 };