/*
  *
  * 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
  * sample format and channel layout conversion audio filter
  */
 
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"

 #include "libavutil/common.h"

 #include "libavutil/mathematics.h"
 #include "libavutil/opt.h"
 
 #include "libavresample/avresample.h"
 
 #include "audio.h"
 #include "avfilter.h"

 #include "formats.h"

 #include "internal.h"
 
 typedef struct ResampleContext {
     AVAudioResampleContext *avr;
 
     int64_t next_pts;

     /* set by filter_frame() to signal an output frame to request_frame() */

     int got_output;

 } ResampleContext;
 
 static av_cold void uninit(AVFilterContext *ctx)
 {
     ResampleContext *s = ctx->priv;
 
     if (s->avr) {
         avresample_close(s->avr);
         avresample_free(&s->avr);
     }
 }
 
 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterLink *inlink  = ctx->inputs[0];
     AVFilterLink *outlink = ctx->outputs[0];
 

     AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
     AVFilterFormats        *out_formats     = ff_all_formats(AVMEDIA_TYPE_AUDIO);

     AVFilterFormats        *in_samplerates  = ff_all_samplerates();
     AVFilterFormats        *out_samplerates = ff_all_samplerates();
     AVFilterChannelLayouts *in_layouts      = ff_all_channel_layouts();
     AVFilterChannelLayouts *out_layouts     = ff_all_channel_layouts();

     ff_formats_ref(in_formats,  &inlink->out_formats);
     ff_formats_ref(out_formats, &outlink->in_formats);

     ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
     ff_formats_ref(out_samplerates, &outlink->in_samplerates);

 
     ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
     ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
 

     return 0;
 }
 
 static int config_output(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     AVFilterLink *inlink = ctx->inputs[0];
     ResampleContext   *s = ctx->priv;
     char buf1[64], buf2[64];
     int ret;
 
     if (s->avr) {
         avresample_close(s->avr);
         avresample_free(&s->avr);
     }
 
     if (inlink->channel_layout == outlink->channel_layout &&
         inlink->sample_rate    == outlink->sample_rate    &&

         (inlink->format        == outlink->format ||
         (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&
          av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
          av_get_planar_sample_fmt(inlink->format) ==
          av_get_planar_sample_fmt(outlink->format))))

         return 0;
 
     if (!(s->avr = avresample_alloc_context()))
         return AVERROR(ENOMEM);
 
     av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
     av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
     av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
     av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
     av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
     av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);
 
     if ((ret = avresample_open(s->avr)) < 0)
         return ret;
 
     outlink->time_base = (AVRational){ 1, outlink->sample_rate };
     s->next_pts        = AV_NOPTS_VALUE;
 
     av_get_channel_layout_string(buf1, sizeof(buf1),
                                  -1, inlink ->channel_layout);
     av_get_channel_layout_string(buf2, sizeof(buf2),
                                  -1, outlink->channel_layout);
     av_log(ctx, AV_LOG_VERBOSE,

            "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",

            av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
            av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
 
     return 0;
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     AVFilterContext *ctx = outlink->src;
     ResampleContext   *s = ctx->priv;

     int ret = 0;
 
     s->got_output = 0;
     while (ret >= 0 && !s->got_output)
         ret = ff_request_frame(ctx->inputs[0]);

 
     /* flush the lavr delay buffer */
     if (ret == AVERROR_EOF && s->avr) {
         AVFilterBufferRef *buf;
         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
                                         outlink->sample_rate,
                                         ctx->inputs[0]->sample_rate,
                                         AV_ROUND_UP);
 
         if (!nb_samples)
             return ret;
 
         buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
         if (!buf)
             return AVERROR(ENOMEM);
 

         ret = avresample_convert(s->avr, buf->extended_data,

                                  buf->linesize[0], nb_samples,
                                  NULL, 0, 0);
         if (ret <= 0) {
             avfilter_unref_buffer(buf);
             return (ret == 0) ? AVERROR_EOF : ret;
         }
 
         buf->pts = s->next_pts;

         return ff_filter_frame(outlink, buf);

     }
     return ret;
 }
 

 static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)

 {
     AVFilterContext  *ctx = inlink->dst;
     ResampleContext    *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];

     int ret;

 
     if (s->avr) {
         AVFilterBufferRef *buf_out;

         int delay, nb_samples;

 
         /* maximum possible samples lavr can output */
         delay      = avresample_get_delay(s->avr);
         nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
                                     outlink->sample_rate, inlink->sample_rate,
                                     AV_ROUND_UP);
 
         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);

         if (!buf_out) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }
 

         ret     = avresample_convert(s->avr, buf_out->extended_data,

                                      buf_out->linesize[0], nb_samples,

                                      buf->extended_data, buf->linesize[0],

                                      buf->audio->nb_samples);

         if (ret <= 0) {

             avfilter_unref_buffer(buf_out);

             if (ret < 0)
                 goto fail;

         }

 
         av_assert0(!avresample_available(s->avr));
 
         if (s->next_pts == AV_NOPTS_VALUE) {
             if (buf->pts == AV_NOPTS_VALUE) {
                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
                        "assuming 0.\n");
                 s->next_pts = 0;
             } else
                 s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
                                            outlink->time_base);
         }
 
         if (ret > 0) {
             buf_out->audio->nb_samples = ret;
             if (buf->pts != AV_NOPTS_VALUE) {
                 buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
                                             outlink->time_base) -
                                av_rescale(delay, outlink->sample_rate,
                                           inlink->sample_rate);
             } else
                 buf_out->pts = s->next_pts;
 
             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
 

             ret = ff_filter_frame(outlink, buf_out);

             s->got_output = 1;

         }

 
 fail:

         avfilter_unref_buffer(buf);

     } else {

         buf->format = outlink->format;

         ret = ff_filter_frame(outlink, buf);

         s->got_output = 1;
     }

 
     return ret;

 }
 

 static const AVFilterPad avfilter_af_resample_inputs[] = {
     {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,

         .filter_frame   = filter_frame,

         .min_perms      = AV_PERM_READ
     },
     { NULL }
 };
 
 static const AVFilterPad avfilter_af_resample_outputs[] = {
     {
         .name          = "default",
         .type          = AVMEDIA_TYPE_AUDIO,
         .config_props  = config_output,
         .request_frame = request_frame
     },
     { NULL }
 };
 

 AVFilter avfilter_af_resample = {
     .name          = "resample",
     .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
     .priv_size     = sizeof(ResampleContext),
 
     .uninit         = uninit,
     .query_formats  = query_formats,
 

     .inputs    = avfilter_af_resample_inputs,
     .outputs   = avfilter_af_resample_outputs,

 };