/*
  * audio resampling with soxr

  * Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.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
  */
 
 /**
  * @file
  * audio resampling with soxr
  */
 
 #include "libavutil/log.h"
 #include "swresample_internal.h"
 
 #include <soxr.h>
 
 static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
         double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){
     soxr_error_t error;
 
     soxr_datatype_t type =
         format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S :
         format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I :
         format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S :
         format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I :
         format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S :
         format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I :
         format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S :
         format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1;
 
     soxr_io_spec_t io_spec = soxr_io_spec(type, type);
 
     soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby);

     q_spec.precision = linear? 0 : precision;

 #if !defined SOXR_VERSION /* Deprecated @ March 2013: */

     q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;

 #else
     q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end;
 #endif

 
     soxr_delete((soxr_t)c);
     c = (struct ResampleContext *)
         soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
     if (!c)
         av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
     return c;
 }
 
 static void destroy(struct ResampleContext * *c){
     soxr_delete((soxr_t)*c);
     *c = NULL;
 }
 
 static int flush(struct SwrContext *s){
     soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
     return 0;
 }
 
 static int process(
         struct ResampleContext * c, AudioData *dst, int dst_size,
         AudioData *src, int src_size, int *consumed){
     size_t idone, odone;
     soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));

     if (!error)
         error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
                              &idone, dst->ch, (size_t)dst_size, &odone);
     else
         idone = 0;
 

     *consumed = (int)idone;
     return error? -1 : odone;
 }
 
 static int64_t get_delay(struct SwrContext *s, int64_t base){
     double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate;
     return (int64_t)(delay_s * base + .5);
 }
 
 struct Resampler const soxr_resampler={
     create, destroy, process, flush, NULL /* set_compensation */, get_delay,
 };