libswresample/swresample.c - third_party/ffmpeg - Git at Google

 /*
  * Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
  *
  * This file is part of libswresample
  *
  * libswresample 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.
  *
  * libswresample 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 libswresample; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 #include "libavutil/opt.h"
 #include "swresample_internal.h"
 #include "audioconvert.h"
 #include "libavutil/avassert.h"
 #include "libavutil/audioconvert.h"

 #define  C30DB  M_SQRT2
 #define  C15DB  1.189207115
 #define C__0DB  1.0
 #define C_15DB  0.840896415
 #define C_30DB  M_SQRT1_2
 #define C_45DB  0.594603558
 #define C_60DB  0.5


 //TODO split options array out?
 #define OFFSET(x) offsetof(SwrContext,x)
 static const AVOption options[]={
 {"ich",  "input channel count", OFFSET( in.ch_count   ), AV_OPT_TYPE_INT, {.dbl=2}, 0, SWR_CH_MAX, 0},
 {"och", "output channel count", OFFSET(out.ch_count   ), AV_OPT_TYPE_INT, {.dbl=2}, 0, SWR_CH_MAX, 0},
 {"uch",   "used channel count", OFFSET(used_ch_count  ), AV_OPT_TYPE_INT, {.dbl=0}, 0, SWR_CH_MAX, 0},
 {"isr",  "input sample rate"  , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
 {"osr", "output sample rate"  , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
 //{"ip" ,  "input planar"       , OFFSET( in.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
 //{"op" , "output planar"       , OFFSET(out.planar     ), AV_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
 {"isf",  "input sample format", OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
 {"osf", "output sample format", OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
 {"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
 {"icl",  "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
 {"ocl",  "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
 {"clev", "center mix level"     , OFFSET(clev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
 {"slev", "sourround mix level"  , OFFSET(slev)         , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
 {"rmvol", "rematrix volume"     , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0}, -1000, 1000, 0},
 {"flags", NULL                  , OFFSET(flags)        , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0,  UINT_MAX, 0, "flags"},
 {"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},

 {0}
 };

 static const char* context_to_name(void* ptr) {
     return "SWR";
 }

 static const AVClass av_class = {
     .class_name                = "SwrContext",
     .item_name                 = context_to_name,
     .option                    = options,
     .version                   = LIBAVUTIL_VERSION_INT,
     .log_level_offset_offset   = OFFSET(log_level_offset),
     .parent_log_context_offset = OFFSET(log_ctx),
 };

 unsigned swresample_version(void)
 {
     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
     return LIBSWRESAMPLE_VERSION_INT;
 }

 const char *swresample_configuration(void)
 {
     return FFMPEG_CONFIGURATION;
 }

 const char *swresample_license(void)
 {
 #define LICENSE_PREFIX "libswresample license: "
     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
 }

 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
     if(!s || s->in_convert) // s needs to be allocated but not initialized
         return AVERROR(EINVAL);
     s->channel_map = channel_map;
     return 0;
 }

 struct SwrContext *swr_alloc(void){
     SwrContext *s= av_mallocz(sizeof(SwrContext));
     if(s){
         s->av_class= &av_class;
         av_opt_set_defaults(s);
     }
     return s;
 }

 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
                                       int log_offset, void *log_ctx){
     if(!s) s= swr_alloc();
     if(!s) return NULL;

     s->log_level_offset= log_offset;
     s->log_ctx= log_ctx;

     av_opt_set_int(s, "ocl", out_ch_layout,   0);
     av_opt_set_int(s, "osf", out_sample_fmt,  0);
     av_opt_set_int(s, "osr", out_sample_rate, 0);
     av_opt_set_int(s, "icl", in_ch_layout,    0);
     av_opt_set_int(s, "isf", in_sample_fmt,   0);
     av_opt_set_int(s, "isr", in_sample_rate,  0);
     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_S16, 0);
     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
     av_opt_set_int(s, "uch", 0, 0);
     return s;
 }


 static void free_temp(AudioData *a){
     av_free(a->data);
     memset(a, 0, sizeof(*a));
 }

 void swr_free(SwrContext **ss){
     SwrContext *s= *ss;
     if(s){
         free_temp(&s->postin);
         free_temp(&s->midbuf);
         free_temp(&s->preout);
         free_temp(&s->in_buffer);
         swri_audio_convert_free(&s-> in_convert);
         swri_audio_convert_free(&s->out_convert);
         swri_audio_convert_free(&s->full_convert);
         swri_resample_free(&s->resample);
     }

     av_freep(ss);
 }

 int swr_init(struct SwrContext *s){
     s->in_buffer_index= 0;
     s->in_buffer_count= 0;
     s->resample_in_constraint= 0;
     free_temp(&s->postin);
     free_temp(&s->midbuf);
     free_temp(&s->preout);
     free_temp(&s->in_buffer);
     swri_audio_convert_free(&s-> in_convert);
     swri_audio_convert_free(&s->out_convert);
     swri_audio_convert_free(&s->full_convert);

     s-> in.planar= av_sample_fmt_is_planar(s-> in_sample_fmt);
     s->out.planar= av_sample_fmt_is_planar(s->out_sample_fmt);
     s-> in_sample_fmt= av_get_alt_sample_fmt(s-> in_sample_fmt, 0);
     s->out_sample_fmt= av_get_alt_sample_fmt(s->out_sample_fmt, 0);

     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
         return AVERROR(EINVAL);
     }
     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
         return AVERROR(EINVAL);
     }

     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16
         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
         av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
         return AVERROR(EINVAL);
     }

     //FIXME should we allow/support using FLT on material that doesnt need it ?
     if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
         s->int_sample_fmt= AV_SAMPLE_FMT_S16;
     }else
         s->int_sample_fmt= AV_SAMPLE_FMT_FLT;


     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
         s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
     }else
         swri_resample_free(&s->resample);
     if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
         return -1;
     }

     if(!s->used_ch_count)
         s->used_ch_count= s->in.ch_count;

     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
         s-> in_ch_layout= 0;
     }

     if(!s-> in_ch_layout)
         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
     if(!s->out_ch_layout)
         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);

     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0;

 #define RSC 1 //FIXME finetune
     if(!s-> in.ch_count)
         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
     if(!s->used_ch_count)
         s->used_ch_count= s->in.ch_count;
     if(!s->out.ch_count)
         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);

     if(!s-> in.ch_count){
         av_assert0(!s->in_ch_layout);
         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
         return -1;
     }

 av_assert0(s->used_ch_count);
 av_assert0(s->out.ch_count);
     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;

     s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
     s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
     s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);

     if(!s->resample && !s->rematrix && !s->channel_map){
         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
         return 0;
     }

     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);


     s->postin= s->in;
     s->preout= s->out;
     s->midbuf= s->in;
     s->in_buffer= s->in;
     if(s->channel_map){
         s->postin.ch_count=
         s->midbuf.ch_count=
         s->in_buffer.ch_count= s->used_ch_count;
     }
     if(!s->resample_first){
         s->midbuf.ch_count= s->out.ch_count;
         s->in_buffer.ch_count = s->out.ch_count;
     }

     s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps =  s->int_bps;
     s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar =  1;


     if(s->rematrix)
         return swri_rematrix_init(s);

     return 0;
 }

 static int realloc_audio(AudioData *a, int count){
     int i, countb;
     AudioData old;

     if(a->count >= count)
         return 0;

     count*=2;

     countb= FFALIGN(count*a->bps, 32);
     old= *a;

     av_assert0(a->planar);
     av_assert0(a->bps);
     av_assert0(a->ch_count);

     a->data= av_malloc(countb*a->ch_count);
     if(!a->data)
         return AVERROR(ENOMEM);
     for(i=0; i<a->ch_count; i++){
         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
     }
     av_free(old.data);
     a->count= count;

     return 1;
 }

 static void copy(AudioData *out, AudioData *in,
                  int count){
     av_assert0(out->planar == in->planar);
     av_assert0(out->bps == in->bps);
     av_assert0(out->ch_count == in->ch_count);
     if(out->planar){
         int ch;
         for(ch=0; ch<out->ch_count; ch++)
             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
     }else
         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
 }

 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
     int i;
     if(out->planar){
         for(i=0; i<out->ch_count; i++)
             out->ch[i]= in_arg[i];
     }else{
         for(i=0; i<out->ch_count; i++)
             out->ch[i]= in_arg[0] + i*out->bps;
     }
 }

 /**
  *
  * out may be equal in.
  */
 static void buf_set(AudioData *out, AudioData *in, int count){
     if(in->planar){
         int ch;
         for(ch=0; ch<out->ch_count; ch++)
             out->ch[ch]= in->ch[ch] + count*out->bps;
     }else
         out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
 }

 /**
  *
  * @return number of samples output per channel
  */
 static int resample(SwrContext *s, AudioData *out_param, int out_count,
                              const AudioData * in_param, int in_count){
     AudioData in, out, tmp;
     int ret_sum=0;
     int border=0;

     tmp=out=*out_param;
     in =  *in_param;

     do{
         int ret, size, consumed;
         if(!s->resample_in_constraint && s->in_buffer_count){
             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
             ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
             out_count -= ret;
             ret_sum += ret;
             buf_set(&out, &out, ret);
             s->in_buffer_count -= consumed;
             s->in_buffer_index += consumed;

             if(!in_count)
                 break;
             if(s->in_buffer_count <= border){
                 buf_set(&in, &in, -s->in_buffer_count);
                 in_count += s->in_buffer_count;
                 s->in_buffer_count=0;
                 s->in_buffer_index=0;
                 border = 0;
             }
         }

         if(in_count && !s->in_buffer_count){
             s->in_buffer_index=0;
             ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
             out_count -= ret;
             ret_sum += ret;
             buf_set(&out, &out, ret);
             in_count -= consumed;
             buf_set(&in, &in, consumed);
         }

         //TODO is this check sane considering the advanced copy avoidance below
         size= s->in_buffer_index + s->in_buffer_count + in_count;
         if(   size > s->in_buffer.count
            && s->in_buffer_count + in_count <= s->in_buffer_index){
             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
             copy(&s->in_buffer, &tmp, s->in_buffer_count);
             s->in_buffer_index=0;
         }else
             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
                 return ret;

         if(in_count){
             int count= in_count;
             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;

             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
             copy(&tmp, &in, /*in_*/count);
             s->in_buffer_count += count;
             in_count -= count;
             border += count;
             buf_set(&in, &in, count);
             s->resample_in_constraint= 0;
             if(s->in_buffer_count != count || in_count)
                 continue;
         }
         break;
     }while(1);

     s->resample_in_constraint= !!out_count;

     return ret_sum;
 }

 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
                          const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
     AudioData *postin, *midbuf, *preout;
     int ret/*, in_max*/;
     AudioData * in= &s->in;
     AudioData *out= &s->out;
     AudioData preout_tmp, midbuf_tmp;

     if(!s->resample){
         if(in_count > out_count)
             return -1;
         out_count = in_count;
     }

     if(!in_arg){
         if(s->in_buffer_count){
             AudioData *a= &s->in_buffer;
             int i, j, ret;
             if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
                 return ret;
             av_assert0(a->planar);
             for(i=0; i<a->ch_count; i++){
                 for(j=0; j<s->in_buffer_count; j++){
                     memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
                            a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
                 }
             }
             s->in_buffer_count += (s->in_buffer_count+1)/2;
             s->resample_in_constraint = 0;
         }else{
             return 0;
         }
     }else
         fill_audiodata(in ,  (void*)in_arg);
     fill_audiodata(out, out_arg);

     if(s->full_convert){
         av_assert0(!s->resample);
         swri_audio_convert(s->full_convert, out, in, in_count);
         return out_count;
     }

 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);

     if((ret=realloc_audio(&s->postin, in_count))<0)
         return ret;
     if(s->resample_first){
         av_assert0(s->midbuf.ch_count == s->used_ch_count);
         if((ret=realloc_audio(&s->midbuf, out_count))<0)
             return ret;
     }else{
         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
             return ret;
     }
     if((ret=realloc_audio(&s->preout, out_count))<0)
         return ret;

     postin= &s->postin;

     midbuf_tmp= s->midbuf;
     midbuf= &midbuf_tmp;
     preout_tmp= s->preout;
     preout= &preout_tmp;

     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
         postin= in;

     if(s->resample_first ? !s->resample : !s->rematrix)
         midbuf= postin;

     if(s->resample_first ? !s->rematrix : !s->resample)
         preout= midbuf;

     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
         if(preout==in){
             out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
             copy(out, in, out_count);
             return out_count;
         }
         else if(preout==postin) preout= midbuf= postin= out;
         else if(preout==midbuf) preout= midbuf= out;
         else                    preout= out;
     }

     if(in != postin){
         swri_audio_convert(s->in_convert, postin, in, in_count);
     }

     if(s->resample_first){
         if(postin != midbuf)
             out_count= resample(s, midbuf, out_count, postin, in_count);
         if(midbuf != preout)
             swri_rematrix(s, preout, midbuf, out_count, preout==out);
     }else{
         if(postin != midbuf)
             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
         if(midbuf != preout)
             out_count= resample(s, preout, out_count, midbuf, in_count);
     }

     if(preout != out){
 //FIXME packed doesnt need more than 1 chan here!
         swri_audio_convert(s->out_convert, out, preout, out_count);
     }
     if(!in_arg)
         s->in_buffer_count = 0;
     return out_count;
 }
	/*
	* Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
	*
	* This file is part of libswresample
	*
	* libswresample 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.
	*
	* libswresample 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 libswresample; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	#include "libavutil/opt.h"
	#include "swresample_internal.h"
	#include "audioconvert.h"
	#include "libavutil/avassert.h"
	#include "libavutil/audioconvert.h"

	#define C30DB M_SQRT2
	#define C15DB 1.189207115
	#define C__0DB 1.0
	#define C_15DB 0.840896415
	#define C_30DB M_SQRT1_2
	#define C_45DB 0.594603558
	#define C_60DB 0.5


	//TODO split options array out?
	#define OFFSET(x) offsetof(SwrContext,x)
	static const AVOption options[]={
	{"ich", "input channel count", OFFSET( in.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 0, SWR_CH_MAX, 0},
	{"och", "output channel count", OFFSET(out.ch_count ), AV_OPT_TYPE_INT, {.dbl=2}, 0, SWR_CH_MAX, 0},
	{"uch", "used channel count", OFFSET(used_ch_count ), AV_OPT_TYPE_INT, {.dbl=0}, 0, SWR_CH_MAX, 0},
	{"isr", "input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
	{"osr", "output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
	//{"ip" , "input planar" , OFFSET( in.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
	//{"op" , "output planar" , OFFSET(out.planar ), AV_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
	{"isf", "input sample format", OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
	{"osf", "output sample format", OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
	{"tsf", "internal sample format", OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
	{"icl", "input channel layout" , OFFSET( in_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
	{"ocl", "output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
	{"clev", "center mix level" , OFFSET(clev) , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
	{"slev", "sourround mix level" , OFFSET(slev) , AV_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
	{"rmvol", "rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0}, -1000, 1000, 0},
	{"flags", NULL , OFFSET(flags) , AV_OPT_TYPE_FLAGS, {.dbl=0}, 0, UINT_MAX, 0, "flags"},
	{"res", "force resampling", 0, AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},

	{0}
	};

	static const char* context_to_name(void* ptr) {
	return "SWR";
	}

	static const AVClass av_class = {
	.class_name = "SwrContext",
	.item_name = context_to_name,
	.option = options,
	.version = LIBAVUTIL_VERSION_INT,
	.log_level_offset_offset = OFFSET(log_level_offset),
	.parent_log_context_offset = OFFSET(log_ctx),
	};

	unsigned swresample_version(void)
	{
	av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
	return LIBSWRESAMPLE_VERSION_INT;
	}

	const char *swresample_configuration(void)
	{
	return FFMPEG_CONFIGURATION;
	}

	const char *swresample_license(void)
	{
	#define LICENSE_PREFIX "libswresample license: "
	return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
	}

	int swr_set_channel_mapping(struct SwrContext s, const int channel_map){
	if(!s \|\| s->in_convert) // s needs to be allocated but not initialized
	return AVERROR(EINVAL);
	s->channel_map = channel_map;
	return 0;
	}

	struct SwrContext *swr_alloc(void){
	SwrContext *s= av_mallocz(sizeof(SwrContext));
	if(s){
	s->av_class= &av_class;
	av_opt_set_defaults(s);
	}
	return s;
	}

	struct SwrContext swr_alloc_set_opts(struct SwrContext s,
	int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
	int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
	int log_offset, void *log_ctx){
	if(!s) s= swr_alloc();
	if(!s) return NULL;

	s->log_level_offset= log_offset;
	s->log_ctx= log_ctx;

	av_opt_set_int(s, "ocl", out_ch_layout, 0);
	av_opt_set_int(s, "osf", out_sample_fmt, 0);
	av_opt_set_int(s, "osr", out_sample_rate, 0);
	av_opt_set_int(s, "icl", in_ch_layout, 0);
	av_opt_set_int(s, "isf", in_sample_fmt, 0);
	av_opt_set_int(s, "isr", in_sample_rate, 0);
	av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_S16, 0);
	av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
	av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
	av_opt_set_int(s, "uch", 0, 0);
	return s;
	}


	static void free_temp(AudioData *a){
	av_free(a->data);
	memset(a, 0, sizeof(*a));
	}

	void swr_free(SwrContext **ss){
	SwrContext s= ss;
	if(s){
	free_temp(&s->postin);
	free_temp(&s->midbuf);
	free_temp(&s->preout);
	free_temp(&s->in_buffer);
	swri_audio_convert_free(&s-> in_convert);
	swri_audio_convert_free(&s->out_convert);
	swri_audio_convert_free(&s->full_convert);
	swri_resample_free(&s->resample);
	}

	av_freep(ss);
	}

	int swr_init(struct SwrContext *s){
	s->in_buffer_index= 0;
	s->in_buffer_count= 0;
	s->resample_in_constraint= 0;
	free_temp(&s->postin);
	free_temp(&s->midbuf);
	free_temp(&s->preout);
	free_temp(&s->in_buffer);
	swri_audio_convert_free(&s-> in_convert);
	swri_audio_convert_free(&s->out_convert);
	swri_audio_convert_free(&s->full_convert);

	s-> in.planar= av_sample_fmt_is_planar(s-> in_sample_fmt);
	s->out.planar= av_sample_fmt_is_planar(s->out_sample_fmt);
	s-> in_sample_fmt= av_get_alt_sample_fmt(s-> in_sample_fmt, 0);
	s->out_sample_fmt= av_get_alt_sample_fmt(s->out_sample_fmt, 0);

	if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
	av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
	return AVERROR(EINVAL);
	}
	if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
	av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
	return AVERROR(EINVAL);
	}

	if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
	&&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
	av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
	return AVERROR(EINVAL);
	}

	//FIXME should we allow/support using FLT on material that doesnt need it ?
	if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 \|\| s->int_sample_fmt==AV_SAMPLE_FMT_S16){
	s->int_sample_fmt= AV_SAMPLE_FMT_S16;
	}else
	s->int_sample_fmt= AV_SAMPLE_FMT_FLT;


	if (s->out_sample_rate!=s->in_sample_rate \|\| (s->flags & SWR_FLAG_RESAMPLE)){
	s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
	}else
	swri_resample_free(&s->resample);
	if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
	av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
	return -1;
	}

	if(!s->used_ch_count)
	s->used_ch_count= s->in.ch_count;

	if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
	av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
	s-> in_ch_layout= 0;
	}

	if(!s-> in_ch_layout)
	s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
	if(!s->out_ch_layout)
	s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);

	s->rematrix= s->out_ch_layout !=s->in_ch_layout \|\| s->rematrix_volume!=1.0;

	#define RSC 1 //FIXME finetune
	if(!s-> in.ch_count)
	s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
	if(!s->used_ch_count)
	s->used_ch_count= s->in.ch_count;
	if(!s->out.ch_count)
	s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);

	if(!s-> in.ch_count){
	av_assert0(!s->in_ch_layout);
	av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
	return -1;
	}

	av_assert0(s->used_ch_count);
	av_assert0(s->out.ch_count);
	s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;

	s-> in.bps= av_get_bytes_per_sample(s-> in_sample_fmt);
	s->int_bps= av_get_bytes_per_sample(s->int_sample_fmt);
	s->out.bps= av_get_bytes_per_sample(s->out_sample_fmt);

	if(!s->resample && !s->rematrix && !s->channel_map){
	s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
	s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
	return 0;
	}

	s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
	s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
	s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
	s->int_sample_fmt, s->out.ch_count, NULL, 0);


	s->postin= s->in;
	s->preout= s->out;
	s->midbuf= s->in;
	s->in_buffer= s->in;
	if(s->channel_map){
	s->postin.ch_count=
	s->midbuf.ch_count=
	s->in_buffer.ch_count= s->used_ch_count;
	}
	if(!s->resample_first){
	s->midbuf.ch_count= s->out.ch_count;
	s->in_buffer.ch_count = s->out.ch_count;
	}

	s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
	s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar = 1;


	if(s->rematrix)
	return swri_rematrix_init(s);

	return 0;
	}

	static int realloc_audio(AudioData *a, int count){
	int i, countb;
	AudioData old;

	if(a->count >= count)
	return 0;

	count*=2;

	countb= FFALIGN(count*a->bps, 32);
	old= *a;

	av_assert0(a->planar);
	av_assert0(a->bps);
	av_assert0(a->ch_count);

	a->data= av_malloc(countb*a->ch_count);
	if(!a->data)
	return AVERROR(ENOMEM);
	for(i=0; i<a->ch_count; i++){
	a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
	if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
	}
	av_free(old.data);
	a->count= count;

	return 1;
	}

	static void copy(AudioData out, AudioData in,
	int count){
	av_assert0(out->planar == in->planar);
	av_assert0(out->bps == in->bps);
	av_assert0(out->ch_count == in->ch_count);
	if(out->planar){
	int ch;
	for(ch=0; ch<out->ch_count; ch++)
	memcpy(out->ch[ch], in->ch[ch], count*out->bps);
	}else
	memcpy(out->ch[0], in->ch[0], countout->ch_countout->bps);
	}

	static void fill_audiodata(AudioData out, uint8_t in_arg [SWR_CH_MAX]){
	int i;
	if(out->planar){
	for(i=0; i<out->ch_count; i++)
	out->ch[i]= in_arg[i];
	}else{
	for(i=0; i<out->ch_count; i++)
	out->ch[i]= in_arg[0] + i*out->bps;
	}
	}

	/**
	*
	* out may be equal in.
	*/
	static void buf_set(AudioData out, AudioData in, int count){
	if(in->planar){
	int ch;
	for(ch=0; ch<out->ch_count; ch++)
	out->ch[ch]= in->ch[ch] + count*out->bps;
	}else
	out->ch[0]= in->ch[0] + countout->ch_countout->bps;
	}

	/**
	*
	* @return number of samples output per channel
	*/
	static int resample(SwrContext s, AudioData out_param, int out_count,
	const AudioData * in_param, int in_count){
	AudioData in, out, tmp;
	int ret_sum=0;
	int border=0;

	tmp=out=*out_param;
	in = *in_param;

	do{
	int ret, size, consumed;
	if(!s->resample_in_constraint && s->in_buffer_count){
	buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
	ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
	out_count -= ret;
	ret_sum += ret;
	buf_set(&out, &out, ret);
	s->in_buffer_count -= consumed;
	s->in_buffer_index += consumed;

	if(!in_count)
	break;
	if(s->in_buffer_count <= border){
	buf_set(&in, &in, -s->in_buffer_count);
	in_count += s->in_buffer_count;
	s->in_buffer_count=0;
	s->in_buffer_index=0;
	border = 0;
	}
	}

	if(in_count && !s->in_buffer_count){
	s->in_buffer_index=0;
	ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
	out_count -= ret;
	ret_sum += ret;
	buf_set(&out, &out, ret);
	in_count -= consumed;
	buf_set(&in, &in, consumed);
	}

	//TODO is this check sane considering the advanced copy avoidance below
	size= s->in_buffer_index + s->in_buffer_count + in_count;
	if( size > s->in_buffer.count
	&& s->in_buffer_count + in_count <= s->in_buffer_index){
	buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
	copy(&s->in_buffer, &tmp, s->in_buffer_count);
	s->in_buffer_index=0;
	}else
	if((ret=realloc_audio(&s->in_buffer, size)) < 0)
	return ret;

	if(in_count){
	int count= in_count;
	if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;

	buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
	copy(&tmp, &in, /in_/count);
	s->in_buffer_count += count;
	in_count -= count;
	border += count;
	buf_set(&in, &in, count);
	s->resample_in_constraint= 0;
	if(s->in_buffer_count != count \|\| in_count)
	continue;
	}
	break;
	}while(1);

	s->resample_in_constraint= !!out_count;

	return ret_sum;
	}

	int swr_convert(struct SwrContext s, uint8_t out_arg[SWR_CH_MAX], int out_count,
	const uint8_t *in_arg [SWR_CH_MAX], int in_count){
	AudioData postin, midbuf, *preout;
	int ret/, in_max/;
	AudioData * in= &s->in;
	AudioData *out= &s->out;
	AudioData preout_tmp, midbuf_tmp;

	if(!s->resample){
	if(in_count > out_count)
	return -1;
	out_count = in_count;
	}

	if(!in_arg){
	if(s->in_buffer_count){
	AudioData *a= &s->in_buffer;
	int i, j, ret;
	if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
	return ret;
	av_assert0(a->planar);
	for(i=0; i<a->ch_count; i++){
	for(j=0; j<s->in_buffer_count; j++){
	memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
	a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
	}
	}
	s->in_buffer_count += (s->in_buffer_count+1)/2;
	s->resample_in_constraint = 0;
	}else{
	return 0;
	}
	}else
	fill_audiodata(in , (void*)in_arg);
	fill_audiodata(out, out_arg);

	if(s->full_convert){
	av_assert0(!s->resample);
	swri_audio_convert(s->full_convert, out, in, in_count);
	return out_count;
	}

	// in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
	// in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);

	if((ret=realloc_audio(&s->postin, in_count))<0)
	return ret;
	if(s->resample_first){
	av_assert0(s->midbuf.ch_count == s->used_ch_count);
	if((ret=realloc_audio(&s->midbuf, out_count))<0)
	return ret;
	}else{
	av_assert0(s->midbuf.ch_count == s->out.ch_count);
	if((ret=realloc_audio(&s->midbuf, in_count))<0)
	return ret;
	}
	if((ret=realloc_audio(&s->preout, out_count))<0)
	return ret;

	postin= &s->postin;

	midbuf_tmp= s->midbuf;
	midbuf= &midbuf_tmp;
	preout_tmp= s->preout;
	preout= &preout_tmp;

	if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
	postin= in;

	if(s->resample_first ? !s->resample : !s->rematrix)
	midbuf= postin;

	if(s->resample_first ? !s->rematrix : !s->resample)
	preout= midbuf;

	if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
	if(preout==in){
	out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
	av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
	copy(out, in, out_count);
	return out_count;
	}
	else if(preout==postin) preout= midbuf= postin= out;
	else if(preout==midbuf) preout= midbuf= out;
	else preout= out;
	}

	if(in != postin){
	swri_audio_convert(s->in_convert, postin, in, in_count);
	}

	if(s->resample_first){
	if(postin != midbuf)
	out_count= resample(s, midbuf, out_count, postin, in_count);
	if(midbuf != preout)
	swri_rematrix(s, preout, midbuf, out_count, preout==out);
	}else{
	if(postin != midbuf)
	swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
	if(midbuf != preout)
	out_count= resample(s, preout, out_count, midbuf, in_count);
	}

	if(preout != out){
	//FIXME packed doesnt need more than 1 chan here!
	swri_audio_convert(s->out_convert, out, preout, out_count);
	}
	if(!in_arg)
	s->in_buffer_count = 0;
	return out_count;
	}