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1 /*
2  * Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * libswresample is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20
21 #include "libavutil/opt.h"
22 #include "swresample_internal.h"
23 #include "audioconvert.h"
24 #include "libavutil/avassert.h"
25
26 #define  C30DB  M_SQRT2
27 #define  C15DB  1.189207115
28 #define C__0DB  1.0
29 #define C_15DB  0.840896415
30 #define C_30DB  M_SQRT1_2
31 #define C_45DB  0.594603558
32 #define C_60DB  0.5
33
34
35 //TODO split options array out?
36 #define OFFSET(x) offsetof(SwrContext,x)
37 static const AVOption options[]={
38 {"ich",  "input channel count", OFFSET( in.ch_count   ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
39 {"och", "output channel count", OFFSET(out.ch_count   ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
40 {"isr",  "input sample rate"  , OFFSET( in_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
41 {"osr", "output sample rate"  , OFFSET(out_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
42 {"ip" ,  "input planar"       , OFFSET( in.planar     ), FF_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
43 {"op" , "output planar"       , OFFSET(out.planar     ), FF_OPT_TYPE_INT, {.dbl=0},    0,       1, 0},
44 {"isf",  "input sample format", OFFSET( in_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1, 0},
45 {"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1, 0},
46 {"tsf", "internal sample format", OFFSET(int_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
47 {"icl",  "input channel layout" , OFFSET( in_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
48 {"ocl",  "output channel layout", OFFSET(out_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
49 {"clev", "center mix level"     , OFFSET(clev)         , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
50 {"slev", "sourround mix level"  , OFFSET(slev)         , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
51 {"flags", NULL                  , OFFSET(flags)        , FF_OPT_TYPE_FLAGS, {.dbl=0}, 0,  UINT_MAX, 0, "flags"},
52 {"res", "force resampling", 0, FF_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
53
54 {0}
55 };
56
57 static const char* context_to_name(void* ptr) {
58     return "SWR";
59 }
60
61 static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
62
63 static int resample(SwrContext *s, AudioData *out_param, int out_count,
64                              const AudioData * in_param, int in_count);
65
66 SwrContext *swr_alloc(void){
67     SwrContext *s= av_mallocz(sizeof(SwrContext));
68     if(s){
69         s->av_class= &av_class;
70         av_opt_set_defaults2(s, 0, 0);
71     }
72     return s;
73 }
74
75 SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
76                        int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
77                        int log_offset, void *log_ctx){
78     if(!s) s= swr_alloc();
79     if(!s) return NULL;
80
81     s->log_level_offset= log_offset;
82     s->log_ctx= log_ctx;
83
84     av_set_int(s, "ocl", out_ch_layout);
85     av_set_int(s, "osf", out_sample_fmt);
86     av_set_int(s, "osr", out_sample_rate);
87     av_set_int(s, "icl", in_ch_layout);
88     av_set_int(s, "isf", in_sample_fmt);
89     av_set_int(s, "isr", in_sample_rate);
90
91     s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
92     s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
93     s->int_sample_fmt = AV_SAMPLE_FMT_S16;
94
95     return s;
96 }
97
98
99 static void free_temp(AudioData *a){
100     av_free(a->data);
101     memset(a, 0, sizeof(*a));
102 }
103
104 void swr_free(SwrContext **ss){
105     SwrContext *s= *ss;
106     if(s){
107         free_temp(&s->postin);
108         free_temp(&s->midbuf);
109         free_temp(&s->preout);
110         free_temp(&s->in_buffer);
111         swr_audio_convert_free(&s-> in_convert);
112         swr_audio_convert_free(&s->out_convert);
113         swr_resample_free(&s->resample);
114     }
115
116     av_freep(ss);
117 }
118
119 static int64_t guess_layout(int ch){
120     switch(ch){
121     case 1: return AV_CH_LAYOUT_MONO;
122     case 2: return AV_CH_LAYOUT_STEREO;
123     case 5: return AV_CH_LAYOUT_5POINT0;
124     case 6: return AV_CH_LAYOUT_5POINT1;
125     case 7: return AV_CH_LAYOUT_7POINT0;
126     case 8: return AV_CH_LAYOUT_7POINT1;
127     default: return 0;
128     }
129 }
130
131 int swr_init(SwrContext *s){
132     s->in_buffer_index= 0;
133     s->in_buffer_count= 0;
134     s->resample_in_constraint= 0;
135     free_temp(&s->postin);
136     free_temp(&s->midbuf);
137     free_temp(&s->preout);
138     free_temp(&s->in_buffer);
139     swr_audio_convert_free(&s-> in_convert);
140     swr_audio_convert_free(&s->out_convert);
141
142     //We assume AVOptions checked the various values and the defaults where allowed
143     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16
144         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
145         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));
146         return AVERROR(EINVAL);
147     }
148
149     //FIXME should we allow/support using FLT on material that doesnt need it ?
150     if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
151         s->int_sample_fmt= AV_SAMPLE_FMT_S16;
152     }else
153         s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
154
155
156     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
157         s->resample = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
158     }else
159         swr_resample_free(&s->resample);
160     if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
161         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
162         return -1;
163     }
164
165     if(!s-> in_ch_layout)
166         s-> in_ch_layout= guess_layout(s->in.ch_count);
167     if(!s->out_ch_layout)
168         s->out_ch_layout= guess_layout(s->out.ch_count);
169
170     s->rematrix= s->out_ch_layout  !=s->in_ch_layout;
171
172 #define RSC 1 //FIXME finetune
173     if(!s-> in.ch_count)
174         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
175     if(!s->out.ch_count)
176         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
177
178 av_assert0(s-> in.ch_count);
179 av_assert0(s->out.ch_count);
180     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
181
182     s-> in.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
183     s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
184     s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
185
186     s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
187                                             s-> in_sample_fmt, s-> in.ch_count, 0);
188     s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
189                                             s->int_sample_fmt, s->out.ch_count, 0);
190
191
192     s->postin= s->in;
193     s->preout= s->out;
194     s->midbuf= s->in;
195     s->in_buffer= s->in;
196     if(!s->resample_first){
197         s->midbuf.ch_count= s->out.ch_count;
198         s->in_buffer.ch_count = s->out.ch_count;
199     }
200
201     s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps =  s->int_bps;
202     s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar =  1;
203
204
205     if(s->rematrix && swr_rematrix_init(s)<0)
206         return -1;
207
208     return 0;
209 }
210
211 static int realloc_audio(AudioData *a, int count){
212     int i, countb;
213     AudioData old;
214
215     if(a->count >= count)
216         return 0;
217
218     count*=2;
219
220     countb= FFALIGN(count*a->bps, 32);
221     old= *a;
222
223     av_assert0(a->planar);
224     av_assert0(a->bps);
225     av_assert0(a->ch_count);
226
227     a->data= av_malloc(countb*a->ch_count);
228     if(!a->data)
229         return AVERROR(ENOMEM);
230     for(i=0; i<a->ch_count; i++){
231         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
232         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
233     }
234     av_free(old.data);
235     a->count= count;
236
237     return 1;
238 }
239
240 static void copy(AudioData *out, AudioData *in,
241                  int count){
242     av_assert0(out->planar == in->planar);
243     av_assert0(out->bps == in->bps);
244     av_assert0(out->ch_count == in->ch_count);
245     if(out->planar){
246         int ch;
247         for(ch=0; ch<out->ch_count; ch++)
248             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
249     }else
250         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
251 }
252
253 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
254                          const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
255     AudioData *postin, *midbuf, *preout;
256     int ret, i/*, in_max*/;
257     AudioData * in= &s->in;
258     AudioData *out= &s->out;
259     AudioData preout_tmp, midbuf_tmp;
260
261     if(!s->resample){
262         if(in_count > out_count)
263             return -1;
264         out_count = in_count;
265     }
266
267     av_assert0(in ->planar == 0);
268     av_assert0(out->planar == 0);
269     for(i=0; i<s-> in.ch_count; i++)
270         in ->ch[i]=  in_arg[0] + i* in->bps;
271     for(i=0; i<s->out.ch_count; i++)
272         out->ch[i]= out_arg[0] + i*out->bps;
273
274 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
275 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
276
277     if((ret=realloc_audio(&s->postin, in_count))<0)
278         return ret;
279     if(s->resample_first){
280         av_assert0(s->midbuf.ch_count ==  s-> in.ch_count);
281         if((ret=realloc_audio(&s->midbuf, out_count))<0)
282             return ret;
283     }else{
284         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
285         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
286             return ret;
287     }
288     if((ret=realloc_audio(&s->preout, out_count))<0)
289         return ret;
290
291     postin= &s->postin;
292
293     midbuf_tmp= s->midbuf;
294     midbuf= &midbuf_tmp;
295     preout_tmp= s->preout;
296     preout= &preout_tmp;
297
298     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
299         postin= in;
300
301     if(s->resample_first ? !s->resample : !s->rematrix)
302         midbuf= postin;
303
304     if(s->resample_first ? !s->rematrix : !s->resample)
305         preout= midbuf;
306
307     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
308         if(preout==in){
309             out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
310             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
311             copy(out, in, out_count);
312             return out_count;
313         }
314         else if(preout==postin) preout= midbuf= postin= out;
315         else if(preout==midbuf) preout= midbuf= out;
316         else                    preout= out;
317     }
318
319     if(in != postin){
320         swr_audio_convert(s->in_convert, postin, in, in_count);
321     }
322
323     if(s->resample_first){
324         if(postin != midbuf)
325             out_count= resample(s, midbuf, out_count, postin, in_count);
326         if(midbuf != preout)
327             swr_rematrix(s, preout, midbuf, out_count, preout==out);
328     }else{
329         if(postin != midbuf)
330             swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
331         if(midbuf != preout)
332             out_count= resample(s, preout, out_count, midbuf, in_count);
333     }
334
335     if(preout != out){
336 //FIXME packed doesnt need more than 1 chan here!
337         swr_audio_convert(s->out_convert, out, preout, out_count);
338     }
339     return out_count;
340 }
341
342 /**
343  *
344  * out may be equal in.
345  */
346 static void buf_set(AudioData *out, AudioData *in, int count){
347     if(in->planar){
348         int ch;
349         for(ch=0; ch<out->ch_count; ch++)
350             out->ch[ch]= in->ch[ch] + count*out->bps;
351     }else
352         out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
353 }
354
355 /**
356  *
357  * @return number of samples output per channel
358  */
359 static int resample(SwrContext *s, AudioData *out_param, int out_count,
360                              const AudioData * in_param, int in_count){
361     AudioData in, out, tmp;
362     int ret_sum=0;
363     int border=0;
364
365     tmp=out=*out_param;
366     in =  *in_param;
367
368     do{
369         int ret, size, consumed;
370         if(!s->resample_in_constraint && s->in_buffer_count){
371             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
372             ret= swr_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
373             out_count -= ret;
374             ret_sum += ret;
375             buf_set(&out, &out, ret);
376             s->in_buffer_count -= consumed;
377             s->in_buffer_index += consumed;
378
379             if(!in_count)
380                 break;
381             if(s->in_buffer_count <= border){
382                 buf_set(&in, &in, -s->in_buffer_count);
383                 in_count += s->in_buffer_count;
384                 s->in_buffer_count=0;
385                 s->in_buffer_index=0;
386                 border = 0;
387             }
388         }
389
390         if(in_count && !s->in_buffer_count){
391             s->in_buffer_index=0;
392             ret= swr_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
393             out_count -= ret;
394             ret_sum += ret;
395             buf_set(&out, &out, ret);
396             in_count -= consumed;
397             buf_set(&in, &in, consumed);
398         }
399
400         //TODO is this check sane considering the advanced copy avoidance below
401         size= s->in_buffer_index + s->in_buffer_count + in_count;
402         if(   size > s->in_buffer.count
403            && s->in_buffer_count + in_count <= s->in_buffer_index){
404             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
405             copy(&s->in_buffer, &tmp, s->in_buffer_count);
406             s->in_buffer_index=0;
407         }else
408             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
409                 return ret;
410
411         if(in_count){
412             int count= in_count;
413             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
414
415             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
416             copy(&tmp, &in, /*in_*/count);
417             s->in_buffer_count += count;
418             in_count -= count;
419             border += count;
420             buf_set(&in, &in, count);
421             s->resample_in_constraint= 0;
422             if(s->in_buffer_count != count || in_count)
423                 continue;
424         }
425         break;
426     }while(1);
427
428     s->resample_in_constraint= !!out_count;
429
430     return ret_sum;
431 }