swr: set default channel count options to 0.
[ffmpeg.git] / libswresample / swresample.c
1 /*
2  * Copyright (C) 2011-2012 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 #include "libavutil/channel_layout.h"
26
27 #include <float.h>
28
29 #define  C30DB  M_SQRT2
30 #define  C15DB  1.189207115
31 #define C__0DB  1.0
32 #define C_15DB  0.840896415
33 #define C_30DB  M_SQRT1_2
34 #define C_45DB  0.594603558
35 #define C_60DB  0.5
36
37 #define ALIGN 32
38
39 //TODO split options array out?
40 #define OFFSET(x) offsetof(SwrContext,x)
41 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
42
43 static const AVOption options[]={
44 {"ich"                  , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
45 {"in_channel_count"     , "set input channel count"     , OFFSET( in.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
46 {"och"                  , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
47 {"out_channel_count"    , "set output channel count"    , OFFSET(out.ch_count   ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
48 {"uch"                  , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
49 {"used_channel_count"   , "set used channel count"      , OFFSET(used_ch_count  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_CH_MAX, PARAM},
50 {"isr"                  , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
51 {"in_sample_rate"       , "set input sample rate"       , OFFSET( in_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
52 {"osr"                  , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
53 {"out_sample_rate"      , "set output sample rate"      , OFFSET(out_sample_rate), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , INT_MAX   , PARAM},
54 {"isf"                  , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
55 {"in_sample_fmt"        , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
56 {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
57 {"out_sample_fmt"       , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_NB-1+256, PARAM},
58 {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_FLTP, PARAM},
59 {"internal_sample_fmt"  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT  , {.i64=AV_SAMPLE_FMT_NONE    }, -1     , AV_SAMPLE_FMT_FLTP, PARAM},
60 {"icl"                  , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
61 {"in_channel_layout"    , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
62 {"ocl"                  , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
63 {"out_channel_layout"   , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_INT64, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
64 {"clev"                 , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
65 {"center_mix_level"     , "set center mix level"        , OFFSET(clev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
66 {"slev"                 , "set surround mix level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
67 {"surround_mix_level"   , "set surround mix Level"      , OFFSET(slev           ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB                }, -32    , 32        , PARAM},
68 {"lfe_mix_level"        , "set LFE mix level"           , OFFSET(lfe_mix_level  ), AV_OPT_TYPE_FLOAT, {.dbl=0                     }, -32    , 32        , PARAM},
69 {"rmvol"                , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
70 {"rematrix_volume"      , "set rematrix volume"         , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0                   }, -1000  , 1000      , PARAM},
71
72 {"flags"                , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
73 {"swr_flags"            , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
74 {"res"                  , "force resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
75
76 {"dither_scale"         , "set dither scale"            , OFFSET(dither_scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},
77
78 {"dither_method"        , "set dither method"           , OFFSET(dither_method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},
79 {"rectangular"          , "select rectangular dither"   , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
80 {"triangular"           , "select triangular dither"    , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},
81 {"triangular_hp"        , "select triangular dither with high pass" , 0                 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
82
83 {"filter_size"          , "set resampling filter size"  , OFFSET(filter_size)    , AV_OPT_TYPE_INT  , {.i64=16                    }, 0      , INT_MAX   , PARAM },
84 {"phase_shift"          , "set resampling phase shift"  , OFFSET(phase_shift)    , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 30        , PARAM },
85 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
86 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.8                   }, 0      , 1         , PARAM },
87 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
88                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
89 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
90                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
91 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
92                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
93 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
94                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
95
96 { "matrix_encoding"     , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT   ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE,     AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
97     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
98     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
99     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
100
101 { "filter_type"         , "select filter type"          , OFFSET(filter_type)    , AV_OPT_TYPE_INT  , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
102     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
103     { "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0             , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
104     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
105
106 { "kaiser_beta"         , "set Kaiser Window Beta"      , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
107
108 {0}
109 };
110
111 static const char* context_to_name(void* ptr) {
112     return "SWR";
113 }
114
115 static const AVClass av_class = {
116     .class_name                = "SWResampler",
117     .item_name                 = context_to_name,
118     .option                    = options,
119     .version                   = LIBAVUTIL_VERSION_INT,
120     .log_level_offset_offset   = OFFSET(log_level_offset),
121     .parent_log_context_offset = OFFSET(log_ctx),
122     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
123 };
124
125 unsigned swresample_version(void)
126 {
127     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
128     return LIBSWRESAMPLE_VERSION_INT;
129 }
130
131 const char *swresample_configuration(void)
132 {
133     return FFMPEG_CONFIGURATION;
134 }
135
136 const char *swresample_license(void)
137 {
138 #define LICENSE_PREFIX "libswresample license: "
139     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
140 }
141
142 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
143     if(!s || s->in_convert) // s needs to be allocated but not initialized
144         return AVERROR(EINVAL);
145     s->channel_map = channel_map;
146     return 0;
147 }
148
149 const AVClass *swr_get_class(void)
150 {
151     return &av_class;
152 }
153
154 av_cold struct SwrContext *swr_alloc(void){
155     SwrContext *s= av_mallocz(sizeof(SwrContext));
156     if(s){
157         s->av_class= &av_class;
158         av_opt_set_defaults(s);
159     }
160     return s;
161 }
162
163 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
164                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
165                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
166                                       int log_offset, void *log_ctx){
167     if(!s) s= swr_alloc();
168     if(!s) return NULL;
169
170     s->log_level_offset= log_offset;
171     s->log_ctx= log_ctx;
172
173     av_opt_set_int(s, "ocl", out_ch_layout,   0);
174     av_opt_set_int(s, "osf", out_sample_fmt,  0);
175     av_opt_set_int(s, "osr", out_sample_rate, 0);
176     av_opt_set_int(s, "icl", in_ch_layout,    0);
177     av_opt_set_int(s, "isf", in_sample_fmt,   0);
178     av_opt_set_int(s, "isr", in_sample_rate,  0);
179     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
180     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
181     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
182     av_opt_set_int(s, "uch", 0, 0);
183     return s;
184 }
185
186 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
187     a->fmt   = fmt;
188     a->bps   = av_get_bytes_per_sample(fmt);
189     a->planar= av_sample_fmt_is_planar(fmt);
190 }
191
192 static void free_temp(AudioData *a){
193     av_free(a->data);
194     memset(a, 0, sizeof(*a));
195 }
196
197 av_cold void swr_free(SwrContext **ss){
198     SwrContext *s= *ss;
199     if(s){
200         free_temp(&s->postin);
201         free_temp(&s->midbuf);
202         free_temp(&s->preout);
203         free_temp(&s->in_buffer);
204         free_temp(&s->dither);
205         swri_audio_convert_free(&s-> in_convert);
206         swri_audio_convert_free(&s->out_convert);
207         swri_audio_convert_free(&s->full_convert);
208         swri_resample_free(&s->resample);
209         swri_rematrix_free(s);
210     }
211
212     av_freep(ss);
213 }
214
215 av_cold int swr_init(struct SwrContext *s){
216     s->in_buffer_index= 0;
217     s->in_buffer_count= 0;
218     s->resample_in_constraint= 0;
219     free_temp(&s->postin);
220     free_temp(&s->midbuf);
221     free_temp(&s->preout);
222     free_temp(&s->in_buffer);
223     free_temp(&s->dither);
224     memset(s->in.ch, 0, sizeof(s->in.ch));
225     memset(s->out.ch, 0, sizeof(s->out.ch));
226     swri_audio_convert_free(&s-> in_convert);
227     swri_audio_convert_free(&s->out_convert);
228     swri_audio_convert_free(&s->full_convert);
229     swri_rematrix_free(s);
230
231     s->flushed = 0;
232
233     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
234         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
235         return AVERROR(EINVAL);
236     }
237     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
238         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
239         return AVERROR(EINVAL);
240     }
241
242     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
243         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
244             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
245         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
246             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
247         }else{
248             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
249             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
250         }
251     }
252
253     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
254         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
255         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
256         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
257         av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
258         return AVERROR(EINVAL);
259     }
260
261     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
262     set_audiodata_fmt(&s->out, s->out_sample_fmt);
263
264     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
265         s->resample = swri_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta);
266     }else
267         swri_resample_free(&s->resample);
268     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
269         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
270         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
271         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
272         && s->resample){
273         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
274         return -1;
275     }
276
277     if(!s->used_ch_count)
278         s->used_ch_count= s->in.ch_count;
279
280     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
281         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
282         s-> in_ch_layout= 0;
283     }
284
285     if(!s-> in_ch_layout)
286         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
287     if(!s->out_ch_layout)
288         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
289
290     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
291                  s->rematrix_custom;
292
293 #define RSC 1 //FIXME finetune
294     if(!s-> in.ch_count)
295         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
296     if(!s->used_ch_count)
297         s->used_ch_count= s->in.ch_count;
298     if(!s->out.ch_count)
299         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
300
301     if(!s-> in.ch_count){
302         av_assert0(!s->in_ch_layout);
303         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
304         return -1;
305     }
306
307     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
308         av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
309         return -1;
310     }
311
312 av_assert0(s->used_ch_count);
313 av_assert0(s->out.ch_count);
314     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
315
316     s->in_buffer= s->in;
317
318     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
319         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
320                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
321         return 0;
322     }
323
324     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
325                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
326     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
327                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
328
329
330     s->postin= s->in;
331     s->preout= s->out;
332     s->midbuf= s->in;
333
334     if(s->channel_map){
335         s->postin.ch_count=
336         s->midbuf.ch_count= s->used_ch_count;
337         if(s->resample)
338             s->in_buffer.ch_count= s->used_ch_count;
339     }
340     if(!s->resample_first){
341         s->midbuf.ch_count= s->out.ch_count;
342         if(s->resample)
343             s->in_buffer.ch_count = s->out.ch_count;
344     }
345
346     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
347     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
348     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
349
350     if(s->resample){
351         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
352     }
353
354     s->dither = s->preout;
355
356     if(s->rematrix || s->dither_method)
357         return swri_rematrix_init(s);
358
359     return 0;
360 }
361
362 static int realloc_audio(AudioData *a, int count){
363     int i, countb;
364     AudioData old;
365
366     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
367         return AVERROR(EINVAL);
368
369     if(a->count >= count)
370         return 0;
371
372     count*=2;
373
374     countb= FFALIGN(count*a->bps, ALIGN);
375     old= *a;
376
377     av_assert0(a->bps);
378     av_assert0(a->ch_count);
379
380     a->data= av_mallocz(countb*a->ch_count);
381     if(!a->data)
382         return AVERROR(ENOMEM);
383     for(i=0; i<a->ch_count; i++){
384         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
385         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
386     }
387     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
388     av_free(old.data);
389     a->count= count;
390
391     return 1;
392 }
393
394 static void copy(AudioData *out, AudioData *in,
395                  int count){
396     av_assert0(out->planar == in->planar);
397     av_assert0(out->bps == in->bps);
398     av_assert0(out->ch_count == in->ch_count);
399     if(out->planar){
400         int ch;
401         for(ch=0; ch<out->ch_count; ch++)
402             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
403     }else
404         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
405 }
406
407 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
408     int i;
409     if(!in_arg){
410         memset(out->ch, 0, sizeof(out->ch));
411     }else if(out->planar){
412         for(i=0; i<out->ch_count; i++)
413             out->ch[i]= in_arg[i];
414     }else{
415         for(i=0; i<out->ch_count; i++)
416             out->ch[i]= in_arg[0] + i*out->bps;
417     }
418 }
419
420 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
421     int i;
422     if(out->planar){
423         for(i=0; i<out->ch_count; i++)
424             in_arg[i]= out->ch[i];
425     }else{
426         in_arg[0]= out->ch[0];
427     }
428 }
429
430 /**
431  *
432  * out may be equal in.
433  */
434 static void buf_set(AudioData *out, AudioData *in, int count){
435     int ch;
436     if(in->planar){
437         for(ch=0; ch<out->ch_count; ch++)
438             out->ch[ch]= in->ch[ch] + count*out->bps;
439     }else{
440         for(ch=out->ch_count-1; ch>=0; ch--)
441             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
442     }
443 }
444
445 /**
446  *
447  * @return number of samples output per channel
448  */
449 static int resample(SwrContext *s, AudioData *out_param, int out_count,
450                              const AudioData * in_param, int in_count){
451     AudioData in, out, tmp;
452     int ret_sum=0;
453     int border=0;
454
455     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
456     av_assert1(s->in_buffer.planar   == in_param->planar);
457     av_assert1(s->in_buffer.fmt      == in_param->fmt);
458
459     tmp=out=*out_param;
460     in =  *in_param;
461
462     do{
463         int ret, size, consumed;
464         if(!s->resample_in_constraint && s->in_buffer_count){
465             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
466             ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
467             out_count -= ret;
468             ret_sum += ret;
469             buf_set(&out, &out, ret);
470             s->in_buffer_count -= consumed;
471             s->in_buffer_index += consumed;
472
473             if(!in_count)
474                 break;
475             if(s->in_buffer_count <= border){
476                 buf_set(&in, &in, -s->in_buffer_count);
477                 in_count += s->in_buffer_count;
478                 s->in_buffer_count=0;
479                 s->in_buffer_index=0;
480                 border = 0;
481             }
482         }
483
484         if(in_count && !s->in_buffer_count){
485             s->in_buffer_index=0;
486             ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
487             out_count -= ret;
488             ret_sum += ret;
489             buf_set(&out, &out, ret);
490             in_count -= consumed;
491             buf_set(&in, &in, consumed);
492         }
493
494         //TODO is this check sane considering the advanced copy avoidance below
495         size= s->in_buffer_index + s->in_buffer_count + in_count;
496         if(   size > s->in_buffer.count
497            && s->in_buffer_count + in_count <= s->in_buffer_index){
498             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
499             copy(&s->in_buffer, &tmp, s->in_buffer_count);
500             s->in_buffer_index=0;
501         }else
502             if((ret=realloc_audio(&s->in_buffer, size)) < 0)
503                 return ret;
504
505         if(in_count){
506             int count= in_count;
507             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
508
509             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
510             copy(&tmp, &in, /*in_*/count);
511             s->in_buffer_count += count;
512             in_count -= count;
513             border += count;
514             buf_set(&in, &in, count);
515             s->resample_in_constraint= 0;
516             if(s->in_buffer_count != count || in_count)
517                 continue;
518         }
519         break;
520     }while(1);
521
522     s->resample_in_constraint= !!out_count;
523
524     return ret_sum;
525 }
526
527 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
528                                                       AudioData *in , int  in_count){
529     AudioData *postin, *midbuf, *preout;
530     int ret/*, in_max*/;
531     AudioData preout_tmp, midbuf_tmp;
532
533     if(s->full_convert){
534         av_assert0(!s->resample);
535         swri_audio_convert(s->full_convert, out, in, in_count);
536         return out_count;
537     }
538
539 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
540 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
541
542     if((ret=realloc_audio(&s->postin, in_count))<0)
543         return ret;
544     if(s->resample_first){
545         av_assert0(s->midbuf.ch_count == s->used_ch_count);
546         if((ret=realloc_audio(&s->midbuf, out_count))<0)
547             return ret;
548     }else{
549         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
550         if((ret=realloc_audio(&s->midbuf,  in_count))<0)
551             return ret;
552     }
553     if((ret=realloc_audio(&s->preout, out_count))<0)
554         return ret;
555
556     postin= &s->postin;
557
558     midbuf_tmp= s->midbuf;
559     midbuf= &midbuf_tmp;
560     preout_tmp= s->preout;
561     preout= &preout_tmp;
562
563     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
564         postin= in;
565
566     if(s->resample_first ? !s->resample : !s->rematrix)
567         midbuf= postin;
568
569     if(s->resample_first ? !s->rematrix : !s->resample)
570         preout= midbuf;
571
572     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
573         if(preout==in){
574             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
575             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
576             copy(out, in, out_count);
577             return out_count;
578         }
579         else if(preout==postin) preout= midbuf= postin= out;
580         else if(preout==midbuf) preout= midbuf= out;
581         else                    preout= out;
582     }
583
584     if(in != postin){
585         swri_audio_convert(s->in_convert, postin, in, in_count);
586     }
587
588     if(s->resample_first){
589         if(postin != midbuf)
590             out_count= resample(s, midbuf, out_count, postin, in_count);
591         if(midbuf != preout)
592             swri_rematrix(s, preout, midbuf, out_count, preout==out);
593     }else{
594         if(postin != midbuf)
595             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
596         if(midbuf != preout)
597             out_count= resample(s, preout, out_count, midbuf, in_count);
598     }
599
600     if(preout != out && out_count){
601         if(s->dither_method){
602             int ch;
603             int dither_count= FFMAX(out_count, 1<<16);
604             av_assert0(preout != in);
605
606             if((ret=realloc_audio(&s->dither, dither_count))<0)
607                 return ret;
608             if(ret)
609                 for(ch=0; ch<s->dither.ch_count; ch++)
610                     swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
611             av_assert0(s->dither.ch_count == preout->ch_count);
612
613             if(s->dither_pos + out_count > s->dither.count)
614                 s->dither_pos = 0;
615
616             for(ch=0; ch<preout->ch_count; ch++)
617                 s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
618
619             s->dither_pos += out_count;
620         }
621 //FIXME packed doesnt need more than 1 chan here!
622         swri_audio_convert(s->out_convert, out, preout, out_count);
623     }
624     return out_count;
625 }
626
627 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
628                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
629     AudioData * in= &s->in;
630     AudioData *out= &s->out;
631
632     if(s->drop_output > 0){
633         int ret;
634         AudioData tmp = s->out;
635         uint8_t *tmp_arg[SWR_CH_MAX];
636         tmp.count = 0;
637         tmp.data  = NULL;
638         if((ret=realloc_audio(&tmp, s->drop_output))<0)
639             return ret;
640
641         reversefill_audiodata(&tmp, tmp_arg);
642         s->drop_output *= -1; //FIXME find a less hackish solution
643         ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
644         s->drop_output *= -1;
645         if(ret>0)
646             s->drop_output -= ret;
647
648         av_freep(&tmp.data);
649         if(s->drop_output || !out_arg)
650             return 0;
651         in_count = 0;
652     }
653
654     if(!in_arg){
655         if(s->in_buffer_count){
656             if (s->resample && !s->flushed) {
657                 AudioData *a= &s->in_buffer;
658                 int i, j, ret;
659                 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
660                     return ret;
661                 av_assert0(a->planar);
662                 for(i=0; i<a->ch_count; i++){
663                     for(j=0; j<s->in_buffer_count; j++){
664                         memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j  )*a->bps,
665                             a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
666                     }
667                 }
668                 s->in_buffer_count += (s->in_buffer_count+1)/2;
669                 s->resample_in_constraint = 0;
670                 s->flushed = 1;
671             }
672         }else{
673             return 0;
674         }
675     }else
676         fill_audiodata(in ,  (void*)in_arg);
677
678     fill_audiodata(out, out_arg);
679
680     if(s->resample){
681         int ret = swr_convert_internal(s, out, out_count, in, in_count);
682         if(ret>0 && !s->drop_output)
683             s->outpts += ret * (int64_t)s->in_sample_rate;
684         return ret;
685     }else{
686         AudioData tmp= *in;
687         int ret2=0;
688         int ret, size;
689         size = FFMIN(out_count, s->in_buffer_count);
690         if(size){
691             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
692             ret= swr_convert_internal(s, out, size, &tmp, size);
693             if(ret<0)
694                 return ret;
695             ret2= ret;
696             s->in_buffer_count -= ret;
697             s->in_buffer_index += ret;
698             buf_set(out, out, ret);
699             out_count -= ret;
700             if(!s->in_buffer_count)
701                 s->in_buffer_index = 0;
702         }
703
704         if(in_count){
705             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
706
707             if(in_count > out_count) { //FIXME move after swr_convert_internal
708                 if(   size > s->in_buffer.count
709                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
710                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
711                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
712                     s->in_buffer_index=0;
713                 }else
714                     if((ret=realloc_audio(&s->in_buffer, size)) < 0)
715                         return ret;
716             }
717
718             if(out_count){
719                 size = FFMIN(in_count, out_count);
720                 ret= swr_convert_internal(s, out, size, in, size);
721                 if(ret<0)
722                     return ret;
723                 buf_set(in, in, ret);
724                 in_count -= ret;
725                 ret2 += ret;
726             }
727             if(in_count){
728                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
729                 copy(&tmp, in, in_count);
730                 s->in_buffer_count += in_count;
731             }
732         }
733         if(ret2>0 && !s->drop_output)
734             s->outpts += ret2 * (int64_t)s->in_sample_rate;
735         return ret2;
736     }
737 }
738
739 int swr_drop_output(struct SwrContext *s, int count){
740     s->drop_output += count;
741
742     if(s->drop_output <= 0)
743         return 0;
744
745     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
746     return swr_convert(s, NULL, s->drop_output, NULL, 0);
747 }
748
749 int swr_inject_silence(struct SwrContext *s, int count){
750     int ret, i;
751     AudioData silence = s->in;
752     uint8_t *tmp_arg[SWR_CH_MAX];
753
754     if(count <= 0)
755         return 0;
756
757     silence.count = 0;
758     silence.data  = NULL;
759     if((ret=realloc_audio(&silence, count))<0)
760         return ret;
761
762     if(silence.planar) for(i=0; i<silence.ch_count; i++) {
763         memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
764     } else
765         memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
766
767     reversefill_audiodata(&silence, tmp_arg);
768     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
769     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
770     av_freep(&silence.data);
771     return ret;
772 }
773
774 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
775     if(pts == INT64_MIN)
776         return s->outpts;
777     if(s->min_compensation >= FLT_MAX) {
778         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
779     } else {
780         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
781         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
782
783         if(fabs(fdelta) > s->min_compensation) {
784             if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
785                 int ret;
786                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
787                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
788                 if(ret<0){
789                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
790                 }
791             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
792                 int duration = s->out_sample_rate * s->soft_compensation_duration;
793                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
794                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
795                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
796                 swr_set_compensation(s, comp, duration);
797             }
798         }
799
800         return s->outpts;
801     }
802 }