cosmetics: Fix "dont" "wont" "doesnt" typos
[ffmpeg.git] / libswresample / swresample.c
1 /*
2  * Copyright (C) 2011-2013 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_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
55 {"in_sample_fmt"        , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
56 {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
57 {"out_sample_fmt"       , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
58 {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, PARAM},
59 {"internal_sample_fmt"  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , AV_SAMPLE_FMT_NB-1, 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 {"lipshitz"             , "select lipshitz noise shaping dither" , 0             , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},
83 {"shibata"              , "select shibata noise shaping dither" , 0              , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
84 {"low_shibata"          , "select low shibata noise shaping dither" , 0          , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
85 {"high_shibata"         , "select high shibata noise shaping dither" , 0         , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
86 {"f_weighted"           , "select f-weighted noise shaping dither" , 0           , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
87 {"modified_e_weighted"  , "select modified-e-weighted noise shaping dither" , 0  , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
88 {"improved_e_weighted"  , "select improved-e-weighted noise shaping dither" , 0  , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},
89
90 {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=32                    }, 0      , INT_MAX   , PARAM },
91 {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 24        , PARAM },
92 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
93 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
94
95 /* duplicate option in order to work with avconv */
96 {"resample_cutoff"      , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
97
98 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
99 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},
100 {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},
101 {"precision"            , "set soxr resampling precision (in bits)"
102                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
103 {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
104                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
105 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
106                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
107 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
108                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
109 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
110                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
111 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
112                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
113 {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
114                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
115 {"first_pts"            , "Assume the first pts should be this value (in samples)."
116                                                         , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE    }, INT64_MIN,INT64_MAX, PARAM },
117
118 { "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" },
119     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
120     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
121     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
122
123 { "filter_type"         , "select swr 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" },
124     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
125     { "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" },
126     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
127
128 { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
129
130 { "output_sample_bits"  , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT  , {.i64=0   }, 0      , 64        , PARAM },
131 {0}
132 };
133
134 static const char* context_to_name(void* ptr) {
135     return "SWR";
136 }
137
138 static const AVClass av_class = {
139     .class_name                = "SWResampler",
140     .item_name                 = context_to_name,
141     .option                    = options,
142     .version                   = LIBAVUTIL_VERSION_INT,
143     .log_level_offset_offset   = OFFSET(log_level_offset),
144     .parent_log_context_offset = OFFSET(log_ctx),
145     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
146 };
147
148 unsigned swresample_version(void)
149 {
150     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
151     return LIBSWRESAMPLE_VERSION_INT;
152 }
153
154 const char *swresample_configuration(void)
155 {
156     return FFMPEG_CONFIGURATION;
157 }
158
159 const char *swresample_license(void)
160 {
161 #define LICENSE_PREFIX "libswresample license: "
162     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
163 }
164
165 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
166     if(!s || s->in_convert) // s needs to be allocated but not initialized
167         return AVERROR(EINVAL);
168     s->channel_map = channel_map;
169     return 0;
170 }
171
172 const AVClass *swr_get_class(void)
173 {
174     return &av_class;
175 }
176
177 av_cold struct SwrContext *swr_alloc(void){
178     SwrContext *s= av_mallocz(sizeof(SwrContext));
179     if(s){
180         s->av_class= &av_class;
181         av_opt_set_defaults(s);
182     }
183     return s;
184 }
185
186 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
187                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
188                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
189                                       int log_offset, void *log_ctx){
190     if(!s) s= swr_alloc();
191     if(!s) return NULL;
192
193     s->log_level_offset= log_offset;
194     s->log_ctx= log_ctx;
195
196     av_opt_set_int(s, "ocl", out_ch_layout,   0);
197     av_opt_set_int(s, "osf", out_sample_fmt,  0);
198     av_opt_set_int(s, "osr", out_sample_rate, 0);
199     av_opt_set_int(s, "icl", in_ch_layout,    0);
200     av_opt_set_int(s, "isf", in_sample_fmt,   0);
201     av_opt_set_int(s, "isr", in_sample_rate,  0);
202     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
203     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
204     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
205     av_opt_set_int(s, "uch", 0, 0);
206     return s;
207 }
208
209 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
210     a->fmt   = fmt;
211     a->bps   = av_get_bytes_per_sample(fmt);
212     a->planar= av_sample_fmt_is_planar(fmt);
213 }
214
215 static void free_temp(AudioData *a){
216     av_free(a->data);
217     memset(a, 0, sizeof(*a));
218 }
219
220 av_cold void swr_free(SwrContext **ss){
221     SwrContext *s= *ss;
222     if(s){
223         free_temp(&s->postin);
224         free_temp(&s->midbuf);
225         free_temp(&s->preout);
226         free_temp(&s->in_buffer);
227         free_temp(&s->silence);
228         free_temp(&s->drop_temp);
229         free_temp(&s->dither.noise);
230         free_temp(&s->dither.temp);
231         swri_audio_convert_free(&s-> in_convert);
232         swri_audio_convert_free(&s->out_convert);
233         swri_audio_convert_free(&s->full_convert);
234         if (s->resampler)
235             s->resampler->free(&s->resample);
236         swri_rematrix_free(s);
237     }
238
239     av_freep(ss);
240 }
241
242 av_cold int swr_init(struct SwrContext *s){
243     int ret;
244     s->in_buffer_index= 0;
245     s->in_buffer_count= 0;
246     s->resample_in_constraint= 0;
247     free_temp(&s->postin);
248     free_temp(&s->midbuf);
249     free_temp(&s->preout);
250     free_temp(&s->in_buffer);
251     free_temp(&s->silence);
252     free_temp(&s->drop_temp);
253     free_temp(&s->dither.noise);
254     free_temp(&s->dither.temp);
255     memset(s->in.ch, 0, sizeof(s->in.ch));
256     memset(s->out.ch, 0, sizeof(s->out.ch));
257     swri_audio_convert_free(&s-> in_convert);
258     swri_audio_convert_free(&s->out_convert);
259     swri_audio_convert_free(&s->full_convert);
260     swri_rematrix_free(s);
261
262     s->flushed = 0;
263
264     if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
265         av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
266         return AVERROR(EINVAL);
267     }
268     if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
269         av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
270         return AVERROR(EINVAL);
271     }
272
273     if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
274         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
275         s->in_ch_layout = 0;
276     }
277
278     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
279         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
280         s->out_ch_layout = 0;
281     }
282
283     switch(s->engine){
284 #if CONFIG_LIBSOXR
285         extern struct Resampler const soxr_resampler;
286         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
287 #endif
288         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
289         default:
290             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
291             return AVERROR(EINVAL);
292     }
293
294     if(!s->used_ch_count)
295         s->used_ch_count= s->in.ch_count;
296
297     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
298         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
299         s-> in_ch_layout= 0;
300     }
301
302     if(!s-> in_ch_layout)
303         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
304     if(!s->out_ch_layout)
305         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
306
307     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
308                  s->rematrix_custom;
309
310     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
311         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
312             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
313         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
314                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
315                  && !s->rematrix
316                  && s->engine != SWR_ENGINE_SOXR){
317             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
318         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
319             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
320         }else{
321             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
322             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
323         }
324     }
325
326     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
327         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
328         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
329         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
330         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));
331         return AVERROR(EINVAL);
332     }
333
334     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
335     set_audiodata_fmt(&s->out, s->out_sample_fmt);
336
337     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
338         if (!s->async && s->min_compensation >= FLT_MAX/2)
339             s->async = 1;
340         s->firstpts =
341         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
342     } else
343         s->firstpts = AV_NOPTS_VALUE;
344
345     if (s->async) {
346         if (s->min_compensation >= FLT_MAX/2)
347             s->min_compensation = 0.001;
348         if (s->async > 1.0001) {
349             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
350         }
351     }
352
353     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
354         s->resample = s->resampler->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, s->precision, s->cheby);
355     }else
356         s->resampler->free(&s->resample);
357     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
358         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
359         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
360         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
361         && s->resample){
362         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
363         return -1;
364     }
365
366 #define RSC 1 //FIXME finetune
367     if(!s-> in.ch_count)
368         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
369     if(!s->used_ch_count)
370         s->used_ch_count= s->in.ch_count;
371     if(!s->out.ch_count)
372         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
373
374     if(!s-> in.ch_count){
375         av_assert0(!s->in_ch_layout);
376         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
377         return -1;
378     }
379
380     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
381         char l1[1024], l2[1024];
382         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
383         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
384         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
385                "but there is not enough information to do it\n", l1, l2);
386         return -1;
387     }
388
389 av_assert0(s->used_ch_count);
390 av_assert0(s->out.ch_count);
391     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
392
393     s->in_buffer= s->in;
394     s->silence  = s->in;
395     s->drop_temp= s->out;
396
397     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
398         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
399                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
400         return 0;
401     }
402
403     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
404                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
405     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
406                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
407
408     if (!s->in_convert || !s->out_convert)
409         return AVERROR(ENOMEM);
410
411     s->postin= s->in;
412     s->preout= s->out;
413     s->midbuf= s->in;
414
415     if(s->channel_map){
416         s->postin.ch_count=
417         s->midbuf.ch_count= s->used_ch_count;
418         if(s->resample)
419             s->in_buffer.ch_count= s->used_ch_count;
420     }
421     if(!s->resample_first){
422         s->midbuf.ch_count= s->out.ch_count;
423         if(s->resample)
424             s->in_buffer.ch_count = s->out.ch_count;
425     }
426
427     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
428     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
429     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
430
431     if(s->resample){
432         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
433     }
434
435     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
436         return ret;
437
438     if(s->rematrix || s->dither.method)
439         return swri_rematrix_init(s);
440
441     return 0;
442 }
443
444 int swri_realloc_audio(AudioData *a, int count){
445     int i, countb;
446     AudioData old;
447
448     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
449         return AVERROR(EINVAL);
450
451     if(a->count >= count)
452         return 0;
453
454     count*=2;
455
456     countb= FFALIGN(count*a->bps, ALIGN);
457     old= *a;
458
459     av_assert0(a->bps);
460     av_assert0(a->ch_count);
461
462     a->data= av_mallocz(countb*a->ch_count);
463     if(!a->data)
464         return AVERROR(ENOMEM);
465     for(i=0; i<a->ch_count; i++){
466         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
467         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
468     }
469     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
470     av_free(old.data);
471     a->count= count;
472
473     return 1;
474 }
475
476 static void copy(AudioData *out, AudioData *in,
477                  int count){
478     av_assert0(out->planar == in->planar);
479     av_assert0(out->bps == in->bps);
480     av_assert0(out->ch_count == in->ch_count);
481     if(out->planar){
482         int ch;
483         for(ch=0; ch<out->ch_count; ch++)
484             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
485     }else
486         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
487 }
488
489 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
490     int i;
491     if(!in_arg){
492         memset(out->ch, 0, sizeof(out->ch));
493     }else if(out->planar){
494         for(i=0; i<out->ch_count; i++)
495             out->ch[i]= in_arg[i];
496     }else{
497         for(i=0; i<out->ch_count; i++)
498             out->ch[i]= in_arg[0] + i*out->bps;
499     }
500 }
501
502 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
503     int i;
504     if(out->planar){
505         for(i=0; i<out->ch_count; i++)
506             in_arg[i]= out->ch[i];
507     }else{
508         in_arg[0]= out->ch[0];
509     }
510 }
511
512 /**
513  *
514  * out may be equal in.
515  */
516 static void buf_set(AudioData *out, AudioData *in, int count){
517     int ch;
518     if(in->planar){
519         for(ch=0; ch<out->ch_count; ch++)
520             out->ch[ch]= in->ch[ch] + count*out->bps;
521     }else{
522         for(ch=out->ch_count-1; ch>=0; ch--)
523             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
524     }
525 }
526
527 /**
528  *
529  * @return number of samples output per channel
530  */
531 static int resample(SwrContext *s, AudioData *out_param, int out_count,
532                              const AudioData * in_param, int in_count){
533     AudioData in, out, tmp;
534     int ret_sum=0;
535     int border=0;
536
537     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
538     av_assert1(s->in_buffer.planar   == in_param->planar);
539     av_assert1(s->in_buffer.fmt      == in_param->fmt);
540
541     tmp=out=*out_param;
542     in =  *in_param;
543
544     do{
545         int ret, size, consumed;
546         if(!s->resample_in_constraint && s->in_buffer_count){
547             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
548             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
549             out_count -= ret;
550             ret_sum += ret;
551             buf_set(&out, &out, ret);
552             s->in_buffer_count -= consumed;
553             s->in_buffer_index += consumed;
554
555             if(!in_count)
556                 break;
557             if(s->in_buffer_count <= border){
558                 buf_set(&in, &in, -s->in_buffer_count);
559                 in_count += s->in_buffer_count;
560                 s->in_buffer_count=0;
561                 s->in_buffer_index=0;
562                 border = 0;
563             }
564         }
565
566         if((s->flushed || in_count) && !s->in_buffer_count){
567             s->in_buffer_index=0;
568             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
569             out_count -= ret;
570             ret_sum += ret;
571             buf_set(&out, &out, ret);
572             in_count -= consumed;
573             buf_set(&in, &in, consumed);
574         }
575
576         //TODO is this check sane considering the advanced copy avoidance below
577         size= s->in_buffer_index + s->in_buffer_count + in_count;
578         if(   size > s->in_buffer.count
579            && s->in_buffer_count + in_count <= s->in_buffer_index){
580             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
581             copy(&s->in_buffer, &tmp, s->in_buffer_count);
582             s->in_buffer_index=0;
583         }else
584             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
585                 return ret;
586
587         if(in_count){
588             int count= in_count;
589             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
590
591             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
592             copy(&tmp, &in, /*in_*/count);
593             s->in_buffer_count += count;
594             in_count -= count;
595             border += count;
596             buf_set(&in, &in, count);
597             s->resample_in_constraint= 0;
598             if(s->in_buffer_count != count || in_count)
599                 continue;
600         }
601         break;
602     }while(1);
603
604     s->resample_in_constraint= !!out_count;
605
606     return ret_sum;
607 }
608
609 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
610                                                       AudioData *in , int  in_count){
611     AudioData *postin, *midbuf, *preout;
612     int ret/*, in_max*/;
613     AudioData preout_tmp, midbuf_tmp;
614
615     if(s->full_convert){
616         av_assert0(!s->resample);
617         swri_audio_convert(s->full_convert, out, in, in_count);
618         return out_count;
619     }
620
621 //     in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
622 //     in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
623
624     if((ret=swri_realloc_audio(&s->postin, in_count))<0)
625         return ret;
626     if(s->resample_first){
627         av_assert0(s->midbuf.ch_count == s->used_ch_count);
628         if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
629             return ret;
630     }else{
631         av_assert0(s->midbuf.ch_count ==  s->out.ch_count);
632         if((ret=swri_realloc_audio(&s->midbuf,  in_count))<0)
633             return ret;
634     }
635     if((ret=swri_realloc_audio(&s->preout, out_count))<0)
636         return ret;
637
638     postin= &s->postin;
639
640     midbuf_tmp= s->midbuf;
641     midbuf= &midbuf_tmp;
642     preout_tmp= s->preout;
643     preout= &preout_tmp;
644
645     if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
646         postin= in;
647
648     if(s->resample_first ? !s->resample : !s->rematrix)
649         midbuf= postin;
650
651     if(s->resample_first ? !s->rematrix : !s->resample)
652         preout= midbuf;
653
654     if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
655        && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
656         if(preout==in){
657             out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
658             av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
659             copy(out, in, out_count);
660             return out_count;
661         }
662         else if(preout==postin) preout= midbuf= postin= out;
663         else if(preout==midbuf) preout= midbuf= out;
664         else                    preout= out;
665     }
666
667     if(in != postin){
668         swri_audio_convert(s->in_convert, postin, in, in_count);
669     }
670
671     if(s->resample_first){
672         if(postin != midbuf)
673             out_count= resample(s, midbuf, out_count, postin, in_count);
674         if(midbuf != preout)
675             swri_rematrix(s, preout, midbuf, out_count, preout==out);
676     }else{
677         if(postin != midbuf)
678             swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
679         if(midbuf != preout)
680             out_count= resample(s, preout, out_count, midbuf, in_count);
681     }
682
683     if(preout != out && out_count){
684         AudioData *conv_src = preout;
685         if(s->dither.method){
686             int ch;
687             int dither_count= FFMAX(out_count, 1<<16);
688
689             if (preout == in) {
690                 conv_src = &s->dither.temp;
691                 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
692                     return ret;
693             }
694
695             if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
696                 return ret;
697             if(ret)
698                 for(ch=0; ch<s->dither.noise.ch_count; ch++)
699                     swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
700             av_assert0(s->dither.noise.ch_count == preout->ch_count);
701
702             if(s->dither.noise_pos + out_count > s->dither.noise.count)
703                 s->dither.noise_pos = 0;
704
705             if (s->dither.method < SWR_DITHER_NS){
706                 if (s->mix_2_1_simd) {
707                     int len1= out_count&~15;
708                     int off = len1 * preout->bps;
709
710                     if(len1)
711                         for(ch=0; ch<preout->ch_count; ch++)
712                             s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);
713                     if(out_count != len1)
714                         for(ch=0; ch<preout->ch_count; ch++)
715                             s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1);
716                 } else {
717                     for(ch=0; ch<preout->ch_count; ch++)
718                         s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
719                 }
720             } else {
721                 switch(s->int_sample_fmt) {
722                 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
723                 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
724                 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
725                 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
726                 }
727             }
728             s->dither.noise_pos += out_count;
729         }
730 //FIXME packed doesn't need more than 1 chan here!
731         swri_audio_convert(s->out_convert, out, conv_src, out_count);
732     }
733     return out_count;
734 }
735
736 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
737                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
738     AudioData * in= &s->in;
739     AudioData *out= &s->out;
740
741     while(s->drop_output > 0){
742         int ret;
743         uint8_t *tmp_arg[SWR_CH_MAX];
744 #define MAX_DROP_STEP 16384
745         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
746             return ret;
747
748         reversefill_audiodata(&s->drop_temp, tmp_arg);
749         s->drop_output *= -1; //FIXME find a less hackish solution
750         ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
751         s->drop_output *= -1;
752         in_count = 0;
753         if(ret>0) {
754             s->drop_output -= ret;
755             continue;
756         }
757
758         if(s->drop_output || !out_arg)
759             return 0;
760     }
761
762     if(!in_arg){
763         if(s->resample){
764             if (!s->flushed)
765                 s->resampler->flush(s);
766             s->resample_in_constraint = 0;
767             s->flushed = 1;
768         }else if(!s->in_buffer_count){
769             return 0;
770         }
771     }else
772         fill_audiodata(in ,  (void*)in_arg);
773
774     fill_audiodata(out, out_arg);
775
776     if(s->resample){
777         int ret = swr_convert_internal(s, out, out_count, in, in_count);
778         if(ret>0 && !s->drop_output)
779             s->outpts += ret * (int64_t)s->in_sample_rate;
780         return ret;
781     }else{
782         AudioData tmp= *in;
783         int ret2=0;
784         int ret, size;
785         size = FFMIN(out_count, s->in_buffer_count);
786         if(size){
787             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
788             ret= swr_convert_internal(s, out, size, &tmp, size);
789             if(ret<0)
790                 return ret;
791             ret2= ret;
792             s->in_buffer_count -= ret;
793             s->in_buffer_index += ret;
794             buf_set(out, out, ret);
795             out_count -= ret;
796             if(!s->in_buffer_count)
797                 s->in_buffer_index = 0;
798         }
799
800         if(in_count){
801             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
802
803             if(in_count > out_count) { //FIXME move after swr_convert_internal
804                 if(   size > s->in_buffer.count
805                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
806                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
807                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
808                     s->in_buffer_index=0;
809                 }else
810                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
811                         return ret;
812             }
813
814             if(out_count){
815                 size = FFMIN(in_count, out_count);
816                 ret= swr_convert_internal(s, out, size, in, size);
817                 if(ret<0)
818                     return ret;
819                 buf_set(in, in, ret);
820                 in_count -= ret;
821                 ret2 += ret;
822             }
823             if(in_count){
824                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
825                 copy(&tmp, in, in_count);
826                 s->in_buffer_count += in_count;
827             }
828         }
829         if(ret2>0 && !s->drop_output)
830             s->outpts += ret2 * (int64_t)s->in_sample_rate;
831         return ret2;
832     }
833 }
834
835 int swr_drop_output(struct SwrContext *s, int count){
836     s->drop_output += count;
837
838     if(s->drop_output <= 0)
839         return 0;
840
841     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
842     return swr_convert(s, NULL, s->drop_output, NULL, 0);
843 }
844
845 int swr_inject_silence(struct SwrContext *s, int count){
846     int ret, i;
847     uint8_t *tmp_arg[SWR_CH_MAX];
848
849     if(count <= 0)
850         return 0;
851
852 #define MAX_SILENCE_STEP 16384
853     while (count > MAX_SILENCE_STEP) {
854         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
855             return ret;
856         count -= MAX_SILENCE_STEP;
857     }
858
859     if((ret=swri_realloc_audio(&s->silence, count))<0)
860         return ret;
861
862     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
863         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
864     } else
865         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
866
867     reversefill_audiodata(&s->silence, tmp_arg);
868     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
869     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
870     return ret;
871 }
872
873 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
874     if (s->resampler && s->resample){
875         return s->resampler->get_delay(s, base);
876     }else{
877         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
878     }
879 }
880
881 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
882     int ret;
883
884     if (!s || compensation_distance < 0)
885         return AVERROR(EINVAL);
886     if (!compensation_distance && sample_delta)
887         return AVERROR(EINVAL);
888     if (!s->resample) {
889         s->flags |= SWR_FLAG_RESAMPLE;
890         ret = swr_init(s);
891         if (ret < 0)
892             return ret;
893     }
894     if (!s->resampler->set_compensation){
895         return AVERROR(EINVAL);
896     }else{
897         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
898     }
899 }
900
901 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
902     if(pts == INT64_MIN)
903         return s->outpts;
904
905     if (s->firstpts == AV_NOPTS_VALUE)
906         s->outpts = s->firstpts = pts;
907
908     if(s->min_compensation >= FLT_MAX) {
909         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
910     } else {
911         int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
912         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
913
914         if(fabs(fdelta) > s->min_compensation) {
915             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
916                 int ret;
917                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
918                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
919                 if(ret<0){
920                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
921                 }
922             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
923                 int duration = s->out_sample_rate * s->soft_compensation_duration;
924                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
925                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
926                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
927                 swr_set_compensation(s, comp, duration);
928             }
929         }
930
931         return s->outpts;
932     }
933 }