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