swresample: add swr_is_initialized()
[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   , INT_MAX, PARAM},
55 {"in_sample_fmt"        , "set input sample format"     , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
56 {"osf"                  , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
57 {"out_sample_fmt"       , "set output sample format"    , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
58 {"tsf"                  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
59 {"internal_sample_fmt"  , "set internal sample format"  , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1   , INT_MAX, PARAM},
60 {"icl"                  , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
61 {"in_channel_layout"    , "set input channel layout"    , OFFSET( in_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
62 {"ocl"                  , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0                     }, 0      , INT64_MAX , PARAM, "channel_layout"},
63 {"out_channel_layout"   , "set output channel layout"   , OFFSET(out_ch_layout  ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.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 {"rematrix_maxval"      , "set rematrix maxval"         , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0                   }, 0      , 1000      , PARAM},
72
73 {"flags"                , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
74 {"swr_flags"            , "set flags"                   , OFFSET(flags          ), AV_OPT_TYPE_FLAGS, {.i64=0                     }, 0      , UINT_MAX  , PARAM, "flags"},
75 {"res"                  , "force resampling"            , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE     }, INT_MIN, INT_MAX   , PARAM, "flags"},
76
77 {"dither_scale"         , "set dither scale"            , OFFSET(dither.scale   ), AV_OPT_TYPE_FLOAT, {.dbl=1                     }, 0      , INT_MAX   , PARAM},
78
79 {"dither_method"        , "set dither method"           , OFFSET(dither.method  ), AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_DITHER_NB-1, PARAM, "dither_method"},
80 {"rectangular"          , "select rectangular dither"   , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX   , PARAM, "dither_method"},
81 {"triangular"           , "select triangular dither"    , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX   , PARAM, "dither_method"},
82 {"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"},
83 {"lipshitz"             , "select lipshitz noise shaping dither" , 0             , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},
84 {"shibata"              , "select shibata noise shaping dither" , 0              , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},
85 {"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"},
86 {"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"},
87 {"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"},
88 {"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"},
89 {"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"},
90
91 {"filter_size"          , "set swr resampling filter size", OFFSET(filter_size)  , AV_OPT_TYPE_INT  , {.i64=32                    }, 0      , INT_MAX   , PARAM },
92 {"phase_shift"          , "set swr resampling phase shift", OFFSET(phase_shift)  , AV_OPT_TYPE_INT  , {.i64=10                    }, 0      , 24        , PARAM },
93 {"linear_interp"        , "enable linear interpolation" , OFFSET(linear_interp)  , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
94 {"cutoff"               , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
95
96 /* duplicate option in order to work with avconv */
97 {"resample_cutoff"      , "set cutoff frequency ratio"  , OFFSET(cutoff)         , AV_OPT_TYPE_DOUBLE,{.dbl=0.                    }, 0      , 1         , PARAM },
98
99 {"resampler"            , "set resampling Engine"       , OFFSET(engine)         , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , SWR_ENGINE_NB-1, PARAM, "resampler"},
100 {"swr"                  , "select SW Resampler"         , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR        }, INT_MIN, INT_MAX   , PARAM, "resampler"},
101 {"soxr"                 , "select SoX Resampler"        , 0                      , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR       }, INT_MIN, INT_MAX   , PARAM, "resampler"},
102 {"precision"            , "set soxr resampling precision (in bits)"
103                                                         , OFFSET(precision)      , AV_OPT_TYPE_DOUBLE,{.dbl=20.0                  }, 15.0   , 33.0      , PARAM },
104 {"cheby"                , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"
105                                                         , OFFSET(cheby)          , AV_OPT_TYPE_INT  , {.i64=0                     }, 0      , 1         , PARAM },
106 {"min_comp"             , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
107                                                         , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX               }, 0      , FLT_MAX   , PARAM },
108 {"min_hard_comp"        , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
109                                                         , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1                   }, 0      , INT_MAX   , PARAM },
110 {"comp_duration"        , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
111                                                         , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1                     }, 0      , INT_MAX   , PARAM },
112 {"max_soft_comp"        , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
113                                                         , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
114 {"async"                , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"
115                                                         , OFFSET(async)          , AV_OPT_TYPE_FLOAT ,{.dbl=0                     }, INT_MIN, INT_MAX   , PARAM },
116 {"first_pts"            , "Assume the first pts should be this value (in samples)."
117                                                         , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE    }, INT64_MIN,INT64_MAX, PARAM },
118
119 { "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" },
120     { "none",  "select none",               0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE  }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
121     { "dolby", "select Dolby",              0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
122     { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
123
124 { "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" },
125     { "cubic"           , "select cubic"                , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC            }, INT_MIN, INT_MAX, PARAM, "filter_type" },
126     { "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" },
127     { "kaiser"          , "select Kaiser Windowed Sinc" , 0                      , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER           }, INT_MIN, INT_MAX, PARAM, "filter_type" },
128
129 { "kaiser_beta"         , "set swr Kaiser Window Beta"  , OFFSET(kaiser_beta)    , AV_OPT_TYPE_INT  , {.i64=9                     }, 2      , 16        , PARAM },
130
131 { "output_sample_bits"  , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT  , {.i64=0   }, 0      , 64        , PARAM },
132 {0}
133 };
134
135 static const char* context_to_name(void* ptr) {
136     return "SWR";
137 }
138
139 static const AVClass av_class = {
140     .class_name                = "SWResampler",
141     .item_name                 = context_to_name,
142     .option                    = options,
143     .version                   = LIBAVUTIL_VERSION_INT,
144     .log_level_offset_offset   = OFFSET(log_level_offset),
145     .parent_log_context_offset = OFFSET(log_ctx),
146     .category                  = AV_CLASS_CATEGORY_SWRESAMPLER,
147 };
148
149 unsigned swresample_version(void)
150 {
151     av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
152     return LIBSWRESAMPLE_VERSION_INT;
153 }
154
155 const char *swresample_configuration(void)
156 {
157     return FFMPEG_CONFIGURATION;
158 }
159
160 const char *swresample_license(void)
161 {
162 #define LICENSE_PREFIX "libswresample license: "
163     return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
164 }
165
166 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
167     if(!s || s->in_convert) // s needs to be allocated but not initialized
168         return AVERROR(EINVAL);
169     s->channel_map = channel_map;
170     return 0;
171 }
172
173 const AVClass *swr_get_class(void)
174 {
175     return &av_class;
176 }
177
178 av_cold struct SwrContext *swr_alloc(void){
179     SwrContext *s= av_mallocz(sizeof(SwrContext));
180     if(s){
181         s->av_class= &av_class;
182         av_opt_set_defaults(s);
183     }
184     return s;
185 }
186
187 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
188                                       int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
189                                       int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
190                                       int log_offset, void *log_ctx){
191     if(!s) s= swr_alloc();
192     if(!s) return NULL;
193
194     s->log_level_offset= log_offset;
195     s->log_ctx= log_ctx;
196
197     av_opt_set_int(s, "ocl", out_ch_layout,   0);
198     av_opt_set_int(s, "osf", out_sample_fmt,  0);
199     av_opt_set_int(s, "osr", out_sample_rate, 0);
200     av_opt_set_int(s, "icl", in_ch_layout,    0);
201     av_opt_set_int(s, "isf", in_sample_fmt,   0);
202     av_opt_set_int(s, "isr", in_sample_rate,  0);
203     av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE,   0);
204     av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
205     av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
206     av_opt_set_int(s, "uch", 0, 0);
207     return s;
208 }
209
210 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
211     a->fmt   = fmt;
212     a->bps   = av_get_bytes_per_sample(fmt);
213     a->planar= av_sample_fmt_is_planar(fmt);
214 }
215
216 static void free_temp(AudioData *a){
217     av_free(a->data);
218     memset(a, 0, sizeof(*a));
219 }
220
221 static void clear_context(SwrContext *s){
222     s->in_buffer_index= 0;
223     s->in_buffer_count= 0;
224     s->resample_in_constraint= 0;
225     memset(s->in.ch, 0, sizeof(s->in.ch));
226     memset(s->out.ch, 0, sizeof(s->out.ch));
227     free_temp(&s->postin);
228     free_temp(&s->midbuf);
229     free_temp(&s->preout);
230     free_temp(&s->in_buffer);
231     free_temp(&s->silence);
232     free_temp(&s->drop_temp);
233     free_temp(&s->dither.noise);
234     free_temp(&s->dither.temp);
235     swri_audio_convert_free(&s-> in_convert);
236     swri_audio_convert_free(&s->out_convert);
237     swri_audio_convert_free(&s->full_convert);
238     swri_rematrix_free(s);
239
240     s->flushed = 0;
241 }
242
243 av_cold void swr_free(SwrContext **ss){
244     SwrContext *s= *ss;
245     if(s){
246         clear_context(s);
247         if (s->resampler)
248             s->resampler->free(&s->resample);
249     }
250
251     av_freep(ss);
252 }
253
254 av_cold int swr_init(struct SwrContext *s){
255     int ret;
256
257     clear_context(s);
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(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
269         av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
270         s->in_ch_layout = 0;
271     }
272
273     if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
274         av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
275         s->out_ch_layout = 0;
276     }
277
278     switch(s->engine){
279 #if CONFIG_LIBSOXR
280         extern struct Resampler const soxr_resampler;
281         case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
282 #endif
283         case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
284         default:
285             av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
286             return AVERROR(EINVAL);
287     }
288
289     if(!s->used_ch_count)
290         s->used_ch_count= s->in.ch_count;
291
292     if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
293         av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
294         s-> in_ch_layout= 0;
295     }
296
297     if(!s-> in_ch_layout)
298         s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
299     if(!s->out_ch_layout)
300         s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
301
302     s->rematrix= s->out_ch_layout  !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
303                  s->rematrix_custom;
304
305     if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
306         if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
307             s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
308         }else if(   av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
309                  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
310                  && !s->rematrix
311                  && s->engine != SWR_ENGINE_SOXR){
312             s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
313         }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
314             s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
315         }else{
316             av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
317             s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
318         }
319     }
320
321     if(   s->int_sample_fmt != AV_SAMPLE_FMT_S16P
322         &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
323         &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
324         &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
325         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));
326         return AVERROR(EINVAL);
327     }
328
329     set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
330     set_audiodata_fmt(&s->out, s->out_sample_fmt);
331
332     if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
333         if (!s->async && s->min_compensation >= FLT_MAX/2)
334             s->async = 1;
335         s->firstpts =
336         s->outpts   = s->firstpts_in_samples * s->out_sample_rate;
337     } else
338         s->firstpts = AV_NOPTS_VALUE;
339
340     if (s->async) {
341         if (s->min_compensation >= FLT_MAX/2)
342             s->min_compensation = 0.001;
343         if (s->async > 1.0001) {
344             s->max_soft_compensation = s->async / (double) s->in_sample_rate;
345         }
346     }
347
348     if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
349         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);
350     }else
351         s->resampler->free(&s->resample);
352     if(    s->int_sample_fmt != AV_SAMPLE_FMT_S16P
353         && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
354         && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
355         && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
356         && s->resample){
357         av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
358         return -1;
359     }
360
361 #define RSC 1 //FIXME finetune
362     if(!s-> in.ch_count)
363         s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
364     if(!s->used_ch_count)
365         s->used_ch_count= s->in.ch_count;
366     if(!s->out.ch_count)
367         s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
368
369     if(!s-> in.ch_count){
370         av_assert0(!s->in_ch_layout);
371         av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
372         return -1;
373     }
374
375     if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
376         char l1[1024], l2[1024];
377         av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
378         av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
379         av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
380                "but there is not enough information to do it\n", l1, l2);
381         return -1;
382     }
383
384 av_assert0(s->used_ch_count);
385 av_assert0(s->out.ch_count);
386     s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
387
388     s->in_buffer= s->in;
389     s->silence  = s->in;
390     s->drop_temp= s->out;
391
392     if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
393         s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
394                                                    s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
395         return 0;
396     }
397
398     s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
399                                              s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
400     s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
401                                              s->int_sample_fmt, s->out.ch_count, NULL, 0);
402
403     if (!s->in_convert || !s->out_convert)
404         return AVERROR(ENOMEM);
405
406     s->postin= s->in;
407     s->preout= s->out;
408     s->midbuf= s->in;
409
410     if(s->channel_map){
411         s->postin.ch_count=
412         s->midbuf.ch_count= s->used_ch_count;
413         if(s->resample)
414             s->in_buffer.ch_count= s->used_ch_count;
415     }
416     if(!s->resample_first){
417         s->midbuf.ch_count= s->out.ch_count;
418         if(s->resample)
419             s->in_buffer.ch_count = s->out.ch_count;
420     }
421
422     set_audiodata_fmt(&s->postin, s->int_sample_fmt);
423     set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
424     set_audiodata_fmt(&s->preout, s->int_sample_fmt);
425
426     if(s->resample){
427         set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
428     }
429
430     if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
431         return ret;
432
433     if(s->rematrix || s->dither.method)
434         return swri_rematrix_init(s);
435
436     return 0;
437 }
438
439 int swri_realloc_audio(AudioData *a, int count){
440     int i, countb;
441     AudioData old;
442
443     if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
444         return AVERROR(EINVAL);
445
446     if(a->count >= count)
447         return 0;
448
449     count*=2;
450
451     countb= FFALIGN(count*a->bps, ALIGN);
452     old= *a;
453
454     av_assert0(a->bps);
455     av_assert0(a->ch_count);
456
457     a->data= av_mallocz(countb*a->ch_count);
458     if(!a->data)
459         return AVERROR(ENOMEM);
460     for(i=0; i<a->ch_count; i++){
461         a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
462         if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
463     }
464     if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
465     av_freep(&old.data);
466     a->count= count;
467
468     return 1;
469 }
470
471 static void copy(AudioData *out, AudioData *in,
472                  int count){
473     av_assert0(out->planar == in->planar);
474     av_assert0(out->bps == in->bps);
475     av_assert0(out->ch_count == in->ch_count);
476     if(out->planar){
477         int ch;
478         for(ch=0; ch<out->ch_count; ch++)
479             memcpy(out->ch[ch], in->ch[ch], count*out->bps);
480     }else
481         memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
482 }
483
484 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
485     int i;
486     if(!in_arg){
487         memset(out->ch, 0, sizeof(out->ch));
488     }else if(out->planar){
489         for(i=0; i<out->ch_count; i++)
490             out->ch[i]= in_arg[i];
491     }else{
492         for(i=0; i<out->ch_count; i++)
493             out->ch[i]= in_arg[0] + i*out->bps;
494     }
495 }
496
497 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
498     int i;
499     if(out->planar){
500         for(i=0; i<out->ch_count; i++)
501             in_arg[i]= out->ch[i];
502     }else{
503         in_arg[0]= out->ch[0];
504     }
505 }
506
507 /**
508  *
509  * out may be equal in.
510  */
511 static void buf_set(AudioData *out, AudioData *in, int count){
512     int ch;
513     if(in->planar){
514         for(ch=0; ch<out->ch_count; ch++)
515             out->ch[ch]= in->ch[ch] + count*out->bps;
516     }else{
517         for(ch=out->ch_count-1; ch>=0; ch--)
518             out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
519     }
520 }
521
522 /**
523  *
524  * @return number of samples output per channel
525  */
526 static int resample(SwrContext *s, AudioData *out_param, int out_count,
527                              const AudioData * in_param, int in_count){
528     AudioData in, out, tmp;
529     int ret_sum=0;
530     int border=0;
531     int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
532
533     av_assert1(s->in_buffer.ch_count == in_param->ch_count);
534     av_assert1(s->in_buffer.planar   == in_param->planar);
535     av_assert1(s->in_buffer.fmt      == in_param->fmt);
536
537     tmp=out=*out_param;
538     in =  *in_param;
539
540     do{
541         int ret, size, consumed;
542         if(!s->resample_in_constraint && s->in_buffer_count){
543             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
544             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
545             out_count -= ret;
546             ret_sum += ret;
547             buf_set(&out, &out, ret);
548             s->in_buffer_count -= consumed;
549             s->in_buffer_index += consumed;
550
551             if(!in_count)
552                 break;
553             if(s->in_buffer_count <= border){
554                 buf_set(&in, &in, -s->in_buffer_count);
555                 in_count += s->in_buffer_count;
556                 s->in_buffer_count=0;
557                 s->in_buffer_index=0;
558                 border = 0;
559             }
560         }
561
562         if((s->flushed || in_count > padless) && !s->in_buffer_count){
563             s->in_buffer_index=0;
564             ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
565             out_count -= ret;
566             ret_sum += ret;
567             buf_set(&out, &out, ret);
568             in_count -= consumed;
569             buf_set(&in, &in, consumed);
570         }
571
572         //TODO is this check sane considering the advanced copy avoidance below
573         size= s->in_buffer_index + s->in_buffer_count + in_count;
574         if(   size > s->in_buffer.count
575            && s->in_buffer_count + in_count <= s->in_buffer_index){
576             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
577             copy(&s->in_buffer, &tmp, s->in_buffer_count);
578             s->in_buffer_index=0;
579         }else
580             if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
581                 return ret;
582
583         if(in_count){
584             int count= in_count;
585             if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
586
587             buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
588             copy(&tmp, &in, /*in_*/count);
589             s->in_buffer_count += count;
590             in_count -= count;
591             border += count;
592             buf_set(&in, &in, count);
593             s->resample_in_constraint= 0;
594             if(s->in_buffer_count != count || in_count)
595                 continue;
596             if (padless) {
597                 padless = 0;
598                 continue;
599             }
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_is_initialized(struct SwrContext *s) {
737     return !!s->in_buffer.ch_count;
738 }
739
740 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
741                                 const uint8_t *in_arg [SWR_CH_MAX], int  in_count){
742     AudioData * in= &s->in;
743     AudioData *out= &s->out;
744
745     while(s->drop_output > 0){
746         int ret;
747         uint8_t *tmp_arg[SWR_CH_MAX];
748 #define MAX_DROP_STEP 16384
749         if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
750             return ret;
751
752         reversefill_audiodata(&s->drop_temp, tmp_arg);
753         s->drop_output *= -1; //FIXME find a less hackish solution
754         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
755         s->drop_output *= -1;
756         in_count = 0;
757         if(ret>0) {
758             s->drop_output -= ret;
759             continue;
760         }
761
762         if(s->drop_output || !out_arg)
763             return 0;
764     }
765
766     if(!in_arg){
767         if(s->resample){
768             if (!s->flushed)
769                 s->resampler->flush(s);
770             s->resample_in_constraint = 0;
771             s->flushed = 1;
772         }else if(!s->in_buffer_count){
773             return 0;
774         }
775     }else
776         fill_audiodata(in ,  (void*)in_arg);
777
778     fill_audiodata(out, out_arg);
779
780     if(s->resample){
781         int ret = swr_convert_internal(s, out, out_count, in, in_count);
782         if(ret>0 && !s->drop_output)
783             s->outpts += ret * (int64_t)s->in_sample_rate;
784         return ret;
785     }else{
786         AudioData tmp= *in;
787         int ret2=0;
788         int ret, size;
789         size = FFMIN(out_count, s->in_buffer_count);
790         if(size){
791             buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
792             ret= swr_convert_internal(s, out, size, &tmp, size);
793             if(ret<0)
794                 return ret;
795             ret2= ret;
796             s->in_buffer_count -= ret;
797             s->in_buffer_index += ret;
798             buf_set(out, out, ret);
799             out_count -= ret;
800             if(!s->in_buffer_count)
801                 s->in_buffer_index = 0;
802         }
803
804         if(in_count){
805             size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
806
807             if(in_count > out_count) { //FIXME move after swr_convert_internal
808                 if(   size > s->in_buffer.count
809                 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
810                     buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
811                     copy(&s->in_buffer, &tmp, s->in_buffer_count);
812                     s->in_buffer_index=0;
813                 }else
814                     if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
815                         return ret;
816             }
817
818             if(out_count){
819                 size = FFMIN(in_count, out_count);
820                 ret= swr_convert_internal(s, out, size, in, size);
821                 if(ret<0)
822                     return ret;
823                 buf_set(in, in, ret);
824                 in_count -= ret;
825                 ret2 += ret;
826             }
827             if(in_count){
828                 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
829                 copy(&tmp, in, in_count);
830                 s->in_buffer_count += in_count;
831             }
832         }
833         if(ret2>0 && !s->drop_output)
834             s->outpts += ret2 * (int64_t)s->in_sample_rate;
835         return ret2;
836     }
837 }
838
839 int swr_drop_output(struct SwrContext *s, int count){
840     s->drop_output += count;
841
842     if(s->drop_output <= 0)
843         return 0;
844
845     av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
846     return swr_convert(s, NULL, s->drop_output, NULL, 0);
847 }
848
849 int swr_inject_silence(struct SwrContext *s, int count){
850     int ret, i;
851     uint8_t *tmp_arg[SWR_CH_MAX];
852
853     if(count <= 0)
854         return 0;
855
856 #define MAX_SILENCE_STEP 16384
857     while (count > MAX_SILENCE_STEP) {
858         if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
859             return ret;
860         count -= MAX_SILENCE_STEP;
861     }
862
863     if((ret=swri_realloc_audio(&s->silence, count))<0)
864         return ret;
865
866     if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
867         memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
868     } else
869         memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
870
871     reversefill_audiodata(&s->silence, tmp_arg);
872     av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
873     ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
874     return ret;
875 }
876
877 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
878     if (s->resampler && s->resample){
879         return s->resampler->get_delay(s, base);
880     }else{
881         return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
882     }
883 }
884
885 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
886     int ret;
887
888     if (!s || compensation_distance < 0)
889         return AVERROR(EINVAL);
890     if (!compensation_distance && sample_delta)
891         return AVERROR(EINVAL);
892     if (!s->resample) {
893         s->flags |= SWR_FLAG_RESAMPLE;
894         ret = swr_init(s);
895         if (ret < 0)
896             return ret;
897     }
898     if (!s->resampler->set_compensation){
899         return AVERROR(EINVAL);
900     }else{
901         return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
902     }
903 }
904
905 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
906     if(pts == INT64_MIN)
907         return s->outpts;
908
909     if (s->firstpts == AV_NOPTS_VALUE)
910         s->outpts = s->firstpts = pts;
911
912     if(s->min_compensation >= FLT_MAX) {
913         return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
914     } else {
915         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;
916         double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
917
918         if(fabs(fdelta) > s->min_compensation) {
919             if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
920                 int ret;
921                 if(delta > 0) ret = swr_inject_silence(s,  delta / s->out_sample_rate);
922                 else          ret = swr_drop_output   (s, -delta / s-> in_sample_rate);
923                 if(ret<0){
924                     av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
925                 }
926             } else if(s->soft_compensation_duration && s->max_soft_compensation) {
927                 int duration = s->out_sample_rate * s->soft_compensation_duration;
928                 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
929                 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
930                 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
931                 swr_set_compensation(s, comp, duration);
932             }
933         }
934
935         return s->outpts;
936     }
937 }