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swresample.c
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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"
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" , "" , OFFSET(dither.output_sample_bits) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 64 , 0 },
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 {
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);
169  return 0;
170 }
171 
172 const AVClass *swr_get_class(void)
173 {
174  return &av_class;
175 }
176 
178  SwrContext *s= av_mallocz(sizeof(SwrContext));
179  if(s){
180  s->av_class= &av_class;
182  }
183  return s;
184 }
185 
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);
205  av_opt_set_int(s, "uch", 0, 0);
206  return s;
207 }
208 
210  a->fmt = fmt;
211  a->bps = av_get_bytes_per_sample(fmt);
213 }
214 
215 static void free_temp(AudioData *a){
216  av_free(a->data);
217  memset(a, 0, sizeof(*a));
218 }
219 
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);
234  if (s->resampler)
235  s->resampler->free(&s->resample);
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;
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));
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  }
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 
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 
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)
306 
307  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
308  s->rematrix_custom;
309 
313  }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
315  && !s->rematrix
316  && s->engine != SWR_ENGINE_SOXR){
320  }else{
321  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
323  }
324  }
325 
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);
336 
338  if (!s->async && s->min_compensation >= FLT_MAX/2)
339  s->async = 1;
340  s->firstpts =
342  } else
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 
355  }else
356  s->resampler->free(&s->resample);
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)
373 
374  if(!s-> in.ch_count){
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 
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){
399  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
400  return 0;
401  }
402 
404  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
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=
418  if(s->resample)
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 
430 
431  if(s->resample){
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 
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;
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);
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){
628  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
629  return ret;
630  }else{
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  if(preout==in){
656  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
657  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
658  copy(out, in, out_count);
659  return out_count;
660  }
661  else if(preout==postin) preout= midbuf= postin= out;
662  else if(preout==midbuf) preout= midbuf= out;
663  else preout= out;
664  }
665 
666  if(in != postin){
667  swri_audio_convert(s->in_convert, postin, in, in_count);
668  }
669 
670  if(s->resample_first){
671  if(postin != midbuf)
672  out_count= resample(s, midbuf, out_count, postin, in_count);
673  if(midbuf != preout)
674  swri_rematrix(s, preout, midbuf, out_count, preout==out);
675  }else{
676  if(postin != midbuf)
677  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
678  if(midbuf != preout)
679  out_count= resample(s, preout, out_count, midbuf, in_count);
680  }
681 
682  if(preout != out && out_count){
683  AudioData *conv_src = preout;
684  if(s->dither.method){
685  int ch;
686  int dither_count= FFMAX(out_count, 1<<16);
687 
688  if (preout == in) {
689  conv_src = &s->dither.temp;
690  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
691  return ret;
692  }
693 
694  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
695  return ret;
696  if(ret)
697  for(ch=0; ch<s->dither.noise.ch_count; ch++)
698  swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
699  av_assert0(s->dither.noise.ch_count == preout->ch_count);
700 
701  if(s->dither.noise_pos + out_count > s->dither.noise.count)
702  s->dither.noise_pos = 0;
703 
704  if (s->dither.method < SWR_DITHER_NS){
705  if (s->mix_2_1_simd) {
706  int len1= out_count&~15;
707  int off = len1 * preout->bps;
708 
709  if(len1)
710  for(ch=0; ch<preout->ch_count; ch++)
711  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);
712  if(out_count != len1)
713  for(ch=0; ch<preout->ch_count; ch++)
714  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);
715  } else {
716  for(ch=0; ch<preout->ch_count; ch++)
717  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);
718  }
719  } else {
720  switch(s->int_sample_fmt) {
721  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
722  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
723  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
724  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
725  }
726  }
727  s->dither.noise_pos += out_count;
728  }
729 //FIXME packed doesnt need more than 1 chan here!
730  swri_audio_convert(s->out_convert, out, conv_src, out_count);
731  }
732  return out_count;
733 }
734 
735 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
736  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
737  AudioData * in= &s->in;
738  AudioData *out= &s->out;
739 
740  while(s->drop_output > 0){
741  int ret;
742  uint8_t *tmp_arg[SWR_CH_MAX];
743 #define MAX_DROP_STEP 16384
745  return ret;
746 
747  reversefill_audiodata(&s->drop_temp, tmp_arg);
748  s->drop_output *= -1; //FIXME find a less hackish solution
749  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
750  s->drop_output *= -1;
751  in_count = 0;
752  if(ret>0) {
753  s->drop_output -= ret;
754  continue;
755  }
756 
757  if(s->drop_output || !out_arg)
758  return 0;
759  }
760 
761  if(!in_arg){
762  if(s->resample){
763  if (!s->flushed)
764  s->resampler->flush(s);
765  s->resample_in_constraint = 0;
766  s->flushed = 1;
767  }else if(!s->in_buffer_count){
768  return 0;
769  }
770  }else
771  fill_audiodata(in , (void*)in_arg);
772 
773  fill_audiodata(out, out_arg);
774 
775  if(s->resample){
776  int ret = swr_convert_internal(s, out, out_count, in, in_count);
777  if(ret>0 && !s->drop_output)
778  s->outpts += ret * (int64_t)s->in_sample_rate;
779  return ret;
780  }else{
781  AudioData tmp= *in;
782  int ret2=0;
783  int ret, size;
784  size = FFMIN(out_count, s->in_buffer_count);
785  if(size){
786  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
787  ret= swr_convert_internal(s, out, size, &tmp, size);
788  if(ret<0)
789  return ret;
790  ret2= ret;
791  s->in_buffer_count -= ret;
792  s->in_buffer_index += ret;
793  buf_set(out, out, ret);
794  out_count -= ret;
795  if(!s->in_buffer_count)
796  s->in_buffer_index = 0;
797  }
798 
799  if(in_count){
800  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
801 
802  if(in_count > out_count) { //FIXME move after swr_convert_internal
803  if( size > s->in_buffer.count
804  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
805  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
806  copy(&s->in_buffer, &tmp, s->in_buffer_count);
807  s->in_buffer_index=0;
808  }else
809  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
810  return ret;
811  }
812 
813  if(out_count){
814  size = FFMIN(in_count, out_count);
815  ret= swr_convert_internal(s, out, size, in, size);
816  if(ret<0)
817  return ret;
818  buf_set(in, in, ret);
819  in_count -= ret;
820  ret2 += ret;
821  }
822  if(in_count){
823  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
824  copy(&tmp, in, in_count);
825  s->in_buffer_count += in_count;
826  }
827  }
828  if(ret2>0 && !s->drop_output)
829  s->outpts += ret2 * (int64_t)s->in_sample_rate;
830  return ret2;
831  }
832 }
833 
834 int swr_drop_output(struct SwrContext *s, int count){
835  s->drop_output += count;
836 
837  if(s->drop_output <= 0)
838  return 0;
839 
840  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
841  return swr_convert(s, NULL, s->drop_output, NULL, 0);
842 }
843 
844 int swr_inject_silence(struct SwrContext *s, int count){
845  int ret, i;
846  uint8_t *tmp_arg[SWR_CH_MAX];
847 
848  if(count <= 0)
849  return 0;
850 
851 #define MAX_SILENCE_STEP 16384
852  while (count > MAX_SILENCE_STEP) {
853  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
854  return ret;
855  count -= MAX_SILENCE_STEP;
856  }
857 
858  if((ret=swri_realloc_audio(&s->silence, count))<0)
859  return ret;
860 
861  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
862  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
863  } else
864  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
865 
866  reversefill_audiodata(&s->silence, tmp_arg);
867  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
868  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
869  return ret;
870 }
871 
872 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
873  if (s->resampler && s->resample){
874  return s->resampler->get_delay(s, base);
875  }else{
876  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
877  }
878 }
879 
880 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
881  int ret;
882 
883  if (!s || compensation_distance < 0)
884  return AVERROR(EINVAL);
885  if (!compensation_distance && sample_delta)
886  return AVERROR(EINVAL);
887  if (!s->resample) {
888  s->flags |= SWR_FLAG_RESAMPLE;
889  ret = swr_init(s);
890  if (ret < 0)
891  return ret;
892  }
893  if (!s->resampler->set_compensation){
894  return AVERROR(EINVAL);
895  }else{
896  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
897  }
898 }
899 
900 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
901  if(pts == INT64_MIN)
902  return s->outpts;
903 
904  if (s->firstpts == AV_NOPTS_VALUE)
905  s->outpts = s->firstpts = pts;
906 
907  if(s->min_compensation >= FLT_MAX) {
908  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
909  } else {
910  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;
911  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
912 
913  if(fabs(fdelta) > s->min_compensation) {
914  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
915  int ret;
916  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
917  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
918  if(ret<0){
919  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
920  }
923  double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
924  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
925  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
926  swr_set_compensation(s, comp, duration);
927  }
928  }
929 
930  return s->outpts;
931  }
932 }