FFmpeg: libswresample/swresample.c Source File

FFmpeg
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"
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_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  av_cold void swr_free(SwrContext **ss){
222  SwrContext *s= *ss;
223  if(s){
224  free_temp(&s->postin);
225  free_temp(&s->midbuf);
226  free_temp(&s->preout);
227  free_temp(&s->in_buffer);
228  free_temp(&s->silence);
229  free_temp(&s->drop_temp);
230  free_temp(&s->dither.noise);
231  free_temp(&s->dither.temp);
232  swri_audio_convert_free(&s-> in_convert);
233  swri_audio_convert_free(&s->out_convert);
234  swri_audio_convert_free(&s->full_convert);
235  if (s->resampler)
236  s->resampler->free(&s->resample);
237  swri_rematrix_free(s);
238  }
239 
240  av_freep(ss);
241 }
242 
243  av_cold int swr_init(struct SwrContext *s){
244  int ret;
245  s->in_buffer_index= 0;
246  s->in_buffer_count= 0;
247  s->resample_in_constraint= 0;
248  free_temp(&s->postin);
249  free_temp(&s->midbuf);
250  free_temp(&s->preout);
251  free_temp(&s->in_buffer);
252  free_temp(&s->silence);
253  free_temp(&s->drop_temp);
254  free_temp(&s->dither.noise);
255  free_temp(&s->dither.temp);
256  memset(s->in.ch, 0, sizeof(s->in.ch));
257  memset(s->out.ch, 0, sizeof(s->out.ch));
258  swri_audio_convert_free(&s-> in_convert);
259  swri_audio_convert_free(&s->out_convert);
260  swri_audio_convert_free(&s->full_convert);
261  swri_rematrix_free(s);
262 
263  s->flushed = 0;
264 
265  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
266  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
267  return AVERROR(EINVAL);
268  }
269  if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
270  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
271  return AVERROR(EINVAL);
272  }
273 
274  if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {
275  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
276  s->in_ch_layout = 0;
277  }
278 
279  if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {
280  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
281  s->out_ch_layout = 0;
282  }
283 
284  switch(s->engine){
285 #if CONFIG_LIBSOXR
286  extern struct Resampler const soxr_resampler;
287  case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
288 #endif
289  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
290  default:
291  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
292  return AVERROR(EINVAL);
293  }
294 
295  if(!s->used_ch_count)
296  s->used_ch_count= s->in.ch_count;
297 
298  if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
299  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
300  s-> in_ch_layout= 0;
301  }
302 
303  if(!s-> in_ch_layout)
304  s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
305  if(!s->out_ch_layout)
306  s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
307 
308  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
309  s->rematrix_custom;
310 
311  if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
312  if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
313  s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
314  }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P
315  && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P
316  && !s->rematrix
317  && s->engine != SWR_ENGINE_SOXR){
318  s->int_sample_fmt= AV_SAMPLE_FMT_S32P;
319  }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
320  s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
321  }else{
322  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
323  s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
324  }
325  }
326 
327  if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
328  &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
329  &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
330  &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
331  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));
332  return AVERROR(EINVAL);
333  }
334 
335  set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
336  set_audiodata_fmt(&s->out, s->out_sample_fmt);
337 
338  if (s->firstpts_in_samples != AV_NOPTS_VALUE) {
339  if (!s->async && s->min_compensation >= FLT_MAX/2)
340  s->async = 1;
341  s->firstpts =
342  s->outpts = s->firstpts_in_samples * s->out_sample_rate;
343  } else
344  s->firstpts = AV_NOPTS_VALUE;
345 
346  if (s->async) {
347  if (s->min_compensation >= FLT_MAX/2)
348  s->min_compensation = 0.001;
349  if (s->async > 1.0001) {
350  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
351  }
352  }
353 
354  if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
355  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);
356  }else
357  s->resampler->free(&s->resample);
358  if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
359  && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
360  && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
361  && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
362  && s->resample){
363  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
364  return -1;
365  }
366 
367 #define RSC 1 //FIXME finetune
368  if(!s-> in.ch_count)
369  s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
370  if(!s->used_ch_count)
371  s->used_ch_count= s->in.ch_count;
372  if(!s->out.ch_count)
373  s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
374 
375  if(!s-> in.ch_count){
376  av_assert0(!s->in_ch_layout);
377  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
378  return -1;
379  }
380 
381  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
382  char l1[1024], l2[1024];
383  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
384  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
385  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
386  "but there is not enough information to do it\n", l1, l2);
387  return -1;
388  }
389 
390 av_assert0(s->used_ch_count);
391 av_assert0(s->out.ch_count);
392  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
393 
394  s->in_buffer= s->in;
395  s->silence = s->in;
396  s->drop_temp= s->out;
397 
398  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
399  s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
400  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
401  return 0;
402  }
403 
404  s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
405  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
406  s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
407  s->int_sample_fmt, s->out.ch_count, NULL, 0);
408 
409  if (!s->in_convert || !s->out_convert)
410  return AVERROR(ENOMEM);
411 
412  s->postin= s->in;
413  s->preout= s->out;
414  s->midbuf= s->in;
415 
416  if(s->channel_map){
417  s->postin.ch_count=
418  s->midbuf.ch_count= s->used_ch_count;
419  if(s->resample)
420  s->in_buffer.ch_count= s->used_ch_count;
421  }
422  if(!s->resample_first){
423  s->midbuf.ch_count= s->out.ch_count;
424  if(s->resample)
425  s->in_buffer.ch_count = s->out.ch_count;
426  }
427 
428  set_audiodata_fmt(&s->postin, s->int_sample_fmt);
429  set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
430  set_audiodata_fmt(&s->preout, s->int_sample_fmt);
431 
432  if(s->resample){
433  set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
434  }
435 
436  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
437  return ret;
438 
439  if(s->rematrix || s->dither.method)
440  return swri_rematrix_init(s);
441 
442  return 0;
443 }
444 
445  int swri_realloc_audio(AudioData *a, int count){
446  int i, countb;
447  AudioData old;
448 
449  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
450  return AVERROR(EINVAL);
451 
452  if(a->count >= count)
453  return 0;
454 
455  count*=2;
456 
457  countb= FFALIGN(count*a->bps, ALIGN);
458  old= *a;
459 
460  av_assert0(a->bps);
461  av_assert0(a->ch_count);
462 
463  a->data= av_mallocz(countb*a->ch_count);
464  if(!a->data)
465  return AVERROR(ENOMEM);
466  for(i=0; i<a->ch_count; i++){
467  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
468  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
469  }
470  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
471  av_freep(&old.data);
472  a->count= count;
473 
474  return 1;
475 }
476 
477  static void copy(AudioData *out, AudioData *in,
478  int count){
479  av_assert0(out->planar == in->planar);
480  av_assert0(out->bps == in->bps);
481  av_assert0(out->ch_count == in->ch_count);
482  if(out->planar){
483  int ch;
484  for(ch=0; ch<out->ch_count; ch++)
485  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
486  }else
487  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
488 }
489 
490  static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
491  int i;
492  if(!in_arg){
493  memset(out->ch, 0, sizeof(out->ch));
494  }else if(out->planar){
495  for(i=0; i<out->ch_count; i++)
496  out->ch[i]= in_arg[i];
497  }else{
498  for(i=0; i<out->ch_count; i++)
499  out->ch[i]= in_arg[0] + i*out->bps;
500  }
501 }
502 
503  static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
504  int i;
505  if(out->planar){
506  for(i=0; i<out->ch_count; i++)
507  in_arg[i]= out->ch[i];
508  }else{
509  in_arg[0]= out->ch[0];
510  }
511 }
512 
513 /**
514  *
515  * out may be equal in.
516  */
517  static void buf_set(AudioData *out, AudioData *in, int count){
518  int ch;
519  if(in->planar){
520  for(ch=0; ch<out->ch_count; ch++)
521  out->ch[ch]= in->ch[ch] + count*out->bps;
522  }else{
523  for(ch=out->ch_count-1; ch>=0; ch--)
524  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
525  }
526 }
527 
528 /**
529  *
530  * @return number of samples output per channel
531  */
532  static int resample(SwrContext *s, AudioData *out_param, int out_count,
533  const AudioData * in_param, int in_count){
534  AudioData in, out, tmp;
535  int ret_sum=0;
536  int border=0;
537 
538  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
539  av_assert1(s->in_buffer.planar == in_param->planar);
540  av_assert1(s->in_buffer.fmt == in_param->fmt);
541 
542  tmp=out=*out_param;
543  in = *in_param;
544 
545  do{
546  int ret, size, consumed;
547  if(!s->resample_in_constraint && s->in_buffer_count){
548  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
549  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
550  out_count -= ret;
551  ret_sum += ret;
552  buf_set(&out, &out, ret);
553  s->in_buffer_count -= consumed;
554  s->in_buffer_index += consumed;
555 
556  if(!in_count)
557  break;
558  if(s->in_buffer_count <= border){
559  buf_set(&in, &in, -s->in_buffer_count);
560  in_count += s->in_buffer_count;
561  s->in_buffer_count=0;
562  s->in_buffer_index=0;
563  border = 0;
564  }
565  }
566 
567  if((s->flushed || in_count) && !s->in_buffer_count){
568  s->in_buffer_index=0;
569  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
570  out_count -= ret;
571  ret_sum += ret;
572  buf_set(&out, &out, ret);
573  in_count -= consumed;
574  buf_set(&in, &in, consumed);
575  }
576 
577  //TODO is this check sane considering the advanced copy avoidance below
578  size= s->in_buffer_index + s->in_buffer_count + in_count;
579  if( size > s->in_buffer.count
580  && s->in_buffer_count + in_count <= s->in_buffer_index){
581  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
582  copy(&s->in_buffer, &tmp, s->in_buffer_count);
583  s->in_buffer_index=0;
584  }else
585  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
586  return ret;
587 
588  if(in_count){
589  int count= in_count;
590  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
591 
592  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
593  copy(&tmp, &in, /*in_*/count);
594  s->in_buffer_count += count;
595  in_count -= count;
596  border += count;
597  buf_set(&in, &in, count);
598  s->resample_in_constraint= 0;
599  if(s->in_buffer_count != count || in_count)
600  continue;
601  }
602  break;
603  }while(1);
604 
605  s->resample_in_constraint= !!out_count;
606 
607  return ret_sum;
608 }
609 
610  static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
611  AudioData *in , int in_count){
612  AudioData *postin, *midbuf, *preout;
613  int ret /*, in_max*/;
614  AudioData preout_tmp, midbuf_tmp;
615 
616  if(s->full_convert){
617  av_assert0(!s->resample);
618  swri_audio_convert(s->full_convert, out, in, in_count);
619  return out_count;
620  }
621 
622 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
623 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
624 
625  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
626  return ret;
627  if(s->resample_first){
628  av_assert0(s->midbuf.ch_count == s->used_ch_count);
629  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
630  return ret;
631  }else{
632  av_assert0(s->midbuf.ch_count == s->out.ch_count);
633  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
634  return ret;
635  }
636  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
637  return ret;
638 
639  postin= &s->postin;
640 
641  midbuf_tmp= s->midbuf;
642  midbuf= &midbuf_tmp;
643  preout_tmp= s->preout;
644  preout= &preout_tmp;
645 
646  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
647  postin= in;
648 
649  if(s->resample_first ? !s->resample : !s->rematrix)
650  midbuf= postin;
651 
652  if(s->resample_first ? !s->rematrix : !s->resample)
653  preout= midbuf;
654 
655  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
656  && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){
657  if(preout==in){
658  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
659  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
660  copy(out, in, out_count);
661  return out_count;
662  }
663  else if(preout==postin) preout= midbuf= postin= out;
664  else if(preout==midbuf) preout= midbuf= out;
665  else preout= out;
666  }
667 
668  if(in != postin){
669  swri_audio_convert(s->in_convert, postin, in, in_count);
670  }
671 
672  if(s->resample_first){
673  if(postin != midbuf)
674  out_count= resample(s, midbuf, out_count, postin, in_count);
675  if(midbuf != preout)
676  swri_rematrix(s, preout, midbuf, out_count, preout==out);
677  }else{
678  if(postin != midbuf)
679  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
680  if(midbuf != preout)
681  out_count= resample(s, preout, out_count, midbuf, in_count);
682  }
683 
684  if(preout != out && out_count){
685  AudioData *conv_src = preout;
686  if(s->dither.method){
687  int ch;
688  int dither_count= FFMAX(out_count, 1<<16);
689 
690  if (preout == in) {
691  conv_src = &s->dither.temp;
692  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
693  return ret;
694  }
695 
696  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
697  return ret;
698  if(ret)
699  for(ch=0; ch<s->dither.noise.ch_count; ch++)
700  swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
701  av_assert0(s->dither.noise.ch_count == preout->ch_count);
702 
703  if(s->dither.noise_pos + out_count > s->dither.noise.count)
704  s->dither.noise_pos = 0;
705 
706  if (s->dither.method < SWR_DITHER_NS){
707  if (s->mix_2_1_simd) {
708  int len1= out_count&~15;
709  int off = len1 * preout->bps;
710 
711  if(len1)
712  for(ch=0; ch<preout->ch_count; ch++)
713  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);
714  if(out_count != len1)
715  for(ch=0; ch<preout->ch_count; ch++)
716  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);
717  } else {
718  for(ch=0; ch<preout->ch_count; ch++)
719  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);
720  }
721  } else {
722  switch(s->int_sample_fmt) {
723  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
724  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
725  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
726  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
727  }
728  }
729  s->dither.noise_pos += out_count;
730  }
731 //FIXME packed doesn't need more than 1 chan here!
732  swri_audio_convert(s->out_convert, out, conv_src, out_count);
733  }
734  return out_count;
735 }
736 
737  int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
738  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
739  AudioData * in= &s->in;
740  AudioData *out= &s->out;
741 
742  while(s->drop_output > 0){
743  int ret;
744  uint8_t *tmp_arg[SWR_CH_MAX];
745 #define MAX_DROP_STEP 16384
746  if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)
747  return ret;
748 
749  reversefill_audiodata(&s->drop_temp, tmp_arg);
750  s->drop_output *= -1; //FIXME find a less hackish solution
751  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
752  s->drop_output *= -1;
753  in_count = 0;
754  if(ret>0) {
755  s->drop_output -= ret;
756  continue;
757  }
758 
759  if(s->drop_output || !out_arg)
760  return 0;
761  }
762 
763  if(!in_arg){
764  if(s->resample){
765  if (!s->flushed)
766  s->resampler->flush(s);
767  s->resample_in_constraint = 0;
768  s->flushed = 1;
769  }else if(!s->in_buffer_count){
770  return 0;
771  }
772  }else
773  fill_audiodata(in , (void*)in_arg);
774 
775  fill_audiodata(out, out_arg);
776 
777  if(s->resample){
778  int ret = swr_convert_internal(s, out, out_count, in, in_count);
779  if(ret>0 && !s->drop_output)
780  s->outpts += ret * (int64_t)s->in_sample_rate;
781  return ret;
782  }else{
783  AudioData tmp= *in;
784  int ret2=0;
785  int ret, size;
786  size = FFMIN(out_count, s->in_buffer_count);
787  if(size){
788  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
789  ret= swr_convert_internal(s, out, size, &tmp, size);
790  if(ret<0)
791  return ret;
792  ret2= ret;
793  s->in_buffer_count -= ret;
794  s->in_buffer_index += ret;
795  buf_set(out, out, ret);
796  out_count -= ret;
797  if(!s->in_buffer_count)
798  s->in_buffer_index = 0;
799  }
800 
801  if(in_count){
802  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
803 
804  if(in_count > out_count) { //FIXME move after swr_convert_internal
805  if( size > s->in_buffer.count
806  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
807  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
808  copy(&s->in_buffer, &tmp, s->in_buffer_count);
809  s->in_buffer_index=0;
810  }else
811  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
812  return ret;
813  }
814 
815  if(out_count){
816  size = FFMIN(in_count, out_count);
817  ret= swr_convert_internal(s, out, size, in, size);
818  if(ret<0)
819  return ret;
820  buf_set(in, in, ret);
821  in_count -= ret;
822  ret2 += ret;
823  }
824  if(in_count){
825  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
826  copy(&tmp, in, in_count);
827  s->in_buffer_count += in_count;
828  }
829  }
830  if(ret2>0 && !s->drop_output)
831  s->outpts += ret2 * (int64_t)s->in_sample_rate;
832  return ret2;
833  }
834 }
835 
836  int swr_drop_output(struct SwrContext *s, int count){
837  s->drop_output += count;
838 
839  if(s->drop_output <= 0)
840  return 0;
841 
842  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
843  return swr_convert(s, NULL, s->drop_output, NULL, 0);
844 }
845 
846  int swr_inject_silence(struct SwrContext *s, int count){
847  int ret, i;
848  uint8_t *tmp_arg[SWR_CH_MAX];
849 
850  if(count <= 0)
851  return 0;
852 
853 #define MAX_SILENCE_STEP 16384
854  while (count > MAX_SILENCE_STEP) {
855  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
856  return ret;
857  count -= MAX_SILENCE_STEP;
858  }
859 
860  if((ret=swri_realloc_audio(&s->silence, count))<0)
861  return ret;
862 
863  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
864  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
865  } else
866  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
867 
868  reversefill_audiodata(&s->silence, tmp_arg);
869  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
870  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
871  return ret;
872 }
873 
874  int64_t swr_get_delay(struct SwrContext *s, int64_t base){
875  if (s->resampler && s->resample){
876  return s->resampler->get_delay(s, base);
877  }else{
878  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
879  }
880 }
881 
882  int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
883  int ret;
884 
885  if (!s || compensation_distance < 0)
886  return AVERROR(EINVAL);
887  if (!compensation_distance && sample_delta)
888  return AVERROR(EINVAL);
889  if (!s->resample) {
890  s->flags |= SWR_FLAG_RESAMPLE;
891  ret = swr_init(s);
892  if (ret < 0)
893  return ret;
894  }
895  if (!s->resampler->set_compensation){
896  return AVERROR(EINVAL);
897  }else{
898  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
899  }
900 }
901 
902  int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
903  if(pts == INT64_MIN)
904  return s->outpts;
905 
906  if (s->firstpts == AV_NOPTS_VALUE)
907  s->outpts = s->firstpts = pts;
908 
909  if(s->min_compensation >= FLT_MAX) {
910  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
911  } else {
912  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;
913  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
914 
915  if(fabs(fdelta) > s->min_compensation) {
916  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
917  int ret;
918  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
919  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
920  if(ret<0){
921  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
922  }
923  } else if(s->soft_compensation_duration && s->max_soft_compensation) {
924  int duration = s->out_sample_rate * s->soft_compensation_duration;
925  double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
926  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
927  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
928  swr_set_compensation(s, comp, duration);
929  }
930  }
931 
932  return s->outpts;
933  }
934 }
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