2 * Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at)
4 * This file is part of libswresample
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.
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.
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
21 #include "libavutil/opt.h"
22 #include "swresample_internal.h"
23 #include "audioconvert.h"
24 #include "libavutil/avassert.h"
25 #include "libavutil/audioconvert.h"
30 #define C15DB 1.189207115
32 #define C_15DB 0.840896415
33 #define C_30DB M_SQRT1_2
34 #define C_45DB 0.594603558
39 //TODO split options array out?
40 #define OFFSET(x) offsetof(SwrContext,x)
41 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
43 static const AVOption options[]={
44 {"ich" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
45 {"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
46 {"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
47 {"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.dbl=2 }, 0 , SWR_CH_MAX, PARAM},
48 {"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_CH_MAX, PARAM},
49 {"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_CH_MAX, PARAM},
50 {"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
51 {"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
52 {"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
53 {"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , INT_MAX , PARAM},
54 {"isf" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
55 {"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
56 {"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
57 {"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
58 {"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
59 {"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.dbl=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
60 {"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
61 {"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
62 {"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
63 {"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
64 {"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
65 {"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
66 {"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
67 {"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
68 {"lfe_mix_level" , "LFE Mix Level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
69 {"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
70 {"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
71 {"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.dbl=0 }, 0 , UINT_MAX , PARAM, "flags"},
72 {"swr_flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.dbl=0 }, 0 , UINT_MAX , PARAM, "flags"},
73 {"res" , "Force Resampling" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
74 {"dither_scale" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
75 {"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.dbl=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
76 {"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
77 {"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
78 {"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.dbl=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
79 {"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.dbl=16 }, 0 , INT_MAX , PARAM },
80 {"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.dbl=10 }, 0 , 30 , PARAM },
81 {"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.dbl=0 }, 0 , 1 , PARAM },
82 {"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
83 {"min_comp" , "Minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
84 , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
85 {"min_hard_comp" , "Minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
86 , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
87 {"comp_duration" , "Duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
88 , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
89 {"max_soft_comp" , "Maximum factor by which data is stretched/squeezed to make it match the timestamps."
90 , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
91 { "filter_type" , "Filter Type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
92 { "cubic" , "Cubic" , 0 , AV_OPT_TYPE_CONST, { SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
93 { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
94 { "kaiser" , "Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
95 { "kaiser_beta" , "Kaiser Window Beta" ,OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.dbl=9 }, 2 , 16 , PARAM },
100 static const char* context_to_name(void* ptr) {
104 static const AVClass av_class = {
105 .class_name = "SWResampler",
106 .item_name = context_to_name,
108 .version = LIBAVUTIL_VERSION_INT,
109 .log_level_offset_offset = OFFSET(log_level_offset),
110 .parent_log_context_offset = OFFSET(log_ctx),
111 .category = AV_CLASS_CATEGORY_SWRESAMPLER,
114 unsigned swresample_version(void)
116 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
117 return LIBSWRESAMPLE_VERSION_INT;
120 const char *swresample_configuration(void)
122 return FFMPEG_CONFIGURATION;
125 const char *swresample_license(void)
127 #define LICENSE_PREFIX "libswresample license: "
128 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
131 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
132 if(!s || s->in_convert) // s needs to be allocated but not initialized
133 return AVERROR(EINVAL);
134 s->channel_map = channel_map;
138 const AVClass *swr_get_class(void)
143 struct SwrContext *swr_alloc(void){
144 SwrContext *s= av_mallocz(sizeof(SwrContext));
146 s->av_class= &av_class;
147 av_opt_set_defaults(s);
152 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
153 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
154 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
155 int log_offset, void *log_ctx){
156 if(!s) s= swr_alloc();
159 s->log_level_offset= log_offset;
162 av_opt_set_int(s, "ocl", out_ch_layout, 0);
163 av_opt_set_int(s, "osf", out_sample_fmt, 0);
164 av_opt_set_int(s, "osr", out_sample_rate, 0);
165 av_opt_set_int(s, "icl", in_ch_layout, 0);
166 av_opt_set_int(s, "isf", in_sample_fmt, 0);
167 av_opt_set_int(s, "isr", in_sample_rate, 0);
168 av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
169 av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
170 av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
171 av_opt_set_int(s, "uch", 0, 0);
175 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
177 a->bps = av_get_bytes_per_sample(fmt);
178 a->planar= av_sample_fmt_is_planar(fmt);
181 static void free_temp(AudioData *a){
183 memset(a, 0, sizeof(*a));
186 void swr_free(SwrContext **ss){
189 free_temp(&s->postin);
190 free_temp(&s->midbuf);
191 free_temp(&s->preout);
192 free_temp(&s->in_buffer);
193 free_temp(&s->dither);
194 swri_audio_convert_free(&s-> in_convert);
195 swri_audio_convert_free(&s->out_convert);
196 swri_audio_convert_free(&s->full_convert);
197 swri_resample_free(&s->resample);
198 swri_rematrix_free(s);
204 int swr_init(struct SwrContext *s){
205 s->in_buffer_index= 0;
206 s->in_buffer_count= 0;
207 s->resample_in_constraint= 0;
208 free_temp(&s->postin);
209 free_temp(&s->midbuf);
210 free_temp(&s->preout);
211 free_temp(&s->in_buffer);
212 free_temp(&s->dither);
213 swri_audio_convert_free(&s-> in_convert);
214 swri_audio_convert_free(&s->out_convert);
215 swri_audio_convert_free(&s->full_convert);
216 swri_rematrix_free(s);
220 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
221 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
222 return AVERROR(EINVAL);
224 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
225 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
226 return AVERROR(EINVAL);
229 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
230 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
231 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
232 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
233 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
235 av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
236 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
240 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
241 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
242 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
243 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
244 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));
245 return AVERROR(EINVAL);
248 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
249 set_audiodata_fmt(&s->out, s->out_sample_fmt);
251 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
252 s->resample = swri_resample_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);
254 swri_resample_free(&s->resample);
255 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
256 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
257 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
258 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
260 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
264 if(!s->used_ch_count)
265 s->used_ch_count= s->in.ch_count;
267 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
268 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
272 if(!s-> in_ch_layout)
273 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
274 if(!s->out_ch_layout)
275 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
277 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
280 #define RSC 1 //FIXME finetune
282 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
283 if(!s->used_ch_count)
284 s->used_ch_count= s->in.ch_count;
286 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
288 if(!s-> in.ch_count){
289 av_assert0(!s->in_ch_layout);
290 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
294 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
295 av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
299 av_assert0(s->used_ch_count);
300 av_assert0(s->out.ch_count);
301 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
305 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
306 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
307 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
311 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
312 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
313 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
314 s->int_sample_fmt, s->out.ch_count, NULL, 0);
323 s->midbuf.ch_count= s->used_ch_count;
325 s->in_buffer.ch_count= s->used_ch_count;
327 if(!s->resample_first){
328 s->midbuf.ch_count= s->out.ch_count;
330 s->in_buffer.ch_count = s->out.ch_count;
333 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
334 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
335 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
338 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
341 s->dither = s->preout;
343 if(s->rematrix || s->dither_method)
344 return swri_rematrix_init(s);
349 static int realloc_audio(AudioData *a, int count){
353 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
354 return AVERROR(EINVAL);
356 if(a->count >= count)
361 countb= FFALIGN(count*a->bps, ALIGN);
365 av_assert0(a->ch_count);
367 a->data= av_mallocz(countb*a->ch_count);
369 return AVERROR(ENOMEM);
370 for(i=0; i<a->ch_count; i++){
371 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
372 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
374 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
381 static void copy(AudioData *out, AudioData *in,
383 av_assert0(out->planar == in->planar);
384 av_assert0(out->bps == in->bps);
385 av_assert0(out->ch_count == in->ch_count);
388 for(ch=0; ch<out->ch_count; ch++)
389 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
391 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
394 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
397 memset(out->ch, 0, sizeof(out->ch));
398 }else if(out->planar){
399 for(i=0; i<out->ch_count; i++)
400 out->ch[i]= in_arg[i];
402 for(i=0; i<out->ch_count; i++)
403 out->ch[i]= in_arg[0] + i*out->bps;
407 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
410 for(i=0; i<out->ch_count; i++)
411 in_arg[i]= out->ch[i];
413 in_arg[0]= out->ch[0];
419 * out may be equal in.
421 static void buf_set(AudioData *out, AudioData *in, int count){
424 for(ch=0; ch<out->ch_count; ch++)
425 out->ch[ch]= in->ch[ch] + count*out->bps;
427 for(ch=out->ch_count-1; ch>=0; ch--)
428 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
434 * @return number of samples output per channel
436 static int resample(SwrContext *s, AudioData *out_param, int out_count,
437 const AudioData * in_param, int in_count){
438 AudioData in, out, tmp;
442 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
443 av_assert1(s->in_buffer.planar == in_param->planar);
444 av_assert1(s->in_buffer.fmt == in_param->fmt);
450 int ret, size, consumed;
451 if(!s->resample_in_constraint && s->in_buffer_count){
452 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
453 ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
456 buf_set(&out, &out, ret);
457 s->in_buffer_count -= consumed;
458 s->in_buffer_index += consumed;
462 if(s->in_buffer_count <= border){
463 buf_set(&in, &in, -s->in_buffer_count);
464 in_count += s->in_buffer_count;
465 s->in_buffer_count=0;
466 s->in_buffer_index=0;
471 if(in_count && !s->in_buffer_count){
472 s->in_buffer_index=0;
473 ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
476 buf_set(&out, &out, ret);
477 in_count -= consumed;
478 buf_set(&in, &in, consumed);
481 //TODO is this check sane considering the advanced copy avoidance below
482 size= s->in_buffer_index + s->in_buffer_count + in_count;
483 if( size > s->in_buffer.count
484 && s->in_buffer_count + in_count <= s->in_buffer_index){
485 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
486 copy(&s->in_buffer, &tmp, s->in_buffer_count);
487 s->in_buffer_index=0;
489 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
494 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
496 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
497 copy(&tmp, &in, /*in_*/count);
498 s->in_buffer_count += count;
501 buf_set(&in, &in, count);
502 s->resample_in_constraint= 0;
503 if(s->in_buffer_count != count || in_count)
509 s->resample_in_constraint= !!out_count;
514 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
515 AudioData *in , int in_count){
516 AudioData *postin, *midbuf, *preout;
518 AudioData preout_tmp, midbuf_tmp;
521 av_assert0(!s->resample);
522 swri_audio_convert(s->full_convert, out, in, in_count);
526 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
527 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
529 if((ret=realloc_audio(&s->postin, in_count))<0)
531 if(s->resample_first){
532 av_assert0(s->midbuf.ch_count == s->used_ch_count);
533 if((ret=realloc_audio(&s->midbuf, out_count))<0)
536 av_assert0(s->midbuf.ch_count == s->out.ch_count);
537 if((ret=realloc_audio(&s->midbuf, in_count))<0)
540 if((ret=realloc_audio(&s->preout, out_count))<0)
545 midbuf_tmp= s->midbuf;
547 preout_tmp= s->preout;
550 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
553 if(s->resample_first ? !s->resample : !s->rematrix)
556 if(s->resample_first ? !s->rematrix : !s->resample)
559 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
561 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
562 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
563 copy(out, in, out_count);
566 else if(preout==postin) preout= midbuf= postin= out;
567 else if(preout==midbuf) preout= midbuf= out;
572 swri_audio_convert(s->in_convert, postin, in, in_count);
575 if(s->resample_first){
577 out_count= resample(s, midbuf, out_count, postin, in_count);
579 swri_rematrix(s, preout, midbuf, out_count, preout==out);
582 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
584 out_count= resample(s, preout, out_count, midbuf, in_count);
587 if(preout != out && out_count){
588 if(s->dither_method){
590 int dither_count= FFMAX(out_count, 1<<16);
591 av_assert0(preout != in);
593 if((ret=realloc_audio(&s->dither, dither_count))<0)
596 for(ch=0; ch<s->dither.ch_count; ch++)
597 swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
598 av_assert0(s->dither.ch_count == preout->ch_count);
600 if(s->dither_pos + out_count > s->dither.count)
603 for(ch=0; ch<preout->ch_count; ch++)
604 s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
606 s->dither_pos += out_count;
608 //FIXME packed doesnt need more than 1 chan here!
609 swri_audio_convert(s->out_convert, out, preout, out_count);
614 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
615 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
616 AudioData * in= &s->in;
617 AudioData *out= &s->out;
619 if(s->drop_output > 0){
621 AudioData tmp = s->out;
622 uint8_t *tmp_arg[SWR_CH_MAX];
625 if((ret=realloc_audio(&tmp, s->drop_output))<0)
628 reversefill_audiodata(&tmp, tmp_arg);
629 s->drop_output *= -1; //FIXME find a less hackish solution
630 ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
631 s->drop_output *= -1;
633 s->drop_output -= ret;
636 if(s->drop_output || !out_arg)
642 if(s->in_buffer_count){
643 if (s->resample && !s->flushed) {
644 AudioData *a= &s->in_buffer;
646 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
648 av_assert0(a->planar);
649 for(i=0; i<a->ch_count; i++){
650 for(j=0; j<s->in_buffer_count; j++){
651 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
652 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
655 s->in_buffer_count += (s->in_buffer_count+1)/2;
656 s->resample_in_constraint = 0;
663 fill_audiodata(in , (void*)in_arg);
665 fill_audiodata(out, out_arg);
668 int ret = swr_convert_internal(s, out, out_count, in, in_count);
669 if(ret>0 && !s->drop_output)
670 s->outpts += ret * (int64_t)s->in_sample_rate;
676 size = FFMIN(out_count, s->in_buffer_count);
678 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
679 ret= swr_convert_internal(s, out, size, &tmp, size);
683 s->in_buffer_count -= ret;
684 s->in_buffer_index += ret;
685 buf_set(out, out, ret);
687 if(!s->in_buffer_count)
688 s->in_buffer_index = 0;
692 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
694 if(in_count > out_count) { //FIXME move after swr_convert_internal
695 if( size > s->in_buffer.count
696 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
697 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
698 copy(&s->in_buffer, &tmp, s->in_buffer_count);
699 s->in_buffer_index=0;
701 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
706 size = FFMIN(in_count, out_count);
707 ret= swr_convert_internal(s, out, size, in, size);
710 buf_set(in, in, ret);
715 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
716 copy(&tmp, in, in_count);
717 s->in_buffer_count += in_count;
720 if(ret2>0 && !s->drop_output)
721 s->outpts += ret2 * (int64_t)s->in_sample_rate;
726 int swr_drop_output(struct SwrContext *s, int count){
727 s->drop_output += count;
729 if(s->drop_output <= 0)
732 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
733 return swr_convert(s, NULL, s->drop_output, NULL, 0);
736 int swr_inject_silence(struct SwrContext *s, int count){
738 AudioData silence = s->in;
739 uint8_t *tmp_arg[SWR_CH_MAX];
746 if((ret=realloc_audio(&silence, count))<0)
749 if(silence.planar) for(i=0; i<silence.ch_count; i++) {
750 memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
752 memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
754 reversefill_audiodata(&silence, tmp_arg);
755 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
756 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
757 av_freep(&silence.data);
761 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
764 if(s->min_compensation >= FLT_MAX) {
765 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
767 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
768 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
770 if(fabs(fdelta) > s->min_compensation) {
771 if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
773 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
774 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
776 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
778 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
779 int duration = s->out_sample_rate * s->soft_compensation_duration;
780 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
781 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
782 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
783 swr_set_compensation(s, comp, duration);