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/channel_layout.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" , "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},
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"},
76 {"dither_scale" , "set dither scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
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"},
83 {"filter_size" , "set resampling filter size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
84 {"phase_shift" , "set resampling phase shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
85 {"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
86 {"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },
87 {"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},
88 {"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},
89 {"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"},
90 {"precision" , "set resampling precision" , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },
91 {"cheby" , "enable Chebyshev passband" , OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
92 {"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
93 , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
94 {"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
95 , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
96 {"comp_duration" , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
97 , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
98 {"max_soft_comp" , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."
99 , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
101 { "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" },
102 { "none", "select none", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
103 { "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
104 { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
106 { "filter_type" , "select 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" },
107 { "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
108 { "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" },
109 { "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
111 { "kaiser_beta" , "set Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
116 static const char* context_to_name(void* ptr) {
120 static const AVClass av_class = {
121 .class_name = "SWResampler",
122 .item_name = context_to_name,
124 .version = LIBAVUTIL_VERSION_INT,
125 .log_level_offset_offset = OFFSET(log_level_offset),
126 .parent_log_context_offset = OFFSET(log_ctx),
127 .category = AV_CLASS_CATEGORY_SWRESAMPLER,
130 unsigned swresample_version(void)
132 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
133 return LIBSWRESAMPLE_VERSION_INT;
136 const char *swresample_configuration(void)
138 return FFMPEG_CONFIGURATION;
141 const char *swresample_license(void)
143 #define LICENSE_PREFIX "libswresample license: "
144 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
147 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
148 if(!s || s->in_convert) // s needs to be allocated but not initialized
149 return AVERROR(EINVAL);
150 s->channel_map = channel_map;
154 const AVClass *swr_get_class(void)
159 av_cold struct SwrContext *swr_alloc(void){
160 SwrContext *s= av_mallocz(sizeof(SwrContext));
162 s->av_class= &av_class;
163 av_opt_set_defaults(s);
168 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
169 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
170 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
171 int log_offset, void *log_ctx){
172 if(!s) s= swr_alloc();
175 s->log_level_offset= log_offset;
178 av_opt_set_int(s, "ocl", out_ch_layout, 0);
179 av_opt_set_int(s, "osf", out_sample_fmt, 0);
180 av_opt_set_int(s, "osr", out_sample_rate, 0);
181 av_opt_set_int(s, "icl", in_ch_layout, 0);
182 av_opt_set_int(s, "isf", in_sample_fmt, 0);
183 av_opt_set_int(s, "isr", in_sample_rate, 0);
184 av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
185 av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
186 av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
187 av_opt_set_int(s, "uch", 0, 0);
191 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
193 a->bps = av_get_bytes_per_sample(fmt);
194 a->planar= av_sample_fmt_is_planar(fmt);
197 static void free_temp(AudioData *a){
199 memset(a, 0, sizeof(*a));
202 av_cold void swr_free(SwrContext **ss){
205 free_temp(&s->postin);
206 free_temp(&s->midbuf);
207 free_temp(&s->preout);
208 free_temp(&s->in_buffer);
209 free_temp(&s->dither);
210 swri_audio_convert_free(&s-> in_convert);
211 swri_audio_convert_free(&s->out_convert);
212 swri_audio_convert_free(&s->full_convert);
214 s->resampler->free(&s->resample);
215 swri_rematrix_free(s);
221 av_cold int swr_init(struct SwrContext *s){
222 s->in_buffer_index= 0;
223 s->in_buffer_count= 0;
224 s->resample_in_constraint= 0;
225 free_temp(&s->postin);
226 free_temp(&s->midbuf);
227 free_temp(&s->preout);
228 free_temp(&s->in_buffer);
229 free_temp(&s->dither);
230 memset(s->in.ch, 0, sizeof(s->in.ch));
231 memset(s->out.ch, 0, sizeof(s->out.ch));
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 swri_rematrix_free(s);
239 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
240 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
241 return AVERROR(EINVAL);
243 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
244 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
245 return AVERROR(EINVAL);
248 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
249 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
250 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
251 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
252 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
254 av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
255 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
259 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
260 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
261 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
262 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
263 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));
264 return AVERROR(EINVAL);
269 extern struct Resampler const soxr_resampler;
270 case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;
272 case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
274 av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
275 return AVERROR(EINVAL);
278 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
279 set_audiodata_fmt(&s->out, s->out_sample_fmt);
281 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
282 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);
284 s->resampler->free(&s->resample);
285 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
286 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
287 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
288 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
290 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
294 if(!s->used_ch_count)
295 s->used_ch_count= s->in.ch_count;
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");
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)
305 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
307 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
310 #define RSC 1 //FIXME finetune
312 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
313 if(!s->used_ch_count)
314 s->used_ch_count= s->in.ch_count;
316 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
318 if(!s-> in.ch_count){
319 av_assert0(!s->in_ch_layout);
320 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
324 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
325 av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
329 av_assert0(s->used_ch_count);
330 av_assert0(s->out.ch_count);
331 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
335 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
336 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
337 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
341 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
342 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
343 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
344 s->int_sample_fmt, s->out.ch_count, NULL, 0);
353 s->midbuf.ch_count= s->used_ch_count;
355 s->in_buffer.ch_count= s->used_ch_count;
357 if(!s->resample_first){
358 s->midbuf.ch_count= s->out.ch_count;
360 s->in_buffer.ch_count = s->out.ch_count;
363 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
364 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
365 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
368 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
371 s->dither = s->preout;
373 if(s->rematrix || s->dither_method)
374 return swri_rematrix_init(s);
379 int swri_realloc_audio(AudioData *a, int count){
383 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
384 return AVERROR(EINVAL);
386 if(a->count >= count)
391 countb= FFALIGN(count*a->bps, ALIGN);
395 av_assert0(a->ch_count);
397 a->data= av_mallocz(countb*a->ch_count);
399 return AVERROR(ENOMEM);
400 for(i=0; i<a->ch_count; i++){
401 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
402 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
404 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
411 static void copy(AudioData *out, AudioData *in,
413 av_assert0(out->planar == in->planar);
414 av_assert0(out->bps == in->bps);
415 av_assert0(out->ch_count == in->ch_count);
418 for(ch=0; ch<out->ch_count; ch++)
419 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
421 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
424 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
427 memset(out->ch, 0, sizeof(out->ch));
428 }else if(out->planar){
429 for(i=0; i<out->ch_count; i++)
430 out->ch[i]= in_arg[i];
432 for(i=0; i<out->ch_count; i++)
433 out->ch[i]= in_arg[0] + i*out->bps;
437 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
440 for(i=0; i<out->ch_count; i++)
441 in_arg[i]= out->ch[i];
443 in_arg[0]= out->ch[0];
449 * out may be equal in.
451 static void buf_set(AudioData *out, AudioData *in, int count){
454 for(ch=0; ch<out->ch_count; ch++)
455 out->ch[ch]= in->ch[ch] + count*out->bps;
457 for(ch=out->ch_count-1; ch>=0; ch--)
458 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
464 * @return number of samples output per channel
466 static int resample(SwrContext *s, AudioData *out_param, int out_count,
467 const AudioData * in_param, int in_count){
468 AudioData in, out, tmp;
472 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
473 av_assert1(s->in_buffer.planar == in_param->planar);
474 av_assert1(s->in_buffer.fmt == in_param->fmt);
480 int ret, size, consumed;
481 if(!s->resample_in_constraint && s->in_buffer_count){
482 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
483 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
486 buf_set(&out, &out, ret);
487 s->in_buffer_count -= consumed;
488 s->in_buffer_index += consumed;
492 if(s->in_buffer_count <= border){
493 buf_set(&in, &in, -s->in_buffer_count);
494 in_count += s->in_buffer_count;
495 s->in_buffer_count=0;
496 s->in_buffer_index=0;
501 if((s->flushed || in_count) && !s->in_buffer_count){
502 s->in_buffer_index=0;
503 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
506 buf_set(&out, &out, ret);
507 in_count -= consumed;
508 buf_set(&in, &in, consumed);
511 //TODO is this check sane considering the advanced copy avoidance below
512 size= s->in_buffer_index + s->in_buffer_count + in_count;
513 if( size > s->in_buffer.count
514 && s->in_buffer_count + in_count <= s->in_buffer_index){
515 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
516 copy(&s->in_buffer, &tmp, s->in_buffer_count);
517 s->in_buffer_index=0;
519 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
524 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
526 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
527 copy(&tmp, &in, /*in_*/count);
528 s->in_buffer_count += count;
531 buf_set(&in, &in, count);
532 s->resample_in_constraint= 0;
533 if(s->in_buffer_count != count || in_count)
539 s->resample_in_constraint= !!out_count;
544 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
545 AudioData *in , int in_count){
546 AudioData *postin, *midbuf, *preout;
548 AudioData preout_tmp, midbuf_tmp;
551 av_assert0(!s->resample);
552 swri_audio_convert(s->full_convert, out, in, in_count);
556 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
557 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
559 if((ret=swri_realloc_audio(&s->postin, in_count))<0)
561 if(s->resample_first){
562 av_assert0(s->midbuf.ch_count == s->used_ch_count);
563 if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
566 av_assert0(s->midbuf.ch_count == s->out.ch_count);
567 if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
570 if((ret=swri_realloc_audio(&s->preout, out_count))<0)
575 midbuf_tmp= s->midbuf;
577 preout_tmp= s->preout;
580 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
583 if(s->resample_first ? !s->resample : !s->rematrix)
586 if(s->resample_first ? !s->rematrix : !s->resample)
589 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
591 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
592 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
593 copy(out, in, out_count);
596 else if(preout==postin) preout= midbuf= postin= out;
597 else if(preout==midbuf) preout= midbuf= out;
602 swri_audio_convert(s->in_convert, postin, in, in_count);
605 if(s->resample_first){
607 out_count= resample(s, midbuf, out_count, postin, in_count);
609 swri_rematrix(s, preout, midbuf, out_count, preout==out);
612 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
614 out_count= resample(s, preout, out_count, midbuf, in_count);
617 if(preout != out && out_count){
618 if(s->dither_method){
620 int dither_count= FFMAX(out_count, 1<<16);
621 av_assert0(preout != in);
623 if((ret=swri_realloc_audio(&s->dither, dither_count))<0)
626 for(ch=0; ch<s->dither.ch_count; ch++)
627 swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
628 av_assert0(s->dither.ch_count == preout->ch_count);
630 if(s->dither_pos + out_count > s->dither.count)
633 for(ch=0; ch<preout->ch_count; ch++)
634 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);
636 s->dither_pos += out_count;
638 //FIXME packed doesnt need more than 1 chan here!
639 swri_audio_convert(s->out_convert, out, preout, out_count);
644 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
645 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
646 AudioData * in= &s->in;
647 AudioData *out= &s->out;
649 if(s->drop_output > 0){
651 AudioData tmp = s->out;
652 uint8_t *tmp_arg[SWR_CH_MAX];
655 if((ret=swri_realloc_audio(&tmp, s->drop_output))<0)
658 reversefill_audiodata(&tmp, tmp_arg);
659 s->drop_output *= -1; //FIXME find a less hackish solution
660 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
661 s->drop_output *= -1;
663 s->drop_output -= ret;
666 if(s->drop_output || !out_arg)
674 s->resampler->flush(s);
675 s->resample_in_constraint = 0;
677 }else if(!s->in_buffer_count){
681 fill_audiodata(in , (void*)in_arg);
683 fill_audiodata(out, out_arg);
686 int ret = swr_convert_internal(s, out, out_count, in, in_count);
687 if(ret>0 && !s->drop_output)
688 s->outpts += ret * (int64_t)s->in_sample_rate;
694 size = FFMIN(out_count, s->in_buffer_count);
696 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
697 ret= swr_convert_internal(s, out, size, &tmp, size);
701 s->in_buffer_count -= ret;
702 s->in_buffer_index += ret;
703 buf_set(out, out, ret);
705 if(!s->in_buffer_count)
706 s->in_buffer_index = 0;
710 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
712 if(in_count > out_count) { //FIXME move after swr_convert_internal
713 if( size > s->in_buffer.count
714 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
715 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
716 copy(&s->in_buffer, &tmp, s->in_buffer_count);
717 s->in_buffer_index=0;
719 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
724 size = FFMIN(in_count, out_count);
725 ret= swr_convert_internal(s, out, size, in, size);
728 buf_set(in, in, ret);
733 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
734 copy(&tmp, in, in_count);
735 s->in_buffer_count += in_count;
738 if(ret2>0 && !s->drop_output)
739 s->outpts += ret2 * (int64_t)s->in_sample_rate;
744 int swr_drop_output(struct SwrContext *s, int count){
745 s->drop_output += count;
747 if(s->drop_output <= 0)
750 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
751 return swr_convert(s, NULL, s->drop_output, NULL, 0);
754 int swr_inject_silence(struct SwrContext *s, int count){
756 AudioData silence = s->in;
757 uint8_t *tmp_arg[SWR_CH_MAX];
764 if((ret=swri_realloc_audio(&silence, count))<0)
767 if(silence.planar) for(i=0; i<silence.ch_count; i++) {
768 memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
770 memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
772 reversefill_audiodata(&silence, tmp_arg);
773 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
774 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
775 av_freep(&silence.data);
779 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
780 if (s->resampler && s->resample){
781 return s->resampler->get_delay(s, base);
783 return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
787 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
790 if (!s || compensation_distance < 0)
791 return AVERROR(EINVAL);
792 if (!compensation_distance && sample_delta)
793 return AVERROR(EINVAL);
795 s->flags |= SWR_FLAG_RESAMPLE;
800 if (!s->resampler->set_compensation){
801 return AVERROR(EINVAL);
803 return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
807 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
810 if(s->min_compensation >= FLT_MAX) {
811 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
813 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
814 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
816 if(fabs(fdelta) > s->min_compensation) {
817 if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
819 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
820 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
822 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
824 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
825 int duration = s->out_sample_rate * s->soft_compensation_duration;
826 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
827 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
828 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
829 swr_set_compensation(s, comp, duration);