2 * Copyright (C) 2011 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"
28 #define C15DB 1.189207115
30 #define C_15DB 0.840896415
31 #define C_30DB M_SQRT1_2
32 #define C_45DB 0.594603558
36 //TODO split options array out?
37 #define OFFSET(x) offsetof(SwrContext,x)
38 static const AVOption options[]={
39 {"ich", "input channel count", OFFSET( in.ch_count ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
40 {"och", "output channel count", OFFSET(out.ch_count ), FF_OPT_TYPE_INT, {.dbl=2}, 1, SWR_CH_MAX, 0},
41 {"isr", "input sample rate" , OFFSET( in_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
42 {"osr", "output sample rate" , OFFSET(out_sample_rate), FF_OPT_TYPE_INT, {.dbl=48000}, 1, INT_MAX, 0},
43 //{"ip" , "input planar" , OFFSET( in.planar ), FF_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
44 //{"op" , "output planar" , OFFSET(out.planar ), FF_OPT_TYPE_INT, {.dbl=0}, 0, 1, 0},
45 {"isf", "input sample format", OFFSET( in_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
46 {"osf", "output sample format", OFFSET(out_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_S16}, 0, AV_SAMPLE_FMT_NB-1+256, 0},
47 {"tsf", "internal sample format", OFFSET(int_sample_fmt ), FF_OPT_TYPE_INT, {.dbl=AV_SAMPLE_FMT_NONE}, -1, AV_SAMPLE_FMT_FLT, 0},
48 {"icl", "input channel layout" , OFFSET( in_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
49 {"ocl", "output channel layout", OFFSET(out_ch_layout), FF_OPT_TYPE_INT64, {.dbl=0}, 0, INT64_MAX, 0, "channel_layout"},
50 {"clev", "center mix level" , OFFSET(clev) , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
51 {"slev", "sourround mix level" , OFFSET(slev) , FF_OPT_TYPE_FLOAT, {.dbl=C_30DB}, 0, 4, 0},
52 {"flags", NULL , OFFSET(flags) , FF_OPT_TYPE_FLAGS, {.dbl=0}, 0, UINT_MAX, 0, "flags"},
53 {"res", "force resampling", 0, FF_OPT_TYPE_CONST, {.dbl=SWR_FLAG_RESAMPLE}, INT_MIN, INT_MAX, 0, "flags"},
58 static const char* context_to_name(void* ptr) {
62 static const AVClass av_class = { "SwrContext", context_to_name, options, LIBAVUTIL_VERSION_INT, OFFSET(log_level_offset), OFFSET(log_ctx) };
64 static int resample(SwrContext *s, AudioData *out_param, int out_count,
65 const AudioData * in_param, int in_count);
67 SwrContext *swr_alloc(void){
68 SwrContext *s= av_mallocz(sizeof(SwrContext));
70 s->av_class= &av_class;
71 av_opt_set_defaults2(s, 0, 0);
76 SwrContext *swr_alloc2(struct SwrContext *s, int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
77 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
78 int log_offset, void *log_ctx){
79 if(!s) s= swr_alloc();
82 s->log_level_offset= log_offset;
85 av_set_int(s, "ocl", out_ch_layout);
86 av_set_int(s, "osf", out_sample_fmt);
87 av_set_int(s, "osr", out_sample_rate);
88 av_set_int(s, "icl", in_ch_layout);
89 av_set_int(s, "isf", in_sample_fmt);
90 av_set_int(s, "isr", in_sample_rate);
92 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
93 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
94 s->int_sample_fmt = AV_SAMPLE_FMT_S16;
100 static void free_temp(AudioData *a){
102 memset(a, 0, sizeof(*a));
105 void swr_free(SwrContext **ss){
108 free_temp(&s->postin);
109 free_temp(&s->midbuf);
110 free_temp(&s->preout);
111 free_temp(&s->in_buffer);
112 swr_audio_convert_free(&s-> in_convert);
113 swr_audio_convert_free(&s->out_convert);
114 swr_audio_convert_free(&s->full_convert);
115 swr_resample_free(&s->resample);
121 int swr_init(SwrContext *s){
122 s->in_buffer_index= 0;
123 s->in_buffer_count= 0;
124 s->resample_in_constraint= 0;
125 free_temp(&s->postin);
126 free_temp(&s->midbuf);
127 free_temp(&s->preout);
128 free_temp(&s->in_buffer);
129 swr_audio_convert_free(&s-> in_convert);
130 swr_audio_convert_free(&s->out_convert);
131 swr_audio_convert_free(&s->full_convert);
133 s-> in.planar= s-> in_sample_fmt >= 0x100;
134 s->out.planar= s->out_sample_fmt >= 0x100;
135 s-> in_sample_fmt &= 0xFF;
136 s->out_sample_fmt &= 0xFF;
138 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
139 av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->in_sample_fmt));
140 return AVERROR(EINVAL);
142 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
143 av_log(s, AV_LOG_ERROR, "Requested sample format %s is invalid\n", av_get_sample_fmt_name(s->out_sample_fmt));
144 return AVERROR(EINVAL);
147 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16
148 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLT){
149 av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, only float & S16 is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
150 return AVERROR(EINVAL);
153 //FIXME should we allow/support using FLT on material that doesnt need it ?
154 if(s->in_sample_fmt <= AV_SAMPLE_FMT_S16 || s->int_sample_fmt==AV_SAMPLE_FMT_S16){
155 s->int_sample_fmt= AV_SAMPLE_FMT_S16;
157 s->int_sample_fmt= AV_SAMPLE_FMT_FLT;
160 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
161 s->resample = swr_resample_init(s->resample, s->out_sample_rate, s->in_sample_rate, 16, 10, 0, 0.8);
163 swr_resample_free(&s->resample);
164 if(s->int_sample_fmt != AV_SAMPLE_FMT_S16 && s->resample){
165 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16 currently\n"); //FIXME
169 if(s-> in.ch_count && s->in.ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
170 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than there actually is, ignoring layout\n");
174 if(!s-> in_ch_layout)
175 s-> in_ch_layout= av_get_default_channel_layout(s->in.ch_count);
176 if(!s->out_ch_layout)
177 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
179 s->rematrix= s->out_ch_layout !=s->in_ch_layout;
181 #define RSC 1 //FIXME finetune
183 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
185 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
187 av_assert0(s-> in.ch_count);
188 av_assert0(s->out.ch_count);
189 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
191 s-> in.bps= av_get_bits_per_sample_fmt(s-> in_sample_fmt)/8;
192 s->int_bps= av_get_bits_per_sample_fmt(s->int_sample_fmt)/8;
193 s->out.bps= av_get_bits_per_sample_fmt(s->out_sample_fmt)/8;
195 if(!s->resample && !s->rematrix){
196 s->full_convert = swr_audio_convert_alloc(s->out_sample_fmt,
197 s-> in_sample_fmt, s-> in.ch_count, 0);
201 s->in_convert = swr_audio_convert_alloc(s->int_sample_fmt,
202 s-> in_sample_fmt, s-> in.ch_count, 0);
203 s->out_convert= swr_audio_convert_alloc(s->out_sample_fmt,
204 s->int_sample_fmt, s->out.ch_count, 0);
211 if(!s->resample_first){
212 s->midbuf.ch_count= s->out.ch_count;
213 s->in_buffer.ch_count = s->out.ch_count;
216 s->in_buffer.bps = s->postin.bps = s->midbuf.bps = s->preout.bps = s->int_bps;
217 s->in_buffer.planar = s->postin.planar = s->midbuf.planar = s->preout.planar = 1;
220 if(s->rematrix && swr_rematrix_init(s)<0)
226 static int realloc_audio(AudioData *a, int count){
230 if(a->count >= count)
235 countb= FFALIGN(count*a->bps, 32);
238 av_assert0(a->planar);
240 av_assert0(a->ch_count);
242 a->data= av_malloc(countb*a->ch_count);
244 return AVERROR(ENOMEM);
245 for(i=0; i<a->ch_count; i++){
246 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
247 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
255 static void copy(AudioData *out, AudioData *in,
257 av_assert0(out->planar == in->planar);
258 av_assert0(out->bps == in->bps);
259 av_assert0(out->ch_count == in->ch_count);
262 for(ch=0; ch<out->ch_count; ch++)
263 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
265 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
268 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
271 for(i=0; i<out->ch_count; i++)
272 out->ch[i]= in_arg[i];
274 for(i=0; i<out->ch_count; i++)
275 out->ch[i]= in_arg[0] + i*out->bps;
279 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
280 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
281 AudioData *postin, *midbuf, *preout;
282 int ret, i/*, in_max*/;
283 AudioData * in= &s->in;
284 AudioData *out= &s->out;
285 AudioData preout_tmp, midbuf_tmp;
288 if(in_count > out_count)
290 out_count = in_count;
293 fill_audiodata(in , in_arg);
294 fill_audiodata(out, out_arg);
297 av_assert0(!s->resample);
298 swr_audio_convert(s->full_convert, out, in, in_count);
302 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
303 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
305 if((ret=realloc_audio(&s->postin, in_count))<0)
307 if(s->resample_first){
308 av_assert0(s->midbuf.ch_count == s-> in.ch_count);
309 if((ret=realloc_audio(&s->midbuf, out_count))<0)
312 av_assert0(s->midbuf.ch_count == s->out.ch_count);
313 if((ret=realloc_audio(&s->midbuf, in_count))<0)
316 if((ret=realloc_audio(&s->preout, out_count))<0)
321 midbuf_tmp= s->midbuf;
323 preout_tmp= s->preout;
326 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar)
329 if(s->resample_first ? !s->resample : !s->rematrix)
332 if(s->resample_first ? !s->rematrix : !s->resample)
335 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
337 out_count= FFMIN(out_count, in_count); //TODO check at teh end if this is needed or redundant
338 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
339 copy(out, in, out_count);
342 else if(preout==postin) preout= midbuf= postin= out;
343 else if(preout==midbuf) preout= midbuf= out;
348 swr_audio_convert(s->in_convert, postin, in, in_count);
351 if(s->resample_first){
353 out_count= resample(s, midbuf, out_count, postin, in_count);
355 swr_rematrix(s, preout, midbuf, out_count, preout==out);
358 swr_rematrix(s, midbuf, postin, in_count, midbuf==out);
360 out_count= resample(s, preout, out_count, midbuf, in_count);
364 //FIXME packed doesnt need more than 1 chan here!
365 swr_audio_convert(s->out_convert, out, preout, out_count);
372 * out may be equal in.
374 static void buf_set(AudioData *out, AudioData *in, int count){
377 for(ch=0; ch<out->ch_count; ch++)
378 out->ch[ch]= in->ch[ch] + count*out->bps;
380 out->ch[0]= in->ch[0] + count*out->ch_count*out->bps;
385 * @return number of samples output per channel
387 static int resample(SwrContext *s, AudioData *out_param, int out_count,
388 const AudioData * in_param, int in_count){
389 AudioData in, out, tmp;
397 int ret, size, consumed;
398 if(!s->resample_in_constraint && s->in_buffer_count){
399 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
400 ret= swr_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
403 buf_set(&out, &out, ret);
404 s->in_buffer_count -= consumed;
405 s->in_buffer_index += consumed;
409 if(s->in_buffer_count <= border){
410 buf_set(&in, &in, -s->in_buffer_count);
411 in_count += s->in_buffer_count;
412 s->in_buffer_count=0;
413 s->in_buffer_index=0;
418 if(in_count && !s->in_buffer_count){
419 s->in_buffer_index=0;
420 ret= swr_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
423 buf_set(&out, &out, ret);
424 in_count -= consumed;
425 buf_set(&in, &in, consumed);
428 //TODO is this check sane considering the advanced copy avoidance below
429 size= s->in_buffer_index + s->in_buffer_count + in_count;
430 if( size > s->in_buffer.count
431 && s->in_buffer_count + in_count <= s->in_buffer_index){
432 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
433 copy(&s->in_buffer, &tmp, s->in_buffer_count);
434 s->in_buffer_index=0;
436 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
441 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
443 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
444 copy(&tmp, &in, /*in_*/count);
445 s->in_buffer_count += count;
448 buf_set(&in, &in, count);
449 s->resample_in_constraint= 0;
450 if(s->in_buffer_count != count || in_count)
456 s->resample_in_constraint= !!out_count;