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00021 #include "libavutil/opt.h"
00022 #include "swresample_internal.h"
00023 #include "audioconvert.h"
00024 #include "libavutil/avassert.h"
00025 #include "libavutil/audioconvert.h"
00026
00027 #include <float.h>
00028
00029 #define C30DB M_SQRT2
00030 #define C15DB 1.189207115
00031 #define C__0DB 1.0
00032 #define C_15DB 0.840896415
00033 #define C_30DB M_SQRT1_2
00034 #define C_45DB 0.594603558
00035 #define C_60DB 0.5
00036
00037 #define ALIGN 32
00038
00039
00040 #define OFFSET(x) offsetof(SwrContext,x)
00041 #define PARAM AV_OPT_FLAG_AUDIO_PARAM
00042
00043 static const AVOption options[]={
00044 {"ich" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
00045 {"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
00046 {"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
00047 {"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
00048 {"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
00049 {"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
00050 {"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
00051 {"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
00052 {"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
00053 {"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
00054 {"isf" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
00055 {"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
00056 {"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
00057 {"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
00058 {"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
00059 {"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
00060 {"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
00061 {"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
00062 {"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
00063 {"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
00064 {"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
00065 {"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
00066 {"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
00067 {"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
00068 {"lfe_mix_level" , "LFE Mix Level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
00069 {"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
00070 {"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
00071 {"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
00072 {"swr_flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
00073 {"res" , "Force Resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},
00074 {"dither_scale" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
00075 {"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
00076 {"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
00077 {"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
00078 {"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
00079 {"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
00080 {"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
00081 {"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
00082 {"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
00083 {"min_comp" , "Minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"
00084 , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },
00085 {"min_hard_comp" , "Minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."
00086 , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },
00087 {"comp_duration" , "Duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."
00088 , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },
00089 {"max_soft_comp" , "Maximum factor by which data is stretched/squeezed to make it match the timestamps."
00090 , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },
00091 { "matrix_encoding" , "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" },
00092 { "none", "None", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
00093 { "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
00094 { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
00095 { "filter_type" , "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" },
00096 { "cubic" , "Cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
00097 { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
00098 { "kaiser" , "Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
00099 { "kaiser_beta" , "Kaiser Window Beta" ,OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
00100
00101 {0}
00102 };
00103
00104 static const char* context_to_name(void* ptr) {
00105 return "SWR";
00106 }
00107
00108 static const AVClass av_class = {
00109 .class_name = "SWResampler",
00110 .item_name = context_to_name,
00111 .option = options,
00112 .version = LIBAVUTIL_VERSION_INT,
00113 .log_level_offset_offset = OFFSET(log_level_offset),
00114 .parent_log_context_offset = OFFSET(log_ctx),
00115 .category = AV_CLASS_CATEGORY_SWRESAMPLER,
00116 };
00117
00118 unsigned swresample_version(void)
00119 {
00120 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);
00121 return LIBSWRESAMPLE_VERSION_INT;
00122 }
00123
00124 const char *swresample_configuration(void)
00125 {
00126 return FFMPEG_CONFIGURATION;
00127 }
00128
00129 const char *swresample_license(void)
00130 {
00131 #define LICENSE_PREFIX "libswresample license: "
00132 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
00133 }
00134
00135 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){
00136 if(!s || s->in_convert)
00137 return AVERROR(EINVAL);
00138 s->channel_map = channel_map;
00139 return 0;
00140 }
00141
00142 const AVClass *swr_get_class(void)
00143 {
00144 return &av_class;
00145 }
00146
00147 av_cold struct SwrContext *swr_alloc(void){
00148 SwrContext *s= av_mallocz(sizeof(SwrContext));
00149 if(s){
00150 s->av_class= &av_class;
00151 av_opt_set_defaults(s);
00152 }
00153 return s;
00154 }
00155
00156 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
00157 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
00158 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
00159 int log_offset, void *log_ctx){
00160 if(!s) s= swr_alloc();
00161 if(!s) return NULL;
00162
00163 s->log_level_offset= log_offset;
00164 s->log_ctx= log_ctx;
00165
00166 av_opt_set_int(s, "ocl", out_ch_layout, 0);
00167 av_opt_set_int(s, "osf", out_sample_fmt, 0);
00168 av_opt_set_int(s, "osr", out_sample_rate, 0);
00169 av_opt_set_int(s, "icl", in_ch_layout, 0);
00170 av_opt_set_int(s, "isf", in_sample_fmt, 0);
00171 av_opt_set_int(s, "isr", in_sample_rate, 0);
00172 av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);
00173 av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);
00174 av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);
00175 av_opt_set_int(s, "uch", 0, 0);
00176 return s;
00177 }
00178
00179 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){
00180 a->fmt = fmt;
00181 a->bps = av_get_bytes_per_sample(fmt);
00182 a->planar= av_sample_fmt_is_planar(fmt);
00183 }
00184
00185 static void free_temp(AudioData *a){
00186 av_free(a->data);
00187 memset(a, 0, sizeof(*a));
00188 }
00189
00190 av_cold void swr_free(SwrContext **ss){
00191 SwrContext *s= *ss;
00192 if(s){
00193 free_temp(&s->postin);
00194 free_temp(&s->midbuf);
00195 free_temp(&s->preout);
00196 free_temp(&s->in_buffer);
00197 free_temp(&s->dither);
00198 swri_audio_convert_free(&s-> in_convert);
00199 swri_audio_convert_free(&s->out_convert);
00200 swri_audio_convert_free(&s->full_convert);
00201 swri_resample_free(&s->resample);
00202 swri_rematrix_free(s);
00203 }
00204
00205 av_freep(ss);
00206 }
00207
00208 av_cold int swr_init(struct SwrContext *s){
00209 s->in_buffer_index= 0;
00210 s->in_buffer_count= 0;
00211 s->resample_in_constraint= 0;
00212 free_temp(&s->postin);
00213 free_temp(&s->midbuf);
00214 free_temp(&s->preout);
00215 free_temp(&s->in_buffer);
00216 free_temp(&s->dither);
00217 memset(s->in.ch, 0, sizeof(s->in.ch));
00218 memset(s->out.ch, 0, sizeof(s->out.ch));
00219 swri_audio_convert_free(&s-> in_convert);
00220 swri_audio_convert_free(&s->out_convert);
00221 swri_audio_convert_free(&s->full_convert);
00222 swri_rematrix_free(s);
00223
00224 s->flushed = 0;
00225
00226 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
00227 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
00228 return AVERROR(EINVAL);
00229 }
00230 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){
00231 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
00232 return AVERROR(EINVAL);
00233 }
00234
00235 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){
00236 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){
00237 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;
00238 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){
00239 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;
00240 }else{
00241 av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
00242 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;
00243 }
00244 }
00245
00246 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
00247 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P
00248 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
00249 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){
00250 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));
00251 return AVERROR(EINVAL);
00252 }
00253
00254 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);
00255 set_audiodata_fmt(&s->out, s->out_sample_fmt);
00256
00257 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
00258 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);
00259 }else
00260 swri_resample_free(&s->resample);
00261 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P
00262 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P
00263 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP
00264 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP
00265 && s->resample){
00266 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
00267 return -1;
00268 }
00269
00270 if(!s->used_ch_count)
00271 s->used_ch_count= s->in.ch_count;
00272
00273 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){
00274 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
00275 s-> in_ch_layout= 0;
00276 }
00277
00278 if(!s-> in_ch_layout)
00279 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);
00280 if(!s->out_ch_layout)
00281 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);
00282
00283 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
00284 s->rematrix_custom;
00285
00286 #define RSC 1 //FIXME finetune
00287 if(!s-> in.ch_count)
00288 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);
00289 if(!s->used_ch_count)
00290 s->used_ch_count= s->in.ch_count;
00291 if(!s->out.ch_count)
00292 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);
00293
00294 if(!s-> in.ch_count){
00295 av_assert0(!s->in_ch_layout);
00296 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
00297 return -1;
00298 }
00299
00300 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
00301 av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
00302 return -1;
00303 }
00304
00305 av_assert0(s->used_ch_count);
00306 av_assert0(s->out.ch_count);
00307 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
00308
00309 s->in_buffer= s->in;
00310
00311 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither_method){
00312 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,
00313 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
00314 return 0;
00315 }
00316
00317 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,
00318 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
00319 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,
00320 s->int_sample_fmt, s->out.ch_count, NULL, 0);
00321
00322
00323 s->postin= s->in;
00324 s->preout= s->out;
00325 s->midbuf= s->in;
00326
00327 if(s->channel_map){
00328 s->postin.ch_count=
00329 s->midbuf.ch_count= s->used_ch_count;
00330 if(s->resample)
00331 s->in_buffer.ch_count= s->used_ch_count;
00332 }
00333 if(!s->resample_first){
00334 s->midbuf.ch_count= s->out.ch_count;
00335 if(s->resample)
00336 s->in_buffer.ch_count = s->out.ch_count;
00337 }
00338
00339 set_audiodata_fmt(&s->postin, s->int_sample_fmt);
00340 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);
00341 set_audiodata_fmt(&s->preout, s->int_sample_fmt);
00342
00343 if(s->resample){
00344 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);
00345 }
00346
00347 s->dither = s->preout;
00348
00349 if(s->rematrix || s->dither_method)
00350 return swri_rematrix_init(s);
00351
00352 return 0;
00353 }
00354
00355 static int realloc_audio(AudioData *a, int count){
00356 int i, countb;
00357 AudioData old;
00358
00359 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
00360 return AVERROR(EINVAL);
00361
00362 if(a->count >= count)
00363 return 0;
00364
00365 count*=2;
00366
00367 countb= FFALIGN(count*a->bps, ALIGN);
00368 old= *a;
00369
00370 av_assert0(a->bps);
00371 av_assert0(a->ch_count);
00372
00373 a->data= av_mallocz(countb*a->ch_count);
00374 if(!a->data)
00375 return AVERROR(ENOMEM);
00376 for(i=0; i<a->ch_count; i++){
00377 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
00378 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
00379 }
00380 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
00381 av_free(old.data);
00382 a->count= count;
00383
00384 return 1;
00385 }
00386
00387 static void copy(AudioData *out, AudioData *in,
00388 int count){
00389 av_assert0(out->planar == in->planar);
00390 av_assert0(out->bps == in->bps);
00391 av_assert0(out->ch_count == in->ch_count);
00392 if(out->planar){
00393 int ch;
00394 for(ch=0; ch<out->ch_count; ch++)
00395 memcpy(out->ch[ch], in->ch[ch], count*out->bps);
00396 }else
00397 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
00398 }
00399
00400 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
00401 int i;
00402 if(!in_arg){
00403 memset(out->ch, 0, sizeof(out->ch));
00404 }else if(out->planar){
00405 for(i=0; i<out->ch_count; i++)
00406 out->ch[i]= in_arg[i];
00407 }else{
00408 for(i=0; i<out->ch_count; i++)
00409 out->ch[i]= in_arg[0] + i*out->bps;
00410 }
00411 }
00412
00413 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
00414 int i;
00415 if(out->planar){
00416 for(i=0; i<out->ch_count; i++)
00417 in_arg[i]= out->ch[i];
00418 }else{
00419 in_arg[0]= out->ch[0];
00420 }
00421 }
00422
00427 static void buf_set(AudioData *out, AudioData *in, int count){
00428 int ch;
00429 if(in->planar){
00430 for(ch=0; ch<out->ch_count; ch++)
00431 out->ch[ch]= in->ch[ch] + count*out->bps;
00432 }else{
00433 for(ch=out->ch_count-1; ch>=0; ch--)
00434 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
00435 }
00436 }
00437
00442 static int resample(SwrContext *s, AudioData *out_param, int out_count,
00443 const AudioData * in_param, int in_count){
00444 AudioData in, out, tmp;
00445 int ret_sum=0;
00446 int border=0;
00447
00448 av_assert1(s->in_buffer.ch_count == in_param->ch_count);
00449 av_assert1(s->in_buffer.planar == in_param->planar);
00450 av_assert1(s->in_buffer.fmt == in_param->fmt);
00451
00452 tmp=out=*out_param;
00453 in = *in_param;
00454
00455 do{
00456 int ret, size, consumed;
00457 if(!s->resample_in_constraint && s->in_buffer_count){
00458 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
00459 ret= swri_multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
00460 out_count -= ret;
00461 ret_sum += ret;
00462 buf_set(&out, &out, ret);
00463 s->in_buffer_count -= consumed;
00464 s->in_buffer_index += consumed;
00465
00466 if(!in_count)
00467 break;
00468 if(s->in_buffer_count <= border){
00469 buf_set(&in, &in, -s->in_buffer_count);
00470 in_count += s->in_buffer_count;
00471 s->in_buffer_count=0;
00472 s->in_buffer_index=0;
00473 border = 0;
00474 }
00475 }
00476
00477 if(in_count && !s->in_buffer_count){
00478 s->in_buffer_index=0;
00479 ret= swri_multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed);
00480 out_count -= ret;
00481 ret_sum += ret;
00482 buf_set(&out, &out, ret);
00483 in_count -= consumed;
00484 buf_set(&in, &in, consumed);
00485 }
00486
00487
00488 size= s->in_buffer_index + s->in_buffer_count + in_count;
00489 if( size > s->in_buffer.count
00490 && s->in_buffer_count + in_count <= s->in_buffer_index){
00491 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
00492 copy(&s->in_buffer, &tmp, s->in_buffer_count);
00493 s->in_buffer_index=0;
00494 }else
00495 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
00496 return ret;
00497
00498 if(in_count){
00499 int count= in_count;
00500 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
00501
00502 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
00503 copy(&tmp, &in, count);
00504 s->in_buffer_count += count;
00505 in_count -= count;
00506 border += count;
00507 buf_set(&in, &in, count);
00508 s->resample_in_constraint= 0;
00509 if(s->in_buffer_count != count || in_count)
00510 continue;
00511 }
00512 break;
00513 }while(1);
00514
00515 s->resample_in_constraint= !!out_count;
00516
00517 return ret_sum;
00518 }
00519
00520 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
00521 AudioData *in , int in_count){
00522 AudioData *postin, *midbuf, *preout;
00523 int ret;
00524 AudioData preout_tmp, midbuf_tmp;
00525
00526 if(s->full_convert){
00527 av_assert0(!s->resample);
00528 swri_audio_convert(s->full_convert, out, in, in_count);
00529 return out_count;
00530 }
00531
00532
00533
00534
00535 if((ret=realloc_audio(&s->postin, in_count))<0)
00536 return ret;
00537 if(s->resample_first){
00538 av_assert0(s->midbuf.ch_count == s->used_ch_count);
00539 if((ret=realloc_audio(&s->midbuf, out_count))<0)
00540 return ret;
00541 }else{
00542 av_assert0(s->midbuf.ch_count == s->out.ch_count);
00543 if((ret=realloc_audio(&s->midbuf, in_count))<0)
00544 return ret;
00545 }
00546 if((ret=realloc_audio(&s->preout, out_count))<0)
00547 return ret;
00548
00549 postin= &s->postin;
00550
00551 midbuf_tmp= s->midbuf;
00552 midbuf= &midbuf_tmp;
00553 preout_tmp= s->preout;
00554 preout= &preout_tmp;
00555
00556 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
00557 postin= in;
00558
00559 if(s->resample_first ? !s->resample : !s->rematrix)
00560 midbuf= postin;
00561
00562 if(s->resample_first ? !s->rematrix : !s->resample)
00563 preout= midbuf;
00564
00565 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){
00566 if(preout==in){
00567 out_count= FFMIN(out_count, in_count);
00568 av_assert0(s->in.planar);
00569 copy(out, in, out_count);
00570 return out_count;
00571 }
00572 else if(preout==postin) preout= midbuf= postin= out;
00573 else if(preout==midbuf) preout= midbuf= out;
00574 else preout= out;
00575 }
00576
00577 if(in != postin){
00578 swri_audio_convert(s->in_convert, postin, in, in_count);
00579 }
00580
00581 if(s->resample_first){
00582 if(postin != midbuf)
00583 out_count= resample(s, midbuf, out_count, postin, in_count);
00584 if(midbuf != preout)
00585 swri_rematrix(s, preout, midbuf, out_count, preout==out);
00586 }else{
00587 if(postin != midbuf)
00588 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
00589 if(midbuf != preout)
00590 out_count= resample(s, preout, out_count, midbuf, in_count);
00591 }
00592
00593 if(preout != out && out_count){
00594 if(s->dither_method){
00595 int ch;
00596 int dither_count= FFMAX(out_count, 1<<16);
00597 av_assert0(preout != in);
00598
00599 if((ret=realloc_audio(&s->dither, dither_count))<0)
00600 return ret;
00601 if(ret)
00602 for(ch=0; ch<s->dither.ch_count; ch++)
00603 swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
00604 av_assert0(s->dither.ch_count == preout->ch_count);
00605
00606 if(s->dither_pos + out_count > s->dither.count)
00607 s->dither_pos = 0;
00608
00609 for(ch=0; ch<preout->ch_count; ch++)
00610 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);
00611
00612 s->dither_pos += out_count;
00613 }
00614
00615 swri_audio_convert(s->out_convert, out, preout, out_count);
00616 }
00617 return out_count;
00618 }
00619
00620 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
00621 const uint8_t *in_arg [SWR_CH_MAX], int in_count){
00622 AudioData * in= &s->in;
00623 AudioData *out= &s->out;
00624
00625 if(s->drop_output > 0){
00626 int ret;
00627 AudioData tmp = s->out;
00628 uint8_t *tmp_arg[SWR_CH_MAX];
00629 tmp.count = 0;
00630 tmp.data = NULL;
00631 if((ret=realloc_audio(&tmp, s->drop_output))<0)
00632 return ret;
00633
00634 reversefill_audiodata(&tmp, tmp_arg);
00635 s->drop_output *= -1;
00636 ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count);
00637 s->drop_output *= -1;
00638 if(ret>0)
00639 s->drop_output -= ret;
00640
00641 av_freep(&tmp.data);
00642 if(s->drop_output || !out_arg)
00643 return 0;
00644 in_count = 0;
00645 }
00646
00647 if(!in_arg){
00648 if(s->in_buffer_count){
00649 if (s->resample && !s->flushed) {
00650 AudioData *a= &s->in_buffer;
00651 int i, j, ret;
00652 if((ret=realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0)
00653 return ret;
00654 av_assert0(a->planar);
00655 for(i=0; i<a->ch_count; i++){
00656 for(j=0; j<s->in_buffer_count; j++){
00657 memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps,
00658 a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps);
00659 }
00660 }
00661 s->in_buffer_count += (s->in_buffer_count+1)/2;
00662 s->resample_in_constraint = 0;
00663 s->flushed = 1;
00664 }
00665 }else{
00666 return 0;
00667 }
00668 }else
00669 fill_audiodata(in , (void*)in_arg);
00670
00671 fill_audiodata(out, out_arg);
00672
00673 if(s->resample){
00674 int ret = swr_convert_internal(s, out, out_count, in, in_count);
00675 if(ret>0 && !s->drop_output)
00676 s->outpts += ret * (int64_t)s->in_sample_rate;
00677 return ret;
00678 }else{
00679 AudioData tmp= *in;
00680 int ret2=0;
00681 int ret, size;
00682 size = FFMIN(out_count, s->in_buffer_count);
00683 if(size){
00684 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
00685 ret= swr_convert_internal(s, out, size, &tmp, size);
00686 if(ret<0)
00687 return ret;
00688 ret2= ret;
00689 s->in_buffer_count -= ret;
00690 s->in_buffer_index += ret;
00691 buf_set(out, out, ret);
00692 out_count -= ret;
00693 if(!s->in_buffer_count)
00694 s->in_buffer_index = 0;
00695 }
00696
00697 if(in_count){
00698 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
00699
00700 if(in_count > out_count) {
00701 if( size > s->in_buffer.count
00702 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
00703 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
00704 copy(&s->in_buffer, &tmp, s->in_buffer_count);
00705 s->in_buffer_index=0;
00706 }else
00707 if((ret=realloc_audio(&s->in_buffer, size)) < 0)
00708 return ret;
00709 }
00710
00711 if(out_count){
00712 size = FFMIN(in_count, out_count);
00713 ret= swr_convert_internal(s, out, size, in, size);
00714 if(ret<0)
00715 return ret;
00716 buf_set(in, in, ret);
00717 in_count -= ret;
00718 ret2 += ret;
00719 }
00720 if(in_count){
00721 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
00722 copy(&tmp, in, in_count);
00723 s->in_buffer_count += in_count;
00724 }
00725 }
00726 if(ret2>0 && !s->drop_output)
00727 s->outpts += ret2 * (int64_t)s->in_sample_rate;
00728 return ret2;
00729 }
00730 }
00731
00732 int swr_drop_output(struct SwrContext *s, int count){
00733 s->drop_output += count;
00734
00735 if(s->drop_output <= 0)
00736 return 0;
00737
00738 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
00739 return swr_convert(s, NULL, s->drop_output, NULL, 0);
00740 }
00741
00742 int swr_inject_silence(struct SwrContext *s, int count){
00743 int ret, i;
00744 AudioData silence = s->in;
00745 uint8_t *tmp_arg[SWR_CH_MAX];
00746
00747 if(count <= 0)
00748 return 0;
00749
00750 silence.count = 0;
00751 silence.data = NULL;
00752 if((ret=realloc_audio(&silence, count))<0)
00753 return ret;
00754
00755 if(silence.planar) for(i=0; i<silence.ch_count; i++) {
00756 memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
00757 } else
00758 memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
00759
00760 reversefill_audiodata(&silence, tmp_arg);
00761 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
00762 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
00763 av_freep(&silence.data);
00764 return ret;
00765 }
00766
00767 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
00768 if(pts == INT64_MIN)
00769 return s->outpts;
00770 if(s->min_compensation >= FLT_MAX) {
00771 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
00772 } else {
00773 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts;
00774 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
00775
00776 if(fabs(fdelta) > s->min_compensation) {
00777 if(!s->outpts || fabs(fdelta) > s->min_hard_compensation){
00778 int ret;
00779 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
00780 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
00781 if(ret<0){
00782 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
00783 }
00784 } else if(s->soft_compensation_duration && s->max_soft_compensation) {
00785 int duration = s->out_sample_rate * s->soft_compensation_duration;
00786 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);
00787 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
00788 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
00789 swr_set_compensation(s, comp, duration);
00790 }
00791 }
00792
00793 return s->outpts;
00794 }
00795 }
