FFmpeg: libswresample/swresample.c Source File

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libswresample

swresample.c

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1 /*

3 *

4 * This file is part of libswresample

5 *

6 * libswresample is free software; you can redistribute it and/or

7 * modify it under the terms of the GNU Lesser General Public

8 * License as published by the Free Software Foundation; either

9 * version 2.1 of the License, or (at your option) any later version.

10 *

11 * libswresample is distributed in the hope that it will be useful,

12 * but WITHOUT ANY WARRANTY; without even the implied warranty of

13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

14 * Lesser General Public License for more details.

15 *

16 * You should have received a copy of the GNU Lesser General Public

17 * License along with libswresample; if not, write to the Free Software

18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

19 */

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"

27 #include <float.h>

29 #define C30DB M_SQRT2

30 #define C15DB 1.189207115

31 #define C__0DB 1.0

32 #define C_15DB 0.840896415

33 #define C_30DB M_SQRT1_2

34 #define C_45DB 0.594603558

35 #define C_60DB 0.5

37 #define ALIGN 32

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 , INT_MAX, PARAM},

55 {"in_sample_fmt" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},

56 {"osf" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},

57 {"out_sample_fmt" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},

58 {"tsf" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},

59 {"internal_sample_fmt" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , INT_MAX, PARAM},

60 {"icl" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},

61 {"in_channel_layout" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},

62 {"ocl" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},

63 {"out_channel_layout" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},

64 {"clev" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},

65 {"center_mix_level" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},

66 {"slev" , "set surround mix level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},

67 {"surround_mix_level" , "set surround mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},

68 {"lfe_mix_level" , "set LFE mix level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},

69 {"rmvol" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},

70 {"rematrix_volume" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},

71 {"rematrix_maxval" , "set rematrix maxval" , OFFSET(rematrix_maxval), AV_OPT_TYPE_FLOAT, {.dbl=0.0 }, 0 , 1000 , PARAM},

73 {"flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},

74 {"swr_flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},

75 {"res" , "force resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"},

77 {"dither_scale" , "set dither scale" , OFFSET(dither.scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},

79 {"dither_method" , "set dither method" , OFFSET(dither.method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},

80 {"rectangular" , "select rectangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},

81 {"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},

82 {"triangular_hp" , "select triangular dither with high pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},

83 {"lipshitz" , "select lipshitz noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"},

84 {"shibata" , "select shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},

85 {"low_shibata" , "select low shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},

86 {"high_shibata" , "select high shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"},

87 {"f_weighted" , "select f-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},

88 {"modified_e_weighted" , "select modified-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},

89 {"improved_e_weighted" , "select improved-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"},

91 {"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM },

92 {"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 24 , PARAM },

93 {"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },

94 {"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },

96 /* duplicate option in order to work with avconv */

97 {"resample_cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM },

99 {"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"},

100 {"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"},

101 {"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"},

102 {"precision" , "set soxr resampling precision (in bits)"

103 , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM },

104 {"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation"

105 , OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },

106 {"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied"

107 , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM },

108 {"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data."

109 , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM },

110 {"comp_duration" , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps."

111 , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM },

112 {"max_soft_comp" , "set maximum factor by which data is stretched/squeezed to make it match the timestamps."

113 , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },

114 {"async" , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)"

115 , OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM },

116 {"first_pts" , "Assume the first pts should be this value (in samples)."

117 , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE }, INT64_MIN,INT64_MAX, PARAM },

118

119 { "matrix_encoding" , "set matrixed stereo encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },

120 { "none", "select none", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },

121 { "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },

122 { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },

123

124 { "filter_type" , "select swr filter type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },

125 { "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },

126 { "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },

127 { "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },

128

129 { "kaiser_beta" , "set swr Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },

130

131 { "output_sample_bits" , "set swr number of output sample bits", OFFSET(dither.output_sample_bits), AV_OPT_TYPE_INT , {.i64=0 }, 0 , 64 , PARAM },

132 {0}

133 };

134

135 static const char* context_to_name(void* ptr) {

136 return "SWR";

137 }

138

139 static const AVClass av_class = {

140 .class_name = "SWResampler",

141 .item_name = context_to_name,

142 .option = options,

143 .version = LIBAVUTIL_VERSION_INT,

144 .log_level_offset_offset = OFFSET(log_level_offset),

145 .parent_log_context_offset = OFFSET(log_ctx),

146 .category = AV_CLASS_CATEGORY_SWRESAMPLER,

147 };

148

149 unsigned swresample_version(void)

150 {

151 av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100);

152 return LIBSWRESAMPLE_VERSION_INT;

153 }

154

155 const char *swresample_configuration(void)

156 {

157 return FFMPEG_CONFIGURATION;

158 }

159

160 const char *swresample_license(void)

161 {

162 #define LICENSE_PREFIX "libswresample license: "

163 return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;

164 }

165

166 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){

167 if(!s || s->in_convert) // s needs to be allocated but not initialized

168 return AVERROR(EINVAL);

169 s->channel_map = channel_map;

170 return 0;

171 }

172

173 const AVClass *swr_get_class(void)

174 {

175 return &av_class;

176 }

177

178 av_cold struct SwrContext *swr_alloc(void){

179 SwrContext *s= av_mallocz(sizeof(SwrContext));

180 if(s){

181 s->av_class= &av_class;

182 av_opt_set_defaults(s);

183 }

184 return s;

185 }

186

187 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,

188 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,

189 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,

190 int log_offset, void *log_ctx){

191 if(!s) s= swr_alloc();

192 if(!s) return NULL;

193

194 s->log_level_offset= log_offset;

195 s->log_ctx= log_ctx;

196

197 av_opt_set_int(s, "ocl", out_ch_layout, 0);

198 av_opt_set_int(s, "osf", out_sample_fmt, 0);

199 av_opt_set_int(s, "osr", out_sample_rate, 0);

200 av_opt_set_int(s, "icl", in_ch_layout, 0);

201 av_opt_set_int(s, "isf", in_sample_fmt, 0);

202 av_opt_set_int(s, "isr", in_sample_rate, 0);

203 av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0);

204 av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0);

205 av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0);

206 av_opt_set_int(s, "uch", 0, 0);

207 return s;

208 }

209

210 static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){

211 a->fmt = fmt;

212 a->bps = av_get_bytes_per_sample(fmt);

213 a->planar= av_sample_fmt_is_planar(fmt);

214 }

215

216 static void free_temp(AudioData *a){

217 av_free(a->data);

218 memset(a, 0, sizeof(*a));

219 }

220

221 static void clear_context(SwrContext *s){

222 s->in_buffer_index= 0;

223 s->in_buffer_count= 0;

224 s->resample_in_constraint= 0;

225 memset(s->in.ch, 0, sizeof(s->in.ch));

226 memset(s->out.ch, 0, sizeof(s->out.ch));

227 free_temp(&s->postin);

228 free_temp(&s->midbuf);

229 free_temp(&s->preout);

230 free_temp(&s->in_buffer);

231 free_temp(&s->silence);

232 free_temp(&s->drop_temp);

233 free_temp(&s->dither.noise);

234 free_temp(&s->dither.temp);

235 swri_audio_convert_free(&s-> in_convert);

236 swri_audio_convert_free(&s->out_convert);

237 swri_audio_convert_free(&s->full_convert);

238 swri_rematrix_free(s);

239

240 s->flushed = 0;

241 }

242

243 av_cold void swr_free(SwrContext **ss){

244 SwrContext *s= *ss;

245 if(s){

246 clear_context(s);

247 if (s->resampler)

248 s->resampler->free(&s->resample);

249 }

250

251 av_freep(ss);

252 }

253

254 av_cold int swr_init(struct SwrContext *s){

255 int ret;

256

257 clear_context(s);

258

259 if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){

260 av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);

261 return AVERROR(EINVAL);

262 }

263 if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){

264 av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);

265 return AVERROR(EINVAL);

266 }

267

268 if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) {

269 av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);

270 s->in_ch_layout = 0;

271 }

272

273 if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) {

274 av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);

275 s->out_ch_layout = 0;

276 }

277

278 switch(s->engine){

279 #if CONFIG_LIBSOXR

280 extern struct Resampler const soxr_resampler;

281 case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break;

282 #endif

283 case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;

284 default:

285 av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");

286 return AVERROR(EINVAL);

287 }

288

289 if(!s->used_ch_count)

290 s->used_ch_count= s->in.ch_count;

291

292 if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){

293 av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");

294 s-> in_ch_layout= 0;

295 }

296

297 if(!s-> in_ch_layout)

298 s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count);

299 if(!s->out_ch_layout)

300 s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count);

301

302 s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||

303 s->rematrix_custom;

304

305 if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){

306 if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){

307 s->int_sample_fmt= AV_SAMPLE_FMT_S16P;

308 }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P

309 && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P

310 && !s->rematrix

311 && s->engine != SWR_ENGINE_SOXR){

312 s->int_sample_fmt= AV_SAMPLE_FMT_S32P;

313 }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){

314 s->int_sample_fmt= AV_SAMPLE_FMT_FLTP;

315 }else{

316 av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");

317 s->int_sample_fmt= AV_SAMPLE_FMT_DBLP;

318 }

319 }

320

321 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P

322 &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P

323 &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

324 &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){

325 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));

326 return AVERROR(EINVAL);

327 }

328

329 set_audiodata_fmt(&s-> in, s-> in_sample_fmt);

330 set_audiodata_fmt(&s->out, s->out_sample_fmt);

331

332 if (s->firstpts_in_samples != AV_NOPTS_VALUE) {

333 if (!s->async && s->min_compensation >= FLT_MAX/2)

334 s->async = 1;

335 s->firstpts =

336 s->outpts = s->firstpts_in_samples * s->out_sample_rate;

337 } else

338 s->firstpts = AV_NOPTS_VALUE;

339

340 if (s->async) {

341 if (s->min_compensation >= FLT_MAX/2)

342 s->min_compensation = 0.001;

343 if (s->async > 1.0001) {

344 s->max_soft_compensation = s->async / (double) s->in_sample_rate;

345 }

346 }

347

348 if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){

349 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);

350 }else

351 s->resampler->free(&s->resample);

352 if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P

353 && s->int_sample_fmt != AV_SAMPLE_FMT_S32P

354 && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP

355 && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP

356 && s->resample){

357 av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");

358 return -1;

359 }

360

361 #define RSC 1 //FIXME finetune

362 if(!s-> in.ch_count)

363 s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout);

364 if(!s->used_ch_count)

365 s->used_ch_count= s->in.ch_count;

366 if(!s->out.ch_count)

367 s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout);

368

369 if(!s-> in.ch_count){

370 av_assert0(!s->in_ch_layout);

371 av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");

372 return -1;

373 }

374

375 if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {

376 char l1[1024], l2[1024];

377 av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);

378 av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);

379 av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "

380 "but there is not enough information to do it\n", l1, l2);

381 return -1;

382 }

383

384 av_assert0(s->used_ch_count);

385 av_assert0(s->out.ch_count);

386 s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;

387

388 s->in_buffer= s->in;

389 s->silence = s->in;

390 s->drop_temp= s->out;

391

392 if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){

393 s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt,

394 s-> in_sample_fmt, s-> in.ch_count, NULL, 0);

395 return 0;

396 }

397

398 s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt,

399 s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);

400 s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt,

401 s->int_sample_fmt, s->out.ch_count, NULL, 0);

402

403 if (!s->in_convert || !s->out_convert)

404 return AVERROR(ENOMEM);

405

406 s->postin= s->in;

407 s->preout= s->out;

408 s->midbuf= s->in;

409

410 if(s->channel_map){

411 s->postin.ch_count=

412 s->midbuf.ch_count= s->used_ch_count;

413 if(s->resample)

414 s->in_buffer.ch_count= s->used_ch_count;

415 }

416 if(!s->resample_first){

417 s->midbuf.ch_count= s->out.ch_count;

418 if(s->resample)

419 s->in_buffer.ch_count = s->out.ch_count;

420 }

421

422 set_audiodata_fmt(&s->postin, s->int_sample_fmt);

423 set_audiodata_fmt(&s->midbuf, s->int_sample_fmt);

424 set_audiodata_fmt(&s->preout, s->int_sample_fmt);

425

426 if(s->resample){

427 set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt);

428 }

429

430 if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)

431 return ret;

432

433 if(s->rematrix || s->dither.method)

434 return swri_rematrix_init(s);

435

436 return 0;

437 }

438

439 int swri_realloc_audio(AudioData *a, int count){

440 int i, countb;

441 AudioData old;

442

443 if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)

444 return AVERROR(EINVAL);

445

446 if(a->count >= count)

447 return 0;

448

449 count*=2;

450

451 countb= FFALIGN(count*a->bps, ALIGN);

452 old= *a;

453

454 av_assert0(a->bps);

455 av_assert0(a->ch_count);

456

457 a->data= av_mallocz(countb*a->ch_count);

458 if(!a->data)

459 return AVERROR(ENOMEM);

460 for(i=0; i<a->ch_count; i++){

461 a->ch[i]= a->data + i*(a->planar ? countb : a->bps);

462 if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);

463 }

464 if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);

465 av_freep(&old.data);

466 a->count= count;

467

468 return 1;

469 }

470

471 static void copy(AudioData *out, AudioData *in,

472 int count){

473 av_assert0(out->planar == in->planar);

474 av_assert0(out->bps == in->bps);

475 av_assert0(out->ch_count == in->ch_count);

476 if(out->planar){

477 int ch;

478 for(ch=0; ch<out->ch_count; ch++)

479 memcpy(out->ch[ch], in->ch[ch], count*out->bps);

480 }else

481 memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);

482 }

483

484 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){

485 int i;

486 if(!in_arg){

487 memset(out->ch, 0, sizeof(out->ch));

488 }else if(out->planar){

489 for(i=0; i<out->ch_count; i++)

490 out->ch[i]= in_arg[i];

491 }else{

492 for(i=0; i<out->ch_count; i++)

493 out->ch[i]= in_arg[0] + i*out->bps;

494 }

495 }

496

497 static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){

498 int i;

499 if(out->planar){

500 for(i=0; i<out->ch_count; i++)

501 in_arg[i]= out->ch[i];

502 }else{

503 in_arg[0]= out->ch[0];

504 }

505 }

506

507 /**

508 *

509 * out may be equal in.

510 */

511 static void buf_set(AudioData *out, AudioData *in, int count){

512 int ch;

513 if(in->planar){

514 for(ch=0; ch<out->ch_count; ch++)

515 out->ch[ch]= in->ch[ch] + count*out->bps;

516 }else{

517 for(ch=out->ch_count-1; ch>=0; ch--)

518 out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;

519 }

520 }

521

522 /**

523 *

524 * @return number of samples output per channel

525 */

526 static int resample(SwrContext *s, AudioData *out_param, int out_count,

527 const AudioData * in_param, int in_count){

528 AudioData in, out, tmp;

529 int ret_sum=0;

530 int border=0;

531 int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;

532

533 av_assert1(s->in_buffer.ch_count == in_param->ch_count);

534 av_assert1(s->in_buffer.planar == in_param->planar);

535 av_assert1(s->in_buffer.fmt == in_param->fmt);

536

537 tmp=out=*out_param;

538 in = *in_param;

539

540 do{

541 int ret, size, consumed;

542 if(!s->resample_in_constraint && s->in_buffer_count){

543 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

544 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);

545 out_count -= ret;

546 ret_sum += ret;

547 buf_set(&out, &out, ret);

548 s->in_buffer_count -= consumed;

549 s->in_buffer_index += consumed;

550

551 if(!in_count)

552 break;

553 if(s->in_buffer_count <= border){

554 buf_set(&in, &in, -s->in_buffer_count);

555 in_count += s->in_buffer_count;

556 s->in_buffer_count=0;

557 s->in_buffer_index=0;

558 border = 0;

559 }

560 }

561

562 if((s->flushed || in_count > padless) && !s->in_buffer_count){

563 s->in_buffer_index=0;

564 ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);

565 out_count -= ret;

566 ret_sum += ret;

567 buf_set(&out, &out, ret);

568 in_count -= consumed;

569 buf_set(&in, &in, consumed);

570 }

571

572 //TODO is this check sane considering the advanced copy avoidance below

573 size= s->in_buffer_index + s->in_buffer_count + in_count;

574 if( size > s->in_buffer.count

575 && s->in_buffer_count + in_count <= s->in_buffer_index){

576 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

577 copy(&s->in_buffer, &tmp, s->in_buffer_count);

578 s->in_buffer_index=0;

579 }else

580 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)

581 return ret;

582

583 if(in_count){

584 int count= in_count;

585 if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;

586

587 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);

588 copy(&tmp, &in, /*in_*/count);

589 s->in_buffer_count += count;

590 in_count -= count;

591 border += count;

592 buf_set(&in, &in, count);

593 s->resample_in_constraint= 0;

594 if(s->in_buffer_count != count || in_count)

595 continue;

596 if (padless) {

597 padless = 0;

598 continue;

599 }

600 }

601 break;

602 }while(1);

603

604 s->resample_in_constraint= !!out_count;

605

606 return ret_sum;

607 }

608

609 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,

610 AudioData *in , int in_count){

611 AudioData *postin, *midbuf, *preout;

612 int ret /*, in_max*/;

613 AudioData preout_tmp, midbuf_tmp;

614

615 if(s->full_convert){

616 av_assert0(!s->resample);

617 swri_audio_convert(s->full_convert, out, in, in_count);

618 return out_count;

619 }

620

621 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;

622 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);

623

624 if((ret=swri_realloc_audio(&s->postin, in_count))<0)

625 return ret;

626 if(s->resample_first){

627 av_assert0(s->midbuf.ch_count == s->used_ch_count);

628 if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)

629 return ret;

630 }else{

631 av_assert0(s->midbuf.ch_count == s->out.ch_count);

632 if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)

633 return ret;

634 }

635 if((ret=swri_realloc_audio(&s->preout, out_count))<0)

636 return ret;

637

638 postin= &s->postin;

639

640 midbuf_tmp= s->midbuf;

641 midbuf= &midbuf_tmp;

642 preout_tmp= s->preout;

643 preout= &preout_tmp;

644

645 if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)

646 postin= in;

647

648 if(s->resample_first ? !s->resample : !s->rematrix)

649 midbuf= postin;

650

651 if(s->resample_first ? !s->rematrix : !s->resample)

652 preout= midbuf;

653

654 if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar

655 && !(s->out_sample_fmt==AV_SAMPLE_FMT_S32P && (s->dither.output_sample_bits&31))){

656 if(preout==in){

657 out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant

658 av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though

659 copy(out, in, out_count);

660 return out_count;

661 }

662 else if(preout==postin) preout= midbuf= postin= out;

663 else if(preout==midbuf) preout= midbuf= out;

664 else preout= out;

665 }

666

667 if(in != postin){

668 swri_audio_convert(s->in_convert, postin, in, in_count);

669 }

670

671 if(s->resample_first){

672 if(postin != midbuf)

673 out_count= resample(s, midbuf, out_count, postin, in_count);

674 if(midbuf != preout)

675 swri_rematrix(s, preout, midbuf, out_count, preout==out);

676 }else{

677 if(postin != midbuf)

678 swri_rematrix(s, midbuf, postin, in_count, midbuf==out);

679 if(midbuf != preout)

680 out_count= resample(s, preout, out_count, midbuf, in_count);

681 }

682

683 if(preout != out && out_count){

684 AudioData *conv_src = preout;

685 if(s->dither.method){

686 int ch;

687 int dither_count= FFMAX(out_count, 1<<16);

688

689 if (preout == in) {

690 conv_src = &s->dither.temp;

691 if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)

692 return ret;

693 }

694

695 if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)

696 return ret;

697 if(ret)

698 for(ch=0; ch<s->dither.noise.ch_count; ch++)

699 swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);

700 av_assert0(s->dither.noise.ch_count == preout->ch_count);

701

702 if(s->dither.noise_pos + out_count > s->dither.noise.count)

703 s->dither.noise_pos = 0;

704

705 if (s->dither.method < SWR_DITHER_NS){

706 if (s->mix_2_1_simd) {

707 int len1= out_count&~15;

708 int off = len1 * preout->bps;

709

710 if(len1)

711 for(ch=0; ch<preout->ch_count; ch++)

712 s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);

713 if(out_count != len1)

714 for(ch=0; ch<preout->ch_count; ch++)

715 s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1);

716 } else {

717 for(ch=0; ch<preout->ch_count; ch++)

718 s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);

719 }

720 } else {

721 switch(s->int_sample_fmt) {

722 case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;

723 case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;

724 case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;

725 case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;

726 }

727 }

728 s->dither.noise_pos += out_count;

729 }

730 //FIXME packed doesn't need more than 1 chan here!

731 swri_audio_convert(s->out_convert, out, conv_src, out_count);

732 }

733 return out_count;

734 }

735

736 int swr_is_initialized(struct SwrContext *s) {

737 return !!s->in_buffer.ch_count;

738 }

739

740 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,

741 const uint8_t *in_arg [SWR_CH_MAX], int in_count){

742 AudioData * in= &s->in;

743 AudioData *out= &s->out;

744

745 if (!swr_is_initialized(s)) {

746 av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");

747 return AVERROR(EINVAL);

748 }

749

750 while(s->drop_output > 0){

751 int ret;

752 uint8_t *tmp_arg[SWR_CH_MAX];

753 #define MAX_DROP_STEP 16384

754 if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0)

755 return ret;

756

757 reversefill_audiodata(&s->drop_temp, tmp_arg);

758 s->drop_output *= -1; //FIXME find a less hackish solution

759 ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter

760 s->drop_output *= -1;

761 in_count = 0;

762 if(ret>0) {

763 s->drop_output -= ret;

764 continue;

765 }

766

767 if(s->drop_output || !out_arg)

768 return 0;

769 }

770

771 if(!in_arg){

772 if(s->resample){

773 if (!s->flushed)

774 s->resampler->flush(s);

775 s->resample_in_constraint = 0;

776 s->flushed = 1;

777 }else if(!s->in_buffer_count){

778 return 0;

779 }

780 }else

781 fill_audiodata(in , (void*)in_arg);

782

783 fill_audiodata(out, out_arg);

784

785 if(s->resample){

786 int ret = swr_convert_internal(s, out, out_count, in, in_count);

787 if(ret>0 && !s->drop_output)

788 s->outpts += ret * (int64_t)s->in_sample_rate;

789 return ret;

790 }else{

791 AudioData tmp= *in;

792 int ret2=0;

793 int ret, size;

794 size = FFMIN(out_count, s->in_buffer_count);

795 if(size){

796 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

797 ret= swr_convert_internal(s, out, size, &tmp, size);

798 if(ret<0)

799 return ret;

800 ret2= ret;

801 s->in_buffer_count -= ret;

802 s->in_buffer_index += ret;

803 buf_set(out, out, ret);

804 out_count -= ret;

805 if(!s->in_buffer_count)

806 s->in_buffer_index = 0;

807 }

808

809 if(in_count){

810 size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;

811

812 if(in_count > out_count) { //FIXME move after swr_convert_internal

813 if( size > s->in_buffer.count

814 && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){

815 buf_set(&tmp, &s->in_buffer, s->in_buffer_index);

816 copy(&s->in_buffer, &tmp, s->in_buffer_count);

817 s->in_buffer_index=0;

818 }else

819 if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)

820 return ret;

821 }

822

823 if(out_count){

824 size = FFMIN(in_count, out_count);

825 ret= swr_convert_internal(s, out, size, in, size);

826 if(ret<0)

827 return ret;

828 buf_set(in, in, ret);

829 in_count -= ret;

830 ret2 += ret;

831 }

832 if(in_count){

833 buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);

834 copy(&tmp, in, in_count);

835 s->in_buffer_count += in_count;

836 }

837 }

838 if(ret2>0 && !s->drop_output)

839 s->outpts += ret2 * (int64_t)s->in_sample_rate;

840 return ret2;

841 }

842 }

843

844 int swr_drop_output(struct SwrContext *s, int count){

845 s->drop_output += count;

846

847 if(s->drop_output <= 0)

848 return 0;

849

850 av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);

851 return swr_convert(s, NULL, s->drop_output, NULL, 0);

852 }

853

854 int swr_inject_silence(struct SwrContext *s, int count){

855 int ret, i;

856 uint8_t *tmp_arg[SWR_CH_MAX];

857

858 if(count <= 0)

859 return 0;

860

861 #define MAX_SILENCE_STEP 16384

862 while (count > MAX_SILENCE_STEP) {

863 if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)

864 return ret;

865 count -= MAX_SILENCE_STEP;

866 }

867

868 if((ret=swri_realloc_audio(&s->silence, count))<0)

869 return ret;

870

871 if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {

872 memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);

873 } else

874 memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);

875

876 reversefill_audiodata(&s->silence, tmp_arg);

877 av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);

878 ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);

879 return ret;

880 }

881

882 int64_t swr_get_delay(struct SwrContext *s, int64_t base){

883 if (s->resampler && s->resample){

884 return s->resampler->get_delay(s, base);

885 }else{

886 return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;

887 }

888 }

889

890 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){

891 int ret;

892

893 if (!s || compensation_distance < 0)

894 return AVERROR(EINVAL);

895 if (!compensation_distance && sample_delta)

896 return AVERROR(EINVAL);

897 if (!s->resample) {

898 s->flags |= SWR_FLAG_RESAMPLE;

899 ret = swr_init(s);

900 if (ret < 0)

901 return ret;

902 }

903 if (!s->resampler->set_compensation){

904 return AVERROR(EINVAL);

905 }else{

906 return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);

907 }

908 }

909

910 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){

911 if(pts == INT64_MIN)

912 return s->outpts;

913

914 if (s->firstpts == AV_NOPTS_VALUE)

915 s->outpts = s->firstpts = pts;

916

917 if(s->min_compensation >= FLT_MAX) {

918 return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));

919 } else {

920 int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;

921 double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);

922

923 if(fabs(fdelta) > s->min_compensation) {

924 if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){

925 int ret;

926 if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);

927 else ret = swr_drop_output (s, -delta / s-> in_sample_rate);

928 if(ret<0){

929 av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);

930 }

931 } else if(s->soft_compensation_duration && s->max_soft_compensation) {

932 int duration = s->out_sample_rate * s->soft_compensation_duration;

933 double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1);

934 int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;

935 av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);

936 swr_set_compensation(s, comp, duration);

937 }

938 }

939

940 return s->outpts;

941 }

942 }

Generated on Sun Mar 23 2014 23:50:15 for FFmpeg by doxygen 1.8.2