henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | //TODO(hlundin): Reformat file to meet style guide. |
| 12 | |
| 13 | /* header includes */ |
| 14 | #include <stdio.h> |
| 15 | #include <stdlib.h> |
| 16 | #include <string.h> |
| 17 | #ifdef WIN32 |
| 18 | #include <winsock2.h> |
| 19 | #endif |
| 20 | #ifdef WEBRTC_LINUX |
| 21 | #include <netinet/in.h> |
| 22 | #endif |
| 23 | |
pbos@webrtc.org | 12dc1a3 | 2013-08-05 16:22:53 +0000 | [diff] [blame] | 24 | #include <assert.h> |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 25 | |
| 26 | #include "webrtc/typedefs.h" |
| 27 | // needed for NetEqDecoder |
kwiberg@webrtc.org | e04a93b | 2014-12-09 10:12:53 +0000 | [diff] [blame] | 28 | #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" |
henrik.lundin@webrtc.org | 9c55f0f | 2014-06-09 08:10:28 +0000 | [diff] [blame] | 29 | #include "webrtc/modules/audio_coding/neteq/interface/neteq.h" |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 30 | |
| 31 | /************************/ |
| 32 | /* Define payload types */ |
| 33 | /************************/ |
| 34 | |
| 35 | #include "PayloadTypes.h" |
| 36 | |
| 37 | |
| 38 | |
| 39 | /*********************/ |
| 40 | /* Misc. definitions */ |
| 41 | /*********************/ |
| 42 | |
| 43 | #define STOPSENDTIME 3000 |
| 44 | #define RESTARTSENDTIME 0 //162500 |
| 45 | #define FIRSTLINELEN 40 |
| 46 | #define CHECK_NOT_NULL(a) if((a)==0){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);} |
| 47 | |
| 48 | //#define MULTIPLE_SAME_TIMESTAMP |
| 49 | #define REPEAT_PACKET_DISTANCE 17 |
| 50 | #define REPEAT_PACKET_COUNT 1 // number of extra packets to send |
| 51 | |
| 52 | //#define INSERT_OLD_PACKETS |
| 53 | #define OLD_PACKET 5 // how many seconds too old should the packet be? |
| 54 | |
| 55 | //#define TIMESTAMP_WRAPAROUND |
| 56 | |
| 57 | //#define RANDOM_DATA |
| 58 | //#define RANDOM_PAYLOAD_DATA |
| 59 | #define RANDOM_SEED 10 |
| 60 | |
| 61 | //#define INSERT_DTMF_PACKETS |
| 62 | //#define NO_DTMF_OVERDUB |
| 63 | #define DTMF_PACKET_INTERVAL 2000 |
| 64 | #define DTMF_DURATION 500 |
| 65 | |
| 66 | #define STEREO_MODE_FRAME 0 |
| 67 | #define STEREO_MODE_SAMPLE_1 1 //1 octet per sample |
| 68 | #define STEREO_MODE_SAMPLE_2 2 //2 octets per sample |
| 69 | |
| 70 | /*************************/ |
| 71 | /* Function declarations */ |
| 72 | /*************************/ |
| 73 | |
| 74 | void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed); |
| 75 | int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels); |
| 76 | void defineCodecs(webrtc::NetEqDecoder *usedCodec, int *noOfCodecs ); |
| 77 | int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels); |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 78 | int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels); |
| 79 | void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc); |
| 80 | int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen, |
| 81 | int seqNo, uint32_t ssrc); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 82 | int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration); |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 83 | void stereoDeInterleave(int16_t* audioSamples, int numSamples); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 84 | void stereoInterleave(unsigned char* data, int dataLen, int stride); |
| 85 | |
| 86 | /*********************/ |
| 87 | /* Codec definitions */ |
| 88 | /*********************/ |
| 89 | |
| 90 | #include "webrtc_vad.h" |
| 91 | |
| 92 | #if ((defined CODEC_PCM16B)||(defined NETEQ_ARBITRARY_CODEC)) |
| 93 | #include "pcm16b.h" |
| 94 | #endif |
| 95 | #ifdef CODEC_G711 |
| 96 | #include "g711_interface.h" |
| 97 | #endif |
| 98 | #ifdef CODEC_G729 |
| 99 | #include "G729Interface.h" |
| 100 | #endif |
| 101 | #ifdef CODEC_G729_1 |
| 102 | #include "G729_1Interface.h" |
| 103 | #endif |
| 104 | #ifdef CODEC_AMR |
| 105 | #include "AMRInterface.h" |
| 106 | #include "AMRCreation.h" |
| 107 | #endif |
| 108 | #ifdef CODEC_AMRWB |
| 109 | #include "AMRWBInterface.h" |
| 110 | #include "AMRWBCreation.h" |
| 111 | #endif |
| 112 | #ifdef CODEC_ILBC |
| 113 | #include "ilbc.h" |
| 114 | #endif |
| 115 | #if (defined CODEC_ISAC || defined CODEC_ISAC_SWB) |
| 116 | #include "isac.h" |
| 117 | #endif |
| 118 | #ifdef NETEQ_ISACFIX_CODEC |
| 119 | #include "isacfix.h" |
| 120 | #ifdef CODEC_ISAC |
| 121 | #error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp |
| 122 | #endif |
| 123 | #endif |
| 124 | #ifdef CODEC_G722 |
| 125 | #include "g722_interface.h" |
| 126 | #endif |
| 127 | #ifdef CODEC_G722_1_24 |
| 128 | #include "G722_1Interface.h" |
| 129 | #endif |
| 130 | #ifdef CODEC_G722_1_32 |
| 131 | #include "G722_1Interface.h" |
| 132 | #endif |
| 133 | #ifdef CODEC_G722_1_16 |
| 134 | #include "G722_1Interface.h" |
| 135 | #endif |
| 136 | #ifdef CODEC_G722_1C_24 |
| 137 | #include "G722_1Interface.h" |
| 138 | #endif |
| 139 | #ifdef CODEC_G722_1C_32 |
| 140 | #include "G722_1Interface.h" |
| 141 | #endif |
| 142 | #ifdef CODEC_G722_1C_48 |
| 143 | #include "G722_1Interface.h" |
| 144 | #endif |
| 145 | #ifdef CODEC_G726 |
| 146 | #include "G726Creation.h" |
| 147 | #include "G726Interface.h" |
| 148 | #endif |
| 149 | #ifdef CODEC_GSMFR |
| 150 | #include "GSMFRInterface.h" |
| 151 | #include "GSMFRCreation.h" |
| 152 | #endif |
| 153 | #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| 154 | defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| 155 | #include "webrtc_cng.h" |
| 156 | #endif |
| 157 | #if ((defined CODEC_SPEEX_8)||(defined CODEC_SPEEX_16)) |
| 158 | #include "SpeexInterface.h" |
| 159 | #endif |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 160 | |
| 161 | /***********************************/ |
| 162 | /* Global codec instance variables */ |
| 163 | /***********************************/ |
| 164 | |
| 165 | WebRtcVadInst *VAD_inst[2]; |
| 166 | |
| 167 | #ifdef CODEC_G722 |
| 168 | G722EncInst *g722EncState[2]; |
| 169 | #endif |
| 170 | |
| 171 | #ifdef CODEC_G722_1_24 |
| 172 | G722_1_24_encinst_t *G722_1_24enc_inst[2]; |
| 173 | #endif |
| 174 | #ifdef CODEC_G722_1_32 |
| 175 | G722_1_32_encinst_t *G722_1_32enc_inst[2]; |
| 176 | #endif |
| 177 | #ifdef CODEC_G722_1_16 |
| 178 | G722_1_16_encinst_t *G722_1_16enc_inst[2]; |
| 179 | #endif |
| 180 | #ifdef CODEC_G722_1C_24 |
| 181 | G722_1C_24_encinst_t *G722_1C_24enc_inst[2]; |
| 182 | #endif |
| 183 | #ifdef CODEC_G722_1C_32 |
| 184 | G722_1C_32_encinst_t *G722_1C_32enc_inst[2]; |
| 185 | #endif |
| 186 | #ifdef CODEC_G722_1C_48 |
| 187 | G722_1C_48_encinst_t *G722_1C_48enc_inst[2]; |
| 188 | #endif |
| 189 | #ifdef CODEC_G726 |
| 190 | G726_encinst_t *G726enc_inst[2]; |
| 191 | #endif |
| 192 | #ifdef CODEC_G729 |
| 193 | G729_encinst_t *G729enc_inst[2]; |
| 194 | #endif |
| 195 | #ifdef CODEC_G729_1 |
| 196 | G729_1_inst_t *G729_1_inst[2]; |
| 197 | #endif |
| 198 | #ifdef CODEC_AMR |
| 199 | AMR_encinst_t *AMRenc_inst[2]; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 200 | int16_t AMR_bitrate; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 201 | #endif |
| 202 | #ifdef CODEC_AMRWB |
| 203 | AMRWB_encinst_t *AMRWBenc_inst[2]; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 204 | int16_t AMRWB_bitrate; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 205 | #endif |
| 206 | #ifdef CODEC_ILBC |
pbos@webrtc.org | e728ee0 | 2014-12-17 13:43:55 +0000 | [diff] [blame] | 207 | IlbcEncoderInstance *iLBCenc_inst[2]; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 208 | #endif |
| 209 | #ifdef CODEC_ISAC |
| 210 | ISACStruct *ISAC_inst[2]; |
| 211 | #endif |
| 212 | #ifdef NETEQ_ISACFIX_CODEC |
| 213 | ISACFIX_MainStruct *ISAC_inst[2]; |
| 214 | #endif |
| 215 | #ifdef CODEC_ISAC_SWB |
| 216 | ISACStruct *ISACSWB_inst[2]; |
| 217 | #endif |
| 218 | #ifdef CODEC_GSMFR |
| 219 | GSMFR_encinst_t *GSMFRenc_inst[2]; |
| 220 | #endif |
| 221 | #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| 222 | defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| 223 | CNG_enc_inst *CNGenc_inst[2]; |
| 224 | #endif |
| 225 | #ifdef CODEC_SPEEX_8 |
| 226 | SPEEX_encinst_t *SPEEX8enc_inst[2]; |
| 227 | #endif |
| 228 | #ifdef CODEC_SPEEX_16 |
| 229 | SPEEX_encinst_t *SPEEX16enc_inst[2]; |
| 230 | #endif |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 231 | |
| 232 | int main(int argc, char* argv[]) |
| 233 | { |
| 234 | int packet_size, fs; |
| 235 | webrtc::NetEqDecoder usedCodec; |
| 236 | int payloadType; |
| 237 | int bitrate = 0; |
| 238 | int useVAD, vad; |
| 239 | int useRed=0; |
| 240 | int len, enc_len; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 241 | int16_t org_data[4000]; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 242 | unsigned char rtp_data[8000]; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 243 | int16_t seqNo=0xFFF; |
| 244 | uint32_t ssrc=1235412312; |
| 245 | uint32_t timestamp=0xAC1245; |
| 246 | uint16_t length, plen; |
| 247 | uint32_t offset; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 248 | double sendtime = 0; |
| 249 | int red_PT[2] = {0}; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 250 | uint32_t red_TS[2] = {0}; |
| 251 | uint16_t red_len[2] = {0}; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 252 | int RTPheaderLen=12; |
kwiberg@webrtc.org | 1172988 | 2014-10-13 10:53:42 +0000 | [diff] [blame] | 253 | uint8_t red_data[8000]; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 254 | #ifdef INSERT_OLD_PACKETS |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 255 | uint16_t old_length, old_plen; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 256 | int old_enc_len; |
| 257 | int first_old_packet=1; |
| 258 | unsigned char old_rtp_data[8000]; |
| 259 | int packet_age=0; |
| 260 | #endif |
| 261 | #ifdef INSERT_DTMF_PACKETS |
| 262 | int NTone = 1; |
| 263 | int DTMFfirst = 1; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 264 | uint32_t DTMFtimestamp; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 265 | bool dtmfSent = false; |
| 266 | #endif |
| 267 | bool usingStereo = false; |
| 268 | int stereoMode = 0; |
| 269 | int numChannels = 1; |
| 270 | |
| 271 | /* check number of parameters */ |
| 272 | if ((argc != 6) && (argc != 7)) { |
| 273 | /* print help text and exit */ |
| 274 | printf("Application to encode speech into an RTP stream.\n"); |
| 275 | printf("The program reads a PCM file and encodes is using the specified codec.\n"); |
| 276 | printf("The coded speech is packetized in RTP packest and written to the output file.\n"); |
| 277 | printf("The format of the RTP stream file is simlilar to that of rtpplay,\n"); |
| 278 | printf("but with the receive time euqal to 0 for all packets.\n"); |
| 279 | printf("Usage:\n\n"); |
| 280 | printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]); |
| 281 | printf("where:\n"); |
| 282 | |
| 283 | printf("PCMfile : PCM speech input file\n\n"); |
| 284 | |
| 285 | printf("RTPfile : RTP stream output file\n\n"); |
| 286 | |
| 287 | printf("frameLen : 80...960... Number of samples per packet (limit depends on codec)\n\n"); |
| 288 | |
| 289 | printf("codecName\n"); |
| 290 | #ifdef CODEC_PCM16B |
| 291 | printf(" : pcm16b 16 bit PCM (8kHz)\n"); |
| 292 | #endif |
| 293 | #ifdef CODEC_PCM16B_WB |
| 294 | printf(" : pcm16b_wb 16 bit PCM (16kHz)\n"); |
| 295 | #endif |
| 296 | #ifdef CODEC_PCM16B_32KHZ |
| 297 | printf(" : pcm16b_swb32 16 bit PCM (32kHz)\n"); |
| 298 | #endif |
| 299 | #ifdef CODEC_PCM16B_48KHZ |
| 300 | printf(" : pcm16b_swb48 16 bit PCM (48kHz)\n"); |
| 301 | #endif |
| 302 | #ifdef CODEC_G711 |
| 303 | printf(" : pcma g711 A-law (8kHz)\n"); |
| 304 | #endif |
| 305 | #ifdef CODEC_G711 |
| 306 | printf(" : pcmu g711 u-law (8kHz)\n"); |
| 307 | #endif |
| 308 | #ifdef CODEC_G729 |
| 309 | printf(" : g729 G729 (8kHz and 8kbps) CELP (One-Three frame(s)/packet)\n"); |
| 310 | #endif |
| 311 | #ifdef CODEC_G729_1 |
| 312 | printf(" : g729.1 G729.1 (16kHz) variable rate (8--32 kbps)\n"); |
| 313 | #endif |
| 314 | #ifdef CODEC_G722_1_16 |
| 315 | printf(" : g722.1_16 G722.1 coder (16kHz) (g722.1 with 16kbps)\n"); |
| 316 | #endif |
| 317 | #ifdef CODEC_G722_1_24 |
| 318 | printf(" : g722.1_24 G722.1 coder (16kHz) (the 24kbps version)\n"); |
| 319 | #endif |
| 320 | #ifdef CODEC_G722_1_32 |
| 321 | printf(" : g722.1_32 G722.1 coder (16kHz) (the 32kbps version)\n"); |
| 322 | #endif |
| 323 | #ifdef CODEC_G722_1C_24 |
| 324 | printf(" : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps version)\n"); |
| 325 | #endif |
| 326 | #ifdef CODEC_G722_1C_32 |
| 327 | printf(" : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps version)\n"); |
| 328 | #endif |
| 329 | #ifdef CODEC_G722_1C_48 |
| 330 | printf(" : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps)\n"); |
| 331 | #endif |
| 332 | |
| 333 | #ifdef CODEC_G726 |
| 334 | printf(" : g726_16 G726 coder (8kHz) 16kbps\n"); |
| 335 | printf(" : g726_24 G726 coder (8kHz) 24kbps\n"); |
| 336 | printf(" : g726_32 G726 coder (8kHz) 32kbps\n"); |
| 337 | printf(" : g726_40 G726 coder (8kHz) 40kbps\n"); |
| 338 | #endif |
| 339 | #ifdef CODEC_AMR |
| 340 | printf(" : AMRXk Adaptive Multi Rate CELP codec (8kHz)\n"); |
| 341 | printf(" X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 or 12.2\n"); |
| 342 | #endif |
| 343 | #ifdef CODEC_AMRWB |
| 344 | printf(" : AMRwbXk Adaptive Multi Rate Wideband CELP codec (16kHz)\n"); |
| 345 | printf(" X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n"); |
| 346 | #endif |
| 347 | #ifdef CODEC_ILBC |
| 348 | printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n"); |
| 349 | #endif |
| 350 | #ifdef CODEC_ISAC |
| 351 | printf(" : isac iSAC (16kHz and 32.0 kbps). To set rate specify a rate parameter as last parameter\n"); |
| 352 | #endif |
| 353 | #ifdef CODEC_ISAC_SWB |
| 354 | printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n"); |
| 355 | #endif |
| 356 | #ifdef CODEC_GSMFR |
| 357 | printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n"); |
| 358 | #endif |
| 359 | #ifdef CODEC_G722 |
| 360 | printf(" : g722 g722 coder (16kHz) (the 64kbps version)\n"); |
| 361 | #endif |
| 362 | #ifdef CODEC_SPEEX_8 |
| 363 | printf(" : speex8 speex coder (8 kHz)\n"); |
| 364 | #endif |
| 365 | #ifdef CODEC_SPEEX_16 |
| 366 | printf(" : speex16 speex coder (16 kHz)\n"); |
| 367 | #endif |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 368 | #ifdef CODEC_RED |
| 369 | #ifdef CODEC_G711 |
| 370 | printf(" : red_pcm Redundancy RTP packet with 2*G711A frames\n"); |
| 371 | #endif |
| 372 | #ifdef CODEC_ISAC |
| 373 | printf(" : red_isac Redundancy RTP packet with 2*iSAC frames\n"); |
| 374 | #endif |
| 375 | #endif |
| 376 | printf("\n"); |
| 377 | |
| 378 | #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| 379 | defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| 380 | printf("useVAD : 0 Voice Activity Detection is switched off\n"); |
| 381 | printf(" : 1 Voice Activity Detection is switched on\n\n"); |
| 382 | #else |
| 383 | printf("useVAD : 0 Voice Activity Detection switched off (on not supported)\n\n"); |
| 384 | #endif |
| 385 | printf("bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n"); |
| 386 | |
| 387 | return(0); |
| 388 | } |
| 389 | |
| 390 | FILE* in_file=fopen(argv[1],"rb"); |
| 391 | CHECK_NOT_NULL(in_file); |
| 392 | printf("Input file: %s\n",argv[1]); |
| 393 | FILE* out_file=fopen(argv[2],"wb"); |
| 394 | CHECK_NOT_NULL(out_file); |
| 395 | printf("Output file: %s\n\n",argv[2]); |
| 396 | packet_size=atoi(argv[3]); |
| 397 | CHECK_NOT_NULL(packet_size); |
| 398 | printf("Packet size: %i\n",packet_size); |
| 399 | |
| 400 | // check for stereo |
| 401 | if(argv[4][strlen(argv[4])-1] == '*') { |
| 402 | // use stereo |
| 403 | usingStereo = true; |
| 404 | numChannels = 2; |
| 405 | argv[4][strlen(argv[4])-1] = '\0'; |
| 406 | } |
| 407 | |
| 408 | NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed); |
| 409 | |
| 410 | if(useRed) { |
| 411 | RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant payload, except last one which is 1 byte */ |
| 412 | } |
| 413 | |
| 414 | useVAD=atoi(argv[5]); |
| 415 | #if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| 416 | defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| 417 | if (useVAD!=0) { |
| 418 | printf("Error: this simulation does not support VAD/DTX/CNG\n"); |
| 419 | } |
| 420 | #endif |
| 421 | |
| 422 | // check stereo type |
| 423 | if(usingStereo) |
| 424 | { |
| 425 | switch(usedCodec) |
| 426 | { |
| 427 | // sample based codecs |
| 428 | case webrtc::kDecoderPCMu: |
| 429 | case webrtc::kDecoderPCMa: |
| 430 | case webrtc::kDecoderG722: |
| 431 | { |
| 432 | // 1 octet per sample |
| 433 | stereoMode = STEREO_MODE_SAMPLE_1; |
| 434 | break; |
| 435 | } |
| 436 | case webrtc::kDecoderPCM16B: |
| 437 | case webrtc::kDecoderPCM16Bwb: |
| 438 | case webrtc::kDecoderPCM16Bswb32kHz: |
| 439 | case webrtc::kDecoderPCM16Bswb48kHz: |
| 440 | { |
| 441 | // 2 octets per sample |
| 442 | stereoMode = STEREO_MODE_SAMPLE_2; |
| 443 | break; |
| 444 | } |
| 445 | |
| 446 | // fixed-rate frame codecs (with internal VAD) |
| 447 | default: |
| 448 | { |
| 449 | printf("Cannot use codec %s as stereo codec\n", argv[4]); |
| 450 | exit(0); |
| 451 | } |
| 452 | } |
| 453 | } |
| 454 | |
| 455 | if ((usedCodec == webrtc::kDecoderISAC) || (usedCodec == webrtc::kDecoderISACswb)) |
| 456 | { |
| 457 | if (argc != 7) |
| 458 | { |
| 459 | if (usedCodec == webrtc::kDecoderISAC) |
| 460 | { |
| 461 | bitrate = 32000; |
| 462 | printf( |
| 463 | "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n"); |
| 464 | } |
| 465 | else // (usedCodec==webrtc::kDecoderISACswb) |
| 466 | { |
| 467 | bitrate = 56000; |
| 468 | printf( |
| 469 | "Running iSAC at default bitrate of 56000 bps (to specify explicitly add the bps as last parameter)\n"); |
| 470 | } |
| 471 | } |
| 472 | else |
| 473 | { |
| 474 | bitrate = atoi(argv[6]); |
| 475 | if (usedCodec == webrtc::kDecoderISAC) |
| 476 | { |
| 477 | if ((bitrate < 10000) || (bitrate > 32000)) |
| 478 | { |
| 479 | printf( |
| 480 | "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is invalid)\n", |
| 481 | bitrate); |
| 482 | exit(0); |
| 483 | } |
| 484 | printf("Running iSAC at bitrate of %i bps\n", bitrate); |
| 485 | } |
| 486 | else // (usedCodec==webrtc::kDecoderISACswb) |
| 487 | { |
| 488 | if ((bitrate < 32000) || (bitrate > 56000)) |
| 489 | { |
| 490 | printf( |
| 491 | "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n", |
| 492 | bitrate); |
| 493 | exit(0); |
| 494 | } |
| 495 | } |
| 496 | } |
| 497 | } |
| 498 | else |
| 499 | { |
| 500 | if (argc == 7) |
| 501 | { |
| 502 | printf( |
| 503 | "Error: Bitrate parameter can only be specified for iSAC, G.723, and G.729.1\n"); |
| 504 | exit(0); |
| 505 | } |
| 506 | } |
| 507 | |
| 508 | if(useRed) { |
| 509 | printf("Redundancy engaged. "); |
| 510 | } |
| 511 | printf("Used codec: %i\n",usedCodec); |
| 512 | printf("Payload type: %i\n",payloadType); |
| 513 | |
| 514 | NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels); |
| 515 | |
| 516 | /* write file header */ |
| 517 | //fprintf(out_file, "#!RTPencode%s\n", "1.0"); |
| 518 | fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 519 | uint32_t dummy_variable = 0; // should be converted to network endian format, but does not matter when 0 |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 520 | if (fwrite(&dummy_variable, 4, 1, out_file) != 1) { |
| 521 | return -1; |
| 522 | } |
| 523 | if (fwrite(&dummy_variable, 4, 1, out_file) != 1) { |
| 524 | return -1; |
| 525 | } |
| 526 | if (fwrite(&dummy_variable, 4, 1, out_file) != 1) { |
| 527 | return -1; |
| 528 | } |
| 529 | if (fwrite(&dummy_variable, 2, 1, out_file) != 1) { |
| 530 | return -1; |
| 531 | } |
| 532 | if (fwrite(&dummy_variable, 2, 1, out_file) != 1) { |
| 533 | return -1; |
| 534 | } |
| 535 | |
| 536 | #ifdef TIMESTAMP_WRAPAROUND |
| 537 | timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */ |
| 538 | #endif |
| 539 | #if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA) |
| 540 | srand(RANDOM_SEED); |
| 541 | #endif |
| 542 | |
| 543 | /* if redundancy is used, the first redundant payload is zero length */ |
| 544 | red_len[0] = 0; |
| 545 | |
| 546 | /* read first frame */ |
| 547 | len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels; |
| 548 | |
| 549 | /* de-interleave if stereo */ |
| 550 | if ( usingStereo ) |
| 551 | { |
| 552 | stereoDeInterleave(org_data, len * numChannels); |
| 553 | } |
| 554 | |
| 555 | while (len==packet_size) { |
| 556 | |
| 557 | #ifdef INSERT_DTMF_PACKETS |
| 558 | dtmfSent = false; |
| 559 | |
| 560 | if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) { |
| 561 | if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) { |
| 562 | // tone has not ended |
| 563 | if (DTMFfirst==1) { |
| 564 | DTMFtimestamp = timestamp; // save this timestamp |
| 565 | DTMFfirst=0; |
| 566 | } |
| 567 | makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc); |
| 568 | enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len); |
| 569 | } |
| 570 | else { |
| 571 | // tone has ended |
| 572 | makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc); |
| 573 | enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4, DTMF_DURATION*(fs/1000)); |
| 574 | NTone++; |
| 575 | DTMFfirst=1; |
| 576 | } |
| 577 | |
| 578 | /* write RTP packet to file */ |
| 579 | length = htons(12 + enc_len + 8); |
| 580 | plen = htons(12 + enc_len); |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 581 | offset = (uint32_t) sendtime; //(timestamp/(fs/1000)); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 582 | offset = htonl(offset); |
| 583 | if (fwrite(&length, 2, 1, out_file) != 1) { |
| 584 | return -1; |
| 585 | } |
| 586 | if (fwrite(&plen, 2, 1, out_file) != 1) { |
| 587 | return -1; |
| 588 | } |
| 589 | if (fwrite(&offset, 4, 1, out_file) != 1) { |
| 590 | return -1; |
| 591 | } |
| 592 | if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) { |
| 593 | return -1; |
| 594 | } |
| 595 | |
| 596 | dtmfSent = true; |
| 597 | } |
| 598 | #endif |
| 599 | |
| 600 | #ifdef NO_DTMF_OVERDUB |
| 601 | /* If DTMF is sent, we should not send any speech packets during the same time */ |
| 602 | if (dtmfSent) { |
| 603 | enc_len = 0; |
| 604 | } |
| 605 | else { |
| 606 | #endif |
| 607 | /* encode frame */ |
| 608 | enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels); |
| 609 | if (enc_len==-1) { |
| 610 | printf("Error encoding frame\n"); |
| 611 | exit(0); |
| 612 | } |
| 613 | |
| 614 | if ( usingStereo && |
| 615 | stereoMode != STEREO_MODE_FRAME && |
| 616 | vad == 1 ) |
| 617 | { |
| 618 | // interleave the encoded payload for sample-based codecs (not for CNG) |
| 619 | stereoInterleave(&rtp_data[12], enc_len, stereoMode); |
| 620 | } |
| 621 | #ifdef NO_DTMF_OVERDUB |
| 622 | } |
| 623 | #endif |
| 624 | |
| 625 | if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) { |
| 626 | if(useRed) { |
| 627 | if(red_len[0] > 0) { |
| 628 | memmove(&rtp_data[RTPheaderLen+red_len[0]], &rtp_data[12], enc_len); |
| 629 | memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]); |
| 630 | |
| 631 | red_len[1] = enc_len; |
| 632 | red_TS[1] = timestamp; |
| 633 | if(vad) |
| 634 | red_PT[1] = payloadType; |
| 635 | else |
| 636 | red_PT[1] = NETEQ_CODEC_CN_PT; |
| 637 | |
| 638 | makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc); |
| 639 | |
| 640 | |
| 641 | enc_len += red_len[0] + RTPheaderLen - 12; |
| 642 | } |
| 643 | else { // do not use redundancy payload for this packet, i.e., only last payload |
| 644 | memmove(&rtp_data[RTPheaderLen-4], &rtp_data[12], enc_len); |
| 645 | //memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]); |
| 646 | |
| 647 | red_len[1] = enc_len; |
| 648 | red_TS[1] = timestamp; |
| 649 | if(vad) |
| 650 | red_PT[1] = payloadType; |
| 651 | else |
| 652 | red_PT[1] = NETEQ_CODEC_CN_PT; |
| 653 | |
| 654 | makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc); |
| 655 | |
| 656 | |
| 657 | enc_len += red_len[0] + RTPheaderLen - 4 - 12; // 4 is length of redundancy header (not used) |
| 658 | } |
| 659 | } |
| 660 | else { |
| 661 | |
| 662 | /* make RTP header */ |
| 663 | if (vad) // regular speech data |
| 664 | makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc); |
| 665 | else // CNG data |
| 666 | makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc); |
| 667 | |
| 668 | } |
| 669 | #ifdef MULTIPLE_SAME_TIMESTAMP |
| 670 | int mult_pack=0; |
| 671 | do { |
| 672 | #endif //MULTIPLE_SAME_TIMESTAMP |
| 673 | /* write RTP packet to file */ |
| 674 | length = htons(12 + enc_len + 8); |
| 675 | plen = htons(12 + enc_len); |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 676 | offset = (uint32_t) sendtime; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 677 | //(timestamp/(fs/1000)); |
| 678 | offset = htonl(offset); |
| 679 | if (fwrite(&length, 2, 1, out_file) != 1) { |
| 680 | return -1; |
| 681 | } |
| 682 | if (fwrite(&plen, 2, 1, out_file) != 1) { |
| 683 | return -1; |
| 684 | } |
| 685 | if (fwrite(&offset, 4, 1, out_file) != 1) { |
| 686 | return -1; |
| 687 | } |
| 688 | #ifdef RANDOM_DATA |
| 689 | for (int k=0; k<12+enc_len; k++) { |
| 690 | rtp_data[k] = rand() + rand(); |
| 691 | } |
| 692 | #endif |
| 693 | #ifdef RANDOM_PAYLOAD_DATA |
| 694 | for (int k=12; k<12+enc_len; k++) { |
| 695 | rtp_data[k] = rand() + rand(); |
| 696 | } |
| 697 | #endif |
| 698 | if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) { |
| 699 | return -1; |
| 700 | } |
| 701 | #ifdef MULTIPLE_SAME_TIMESTAMP |
| 702 | } while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) ); |
| 703 | #endif //MULTIPLE_SAME_TIMESTAMP |
| 704 | |
| 705 | #ifdef INSERT_OLD_PACKETS |
| 706 | if (packet_age >= OLD_PACKET*fs) { |
| 707 | if (!first_old_packet) { |
| 708 | // send the old packet |
| 709 | if (fwrite(&old_length, 2, 1, |
| 710 | out_file) != 1) { |
| 711 | return -1; |
| 712 | } |
| 713 | if (fwrite(&old_plen, 2, 1, |
| 714 | out_file) != 1) { |
| 715 | return -1; |
| 716 | } |
| 717 | if (fwrite(&offset, 4, 1, |
| 718 | out_file) != 1) { |
| 719 | return -1; |
| 720 | } |
| 721 | if (fwrite(old_rtp_data, 12 + old_enc_len, |
| 722 | 1, out_file) != 1) { |
| 723 | return -1; |
| 724 | } |
| 725 | } |
| 726 | // store current packet as old |
| 727 | old_length=length; |
| 728 | old_plen=plen; |
| 729 | memcpy(old_rtp_data,rtp_data,12+enc_len); |
| 730 | old_enc_len=enc_len; |
| 731 | first_old_packet=0; |
| 732 | packet_age=0; |
| 733 | |
| 734 | } |
| 735 | packet_age += packet_size; |
| 736 | #endif |
| 737 | |
| 738 | if(useRed) { |
| 739 | /* move data to redundancy store */ |
| 740 | #ifdef CODEC_ISAC |
| 741 | if(usedCodec==webrtc::kDecoderISAC) |
| 742 | { |
| 743 | assert(!usingStereo); // Cannot handle stereo yet |
kwiberg@webrtc.org | 1172988 | 2014-10-13 10:53:42 +0000 | [diff] [blame] | 744 | red_len[0] = |
| 745 | WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 746 | } |
| 747 | else |
| 748 | { |
| 749 | #endif |
| 750 | memcpy(red_data, &rtp_data[RTPheaderLen+red_len[0]], enc_len); |
| 751 | red_len[0]=red_len[1]; |
| 752 | #ifdef CODEC_ISAC |
| 753 | } |
| 754 | #endif |
| 755 | red_TS[0]=red_TS[1]; |
| 756 | red_PT[0]=red_PT[1]; |
| 757 | } |
| 758 | |
| 759 | } |
| 760 | |
| 761 | /* read next frame */ |
| 762 | len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels; |
| 763 | /* de-interleave if stereo */ |
| 764 | if ( usingStereo ) |
| 765 | { |
| 766 | stereoDeInterleave(org_data, len * numChannels); |
| 767 | } |
| 768 | |
| 769 | if (payloadType==NETEQ_CODEC_G722_PT) |
| 770 | timestamp+=len>>1; |
| 771 | else |
| 772 | timestamp+=len; |
| 773 | |
| 774 | sendtime += (double) len/(fs/1000); |
| 775 | } |
| 776 | |
| 777 | NetEQTest_free_coders(usedCodec, numChannels); |
| 778 | fclose(in_file); |
| 779 | fclose(out_file); |
| 780 | printf("Done!\n"); |
| 781 | |
| 782 | return(0); |
| 783 | } |
| 784 | |
| 785 | |
| 786 | |
| 787 | |
| 788 | /****************/ |
| 789 | /* Subfunctions */ |
| 790 | /****************/ |
| 791 | |
| 792 | void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed) { |
| 793 | |
| 794 | *bitrate = 0; /* Default bitrate setting */ |
| 795 | *useRed = 0; /* Default no redundancy */ |
| 796 | |
| 797 | if(!strcmp(name,"pcmu")){ |
| 798 | *codec=webrtc::kDecoderPCMu; |
| 799 | *PT=NETEQ_CODEC_PCMU_PT; |
| 800 | *fs=8000; |
| 801 | } |
| 802 | else if(!strcmp(name,"pcma")){ |
| 803 | *codec=webrtc::kDecoderPCMa; |
| 804 | *PT=NETEQ_CODEC_PCMA_PT; |
| 805 | *fs=8000; |
| 806 | } |
| 807 | else if(!strcmp(name,"pcm16b")){ |
| 808 | *codec=webrtc::kDecoderPCM16B; |
| 809 | *PT=NETEQ_CODEC_PCM16B_PT; |
| 810 | *fs=8000; |
| 811 | } |
| 812 | else if(!strcmp(name,"pcm16b_wb")){ |
| 813 | *codec=webrtc::kDecoderPCM16Bwb; |
| 814 | *PT=NETEQ_CODEC_PCM16B_WB_PT; |
| 815 | *fs=16000; |
| 816 | } |
| 817 | else if(!strcmp(name,"pcm16b_swb32")){ |
| 818 | *codec=webrtc::kDecoderPCM16Bswb32kHz; |
| 819 | *PT=NETEQ_CODEC_PCM16B_SWB32KHZ_PT; |
| 820 | *fs=32000; |
| 821 | } |
| 822 | else if(!strcmp(name,"pcm16b_swb48")){ |
| 823 | *codec=webrtc::kDecoderPCM16Bswb48kHz; |
| 824 | *PT=NETEQ_CODEC_PCM16B_SWB48KHZ_PT; |
| 825 | *fs=48000; |
| 826 | } |
| 827 | else if(!strcmp(name,"g722")){ |
| 828 | *codec=webrtc::kDecoderG722; |
| 829 | *PT=NETEQ_CODEC_G722_PT; |
| 830 | *fs=16000; |
| 831 | } |
| 832 | else if((!strcmp(name,"ilbc"))&&((frameLen%240==0)||(frameLen%160==0))){ |
| 833 | *fs=8000; |
| 834 | *codec=webrtc::kDecoderILBC; |
| 835 | *PT=NETEQ_CODEC_ILBC_PT; |
| 836 | } |
| 837 | else if(!strcmp(name,"isac")){ |
| 838 | *fs=16000; |
| 839 | *codec=webrtc::kDecoderISAC; |
| 840 | *PT=NETEQ_CODEC_ISAC_PT; |
| 841 | } |
| 842 | else if(!strcmp(name,"isacswb")){ |
| 843 | *fs=32000; |
| 844 | *codec=webrtc::kDecoderISACswb; |
| 845 | *PT=NETEQ_CODEC_ISACSWB_PT; |
| 846 | } |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 847 | else if(!strcmp(name,"red_pcm")){ |
| 848 | *codec=webrtc::kDecoderPCMa; |
| 849 | *PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */ |
| 850 | *fs=8000; |
| 851 | *useRed = 1; |
| 852 | } else if(!strcmp(name,"red_isac")){ |
| 853 | *codec=webrtc::kDecoderISAC; |
| 854 | *PT=NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */ |
| 855 | *fs=16000; |
| 856 | *useRed = 1; |
| 857 | } else { |
| 858 | printf("Error: Not a supported codec (%s)\n", name); |
| 859 | exit(0); |
| 860 | } |
| 861 | |
| 862 | } |
| 863 | |
| 864 | |
| 865 | |
| 866 | |
| 867 | int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels){ |
| 868 | |
| 869 | int ok=0; |
| 870 | |
| 871 | for (int k = 0; k < numChannels; k++) |
| 872 | { |
| 873 | ok=WebRtcVad_Create(&VAD_inst[k]); |
| 874 | if (ok!=0) { |
| 875 | printf("Error: Couldn't allocate memory for VAD instance\n"); |
| 876 | exit(0); |
| 877 | } |
| 878 | ok=WebRtcVad_Init(VAD_inst[k]); |
| 879 | if (ok==-1) { |
| 880 | printf("Error: Initialization of VAD struct failed\n"); |
| 881 | exit(0); |
| 882 | } |
| 883 | |
| 884 | |
| 885 | #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| 886 | defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| 887 | ok=WebRtcCng_CreateEnc(&CNGenc_inst[k]); |
| 888 | if (ok!=0) { |
| 889 | printf("Error: Couldn't allocate memory for CNG encoding instance\n"); |
| 890 | exit(0); |
| 891 | } |
| 892 | if(sampfreq <= 16000) { |
| 893 | ok=WebRtcCng_InitEnc(CNGenc_inst[k],sampfreq, 200, 5); |
| 894 | if (ok==-1) { |
| 895 | printf("Error: Initialization of CNG struct failed. Error code %d\n", |
| 896 | WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k])); |
| 897 | exit(0); |
| 898 | } |
| 899 | } |
| 900 | #endif |
| 901 | |
| 902 | switch (coder) { |
| 903 | #ifdef CODEC_PCM16B |
| 904 | case webrtc::kDecoderPCM16B : |
| 905 | #endif |
| 906 | #ifdef CODEC_PCM16B_WB |
| 907 | case webrtc::kDecoderPCM16Bwb : |
| 908 | #endif |
| 909 | #ifdef CODEC_PCM16B_32KHZ |
| 910 | case webrtc::kDecoderPCM16Bswb32kHz : |
| 911 | #endif |
| 912 | #ifdef CODEC_PCM16B_48KHZ |
| 913 | case webrtc::kDecoderPCM16Bswb48kHz : |
| 914 | #endif |
| 915 | #ifdef CODEC_G711 |
| 916 | case webrtc::kDecoderPCMu : |
| 917 | case webrtc::kDecoderPCMa : |
| 918 | #endif |
| 919 | // do nothing |
| 920 | break; |
| 921 | #ifdef CODEC_G729 |
| 922 | case webrtc::kDecoderG729: |
| 923 | if (sampfreq==8000) { |
| 924 | if ((enc_frameSize==80)||(enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==400)||(enc_frameSize==480)) { |
| 925 | ok=WebRtcG729_CreateEnc(&G729enc_inst[k]); |
| 926 | if (ok!=0) { |
| 927 | printf("Error: Couldn't allocate memory for G729 encoding instance\n"); |
| 928 | exit(0); |
| 929 | } |
| 930 | } else { |
| 931 | printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 ms!!\n\n"); |
| 932 | exit(0); |
| 933 | } |
| 934 | WebRtcG729_EncoderInit(G729enc_inst[k], vad); |
| 935 | if ((vad==1)&&(enc_frameSize!=80)) { |
| 936 | printf("\nError - This simulation only supports VAD for G729 at 10ms packets (not %dms)\n", (enc_frameSize>>3)); |
| 937 | } |
| 938 | } else { |
| 939 | printf("\nError - g729 is only developed for 8kHz \n"); |
| 940 | exit(0); |
| 941 | } |
| 942 | break; |
| 943 | #endif |
| 944 | #ifdef CODEC_G729_1 |
| 945 | case webrtc::kDecoderG729_1: |
| 946 | if (sampfreq==16000) { |
| 947 | if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960) |
| 948 | ) { |
| 949 | ok=WebRtcG7291_Create(&G729_1_inst[k]); |
| 950 | if (ok!=0) { |
| 951 | printf("Error: Couldn't allocate memory for G.729.1 codec instance\n"); |
| 952 | exit(0); |
| 953 | } |
| 954 | } else { |
| 955 | printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n"); |
| 956 | exit(0); |
| 957 | } |
| 958 | if (!(((bitrate >= 12000) && (bitrate <= 32000) && (bitrate%2000 == 0)) || (bitrate == 8000))) { |
| 959 | /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */ |
| 960 | printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in steps of 2000 bps\n"); |
| 961 | exit(0); |
| 962 | } |
| 963 | WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/, 0 /*flagG729mode*/); |
| 964 | } else { |
| 965 | printf("\nError - G.729.1 input is always 16 kHz \n"); |
| 966 | exit(0); |
| 967 | } |
| 968 | break; |
| 969 | #endif |
| 970 | #ifdef CODEC_SPEEX_8 |
| 971 | case webrtc::kDecoderSPEEX_8 : |
| 972 | if (sampfreq==8000) { |
| 973 | if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) { |
| 974 | ok=WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq); |
| 975 | if (ok!=0) { |
| 976 | printf("Error: Couldn't allocate memory for Speex encoding instance\n"); |
| 977 | exit(0); |
| 978 | } |
| 979 | } else { |
| 980 | printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n"); |
| 981 | exit(0); |
| 982 | } |
| 983 | if ((vad==1)&&(enc_frameSize!=160)) { |
| 984 | printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>3)); |
| 985 | vad=0; |
| 986 | } |
| 987 | ok=WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad); |
| 988 | if (ok!=0) exit(0); |
| 989 | } else { |
| 990 | printf("\nError - Speex8 called with sample frequency other than 8 kHz.\n\n"); |
| 991 | } |
| 992 | break; |
| 993 | #endif |
| 994 | #ifdef CODEC_SPEEX_16 |
| 995 | case webrtc::kDecoderSPEEX_16 : |
| 996 | if (sampfreq==16000) { |
| 997 | if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)) { |
| 998 | ok=WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq); |
| 999 | if (ok!=0) { |
| 1000 | printf("Error: Couldn't allocate memory for Speex encoding instance\n"); |
| 1001 | exit(0); |
| 1002 | } |
| 1003 | } else { |
| 1004 | printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n"); |
| 1005 | exit(0); |
| 1006 | } |
| 1007 | if ((vad==1)&&(enc_frameSize!=320)) { |
| 1008 | printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>4)); |
| 1009 | vad=0; |
| 1010 | } |
| 1011 | ok=WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad); |
| 1012 | if (ok!=0) exit(0); |
| 1013 | } else { |
| 1014 | printf("\nError - Speex16 called with sample frequency other than 16 kHz.\n\n"); |
| 1015 | } |
| 1016 | break; |
| 1017 | #endif |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1018 | |
| 1019 | #ifdef CODEC_G722_1_16 |
| 1020 | case webrtc::kDecoderG722_1_16 : |
| 1021 | if (sampfreq==16000) { |
| 1022 | ok=WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]); |
| 1023 | if (ok!=0) { |
| 1024 | printf("Error: Couldn't allocate memory for G.722.1 instance\n"); |
| 1025 | exit(0); |
| 1026 | } |
| 1027 | if (enc_frameSize==320) { |
| 1028 | } else { |
| 1029 | printf("\nError: G722.1 only supports 20 ms!!\n\n"); |
| 1030 | exit(0); |
| 1031 | } |
| 1032 | WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]); |
| 1033 | } else { |
| 1034 | printf("\nError - G722.1 is only developed for 16kHz \n"); |
| 1035 | exit(0); |
| 1036 | } |
| 1037 | break; |
| 1038 | #endif |
| 1039 | #ifdef CODEC_G722_1_24 |
| 1040 | case webrtc::kDecoderG722_1_24 : |
| 1041 | if (sampfreq==16000) { |
| 1042 | ok=WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]); |
| 1043 | if (ok!=0) { |
| 1044 | printf("Error: Couldn't allocate memory for G.722.1 instance\n"); |
| 1045 | exit(0); |
| 1046 | } |
| 1047 | if (enc_frameSize==320) { |
| 1048 | } else { |
| 1049 | printf("\nError: G722.1 only supports 20 ms!!\n\n"); |
| 1050 | exit(0); |
| 1051 | } |
| 1052 | WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]); |
| 1053 | } else { |
| 1054 | printf("\nError - G722.1 is only developed for 16kHz \n"); |
| 1055 | exit(0); |
| 1056 | } |
| 1057 | break; |
| 1058 | #endif |
| 1059 | #ifdef CODEC_G722_1_32 |
| 1060 | case webrtc::kDecoderG722_1_32 : |
| 1061 | if (sampfreq==16000) { |
| 1062 | ok=WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]); |
| 1063 | if (ok!=0) { |
| 1064 | printf("Error: Couldn't allocate memory for G.722.1 instance\n"); |
| 1065 | exit(0); |
| 1066 | } |
| 1067 | if (enc_frameSize==320) { |
| 1068 | } else { |
| 1069 | printf("\nError: G722.1 only supports 20 ms!!\n\n"); |
| 1070 | exit(0); |
| 1071 | } |
| 1072 | WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]); |
| 1073 | } else { |
| 1074 | printf("\nError - G722.1 is only developed for 16kHz \n"); |
| 1075 | exit(0); |
| 1076 | } |
| 1077 | break; |
| 1078 | #endif |
| 1079 | #ifdef CODEC_G722_1C_24 |
| 1080 | case webrtc::kDecoderG722_1C_24 : |
| 1081 | if (sampfreq==32000) { |
| 1082 | ok=WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]); |
| 1083 | if (ok!=0) { |
| 1084 | printf("Error: Couldn't allocate memory for G.722.1C instance\n"); |
| 1085 | exit(0); |
| 1086 | } |
| 1087 | if (enc_frameSize==640) { |
| 1088 | } else { |
| 1089 | printf("\nError: G722.1 C only supports 20 ms!!\n\n"); |
| 1090 | exit(0); |
| 1091 | } |
| 1092 | WebRtcG7221C_EncoderInit24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]); |
| 1093 | } else { |
| 1094 | printf("\nError - G722.1 C is only developed for 32kHz \n"); |
| 1095 | exit(0); |
| 1096 | } |
| 1097 | break; |
| 1098 | #endif |
| 1099 | #ifdef CODEC_G722_1C_32 |
| 1100 | case webrtc::kDecoderG722_1C_32 : |
| 1101 | if (sampfreq==32000) { |
| 1102 | ok=WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]); |
| 1103 | if (ok!=0) { |
| 1104 | printf("Error: Couldn't allocate memory for G.722.1C instance\n"); |
| 1105 | exit(0); |
| 1106 | } |
| 1107 | if (enc_frameSize==640) { |
| 1108 | } else { |
| 1109 | printf("\nError: G722.1 C only supports 20 ms!!\n\n"); |
| 1110 | exit(0); |
| 1111 | } |
| 1112 | WebRtcG7221C_EncoderInit32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]); |
| 1113 | } else { |
| 1114 | printf("\nError - G722.1 C is only developed for 32kHz \n"); |
| 1115 | exit(0); |
| 1116 | } |
| 1117 | break; |
| 1118 | #endif |
| 1119 | #ifdef CODEC_G722_1C_48 |
| 1120 | case webrtc::kDecoderG722_1C_48 : |
| 1121 | if (sampfreq==32000) { |
| 1122 | ok=WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]); |
| 1123 | if (ok!=0) { |
| 1124 | printf("Error: Couldn't allocate memory for G.722.1C instance\n"); |
| 1125 | exit(0); |
| 1126 | } |
| 1127 | if (enc_frameSize==640) { |
| 1128 | } else { |
| 1129 | printf("\nError: G722.1 C only supports 20 ms!!\n\n"); |
| 1130 | exit(0); |
| 1131 | } |
| 1132 | WebRtcG7221C_EncoderInit48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]); |
| 1133 | } else { |
| 1134 | printf("\nError - G722.1 C is only developed for 32kHz \n"); |
| 1135 | exit(0); |
| 1136 | } |
| 1137 | break; |
| 1138 | #endif |
| 1139 | #ifdef CODEC_G722 |
| 1140 | case webrtc::kDecoderG722 : |
| 1141 | if (sampfreq==16000) { |
| 1142 | if (enc_frameSize%2==0) { |
| 1143 | } else { |
| 1144 | printf("\nError - g722 frames must have an even number of enc_frameSize\n"); |
| 1145 | exit(0); |
| 1146 | } |
| 1147 | WebRtcG722_CreateEncoder(&g722EncState[k]); |
| 1148 | WebRtcG722_EncoderInit(g722EncState[k]); |
| 1149 | } else { |
| 1150 | printf("\nError - g722 is only developed for 16kHz \n"); |
| 1151 | exit(0); |
| 1152 | } |
| 1153 | break; |
| 1154 | #endif |
| 1155 | #ifdef CODEC_AMR |
| 1156 | case webrtc::kDecoderAMR : |
| 1157 | if (sampfreq==8000) { |
| 1158 | ok=WebRtcAmr_CreateEnc(&AMRenc_inst[k]); |
| 1159 | if (ok!=0) { |
| 1160 | printf("Error: Couldn't allocate memory for AMR encoding instance\n"); |
| 1161 | exit(0); |
| 1162 | }if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) { |
| 1163 | } else { |
| 1164 | printf("\nError - AMR must have a multiple of 160 enc_frameSize\n"); |
| 1165 | exit(0); |
| 1166 | } |
| 1167 | WebRtcAmr_EncoderInit(AMRenc_inst[k], vad); |
| 1168 | WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient); |
| 1169 | AMR_bitrate = bitrate; |
| 1170 | } else { |
| 1171 | printf("\nError - AMR is only developed for 8kHz \n"); |
| 1172 | exit(0); |
| 1173 | } |
| 1174 | break; |
| 1175 | #endif |
| 1176 | #ifdef CODEC_AMRWB |
| 1177 | case webrtc::kDecoderAMRWB : |
| 1178 | if (sampfreq==16000) { |
| 1179 | ok=WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]); |
| 1180 | if (ok!=0) { |
| 1181 | printf("Error: Couldn't allocate memory for AMRWB encoding instance\n"); |
| 1182 | exit(0); |
| 1183 | } |
| 1184 | if (((enc_frameSize/320)<0)||((enc_frameSize/320)>3)||((enc_frameSize%320)!=0)) { |
| 1185 | printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n"); |
| 1186 | exit(0); |
| 1187 | } |
| 1188 | WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad); |
| 1189 | if (bitrate==7000) { |
| 1190 | AMRWB_bitrate = AMRWB_MODE_7k; |
| 1191 | } else if (bitrate==9000) { |
| 1192 | AMRWB_bitrate = AMRWB_MODE_9k; |
| 1193 | } else if (bitrate==12000) { |
| 1194 | AMRWB_bitrate = AMRWB_MODE_12k; |
| 1195 | } else if (bitrate==14000) { |
| 1196 | AMRWB_bitrate = AMRWB_MODE_14k; |
| 1197 | } else if (bitrate==16000) { |
| 1198 | AMRWB_bitrate = AMRWB_MODE_16k; |
| 1199 | } else if (bitrate==18000) { |
| 1200 | AMRWB_bitrate = AMRWB_MODE_18k; |
| 1201 | } else if (bitrate==20000) { |
| 1202 | AMRWB_bitrate = AMRWB_MODE_20k; |
| 1203 | } else if (bitrate==23000) { |
| 1204 | AMRWB_bitrate = AMRWB_MODE_23k; |
| 1205 | } else if (bitrate==24000) { |
| 1206 | AMRWB_bitrate = AMRWB_MODE_24k; |
| 1207 | } |
| 1208 | WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient); |
| 1209 | |
| 1210 | } else { |
| 1211 | printf("\nError - AMRwb is only developed for 16kHz \n"); |
| 1212 | exit(0); |
| 1213 | } |
| 1214 | break; |
| 1215 | #endif |
| 1216 | #ifdef CODEC_ILBC |
| 1217 | case webrtc::kDecoderILBC : |
| 1218 | if (sampfreq==8000) { |
| 1219 | ok=WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]); |
| 1220 | if (ok!=0) { |
| 1221 | printf("Error: Couldn't allocate memory for iLBC encoding instance\n"); |
| 1222 | exit(0); |
| 1223 | } |
| 1224 | if ((enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==480)) { |
| 1225 | } else { |
| 1226 | printf("\nError - iLBC only supports 160, 240, 320 and 480 enc_frameSize (20, 30, 40 and 60 ms)\n"); |
| 1227 | exit(0); |
| 1228 | } |
| 1229 | if ((enc_frameSize==160)||(enc_frameSize==320)) { |
| 1230 | /* 20 ms version */ |
| 1231 | WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20); |
| 1232 | } else { |
| 1233 | /* 30 ms version */ |
| 1234 | WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30); |
| 1235 | } |
| 1236 | } else { |
| 1237 | printf("\nError - iLBC is only developed for 8kHz \n"); |
| 1238 | exit(0); |
| 1239 | } |
| 1240 | break; |
| 1241 | #endif |
| 1242 | #ifdef CODEC_ISAC |
| 1243 | case webrtc::kDecoderISAC: |
| 1244 | if (sampfreq==16000) { |
| 1245 | ok=WebRtcIsac_Create(&ISAC_inst[k]); |
| 1246 | if (ok!=0) { |
| 1247 | printf("Error: Couldn't allocate memory for iSAC instance\n"); |
| 1248 | exit(0); |
| 1249 | }if ((enc_frameSize==480)||(enc_frameSize==960)) { |
| 1250 | } else { |
| 1251 | printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n"); |
| 1252 | exit(0); |
| 1253 | } |
| 1254 | WebRtcIsac_EncoderInit(ISAC_inst[k],1); |
| 1255 | if ((bitrate<10000)||(bitrate>32000)) { |
| 1256 | printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate); |
| 1257 | exit(0); |
| 1258 | } |
| 1259 | WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize>>4); |
| 1260 | } else { |
| 1261 | printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n"); |
| 1262 | exit(0); |
| 1263 | } |
| 1264 | break; |
| 1265 | #endif |
| 1266 | #ifdef NETEQ_ISACFIX_CODEC |
| 1267 | case webrtc::kDecoderISAC: |
| 1268 | if (sampfreq==16000) { |
| 1269 | ok=WebRtcIsacfix_Create(&ISAC_inst[k]); |
| 1270 | if (ok!=0) { |
| 1271 | printf("Error: Couldn't allocate memory for iSAC instance\n"); |
| 1272 | exit(0); |
| 1273 | }if ((enc_frameSize==480)||(enc_frameSize==960)) { |
| 1274 | } else { |
| 1275 | printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n"); |
| 1276 | exit(0); |
| 1277 | } |
| 1278 | WebRtcIsacfix_EncoderInit(ISAC_inst[k],1); |
| 1279 | if ((bitrate<10000)||(bitrate>32000)) { |
| 1280 | printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate); |
| 1281 | exit(0); |
| 1282 | } |
| 1283 | WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize>>4); |
| 1284 | } else { |
| 1285 | printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n"); |
| 1286 | exit(0); |
| 1287 | } |
| 1288 | break; |
| 1289 | #endif |
| 1290 | #ifdef CODEC_ISAC_SWB |
| 1291 | case webrtc::kDecoderISACswb: |
| 1292 | if (sampfreq==32000) { |
| 1293 | ok=WebRtcIsac_Create(&ISACSWB_inst[k]); |
| 1294 | if (ok!=0) { |
| 1295 | printf("Error: Couldn't allocate memory for iSAC SWB instance\n"); |
| 1296 | exit(0); |
| 1297 | }if (enc_frameSize==960) { |
| 1298 | } else { |
| 1299 | printf("\nError - iSAC SWB only supports frameSize 30 ms\n"); |
| 1300 | exit(0); |
| 1301 | } |
| 1302 | ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000); |
| 1303 | if (ok!=0) { |
| 1304 | printf("Error: Couldn't set sample rate for iSAC SWB instance\n"); |
| 1305 | exit(0); |
| 1306 | } |
| 1307 | WebRtcIsac_EncoderInit(ISACSWB_inst[k],1); |
| 1308 | if ((bitrate<32000)||(bitrate>56000)) { |
| 1309 | printf("\nError - iSAC SWB bitrate has to be between 32000 and 56000 bps (not %i)\n", bitrate); |
| 1310 | exit(0); |
| 1311 | } |
| 1312 | WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize>>5); |
| 1313 | } else { |
| 1314 | printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n"); |
| 1315 | exit(0); |
| 1316 | } |
| 1317 | break; |
| 1318 | #endif |
| 1319 | #ifdef CODEC_GSMFR |
| 1320 | case webrtc::kDecoderGSMFR: |
| 1321 | if (sampfreq==8000) { |
| 1322 | ok=WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]); |
| 1323 | if (ok!=0) { |
| 1324 | printf("Error: Couldn't allocate memory for GSM FR encoding instance\n"); |
| 1325 | exit(0); |
| 1326 | } |
| 1327 | if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) { |
| 1328 | } else { |
| 1329 | printf("\nError - GSM FR must have a multiple of 160 enc_frameSize\n"); |
| 1330 | exit(0); |
| 1331 | } |
| 1332 | WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0); |
| 1333 | } else { |
| 1334 | printf("\nError - GSM FR is only developed for 8kHz \n"); |
| 1335 | exit(0); |
| 1336 | } |
| 1337 | break; |
| 1338 | #endif |
| 1339 | default : |
| 1340 | printf("Error: unknown codec in call to NetEQTest_init_coders.\n"); |
| 1341 | exit(0); |
| 1342 | break; |
| 1343 | } |
| 1344 | |
| 1345 | if (ok != 0) { |
| 1346 | return(ok); |
| 1347 | } |
pbos@webrtc.org | d900e8b | 2013-07-03 15:12:26 +0000 | [diff] [blame] | 1348 | } // end for |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1349 | |
| 1350 | return(0); |
| 1351 | } |
| 1352 | |
| 1353 | |
| 1354 | |
| 1355 | |
| 1356 | int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels) { |
| 1357 | |
| 1358 | for (int k = 0; k < numChannels; k++) |
| 1359 | { |
| 1360 | WebRtcVad_Free(VAD_inst[k]); |
| 1361 | #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| 1362 | defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| 1363 | WebRtcCng_FreeEnc(CNGenc_inst[k]); |
| 1364 | #endif |
| 1365 | |
| 1366 | switch (coder) |
| 1367 | { |
| 1368 | #ifdef CODEC_PCM16B |
| 1369 | case webrtc::kDecoderPCM16B : |
| 1370 | #endif |
| 1371 | #ifdef CODEC_PCM16B_WB |
| 1372 | case webrtc::kDecoderPCM16Bwb : |
| 1373 | #endif |
| 1374 | #ifdef CODEC_PCM16B_32KHZ |
| 1375 | case webrtc::kDecoderPCM16Bswb32kHz : |
| 1376 | #endif |
| 1377 | #ifdef CODEC_PCM16B_48KHZ |
| 1378 | case webrtc::kDecoderPCM16Bswb48kHz : |
| 1379 | #endif |
| 1380 | #ifdef CODEC_G711 |
| 1381 | case webrtc::kDecoderPCMu : |
| 1382 | case webrtc::kDecoderPCMa : |
| 1383 | #endif |
| 1384 | // do nothing |
| 1385 | break; |
| 1386 | #ifdef CODEC_G729 |
| 1387 | case webrtc::kDecoderG729: |
| 1388 | WebRtcG729_FreeEnc(G729enc_inst[k]); |
| 1389 | break; |
| 1390 | #endif |
| 1391 | #ifdef CODEC_G729_1 |
| 1392 | case webrtc::kDecoderG729_1: |
| 1393 | WebRtcG7291_Free(G729_1_inst[k]); |
| 1394 | break; |
| 1395 | #endif |
| 1396 | #ifdef CODEC_SPEEX_8 |
| 1397 | case webrtc::kDecoderSPEEX_8 : |
| 1398 | WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]); |
| 1399 | break; |
| 1400 | #endif |
| 1401 | #ifdef CODEC_SPEEX_16 |
| 1402 | case webrtc::kDecoderSPEEX_16 : |
| 1403 | WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]); |
| 1404 | break; |
| 1405 | #endif |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1406 | |
| 1407 | #ifdef CODEC_G722_1_16 |
| 1408 | case webrtc::kDecoderG722_1_16 : |
| 1409 | WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]); |
| 1410 | break; |
| 1411 | #endif |
| 1412 | #ifdef CODEC_G722_1_24 |
| 1413 | case webrtc::kDecoderG722_1_24 : |
| 1414 | WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]); |
| 1415 | break; |
| 1416 | #endif |
| 1417 | #ifdef CODEC_G722_1_32 |
| 1418 | case webrtc::kDecoderG722_1_32 : |
| 1419 | WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]); |
| 1420 | break; |
| 1421 | #endif |
| 1422 | #ifdef CODEC_G722_1C_24 |
| 1423 | case webrtc::kDecoderG722_1C_24 : |
| 1424 | WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]); |
| 1425 | break; |
| 1426 | #endif |
| 1427 | #ifdef CODEC_G722_1C_32 |
| 1428 | case webrtc::kDecoderG722_1C_32 : |
| 1429 | WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]); |
| 1430 | break; |
| 1431 | #endif |
| 1432 | #ifdef CODEC_G722_1C_48 |
| 1433 | case webrtc::kDecoderG722_1C_48 : |
| 1434 | WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]); |
| 1435 | break; |
| 1436 | #endif |
| 1437 | #ifdef CODEC_G722 |
| 1438 | case webrtc::kDecoderG722 : |
| 1439 | WebRtcG722_FreeEncoder(g722EncState[k]); |
| 1440 | break; |
| 1441 | #endif |
| 1442 | #ifdef CODEC_AMR |
| 1443 | case webrtc::kDecoderAMR : |
| 1444 | WebRtcAmr_FreeEnc(AMRenc_inst[k]); |
| 1445 | break; |
| 1446 | #endif |
| 1447 | #ifdef CODEC_AMRWB |
| 1448 | case webrtc::kDecoderAMRWB : |
| 1449 | WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]); |
| 1450 | break; |
| 1451 | #endif |
| 1452 | #ifdef CODEC_ILBC |
| 1453 | case webrtc::kDecoderILBC : |
| 1454 | WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]); |
| 1455 | break; |
| 1456 | #endif |
| 1457 | #ifdef CODEC_ISAC |
| 1458 | case webrtc::kDecoderISAC: |
| 1459 | WebRtcIsac_Free(ISAC_inst[k]); |
| 1460 | break; |
| 1461 | #endif |
| 1462 | #ifdef NETEQ_ISACFIX_CODEC |
| 1463 | case webrtc::kDecoderISAC: |
| 1464 | WebRtcIsacfix_Free(ISAC_inst[k]); |
| 1465 | break; |
| 1466 | #endif |
| 1467 | #ifdef CODEC_ISAC_SWB |
| 1468 | case webrtc::kDecoderISACswb: |
| 1469 | WebRtcIsac_Free(ISACSWB_inst[k]); |
| 1470 | break; |
| 1471 | #endif |
| 1472 | #ifdef CODEC_GSMFR |
| 1473 | case webrtc::kDecoderGSMFR: |
| 1474 | WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]); |
| 1475 | break; |
| 1476 | #endif |
| 1477 | default : |
| 1478 | printf("Error: unknown codec in call to NetEQTest_init_coders.\n"); |
| 1479 | exit(0); |
| 1480 | break; |
| 1481 | } |
| 1482 | } |
| 1483 | |
| 1484 | return(0); |
| 1485 | } |
| 1486 | |
| 1487 | |
| 1488 | |
| 1489 | |
| 1490 | |
| 1491 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1492 | int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1493 | int * vad, int useVAD, int bitrate, int numChannels){ |
| 1494 | |
| 1495 | short cdlen = 0; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1496 | int16_t *tempdata; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1497 | static int first_cng=1; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1498 | int16_t tempLen; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1499 | |
| 1500 | *vad =1; |
| 1501 | |
| 1502 | // check VAD first |
| 1503 | if(useVAD) |
| 1504 | { |
| 1505 | *vad = 0; |
| 1506 | |
| 1507 | for (int k = 0; k < numChannels; k++) |
| 1508 | { |
| 1509 | tempLen = frameLen; |
| 1510 | tempdata = &indata[k*frameLen]; |
| 1511 | int localVad=0; |
| 1512 | /* Partition the signal and test each chunk for VAD. |
| 1513 | All chunks must be VAD=0 to produce a total VAD=0. */ |
| 1514 | while (tempLen >= 10*sampleRate/1000) { |
| 1515 | if ((tempLen % 30*sampleRate/1000) == 0) { // tempLen is multiple of 30ms |
| 1516 | localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 30*sampleRate/1000); |
| 1517 | tempdata += 30*sampleRate/1000; |
| 1518 | tempLen -= 30*sampleRate/1000; |
| 1519 | } |
| 1520 | else if (tempLen >= 20*sampleRate/1000) { // tempLen >= 20ms |
| 1521 | localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 20*sampleRate/1000); |
| 1522 | tempdata += 20*sampleRate/1000; |
| 1523 | tempLen -= 20*sampleRate/1000; |
| 1524 | } |
| 1525 | else { // use 10ms |
| 1526 | localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 10*sampleRate/1000); |
| 1527 | tempdata += 10*sampleRate/1000; |
| 1528 | tempLen -= 10*sampleRate/1000; |
| 1529 | } |
| 1530 | } |
| 1531 | |
| 1532 | // aggregate all VAD decisions over all channels |
| 1533 | *vad |= localVad; |
| 1534 | } |
| 1535 | |
| 1536 | if(!*vad){ |
| 1537 | // all channels are silent |
| 1538 | cdlen = 0; |
| 1539 | for (int k = 0; k < numChannels; k++) |
| 1540 | { |
| 1541 | WebRtcCng_Encode(CNGenc_inst[k],&indata[k*frameLen], (frameLen <= 640 ? frameLen : 640) /* max 640 */, |
| 1542 | encoded,&tempLen,first_cng); |
| 1543 | encoded += tempLen; |
| 1544 | cdlen += tempLen; |
| 1545 | } |
| 1546 | *vad=0; |
| 1547 | first_cng=0; |
| 1548 | return(cdlen); |
| 1549 | } |
| 1550 | } |
| 1551 | |
| 1552 | |
| 1553 | // loop over all channels |
| 1554 | int totalLen = 0; |
| 1555 | |
| 1556 | for (int k = 0; k < numChannels; k++) |
| 1557 | { |
| 1558 | /* Encode with the selected coder type */ |
| 1559 | if (coder==webrtc::kDecoderPCMu) { /*g711 u-law */ |
| 1560 | #ifdef CODEC_G711 |
kwiberg@webrtc.org | 1c6239a | 2015-02-09 12:55:48 +0000 | [diff] [blame] | 1561 | cdlen = WebRtcG711_EncodeU(indata, frameLen, encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1562 | #endif |
| 1563 | } |
| 1564 | else if (coder==webrtc::kDecoderPCMa) { /*g711 A-law */ |
| 1565 | #ifdef CODEC_G711 |
kwiberg@webrtc.org | 1c6239a | 2015-02-09 12:55:48 +0000 | [diff] [blame] | 1566 | cdlen = WebRtcG711_EncodeA(indata, frameLen, encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1567 | } |
| 1568 | #endif |
| 1569 | #ifdef CODEC_PCM16B |
| 1570 | else if ((coder==webrtc::kDecoderPCM16B)||(coder==webrtc::kDecoderPCM16Bwb)|| |
| 1571 | (coder==webrtc::kDecoderPCM16Bswb32kHz)||(coder==webrtc::kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */ |
kwiberg@webrtc.org | 648f5d6 | 2015-02-10 09:18:28 +0000 | [diff] [blame^] | 1572 | cdlen = WebRtcPcm16b_Encode(indata, frameLen, encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1573 | } |
| 1574 | #endif |
| 1575 | #ifdef CODEC_G722 |
| 1576 | else if (coder==webrtc::kDecoderG722) { /*g722 */ |
kwiberg@webrtc.org | 0cd5558 | 2014-12-02 11:45:51 +0000 | [diff] [blame] | 1577 | cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded); |
turaj@webrtc.org | 58cd316 | 2013-10-31 15:15:55 +0000 | [diff] [blame] | 1578 | assert(cdlen == frameLen>>1); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1579 | } |
| 1580 | #endif |
| 1581 | #ifdef CODEC_ILBC |
| 1582 | else if (coder==webrtc::kDecoderILBC) { /*iLBC */ |
kwiberg@webrtc.org | cb858ba | 2014-12-08 17:11:44 +0000 | [diff] [blame] | 1583 | cdlen = WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata, |
| 1584 | frameLen, encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1585 | } |
| 1586 | #endif |
| 1587 | #if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC |
| 1588 | else if (coder==webrtc::kDecoderISAC) { /*iSAC */ |
| 1589 | int noOfCalls=0; |
| 1590 | cdlen=0; |
| 1591 | while (cdlen<=0) { |
| 1592 | #ifdef CODEC_ISAC /* floating point */ |
kwiberg@webrtc.org | 7ee24a7 | 2014-09-24 10:31:02 +0000 | [diff] [blame] | 1593 | cdlen = WebRtcIsac_Encode(ISAC_inst[k], |
| 1594 | &indata[noOfCalls * 160], |
| 1595 | encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1596 | #else /* fixed point */ |
kwiberg@webrtc.org | 7ee24a7 | 2014-09-24 10:31:02 +0000 | [diff] [blame] | 1597 | cdlen = WebRtcIsacfix_Encode(ISAC_inst[k], |
| 1598 | &indata[noOfCalls * 160], |
| 1599 | encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1600 | #endif |
| 1601 | noOfCalls++; |
| 1602 | } |
| 1603 | } |
| 1604 | #endif |
| 1605 | #ifdef CODEC_ISAC_SWB |
| 1606 | else if (coder==webrtc::kDecoderISACswb) { /* iSAC SWB */ |
| 1607 | int noOfCalls=0; |
| 1608 | cdlen=0; |
| 1609 | while (cdlen<=0) { |
kwiberg@webrtc.org | 7ee24a7 | 2014-09-24 10:31:02 +0000 | [diff] [blame] | 1610 | cdlen = WebRtcIsac_Encode(ISACSWB_inst[k], |
| 1611 | &indata[noOfCalls * 320], |
| 1612 | encoded); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1613 | noOfCalls++; |
| 1614 | } |
| 1615 | } |
| 1616 | #endif |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1617 | indata += frameLen; |
| 1618 | encoded += cdlen; |
| 1619 | totalLen += cdlen; |
| 1620 | |
pbos@webrtc.org | d900e8b | 2013-07-03 15:12:26 +0000 | [diff] [blame] | 1621 | } // end for |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1622 | |
| 1623 | first_cng=1; |
| 1624 | return(totalLen); |
| 1625 | } |
| 1626 | |
| 1627 | |
| 1628 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1629 | void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc){ |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1630 | |
| 1631 | rtp_data[0]=(unsigned char)0x80; |
| 1632 | rtp_data[1]=(unsigned char)(payloadType & 0xFF); |
| 1633 | rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF); |
| 1634 | rtp_data[3]=(unsigned char)((seqNo)&0xFF); |
| 1635 | rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF); |
| 1636 | rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF); |
| 1637 | |
| 1638 | rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF); |
| 1639 | rtp_data[7]=(unsigned char)(timestamp & 0xFF); |
| 1640 | |
| 1641 | rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF); |
| 1642 | rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF); |
| 1643 | |
| 1644 | rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF); |
| 1645 | rtp_data[11]=(unsigned char)(ssrc & 0xFF); |
| 1646 | } |
| 1647 | |
| 1648 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1649 | int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen, |
| 1650 | int seqNo, uint32_t ssrc) |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1651 | { |
| 1652 | |
| 1653 | int i; |
| 1654 | unsigned char *rtpPointer; |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1655 | uint16_t offset; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1656 | |
| 1657 | /* first create "standard" RTP header */ |
| 1658 | makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc); |
| 1659 | |
| 1660 | rtpPointer = &rtp_data[12]; |
| 1661 | |
| 1662 | /* add one sub-header for each redundant payload (not the primary) */ |
| 1663 | for(i=0; i<numPayloads-1; i++) { /* |0 1 2 3 4 5 6 7| */ |
| 1664 | if(blockLen[i] > 0) { |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1665 | offset = (uint16_t) (timestamp[numPayloads-1] - timestamp[i]); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1666 | |
| 1667 | rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F| block PT | */ |
| 1668 | rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF); /* | timestamp- | */ |
| 1669 | rtpPointer[2] = (unsigned char) ( ((offset & 0x3F)<<2) | |
| 1670 | ( (blockLen[i]>>8) & 0x03 ) ); /* | -offset |bl-| */ |
| 1671 | rtpPointer[3] = (unsigned char) ( blockLen[i] & 0xFF ); /* | -ock length | */ |
| 1672 | |
| 1673 | rtpPointer += 4; |
| 1674 | } |
| 1675 | } |
| 1676 | |
| 1677 | /* last sub-header */ |
| 1678 | rtpPointer[0]= (unsigned char) (0x00 | (0x7F&payloadType[numPayloads-1]));/* |F| block PT | */ |
| 1679 | rtpPointer += 1; |
| 1680 | |
| 1681 | return(rtpPointer - rtp_data); /* length of header in bytes */ |
| 1682 | } |
| 1683 | |
| 1684 | |
| 1685 | |
| 1686 | int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) { |
| 1687 | unsigned char E,R,V; |
| 1688 | R=0; |
| 1689 | V=(unsigned char)Volume; |
| 1690 | if (End==0) { |
| 1691 | E = 0x00; |
| 1692 | } else { |
| 1693 | E = 0x80; |
| 1694 | } |
| 1695 | payload_data[0]=(unsigned char)Event; |
| 1696 | payload_data[1]=(unsigned char)(E|R|V); |
| 1697 | //Duration equals 8 times time_ms, default is 8000 Hz. |
| 1698 | payload_data[2]=(unsigned char)((Duration>>8)&0xFF); |
| 1699 | payload_data[3]=(unsigned char)(Duration&0xFF); |
| 1700 | return(4); |
| 1701 | } |
| 1702 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1703 | void stereoDeInterleave(int16_t* audioSamples, int numSamples) |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1704 | { |
| 1705 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1706 | int16_t *tempVec; |
| 1707 | int16_t *readPtr, *writeL, *writeR; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1708 | |
| 1709 | if (numSamples <= 0) |
| 1710 | return; |
| 1711 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1712 | tempVec = (int16_t *) malloc(sizeof(int16_t) * numSamples); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1713 | if (tempVec == NULL) { |
| 1714 | printf("Error allocating memory\n"); |
| 1715 | exit(0); |
| 1716 | } |
| 1717 | |
pbos@webrtc.org | 0946a56 | 2013-04-09 00:28:06 +0000 | [diff] [blame] | 1718 | memcpy(tempVec, audioSamples, numSamples*sizeof(int16_t)); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 1719 | |
| 1720 | writeL = audioSamples; |
| 1721 | writeR = &audioSamples[numSamples/2]; |
| 1722 | readPtr = tempVec; |
| 1723 | |
| 1724 | for (int k = 0; k < numSamples; k += 2) |
| 1725 | { |
| 1726 | *writeL = *readPtr; |
| 1727 | readPtr++; |
| 1728 | *writeR = *readPtr; |
| 1729 | readPtr++; |
| 1730 | writeL++; |
| 1731 | writeR++; |
| 1732 | } |
| 1733 | |
| 1734 | free(tempVec); |
| 1735 | |
| 1736 | } |
| 1737 | |
| 1738 | |
| 1739 | void stereoInterleave(unsigned char* data, int dataLen, int stride) |
| 1740 | { |
| 1741 | |
| 1742 | unsigned char *ptrL, *ptrR; |
| 1743 | unsigned char temp[10]; |
| 1744 | |
| 1745 | if (stride > 10) |
| 1746 | { |
| 1747 | exit(0); |
| 1748 | } |
| 1749 | |
| 1750 | if (dataLen%1 != 0) |
| 1751 | { |
| 1752 | // must be even number of samples |
| 1753 | printf("Error: cannot interleave odd sample number\n"); |
| 1754 | exit(0); |
| 1755 | } |
| 1756 | |
| 1757 | ptrL = data + stride; |
| 1758 | ptrR = &data[dataLen/2]; |
| 1759 | |
| 1760 | while (ptrL < ptrR) { |
| 1761 | // copy from right pointer to temp |
| 1762 | memcpy(temp, ptrR, stride); |
| 1763 | |
| 1764 | // shift data between pointers |
| 1765 | memmove(ptrL + stride, ptrL, ptrR - ptrL); |
| 1766 | |
| 1767 | // copy from temp to left pointer |
| 1768 | memcpy(ptrL, temp, stride); |
| 1769 | |
| 1770 | // advance pointers |
| 1771 | ptrL += stride*2; |
| 1772 | ptrR += stride; |
| 1773 | } |
| 1774 | |
| 1775 | } |