blob: efd00691505439c1dd6f685d59d84488b6a9bf82 [file] [log] [blame]
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001/*
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.org12dc1a32013-08-05 16:22:53 +000024#include <assert.h>
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000025
26#include "webrtc/typedefs.h"
27// needed for NetEqDecoder
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000028#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
29#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000030
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
74void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed);
75int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels);
76void defineCodecs(webrtc::NetEqDecoder *usedCodec, int *noOfCodecs );
77int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels);
pbos@webrtc.org0946a562013-04-09 00:28:06 +000078int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
79void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc);
80int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen,
81 int seqNo, uint32_t ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000082int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration);
pbos@webrtc.org0946a562013-04-09 00:28:06 +000083void stereoDeInterleave(int16_t* audioSamples, int numSamples);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000084void 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
160#ifdef CODEC_CELT_32
161#include "celt_interface.h"
162#endif
163
164
165/***********************************/
166/* Global codec instance variables */
167/***********************************/
168
169WebRtcVadInst *VAD_inst[2];
170
171#ifdef CODEC_G722
172 G722EncInst *g722EncState[2];
173#endif
174
175#ifdef CODEC_G722_1_24
176 G722_1_24_encinst_t *G722_1_24enc_inst[2];
177#endif
178#ifdef CODEC_G722_1_32
179 G722_1_32_encinst_t *G722_1_32enc_inst[2];
180#endif
181#ifdef CODEC_G722_1_16
182 G722_1_16_encinst_t *G722_1_16enc_inst[2];
183#endif
184#ifdef CODEC_G722_1C_24
185 G722_1C_24_encinst_t *G722_1C_24enc_inst[2];
186#endif
187#ifdef CODEC_G722_1C_32
188 G722_1C_32_encinst_t *G722_1C_32enc_inst[2];
189#endif
190#ifdef CODEC_G722_1C_48
191 G722_1C_48_encinst_t *G722_1C_48enc_inst[2];
192#endif
193#ifdef CODEC_G726
194 G726_encinst_t *G726enc_inst[2];
195#endif
196#ifdef CODEC_G729
197 G729_encinst_t *G729enc_inst[2];
198#endif
199#ifdef CODEC_G729_1
200 G729_1_inst_t *G729_1_inst[2];
201#endif
202#ifdef CODEC_AMR
203 AMR_encinst_t *AMRenc_inst[2];
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000204 int16_t AMR_bitrate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000205#endif
206#ifdef CODEC_AMRWB
207 AMRWB_encinst_t *AMRWBenc_inst[2];
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000208 int16_t AMRWB_bitrate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000209#endif
210#ifdef CODEC_ILBC
211 iLBC_encinst_t *iLBCenc_inst[2];
212#endif
213#ifdef CODEC_ISAC
214 ISACStruct *ISAC_inst[2];
215#endif
216#ifdef NETEQ_ISACFIX_CODEC
217 ISACFIX_MainStruct *ISAC_inst[2];
218#endif
219#ifdef CODEC_ISAC_SWB
220 ISACStruct *ISACSWB_inst[2];
221#endif
222#ifdef CODEC_GSMFR
223 GSMFR_encinst_t *GSMFRenc_inst[2];
224#endif
225#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
226 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
227 CNG_enc_inst *CNGenc_inst[2];
228#endif
229#ifdef CODEC_SPEEX_8
230 SPEEX_encinst_t *SPEEX8enc_inst[2];
231#endif
232#ifdef CODEC_SPEEX_16
233 SPEEX_encinst_t *SPEEX16enc_inst[2];
234#endif
235#ifdef CODEC_CELT_32
236 CELT_encinst_t *CELT32enc_inst[2];
237#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000238
239
240int main(int argc, char* argv[])
241{
242 int packet_size, fs;
243 webrtc::NetEqDecoder usedCodec;
244 int payloadType;
245 int bitrate = 0;
246 int useVAD, vad;
247 int useRed=0;
248 int len, enc_len;
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000249 int16_t org_data[4000];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000250 unsigned char rtp_data[8000];
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000251 int16_t seqNo=0xFFF;
252 uint32_t ssrc=1235412312;
253 uint32_t timestamp=0xAC1245;
254 uint16_t length, plen;
255 uint32_t offset;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000256 double sendtime = 0;
257 int red_PT[2] = {0};
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000258 uint32_t red_TS[2] = {0};
259 uint16_t red_len[2] = {0};
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000260 int RTPheaderLen=12;
kwiberg@webrtc.org11729882014-10-13 10:53:42 +0000261 uint8_t red_data[8000];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000262#ifdef INSERT_OLD_PACKETS
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000263 uint16_t old_length, old_plen;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000264 int old_enc_len;
265 int first_old_packet=1;
266 unsigned char old_rtp_data[8000];
267 int packet_age=0;
268#endif
269#ifdef INSERT_DTMF_PACKETS
270 int NTone = 1;
271 int DTMFfirst = 1;
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000272 uint32_t DTMFtimestamp;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000273 bool dtmfSent = false;
274#endif
275 bool usingStereo = false;
276 int stereoMode = 0;
277 int numChannels = 1;
278
279 /* check number of parameters */
280 if ((argc != 6) && (argc != 7)) {
281 /* print help text and exit */
282 printf("Application to encode speech into an RTP stream.\n");
283 printf("The program reads a PCM file and encodes is using the specified codec.\n");
284 printf("The coded speech is packetized in RTP packest and written to the output file.\n");
285 printf("The format of the RTP stream file is simlilar to that of rtpplay,\n");
286 printf("but with the receive time euqal to 0 for all packets.\n");
287 printf("Usage:\n\n");
288 printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
289 printf("where:\n");
290
291 printf("PCMfile : PCM speech input file\n\n");
292
293 printf("RTPfile : RTP stream output file\n\n");
294
295 printf("frameLen : 80...960... Number of samples per packet (limit depends on codec)\n\n");
296
297 printf("codecName\n");
298#ifdef CODEC_PCM16B
299 printf(" : pcm16b 16 bit PCM (8kHz)\n");
300#endif
301#ifdef CODEC_PCM16B_WB
302 printf(" : pcm16b_wb 16 bit PCM (16kHz)\n");
303#endif
304#ifdef CODEC_PCM16B_32KHZ
305 printf(" : pcm16b_swb32 16 bit PCM (32kHz)\n");
306#endif
307#ifdef CODEC_PCM16B_48KHZ
308 printf(" : pcm16b_swb48 16 bit PCM (48kHz)\n");
309#endif
310#ifdef CODEC_G711
311 printf(" : pcma g711 A-law (8kHz)\n");
312#endif
313#ifdef CODEC_G711
314 printf(" : pcmu g711 u-law (8kHz)\n");
315#endif
316#ifdef CODEC_G729
317 printf(" : g729 G729 (8kHz and 8kbps) CELP (One-Three frame(s)/packet)\n");
318#endif
319#ifdef CODEC_G729_1
320 printf(" : g729.1 G729.1 (16kHz) variable rate (8--32 kbps)\n");
321#endif
322#ifdef CODEC_G722_1_16
323 printf(" : g722.1_16 G722.1 coder (16kHz) (g722.1 with 16kbps)\n");
324#endif
325#ifdef CODEC_G722_1_24
326 printf(" : g722.1_24 G722.1 coder (16kHz) (the 24kbps version)\n");
327#endif
328#ifdef CODEC_G722_1_32
329 printf(" : g722.1_32 G722.1 coder (16kHz) (the 32kbps version)\n");
330#endif
331#ifdef CODEC_G722_1C_24
332 printf(" : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps version)\n");
333#endif
334#ifdef CODEC_G722_1C_32
335 printf(" : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps version)\n");
336#endif
337#ifdef CODEC_G722_1C_48
338 printf(" : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps)\n");
339#endif
340
341#ifdef CODEC_G726
342 printf(" : g726_16 G726 coder (8kHz) 16kbps\n");
343 printf(" : g726_24 G726 coder (8kHz) 24kbps\n");
344 printf(" : g726_32 G726 coder (8kHz) 32kbps\n");
345 printf(" : g726_40 G726 coder (8kHz) 40kbps\n");
346#endif
347#ifdef CODEC_AMR
348 printf(" : AMRXk Adaptive Multi Rate CELP codec (8kHz)\n");
349 printf(" X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 or 12.2\n");
350#endif
351#ifdef CODEC_AMRWB
352 printf(" : AMRwbXk Adaptive Multi Rate Wideband CELP codec (16kHz)\n");
353 printf(" X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n");
354#endif
355#ifdef CODEC_ILBC
356 printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n");
357#endif
358#ifdef CODEC_ISAC
359 printf(" : isac iSAC (16kHz and 32.0 kbps). To set rate specify a rate parameter as last parameter\n");
360#endif
361#ifdef CODEC_ISAC_SWB
362 printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
363#endif
364#ifdef CODEC_GSMFR
365 printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
366#endif
367#ifdef CODEC_G722
368 printf(" : g722 g722 coder (16kHz) (the 64kbps version)\n");
369#endif
370#ifdef CODEC_SPEEX_8
371 printf(" : speex8 speex coder (8 kHz)\n");
372#endif
373#ifdef CODEC_SPEEX_16
374 printf(" : speex16 speex coder (16 kHz)\n");
375#endif
376#ifdef CODEC_CELT_32
377 printf(" : celt32 celt coder (32 kHz)\n");
378#endif
379#ifdef CODEC_RED
380#ifdef CODEC_G711
381 printf(" : red_pcm Redundancy RTP packet with 2*G711A frames\n");
382#endif
383#ifdef CODEC_ISAC
384 printf(" : red_isac Redundancy RTP packet with 2*iSAC frames\n");
385#endif
386#endif
387 printf("\n");
388
389#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
390 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
391 printf("useVAD : 0 Voice Activity Detection is switched off\n");
392 printf(" : 1 Voice Activity Detection is switched on\n\n");
393#else
394 printf("useVAD : 0 Voice Activity Detection switched off (on not supported)\n\n");
395#endif
396 printf("bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n");
397
398 return(0);
399 }
400
401 FILE* in_file=fopen(argv[1],"rb");
402 CHECK_NOT_NULL(in_file);
403 printf("Input file: %s\n",argv[1]);
404 FILE* out_file=fopen(argv[2],"wb");
405 CHECK_NOT_NULL(out_file);
406 printf("Output file: %s\n\n",argv[2]);
407 packet_size=atoi(argv[3]);
408 CHECK_NOT_NULL(packet_size);
409 printf("Packet size: %i\n",packet_size);
410
411 // check for stereo
412 if(argv[4][strlen(argv[4])-1] == '*') {
413 // use stereo
414 usingStereo = true;
415 numChannels = 2;
416 argv[4][strlen(argv[4])-1] = '\0';
417 }
418
419 NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed);
420
421 if(useRed) {
422 RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant payload, except last one which is 1 byte */
423 }
424
425 useVAD=atoi(argv[5]);
426#if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
427 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
428 if (useVAD!=0) {
429 printf("Error: this simulation does not support VAD/DTX/CNG\n");
430 }
431#endif
432
433 // check stereo type
434 if(usingStereo)
435 {
436 switch(usedCodec)
437 {
438 // sample based codecs
439 case webrtc::kDecoderPCMu:
440 case webrtc::kDecoderPCMa:
441 case webrtc::kDecoderG722:
442 {
443 // 1 octet per sample
444 stereoMode = STEREO_MODE_SAMPLE_1;
445 break;
446 }
447 case webrtc::kDecoderPCM16B:
448 case webrtc::kDecoderPCM16Bwb:
449 case webrtc::kDecoderPCM16Bswb32kHz:
450 case webrtc::kDecoderPCM16Bswb48kHz:
451 {
452 // 2 octets per sample
453 stereoMode = STEREO_MODE_SAMPLE_2;
454 break;
455 }
456
457 // fixed-rate frame codecs (with internal VAD)
458 default:
459 {
460 printf("Cannot use codec %s as stereo codec\n", argv[4]);
461 exit(0);
462 }
463 }
464 }
465
466 if ((usedCodec == webrtc::kDecoderISAC) || (usedCodec == webrtc::kDecoderISACswb))
467 {
468 if (argc != 7)
469 {
470 if (usedCodec == webrtc::kDecoderISAC)
471 {
472 bitrate = 32000;
473 printf(
474 "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n");
475 }
476 else // (usedCodec==webrtc::kDecoderISACswb)
477 {
478 bitrate = 56000;
479 printf(
480 "Running iSAC at default bitrate of 56000 bps (to specify explicitly add the bps as last parameter)\n");
481 }
482 }
483 else
484 {
485 bitrate = atoi(argv[6]);
486 if (usedCodec == webrtc::kDecoderISAC)
487 {
488 if ((bitrate < 10000) || (bitrate > 32000))
489 {
490 printf(
491 "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is invalid)\n",
492 bitrate);
493 exit(0);
494 }
495 printf("Running iSAC at bitrate of %i bps\n", bitrate);
496 }
497 else // (usedCodec==webrtc::kDecoderISACswb)
498 {
499 if ((bitrate < 32000) || (bitrate > 56000))
500 {
501 printf(
502 "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
503 bitrate);
504 exit(0);
505 }
506 }
507 }
508 }
509 else
510 {
511 if (argc == 7)
512 {
513 printf(
514 "Error: Bitrate parameter can only be specified for iSAC, G.723, and G.729.1\n");
515 exit(0);
516 }
517 }
518
519 if(useRed) {
520 printf("Redundancy engaged. ");
521 }
522 printf("Used codec: %i\n",usedCodec);
523 printf("Payload type: %i\n",payloadType);
524
525 NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels);
526
527 /* write file header */
528 //fprintf(out_file, "#!RTPencode%s\n", "1.0");
529 fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000530 uint32_t dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000531 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
532 return -1;
533 }
534 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
535 return -1;
536 }
537 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
538 return -1;
539 }
540 if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
541 return -1;
542 }
543 if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
544 return -1;
545 }
546
547#ifdef TIMESTAMP_WRAPAROUND
548 timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */
549#endif
550#if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
551 srand(RANDOM_SEED);
552#endif
553
554 /* if redundancy is used, the first redundant payload is zero length */
555 red_len[0] = 0;
556
557 /* read first frame */
558 len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
559
560 /* de-interleave if stereo */
561 if ( usingStereo )
562 {
563 stereoDeInterleave(org_data, len * numChannels);
564 }
565
566 while (len==packet_size) {
567
568#ifdef INSERT_DTMF_PACKETS
569 dtmfSent = false;
570
571 if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) {
572 if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) {
573 // tone has not ended
574 if (DTMFfirst==1) {
575 DTMFtimestamp = timestamp; // save this timestamp
576 DTMFfirst=0;
577 }
578 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
579 enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len);
580 }
581 else {
582 // tone has ended
583 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
584 enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4, DTMF_DURATION*(fs/1000));
585 NTone++;
586 DTMFfirst=1;
587 }
588
589 /* write RTP packet to file */
590 length = htons(12 + enc_len + 8);
591 plen = htons(12 + enc_len);
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000592 offset = (uint32_t) sendtime; //(timestamp/(fs/1000));
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000593 offset = htonl(offset);
594 if (fwrite(&length, 2, 1, out_file) != 1) {
595 return -1;
596 }
597 if (fwrite(&plen, 2, 1, out_file) != 1) {
598 return -1;
599 }
600 if (fwrite(&offset, 4, 1, out_file) != 1) {
601 return -1;
602 }
603 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
604 return -1;
605 }
606
607 dtmfSent = true;
608 }
609#endif
610
611#ifdef NO_DTMF_OVERDUB
612 /* If DTMF is sent, we should not send any speech packets during the same time */
613 if (dtmfSent) {
614 enc_len = 0;
615 }
616 else {
617#endif
618 /* encode frame */
619 enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels);
620 if (enc_len==-1) {
621 printf("Error encoding frame\n");
622 exit(0);
623 }
624
625 if ( usingStereo &&
626 stereoMode != STEREO_MODE_FRAME &&
627 vad == 1 )
628 {
629 // interleave the encoded payload for sample-based codecs (not for CNG)
630 stereoInterleave(&rtp_data[12], enc_len, stereoMode);
631 }
632#ifdef NO_DTMF_OVERDUB
633 }
634#endif
635
636 if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
637 if(useRed) {
638 if(red_len[0] > 0) {
639 memmove(&rtp_data[RTPheaderLen+red_len[0]], &rtp_data[12], enc_len);
640 memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
641
642 red_len[1] = enc_len;
643 red_TS[1] = timestamp;
644 if(vad)
645 red_PT[1] = payloadType;
646 else
647 red_PT[1] = NETEQ_CODEC_CN_PT;
648
649 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
650
651
652 enc_len += red_len[0] + RTPheaderLen - 12;
653 }
654 else { // do not use redundancy payload for this packet, i.e., only last payload
655 memmove(&rtp_data[RTPheaderLen-4], &rtp_data[12], enc_len);
656 //memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
657
658 red_len[1] = enc_len;
659 red_TS[1] = timestamp;
660 if(vad)
661 red_PT[1] = payloadType;
662 else
663 red_PT[1] = NETEQ_CODEC_CN_PT;
664
665 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
666
667
668 enc_len += red_len[0] + RTPheaderLen - 4 - 12; // 4 is length of redundancy header (not used)
669 }
670 }
671 else {
672
673 /* make RTP header */
674 if (vad) // regular speech data
675 makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc);
676 else // CNG data
677 makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc);
678
679 }
680#ifdef MULTIPLE_SAME_TIMESTAMP
681 int mult_pack=0;
682 do {
683#endif //MULTIPLE_SAME_TIMESTAMP
684 /* write RTP packet to file */
685 length = htons(12 + enc_len + 8);
686 plen = htons(12 + enc_len);
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000687 offset = (uint32_t) sendtime;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000688 //(timestamp/(fs/1000));
689 offset = htonl(offset);
690 if (fwrite(&length, 2, 1, out_file) != 1) {
691 return -1;
692 }
693 if (fwrite(&plen, 2, 1, out_file) != 1) {
694 return -1;
695 }
696 if (fwrite(&offset, 4, 1, out_file) != 1) {
697 return -1;
698 }
699#ifdef RANDOM_DATA
700 for (int k=0; k<12+enc_len; k++) {
701 rtp_data[k] = rand() + rand();
702 }
703#endif
704#ifdef RANDOM_PAYLOAD_DATA
705 for (int k=12; k<12+enc_len; k++) {
706 rtp_data[k] = rand() + rand();
707 }
708#endif
709 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
710 return -1;
711 }
712#ifdef MULTIPLE_SAME_TIMESTAMP
713 } while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) );
714#endif //MULTIPLE_SAME_TIMESTAMP
715
716#ifdef INSERT_OLD_PACKETS
717 if (packet_age >= OLD_PACKET*fs) {
718 if (!first_old_packet) {
719 // send the old packet
720 if (fwrite(&old_length, 2, 1,
721 out_file) != 1) {
722 return -1;
723 }
724 if (fwrite(&old_plen, 2, 1,
725 out_file) != 1) {
726 return -1;
727 }
728 if (fwrite(&offset, 4, 1,
729 out_file) != 1) {
730 return -1;
731 }
732 if (fwrite(old_rtp_data, 12 + old_enc_len,
733 1, out_file) != 1) {
734 return -1;
735 }
736 }
737 // store current packet as old
738 old_length=length;
739 old_plen=plen;
740 memcpy(old_rtp_data,rtp_data,12+enc_len);
741 old_enc_len=enc_len;
742 first_old_packet=0;
743 packet_age=0;
744
745 }
746 packet_age += packet_size;
747#endif
748
749 if(useRed) {
750 /* move data to redundancy store */
751#ifdef CODEC_ISAC
752 if(usedCodec==webrtc::kDecoderISAC)
753 {
754 assert(!usingStereo); // Cannot handle stereo yet
kwiberg@webrtc.org11729882014-10-13 10:53:42 +0000755 red_len[0] =
756 WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000757 }
758 else
759 {
760#endif
761 memcpy(red_data, &rtp_data[RTPheaderLen+red_len[0]], enc_len);
762 red_len[0]=red_len[1];
763#ifdef CODEC_ISAC
764 }
765#endif
766 red_TS[0]=red_TS[1];
767 red_PT[0]=red_PT[1];
768 }
769
770 }
771
772 /* read next frame */
773 len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
774 /* de-interleave if stereo */
775 if ( usingStereo )
776 {
777 stereoDeInterleave(org_data, len * numChannels);
778 }
779
780 if (payloadType==NETEQ_CODEC_G722_PT)
781 timestamp+=len>>1;
782 else
783 timestamp+=len;
784
785 sendtime += (double) len/(fs/1000);
786 }
787
788 NetEQTest_free_coders(usedCodec, numChannels);
789 fclose(in_file);
790 fclose(out_file);
791 printf("Done!\n");
792
793 return(0);
794}
795
796
797
798
799/****************/
800/* Subfunctions */
801/****************/
802
803void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed) {
804
805 *bitrate = 0; /* Default bitrate setting */
806 *useRed = 0; /* Default no redundancy */
807
808 if(!strcmp(name,"pcmu")){
809 *codec=webrtc::kDecoderPCMu;
810 *PT=NETEQ_CODEC_PCMU_PT;
811 *fs=8000;
812 }
813 else if(!strcmp(name,"pcma")){
814 *codec=webrtc::kDecoderPCMa;
815 *PT=NETEQ_CODEC_PCMA_PT;
816 *fs=8000;
817 }
818 else if(!strcmp(name,"pcm16b")){
819 *codec=webrtc::kDecoderPCM16B;
820 *PT=NETEQ_CODEC_PCM16B_PT;
821 *fs=8000;
822 }
823 else if(!strcmp(name,"pcm16b_wb")){
824 *codec=webrtc::kDecoderPCM16Bwb;
825 *PT=NETEQ_CODEC_PCM16B_WB_PT;
826 *fs=16000;
827 }
828 else if(!strcmp(name,"pcm16b_swb32")){
829 *codec=webrtc::kDecoderPCM16Bswb32kHz;
830 *PT=NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
831 *fs=32000;
832 }
833 else if(!strcmp(name,"pcm16b_swb48")){
834 *codec=webrtc::kDecoderPCM16Bswb48kHz;
835 *PT=NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
836 *fs=48000;
837 }
838 else if(!strcmp(name,"g722")){
839 *codec=webrtc::kDecoderG722;
840 *PT=NETEQ_CODEC_G722_PT;
841 *fs=16000;
842 }
843 else if((!strcmp(name,"ilbc"))&&((frameLen%240==0)||(frameLen%160==0))){
844 *fs=8000;
845 *codec=webrtc::kDecoderILBC;
846 *PT=NETEQ_CODEC_ILBC_PT;
847 }
848 else if(!strcmp(name,"isac")){
849 *fs=16000;
850 *codec=webrtc::kDecoderISAC;
851 *PT=NETEQ_CODEC_ISAC_PT;
852 }
853 else if(!strcmp(name,"isacswb")){
854 *fs=32000;
855 *codec=webrtc::kDecoderISACswb;
856 *PT=NETEQ_CODEC_ISACSWB_PT;
857 }
858 else if(!strcmp(name,"celt32")){
859 *fs=32000;
860 *codec=webrtc::kDecoderCELT_32;
861 *PT=NETEQ_CODEC_CELT32_PT;
862 }
863 else if(!strcmp(name,"red_pcm")){
864 *codec=webrtc::kDecoderPCMa;
865 *PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
866 *fs=8000;
867 *useRed = 1;
868 } else if(!strcmp(name,"red_isac")){
869 *codec=webrtc::kDecoderISAC;
870 *PT=NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
871 *fs=16000;
872 *useRed = 1;
873 } else {
874 printf("Error: Not a supported codec (%s)\n", name);
875 exit(0);
876 }
877
878}
879
880
881
882
883int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels){
884
885 int ok=0;
886
887 for (int k = 0; k < numChannels; k++)
888 {
889 ok=WebRtcVad_Create(&VAD_inst[k]);
890 if (ok!=0) {
891 printf("Error: Couldn't allocate memory for VAD instance\n");
892 exit(0);
893 }
894 ok=WebRtcVad_Init(VAD_inst[k]);
895 if (ok==-1) {
896 printf("Error: Initialization of VAD struct failed\n");
897 exit(0);
898 }
899
900
901#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
902 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
903 ok=WebRtcCng_CreateEnc(&CNGenc_inst[k]);
904 if (ok!=0) {
905 printf("Error: Couldn't allocate memory for CNG encoding instance\n");
906 exit(0);
907 }
908 if(sampfreq <= 16000) {
909 ok=WebRtcCng_InitEnc(CNGenc_inst[k],sampfreq, 200, 5);
910 if (ok==-1) {
911 printf("Error: Initialization of CNG struct failed. Error code %d\n",
912 WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
913 exit(0);
914 }
915 }
916#endif
917
918 switch (coder) {
919#ifdef CODEC_PCM16B
920 case webrtc::kDecoderPCM16B :
921#endif
922#ifdef CODEC_PCM16B_WB
923 case webrtc::kDecoderPCM16Bwb :
924#endif
925#ifdef CODEC_PCM16B_32KHZ
926 case webrtc::kDecoderPCM16Bswb32kHz :
927#endif
928#ifdef CODEC_PCM16B_48KHZ
929 case webrtc::kDecoderPCM16Bswb48kHz :
930#endif
931#ifdef CODEC_G711
932 case webrtc::kDecoderPCMu :
933 case webrtc::kDecoderPCMa :
934#endif
935 // do nothing
936 break;
937#ifdef CODEC_G729
938 case webrtc::kDecoderG729:
939 if (sampfreq==8000) {
940 if ((enc_frameSize==80)||(enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==400)||(enc_frameSize==480)) {
941 ok=WebRtcG729_CreateEnc(&G729enc_inst[k]);
942 if (ok!=0) {
943 printf("Error: Couldn't allocate memory for G729 encoding instance\n");
944 exit(0);
945 }
946 } else {
947 printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 ms!!\n\n");
948 exit(0);
949 }
950 WebRtcG729_EncoderInit(G729enc_inst[k], vad);
951 if ((vad==1)&&(enc_frameSize!=80)) {
952 printf("\nError - This simulation only supports VAD for G729 at 10ms packets (not %dms)\n", (enc_frameSize>>3));
953 }
954 } else {
955 printf("\nError - g729 is only developed for 8kHz \n");
956 exit(0);
957 }
958 break;
959#endif
960#ifdef CODEC_G729_1
961 case webrtc::kDecoderG729_1:
962 if (sampfreq==16000) {
963 if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)
964 ) {
965 ok=WebRtcG7291_Create(&G729_1_inst[k]);
966 if (ok!=0) {
967 printf("Error: Couldn't allocate memory for G.729.1 codec instance\n");
968 exit(0);
969 }
970 } else {
971 printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
972 exit(0);
973 }
974 if (!(((bitrate >= 12000) && (bitrate <= 32000) && (bitrate%2000 == 0)) || (bitrate == 8000))) {
975 /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
976 printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in steps of 2000 bps\n");
977 exit(0);
978 }
979 WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/, 0 /*flagG729mode*/);
980 } else {
981 printf("\nError - G.729.1 input is always 16 kHz \n");
982 exit(0);
983 }
984 break;
985#endif
986#ifdef CODEC_SPEEX_8
987 case webrtc::kDecoderSPEEX_8 :
988 if (sampfreq==8000) {
989 if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
990 ok=WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
991 if (ok!=0) {
992 printf("Error: Couldn't allocate memory for Speex encoding instance\n");
993 exit(0);
994 }
995 } else {
996 printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
997 exit(0);
998 }
999 if ((vad==1)&&(enc_frameSize!=160)) {
1000 printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>3));
1001 vad=0;
1002 }
1003 ok=WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
1004 if (ok!=0) exit(0);
1005 } else {
1006 printf("\nError - Speex8 called with sample frequency other than 8 kHz.\n\n");
1007 }
1008 break;
1009#endif
1010#ifdef CODEC_SPEEX_16
1011 case webrtc::kDecoderSPEEX_16 :
1012 if (sampfreq==16000) {
1013 if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)) {
1014 ok=WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
1015 if (ok!=0) {
1016 printf("Error: Couldn't allocate memory for Speex encoding instance\n");
1017 exit(0);
1018 }
1019 } else {
1020 printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
1021 exit(0);
1022 }
1023 if ((vad==1)&&(enc_frameSize!=320)) {
1024 printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>4));
1025 vad=0;
1026 }
1027 ok=WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
1028 if (ok!=0) exit(0);
1029 } else {
1030 printf("\nError - Speex16 called with sample frequency other than 16 kHz.\n\n");
1031 }
1032 break;
1033#endif
1034#ifdef CODEC_CELT_32
1035 case webrtc::kDecoderCELT_32 :
1036 if (sampfreq==32000) {
1037 if (enc_frameSize==320) {
1038 ok=WebRtcCelt_CreateEnc(&CELT32enc_inst[k], 1 /*mono*/);
1039 if (ok!=0) {
1040 printf("Error: Couldn't allocate memory for Celt encoding instance\n");
1041 exit(0);
1042 }
1043 } else {
1044 printf("\nError: Celt only supports 10 ms!!\n\n");
1045 exit(0);
1046 }
1047 ok=WebRtcCelt_EncoderInit(CELT32enc_inst[k], 1 /*mono*/, 48000 /*bitrate*/);
1048 if (ok!=0) exit(0);
1049 } else {
1050 printf("\nError - Celt32 called with sample frequency other than 32 kHz.\n\n");
1051 }
1052 break;
1053#endif
1054
1055#ifdef CODEC_G722_1_16
1056 case webrtc::kDecoderG722_1_16 :
1057 if (sampfreq==16000) {
1058 ok=WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
1059 if (ok!=0) {
1060 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
1061 exit(0);
1062 }
1063 if (enc_frameSize==320) {
1064 } else {
1065 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1066 exit(0);
1067 }
1068 WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
1069 } else {
1070 printf("\nError - G722.1 is only developed for 16kHz \n");
1071 exit(0);
1072 }
1073 break;
1074#endif
1075#ifdef CODEC_G722_1_24
1076 case webrtc::kDecoderG722_1_24 :
1077 if (sampfreq==16000) {
1078 ok=WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
1079 if (ok!=0) {
1080 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
1081 exit(0);
1082 }
1083 if (enc_frameSize==320) {
1084 } else {
1085 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1086 exit(0);
1087 }
1088 WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
1089 } else {
1090 printf("\nError - G722.1 is only developed for 16kHz \n");
1091 exit(0);
1092 }
1093 break;
1094#endif
1095#ifdef CODEC_G722_1_32
1096 case webrtc::kDecoderG722_1_32 :
1097 if (sampfreq==16000) {
1098 ok=WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
1099 if (ok!=0) {
1100 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
1101 exit(0);
1102 }
1103 if (enc_frameSize==320) {
1104 } else {
1105 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1106 exit(0);
1107 }
1108 WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
1109 } else {
1110 printf("\nError - G722.1 is only developed for 16kHz \n");
1111 exit(0);
1112 }
1113 break;
1114#endif
1115#ifdef CODEC_G722_1C_24
1116 case webrtc::kDecoderG722_1C_24 :
1117 if (sampfreq==32000) {
1118 ok=WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
1119 if (ok!=0) {
1120 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
1121 exit(0);
1122 }
1123 if (enc_frameSize==640) {
1124 } else {
1125 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1126 exit(0);
1127 }
1128 WebRtcG7221C_EncoderInit24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
1129 } else {
1130 printf("\nError - G722.1 C is only developed for 32kHz \n");
1131 exit(0);
1132 }
1133 break;
1134#endif
1135#ifdef CODEC_G722_1C_32
1136 case webrtc::kDecoderG722_1C_32 :
1137 if (sampfreq==32000) {
1138 ok=WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
1139 if (ok!=0) {
1140 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
1141 exit(0);
1142 }
1143 if (enc_frameSize==640) {
1144 } else {
1145 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1146 exit(0);
1147 }
1148 WebRtcG7221C_EncoderInit32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
1149 } else {
1150 printf("\nError - G722.1 C is only developed for 32kHz \n");
1151 exit(0);
1152 }
1153 break;
1154#endif
1155#ifdef CODEC_G722_1C_48
1156 case webrtc::kDecoderG722_1C_48 :
1157 if (sampfreq==32000) {
1158 ok=WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
1159 if (ok!=0) {
1160 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
1161 exit(0);
1162 }
1163 if (enc_frameSize==640) {
1164 } else {
1165 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1166 exit(0);
1167 }
1168 WebRtcG7221C_EncoderInit48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
1169 } else {
1170 printf("\nError - G722.1 C is only developed for 32kHz \n");
1171 exit(0);
1172 }
1173 break;
1174#endif
1175#ifdef CODEC_G722
1176 case webrtc::kDecoderG722 :
1177 if (sampfreq==16000) {
1178 if (enc_frameSize%2==0) {
1179 } else {
1180 printf("\nError - g722 frames must have an even number of enc_frameSize\n");
1181 exit(0);
1182 }
1183 WebRtcG722_CreateEncoder(&g722EncState[k]);
1184 WebRtcG722_EncoderInit(g722EncState[k]);
1185 } else {
1186 printf("\nError - g722 is only developed for 16kHz \n");
1187 exit(0);
1188 }
1189 break;
1190#endif
1191#ifdef CODEC_AMR
1192 case webrtc::kDecoderAMR :
1193 if (sampfreq==8000) {
1194 ok=WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
1195 if (ok!=0) {
1196 printf("Error: Couldn't allocate memory for AMR encoding instance\n");
1197 exit(0);
1198 }if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
1199 } else {
1200 printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
1201 exit(0);
1202 }
1203 WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
1204 WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
1205 AMR_bitrate = bitrate;
1206 } else {
1207 printf("\nError - AMR is only developed for 8kHz \n");
1208 exit(0);
1209 }
1210 break;
1211#endif
1212#ifdef CODEC_AMRWB
1213 case webrtc::kDecoderAMRWB :
1214 if (sampfreq==16000) {
1215 ok=WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
1216 if (ok!=0) {
1217 printf("Error: Couldn't allocate memory for AMRWB encoding instance\n");
1218 exit(0);
1219 }
1220 if (((enc_frameSize/320)<0)||((enc_frameSize/320)>3)||((enc_frameSize%320)!=0)) {
1221 printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
1222 exit(0);
1223 }
1224 WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
1225 if (bitrate==7000) {
1226 AMRWB_bitrate = AMRWB_MODE_7k;
1227 } else if (bitrate==9000) {
1228 AMRWB_bitrate = AMRWB_MODE_9k;
1229 } else if (bitrate==12000) {
1230 AMRWB_bitrate = AMRWB_MODE_12k;
1231 } else if (bitrate==14000) {
1232 AMRWB_bitrate = AMRWB_MODE_14k;
1233 } else if (bitrate==16000) {
1234 AMRWB_bitrate = AMRWB_MODE_16k;
1235 } else if (bitrate==18000) {
1236 AMRWB_bitrate = AMRWB_MODE_18k;
1237 } else if (bitrate==20000) {
1238 AMRWB_bitrate = AMRWB_MODE_20k;
1239 } else if (bitrate==23000) {
1240 AMRWB_bitrate = AMRWB_MODE_23k;
1241 } else if (bitrate==24000) {
1242 AMRWB_bitrate = AMRWB_MODE_24k;
1243 }
1244 WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
1245
1246 } else {
1247 printf("\nError - AMRwb is only developed for 16kHz \n");
1248 exit(0);
1249 }
1250 break;
1251#endif
1252#ifdef CODEC_ILBC
1253 case webrtc::kDecoderILBC :
1254 if (sampfreq==8000) {
1255 ok=WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
1256 if (ok!=0) {
1257 printf("Error: Couldn't allocate memory for iLBC encoding instance\n");
1258 exit(0);
1259 }
1260 if ((enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==480)) {
1261 } else {
1262 printf("\nError - iLBC only supports 160, 240, 320 and 480 enc_frameSize (20, 30, 40 and 60 ms)\n");
1263 exit(0);
1264 }
1265 if ((enc_frameSize==160)||(enc_frameSize==320)) {
1266 /* 20 ms version */
1267 WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
1268 } else {
1269 /* 30 ms version */
1270 WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
1271 }
1272 } else {
1273 printf("\nError - iLBC is only developed for 8kHz \n");
1274 exit(0);
1275 }
1276 break;
1277#endif
1278#ifdef CODEC_ISAC
1279 case webrtc::kDecoderISAC:
1280 if (sampfreq==16000) {
1281 ok=WebRtcIsac_Create(&ISAC_inst[k]);
1282 if (ok!=0) {
1283 printf("Error: Couldn't allocate memory for iSAC instance\n");
1284 exit(0);
1285 }if ((enc_frameSize==480)||(enc_frameSize==960)) {
1286 } else {
1287 printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
1288 exit(0);
1289 }
1290 WebRtcIsac_EncoderInit(ISAC_inst[k],1);
1291 if ((bitrate<10000)||(bitrate>32000)) {
1292 printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
1293 exit(0);
1294 }
1295 WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
1296 } else {
1297 printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
1298 exit(0);
1299 }
1300 break;
1301#endif
1302#ifdef NETEQ_ISACFIX_CODEC
1303 case webrtc::kDecoderISAC:
1304 if (sampfreq==16000) {
1305 ok=WebRtcIsacfix_Create(&ISAC_inst[k]);
1306 if (ok!=0) {
1307 printf("Error: Couldn't allocate memory for iSAC instance\n");
1308 exit(0);
1309 }if ((enc_frameSize==480)||(enc_frameSize==960)) {
1310 } else {
1311 printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
1312 exit(0);
1313 }
1314 WebRtcIsacfix_EncoderInit(ISAC_inst[k],1);
1315 if ((bitrate<10000)||(bitrate>32000)) {
1316 printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
1317 exit(0);
1318 }
1319 WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
1320 } else {
1321 printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
1322 exit(0);
1323 }
1324 break;
1325#endif
1326#ifdef CODEC_ISAC_SWB
1327 case webrtc::kDecoderISACswb:
1328 if (sampfreq==32000) {
1329 ok=WebRtcIsac_Create(&ISACSWB_inst[k]);
1330 if (ok!=0) {
1331 printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
1332 exit(0);
1333 }if (enc_frameSize==960) {
1334 } else {
1335 printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
1336 exit(0);
1337 }
1338 ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
1339 if (ok!=0) {
1340 printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
1341 exit(0);
1342 }
1343 WebRtcIsac_EncoderInit(ISACSWB_inst[k],1);
1344 if ((bitrate<32000)||(bitrate>56000)) {
1345 printf("\nError - iSAC SWB bitrate has to be between 32000 and 56000 bps (not %i)\n", bitrate);
1346 exit(0);
1347 }
1348 WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize>>5);
1349 } else {
1350 printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n");
1351 exit(0);
1352 }
1353 break;
1354#endif
1355#ifdef CODEC_GSMFR
1356 case webrtc::kDecoderGSMFR:
1357 if (sampfreq==8000) {
1358 ok=WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
1359 if (ok!=0) {
1360 printf("Error: Couldn't allocate memory for GSM FR encoding instance\n");
1361 exit(0);
1362 }
1363 if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
1364 } else {
1365 printf("\nError - GSM FR must have a multiple of 160 enc_frameSize\n");
1366 exit(0);
1367 }
1368 WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
1369 } else {
1370 printf("\nError - GSM FR is only developed for 8kHz \n");
1371 exit(0);
1372 }
1373 break;
1374#endif
1375 default :
1376 printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
1377 exit(0);
1378 break;
1379 }
1380
1381 if (ok != 0) {
1382 return(ok);
1383 }
pbos@webrtc.orgd900e8b2013-07-03 15:12:26 +00001384 } // end for
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001385
1386 return(0);
1387}
1388
1389
1390
1391
1392int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels) {
1393
1394 for (int k = 0; k < numChannels; k++)
1395 {
1396 WebRtcVad_Free(VAD_inst[k]);
1397#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
1398 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
1399 WebRtcCng_FreeEnc(CNGenc_inst[k]);
1400#endif
1401
1402 switch (coder)
1403 {
1404#ifdef CODEC_PCM16B
1405 case webrtc::kDecoderPCM16B :
1406#endif
1407#ifdef CODEC_PCM16B_WB
1408 case webrtc::kDecoderPCM16Bwb :
1409#endif
1410#ifdef CODEC_PCM16B_32KHZ
1411 case webrtc::kDecoderPCM16Bswb32kHz :
1412#endif
1413#ifdef CODEC_PCM16B_48KHZ
1414 case webrtc::kDecoderPCM16Bswb48kHz :
1415#endif
1416#ifdef CODEC_G711
1417 case webrtc::kDecoderPCMu :
1418 case webrtc::kDecoderPCMa :
1419#endif
1420 // do nothing
1421 break;
1422#ifdef CODEC_G729
1423 case webrtc::kDecoderG729:
1424 WebRtcG729_FreeEnc(G729enc_inst[k]);
1425 break;
1426#endif
1427#ifdef CODEC_G729_1
1428 case webrtc::kDecoderG729_1:
1429 WebRtcG7291_Free(G729_1_inst[k]);
1430 break;
1431#endif
1432#ifdef CODEC_SPEEX_8
1433 case webrtc::kDecoderSPEEX_8 :
1434 WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
1435 break;
1436#endif
1437#ifdef CODEC_SPEEX_16
1438 case webrtc::kDecoderSPEEX_16 :
1439 WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
1440 break;
1441#endif
1442#ifdef CODEC_CELT_32
1443 case webrtc::kDecoderCELT_32 :
1444 WebRtcCelt_FreeEnc(CELT32enc_inst[k]);
1445 break;
1446#endif
1447
1448#ifdef CODEC_G722_1_16
1449 case webrtc::kDecoderG722_1_16 :
1450 WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
1451 break;
1452#endif
1453#ifdef CODEC_G722_1_24
1454 case webrtc::kDecoderG722_1_24 :
1455 WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
1456 break;
1457#endif
1458#ifdef CODEC_G722_1_32
1459 case webrtc::kDecoderG722_1_32 :
1460 WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
1461 break;
1462#endif
1463#ifdef CODEC_G722_1C_24
1464 case webrtc::kDecoderG722_1C_24 :
1465 WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
1466 break;
1467#endif
1468#ifdef CODEC_G722_1C_32
1469 case webrtc::kDecoderG722_1C_32 :
1470 WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
1471 break;
1472#endif
1473#ifdef CODEC_G722_1C_48
1474 case webrtc::kDecoderG722_1C_48 :
1475 WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
1476 break;
1477#endif
1478#ifdef CODEC_G722
1479 case webrtc::kDecoderG722 :
1480 WebRtcG722_FreeEncoder(g722EncState[k]);
1481 break;
1482#endif
1483#ifdef CODEC_AMR
1484 case webrtc::kDecoderAMR :
1485 WebRtcAmr_FreeEnc(AMRenc_inst[k]);
1486 break;
1487#endif
1488#ifdef CODEC_AMRWB
1489 case webrtc::kDecoderAMRWB :
1490 WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
1491 break;
1492#endif
1493#ifdef CODEC_ILBC
1494 case webrtc::kDecoderILBC :
1495 WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
1496 break;
1497#endif
1498#ifdef CODEC_ISAC
1499 case webrtc::kDecoderISAC:
1500 WebRtcIsac_Free(ISAC_inst[k]);
1501 break;
1502#endif
1503#ifdef NETEQ_ISACFIX_CODEC
1504 case webrtc::kDecoderISAC:
1505 WebRtcIsacfix_Free(ISAC_inst[k]);
1506 break;
1507#endif
1508#ifdef CODEC_ISAC_SWB
1509 case webrtc::kDecoderISACswb:
1510 WebRtcIsac_Free(ISACSWB_inst[k]);
1511 break;
1512#endif
1513#ifdef CODEC_GSMFR
1514 case webrtc::kDecoderGSMFR:
1515 WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
1516 break;
1517#endif
1518 default :
1519 printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
1520 exit(0);
1521 break;
1522 }
1523 }
1524
1525 return(0);
1526}
1527
1528
1529
1530
1531
1532
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001533int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate ,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001534 int * vad, int useVAD, int bitrate, int numChannels){
1535
1536 short cdlen = 0;
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001537 int16_t *tempdata;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001538 static int first_cng=1;
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001539 int16_t tempLen;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001540
1541 *vad =1;
1542
1543 // check VAD first
1544 if(useVAD)
1545 {
1546 *vad = 0;
1547
1548 for (int k = 0; k < numChannels; k++)
1549 {
1550 tempLen = frameLen;
1551 tempdata = &indata[k*frameLen];
1552 int localVad=0;
1553 /* Partition the signal and test each chunk for VAD.
1554 All chunks must be VAD=0 to produce a total VAD=0. */
1555 while (tempLen >= 10*sampleRate/1000) {
1556 if ((tempLen % 30*sampleRate/1000) == 0) { // tempLen is multiple of 30ms
1557 localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 30*sampleRate/1000);
1558 tempdata += 30*sampleRate/1000;
1559 tempLen -= 30*sampleRate/1000;
1560 }
1561 else if (tempLen >= 20*sampleRate/1000) { // tempLen >= 20ms
1562 localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 20*sampleRate/1000);
1563 tempdata += 20*sampleRate/1000;
1564 tempLen -= 20*sampleRate/1000;
1565 }
1566 else { // use 10ms
1567 localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 10*sampleRate/1000);
1568 tempdata += 10*sampleRate/1000;
1569 tempLen -= 10*sampleRate/1000;
1570 }
1571 }
1572
1573 // aggregate all VAD decisions over all channels
1574 *vad |= localVad;
1575 }
1576
1577 if(!*vad){
1578 // all channels are silent
1579 cdlen = 0;
1580 for (int k = 0; k < numChannels; k++)
1581 {
1582 WebRtcCng_Encode(CNGenc_inst[k],&indata[k*frameLen], (frameLen <= 640 ? frameLen : 640) /* max 640 */,
1583 encoded,&tempLen,first_cng);
1584 encoded += tempLen;
1585 cdlen += tempLen;
1586 }
1587 *vad=0;
1588 first_cng=0;
1589 return(cdlen);
1590 }
1591 }
1592
1593
1594 // loop over all channels
1595 int totalLen = 0;
1596
1597 for (int k = 0; k < numChannels; k++)
1598 {
1599 /* Encode with the selected coder type */
1600 if (coder==webrtc::kDecoderPCMu) { /*g711 u-law */
1601#ifdef CODEC_G711
kwiberg@webrtc.orgc78cf972014-11-04 13:23:36 +00001602 cdlen = WebRtcG711_EncodeU(indata, frameLen, (int16_t*) encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001603#endif
1604 }
1605 else if (coder==webrtc::kDecoderPCMa) { /*g711 A-law */
1606#ifdef CODEC_G711
kwiberg@webrtc.orgc78cf972014-11-04 13:23:36 +00001607 cdlen = WebRtcG711_EncodeA(indata, frameLen, (int16_t*) encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001608 }
1609#endif
1610#ifdef CODEC_PCM16B
1611 else if ((coder==webrtc::kDecoderPCM16B)||(coder==webrtc::kDecoderPCM16Bwb)||
1612 (coder==webrtc::kDecoderPCM16Bswb32kHz)||(coder==webrtc::kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001613 cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (int16_t*) encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001614 }
1615#endif
1616#ifdef CODEC_G722
1617 else if (coder==webrtc::kDecoderG722) { /*g722 */
kwiberg@webrtc.org0cd55582014-12-02 11:45:51 +00001618 cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded);
turaj@webrtc.org58cd3162013-10-31 15:15:55 +00001619 assert(cdlen == frameLen>>1);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001620 }
1621#endif
1622#ifdef CODEC_ILBC
1623 else if (coder==webrtc::kDecoderILBC) { /*iLBC */
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001624 cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(int16_t*)encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001625 }
1626#endif
1627#if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
1628 else if (coder==webrtc::kDecoderISAC) { /*iSAC */
1629 int noOfCalls=0;
1630 cdlen=0;
1631 while (cdlen<=0) {
1632#ifdef CODEC_ISAC /* floating point */
kwiberg@webrtc.org7ee24a72014-09-24 10:31:02 +00001633 cdlen = WebRtcIsac_Encode(ISAC_inst[k],
1634 &indata[noOfCalls * 160],
1635 encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001636#else /* fixed point */
kwiberg@webrtc.org7ee24a72014-09-24 10:31:02 +00001637 cdlen = WebRtcIsacfix_Encode(ISAC_inst[k],
1638 &indata[noOfCalls * 160],
1639 encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001640#endif
1641 noOfCalls++;
1642 }
1643 }
1644#endif
1645#ifdef CODEC_ISAC_SWB
1646 else if (coder==webrtc::kDecoderISACswb) { /* iSAC SWB */
1647 int noOfCalls=0;
1648 cdlen=0;
1649 while (cdlen<=0) {
kwiberg@webrtc.org7ee24a72014-09-24 10:31:02 +00001650 cdlen = WebRtcIsac_Encode(ISACSWB_inst[k],
1651 &indata[noOfCalls * 320],
1652 encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001653 noOfCalls++;
1654 }
1655 }
1656#endif
1657#ifdef CODEC_CELT_32
1658 else if (coder==webrtc::kDecoderCELT_32) { /* Celt */
1659 int encodedLen = 0;
1660 cdlen = 0;
1661 while (cdlen <= 0) {
1662 cdlen = WebRtcCelt_Encode(CELT32enc_inst[k], &indata[encodedLen], encoded);
1663 encodedLen += 10*32; /* 10 ms */
1664 }
1665 if( (encodedLen != frameLen) || cdlen < 0) {
1666 printf("Error encoding Celt frame!\n");
1667 exit(0);
1668 }
1669 }
1670#endif
1671
1672 indata += frameLen;
1673 encoded += cdlen;
1674 totalLen += cdlen;
1675
pbos@webrtc.orgd900e8b2013-07-03 15:12:26 +00001676 } // end for
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001677
1678 first_cng=1;
1679 return(totalLen);
1680}
1681
1682
1683
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001684void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc){
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001685
1686 rtp_data[0]=(unsigned char)0x80;
1687 rtp_data[1]=(unsigned char)(payloadType & 0xFF);
1688 rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF);
1689 rtp_data[3]=(unsigned char)((seqNo)&0xFF);
1690 rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF);
1691 rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF);
1692
1693 rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF);
1694 rtp_data[7]=(unsigned char)(timestamp & 0xFF);
1695
1696 rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF);
1697 rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF);
1698
1699 rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF);
1700 rtp_data[11]=(unsigned char)(ssrc & 0xFF);
1701}
1702
1703
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001704int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen,
1705 int seqNo, uint32_t ssrc)
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001706{
1707
1708 int i;
1709 unsigned char *rtpPointer;
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001710 uint16_t offset;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001711
1712 /* first create "standard" RTP header */
1713 makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc);
1714
1715 rtpPointer = &rtp_data[12];
1716
1717 /* add one sub-header for each redundant payload (not the primary) */
1718 for(i=0; i<numPayloads-1; i++) { /* |0 1 2 3 4 5 6 7| */
1719 if(blockLen[i] > 0) {
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001720 offset = (uint16_t) (timestamp[numPayloads-1] - timestamp[i]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001721
1722 rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F| block PT | */
1723 rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF); /* | timestamp- | */
1724 rtpPointer[2] = (unsigned char) ( ((offset & 0x3F)<<2) |
1725 ( (blockLen[i]>>8) & 0x03 ) ); /* | -offset |bl-| */
1726 rtpPointer[3] = (unsigned char) ( blockLen[i] & 0xFF ); /* | -ock length | */
1727
1728 rtpPointer += 4;
1729 }
1730 }
1731
1732 /* last sub-header */
1733 rtpPointer[0]= (unsigned char) (0x00 | (0x7F&payloadType[numPayloads-1]));/* |F| block PT | */
1734 rtpPointer += 1;
1735
1736 return(rtpPointer - rtp_data); /* length of header in bytes */
1737}
1738
1739
1740
1741int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) {
1742 unsigned char E,R,V;
1743 R=0;
1744 V=(unsigned char)Volume;
1745 if (End==0) {
1746 E = 0x00;
1747 } else {
1748 E = 0x80;
1749 }
1750 payload_data[0]=(unsigned char)Event;
1751 payload_data[1]=(unsigned char)(E|R|V);
1752 //Duration equals 8 times time_ms, default is 8000 Hz.
1753 payload_data[2]=(unsigned char)((Duration>>8)&0xFF);
1754 payload_data[3]=(unsigned char)(Duration&0xFF);
1755 return(4);
1756}
1757
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001758void stereoDeInterleave(int16_t* audioSamples, int numSamples)
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001759{
1760
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001761 int16_t *tempVec;
1762 int16_t *readPtr, *writeL, *writeR;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001763
1764 if (numSamples <= 0)
1765 return;
1766
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001767 tempVec = (int16_t *) malloc(sizeof(int16_t) * numSamples);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001768 if (tempVec == NULL) {
1769 printf("Error allocating memory\n");
1770 exit(0);
1771 }
1772
pbos@webrtc.org0946a562013-04-09 00:28:06 +00001773 memcpy(tempVec, audioSamples, numSamples*sizeof(int16_t));
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001774
1775 writeL = audioSamples;
1776 writeR = &audioSamples[numSamples/2];
1777 readPtr = tempVec;
1778
1779 for (int k = 0; k < numSamples; k += 2)
1780 {
1781 *writeL = *readPtr;
1782 readPtr++;
1783 *writeR = *readPtr;
1784 readPtr++;
1785 writeL++;
1786 writeR++;
1787 }
1788
1789 free(tempVec);
1790
1791}
1792
1793
1794void stereoInterleave(unsigned char* data, int dataLen, int stride)
1795{
1796
1797 unsigned char *ptrL, *ptrR;
1798 unsigned char temp[10];
1799
1800 if (stride > 10)
1801 {
1802 exit(0);
1803 }
1804
1805 if (dataLen%1 != 0)
1806 {
1807 // must be even number of samples
1808 printf("Error: cannot interleave odd sample number\n");
1809 exit(0);
1810 }
1811
1812 ptrL = data + stride;
1813 ptrR = &data[dataLen/2];
1814
1815 while (ptrL < ptrR) {
1816 // copy from right pointer to temp
1817 memcpy(temp, ptrR, stride);
1818
1819 // shift data between pointers
1820 memmove(ptrL + stride, ptrL, ptrR - ptrL);
1821
1822 // copy from temp to left pointer
1823 memcpy(ptrL, temp, stride);
1824
1825 // advance pointers
1826 ptrL += stride*2;
1827 ptrR += stride;
1828 }
1829
1830}