blob: 87f9494f0e5f1ef7417570c8089a85f51c47fc46 [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
Peter Kasting248b0b02015-06-03 12:32:41 -070011// TODO(hlundin): Reformat file to meet style guide.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000012
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
kwiberg@webrtc.orge04a93b2014-12-09 10:12:53 +000028#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000029#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
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000037/*********************/
38/* Misc. definitions */
39/*********************/
40
41#define STOPSENDTIME 3000
Peter Kasting248b0b02015-06-03 12:32:41 -070042#define RESTARTSENDTIME 0 // 162500
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000043#define FIRSTLINELEN 40
Peter Kasting248b0b02015-06-03 12:32:41 -070044#define CHECK_NOT_NULL(a) \
45 if ((a) == 0) { \
46 printf("\n %s \n line: %d \nerror at %s\n", __FILE__, __LINE__, #a); \
47 return (-1); \
48 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000049
50//#define MULTIPLE_SAME_TIMESTAMP
51#define REPEAT_PACKET_DISTANCE 17
52#define REPEAT_PACKET_COUNT 1 // number of extra packets to send
53
54//#define INSERT_OLD_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -070055#define OLD_PACKET 5 // how many seconds too old should the packet be?
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000056
57//#define TIMESTAMP_WRAPAROUND
58
59//#define RANDOM_DATA
60//#define RANDOM_PAYLOAD_DATA
61#define RANDOM_SEED 10
62
63//#define INSERT_DTMF_PACKETS
64//#define NO_DTMF_OVERDUB
65#define DTMF_PACKET_INTERVAL 2000
66#define DTMF_DURATION 500
67
68#define STEREO_MODE_FRAME 0
Peter Kasting248b0b02015-06-03 12:32:41 -070069#define STEREO_MODE_SAMPLE_1 1 // 1 octet per sample
70#define STEREO_MODE_SAMPLE_2 2 // 2 octets per sample
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000071
72/*************************/
73/* Function declarations */
74/*************************/
75
pkasting@chromium.orgd3245462015-02-23 21:28:22 +000076void NetEQTest_GetCodec_and_PT(char* name,
77 webrtc::NetEqDecoder* codec,
78 int* PT,
79 int frameLen,
80 int* fs,
81 int* bitrate,
82 int* useRed);
83int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
84 int enc_frameSize,
85 int bitrate,
86 int sampfreq,
87 int vad,
88 int numChannels);
89void defineCodecs(webrtc::NetEqDecoder* usedCodec, int* noOfCodecs);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000090int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels);
pkasting@chromium.orgd3245462015-02-23 21:28:22 +000091int NetEQTest_encode(int coder,
92 int16_t* indata,
93 int frameLen,
94 unsigned char* encoded,
95 int sampleRate,
96 int* vad,
97 int useVAD,
98 int bitrate,
99 int numChannels);
100void makeRTPheader(unsigned char* rtp_data,
101 int payloadType,
102 int seqNo,
103 uint32_t timestamp,
104 uint32_t ssrc);
105int makeRedundantHeader(unsigned char* rtp_data,
106 int* payloadType,
107 int numPayloads,
108 uint32_t* timestamp,
109 uint16_t* blockLen,
110 int seqNo,
111 uint32_t ssrc);
112int makeDTMFpayload(unsigned char* payload_data,
113 int Event,
114 int End,
115 int Volume,
116 int Duration);
pbos@webrtc.org0946a562013-04-09 00:28:06 +0000117void stereoDeInterleave(int16_t* audioSamples, int numSamples);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000118void stereoInterleave(unsigned char* data, int dataLen, int stride);
119
120/*********************/
121/* Codec definitions */
122/*********************/
123
124#include "webrtc_vad.h"
125
Peter Kasting248b0b02015-06-03 12:32:41 -0700126#if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
127#include "pcm16b.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000128#endif
129#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700130#include "g711_interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000131#endif
132#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700133#include "G729Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000134#endif
135#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -0700136#include "G729_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000137#endif
138#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -0700139#include "AMRInterface.h"
140#include "AMRCreation.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000141#endif
142#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700143#include "AMRWBInterface.h"
144#include "AMRWBCreation.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000145#endif
146#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -0700147#include "ilbc.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000148#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700149#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
150#include "isac.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000151#endif
152#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -0700153#include "isacfix.h"
154#ifdef CODEC_ISAC
155#error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp
156#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000157#endif
158#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -0700159#include "g722_interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000160#endif
161#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700162#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000163#endif
164#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700165#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000166#endif
167#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700168#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000169#endif
170#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700171#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000172#endif
173#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700174#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000175#endif
176#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -0700177#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000178#endif
179#ifdef CODEC_G726
Peter Kasting248b0b02015-06-03 12:32:41 -0700180#include "G726Creation.h"
181#include "G726Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000182#endif
183#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -0700184#include "GSMFRInterface.h"
185#include "GSMFRCreation.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000186#endif
187#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700188 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
189#include "webrtc_cng.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000190#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700191#if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
192#include "SpeexInterface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000193#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000194
195/***********************************/
196/* Global codec instance variables */
197/***********************************/
198
Peter Kasting248b0b02015-06-03 12:32:41 -0700199WebRtcVadInst* VAD_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000200
201#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -0700202G722EncInst* g722EncState[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000203#endif
204
205#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700206G722_1_24_encinst_t* G722_1_24enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000207#endif
208#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700209G722_1_32_encinst_t* G722_1_32enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000210#endif
211#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700212G722_1_16_encinst_t* G722_1_16enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000213#endif
214#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700215G722_1C_24_encinst_t* G722_1C_24enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000216#endif
217#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700218G722_1C_32_encinst_t* G722_1C_32enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000219#endif
220#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -0700221G722_1C_48_encinst_t* G722_1C_48enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000222#endif
223#ifdef CODEC_G726
Peter Kasting248b0b02015-06-03 12:32:41 -0700224G726_encinst_t* G726enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000225#endif
226#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700227G729_encinst_t* G729enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000228#endif
229#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -0700230G729_1_inst_t* G729_1_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000231#endif
232#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -0700233AMR_encinst_t* AMRenc_inst[2];
234int16_t AMR_bitrate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000235#endif
236#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700237AMRWB_encinst_t* AMRWBenc_inst[2];
238int16_t AMRWB_bitrate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000239#endif
240#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -0700241IlbcEncoderInstance* iLBCenc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000242#endif
243#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -0700244ISACStruct* ISAC_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000245#endif
246#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -0700247ISACFIX_MainStruct* ISAC_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000248#endif
249#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700250ISACStruct* ISACSWB_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000251#endif
252#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -0700253GSMFR_encinst_t* GSMFRenc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000254#endif
255#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700256 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
257CNG_enc_inst* CNGenc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000258#endif
259#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -0700260SPEEX_encinst_t* SPEEX8enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000261#endif
262#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700263SPEEX_encinst_t* SPEEX16enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000264#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000265
Peter Kasting248b0b02015-06-03 12:32:41 -0700266int main(int argc, char* argv[]) {
267 int packet_size, fs;
268 webrtc::NetEqDecoder usedCodec;
269 int payloadType;
270 int bitrate = 0;
271 int useVAD, vad;
272 int useRed = 0;
273 int len, enc_len;
274 int16_t org_data[4000];
275 unsigned char rtp_data[8000];
276 int16_t seqNo = 0xFFF;
277 uint32_t ssrc = 1235412312;
278 uint32_t timestamp = 0xAC1245;
279 uint16_t length, plen;
280 uint32_t offset;
281 double sendtime = 0;
282 int red_PT[2] = {0};
283 uint32_t red_TS[2] = {0};
284 uint16_t red_len[2] = {0};
285 int RTPheaderLen = 12;
286 uint8_t red_data[8000];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000287#ifdef INSERT_OLD_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700288 uint16_t old_length, old_plen;
289 int old_enc_len;
290 int first_old_packet = 1;
291 unsigned char old_rtp_data[8000];
292 int packet_age = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000293#endif
294#ifdef INSERT_DTMF_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700295 int NTone = 1;
296 int DTMFfirst = 1;
297 uint32_t DTMFtimestamp;
298 bool dtmfSent = false;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000299#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700300 bool usingStereo = false;
301 int stereoMode = 0;
302 int numChannels = 1;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000303
Peter Kasting248b0b02015-06-03 12:32:41 -0700304 /* check number of parameters */
305 if ((argc != 6) && (argc != 7)) {
306 /* print help text and exit */
307 printf("Application to encode speech into an RTP stream.\n");
308 printf(
309 "The program reads a PCM file and encodes is using the specified "
310 "codec.\n");
311 printf(
312 "The coded speech is packetized in RTP packest and written to the "
313 "output file.\n");
314 printf(
315 "The format of the RTP stream file is simlilar to that of rtpplay,\n");
316 printf("but with the receive time euqal to 0 for all packets.\n");
317 printf("Usage:\n\n");
318 printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
319 printf("where:\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000320
Peter Kasting248b0b02015-06-03 12:32:41 -0700321 printf("PCMfile : PCM speech input file\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000322
Peter Kasting248b0b02015-06-03 12:32:41 -0700323 printf("RTPfile : RTP stream output file\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000324
Peter Kasting248b0b02015-06-03 12:32:41 -0700325 printf(
326 "frameLen : 80...960... Number of samples per packet (limit "
327 "depends on codec)\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000328
Peter Kasting248b0b02015-06-03 12:32:41 -0700329 printf("codecName\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000330#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -0700331 printf(" : pcm16b 16 bit PCM (8kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000332#endif
333#ifdef CODEC_PCM16B_WB
Peter Kasting248b0b02015-06-03 12:32:41 -0700334 printf(" : pcm16b_wb 16 bit PCM (16kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000335#endif
336#ifdef CODEC_PCM16B_32KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -0700337 printf(" : pcm16b_swb32 16 bit PCM (32kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000338#endif
339#ifdef CODEC_PCM16B_48KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -0700340 printf(" : pcm16b_swb48 16 bit PCM (48kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000341#endif
342#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700343 printf(" : pcma g711 A-law (8kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000344#endif
345#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700346 printf(" : pcmu g711 u-law (8kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000347#endif
348#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700349 printf(
350 " : g729 G729 (8kHz and 8kbps) CELP (One-Three "
351 "frame(s)/packet)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000352#endif
353#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -0700354 printf(
355 " : g729.1 G729.1 (16kHz) variable rate (8--32 "
356 "kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000357#endif
358#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700359 printf(
360 " : g722.1_16 G722.1 coder (16kHz) (g722.1 with "
361 "16kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000362#endif
363#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700364 printf(
365 " : g722.1_24 G722.1 coder (16kHz) (the 24kbps "
366 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000367#endif
368#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700369 printf(
370 " : g722.1_32 G722.1 coder (16kHz) (the 32kbps "
371 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000372#endif
373#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700374 printf(
375 " : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps "
376 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000377#endif
378#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700379 printf(
380 " : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps "
381 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000382#endif
383#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -0700384 printf(
385 " : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000386#endif
387
388#ifdef CODEC_G726
Peter Kasting248b0b02015-06-03 12:32:41 -0700389 printf(" : g726_16 G726 coder (8kHz) 16kbps\n");
390 printf(" : g726_24 G726 coder (8kHz) 24kbps\n");
391 printf(" : g726_32 G726 coder (8kHz) 32kbps\n");
392 printf(" : g726_40 G726 coder (8kHz) 40kbps\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000393#endif
394#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -0700395 printf(
396 " : AMRXk Adaptive Multi Rate CELP codec (8kHz)\n");
397 printf(
398 " X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 "
399 "or 12.2\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000400#endif
401#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700402 printf(
403 " : AMRwbXk Adaptive Multi Rate Wideband CELP codec "
404 "(16kHz)\n");
405 printf(
406 " X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000407#endif
408#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -0700409 printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000410#endif
411#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -0700412 printf(
413 " : isac iSAC (16kHz and 32.0 kbps). To set rate "
414 "specify a rate parameter as last parameter\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000415#endif
416#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700417 printf(
418 " : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To "
419 "set rate specify a rate parameter as last parameter\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000420#endif
421#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -0700422 printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000423#endif
424#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -0700425 printf(
426 " : g722 g722 coder (16kHz) (the 64kbps "
427 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000428#endif
429#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -0700430 printf(" : speex8 speex coder (8 kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000431#endif
432#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700433 printf(" : speex16 speex coder (16 kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000434#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000435#ifdef CODEC_RED
436#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700437 printf(
438 " : red_pcm Redundancy RTP packet with 2*G711A "
439 "frames\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000440#endif
441#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -0700442 printf(
443 " : red_isac Redundancy RTP packet with 2*iSAC "
444 "frames\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000445#endif
446#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700447 printf("\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000448
449#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700450 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
451 printf("useVAD : 0 Voice Activity Detection is switched off\n");
452 printf(" : 1 Voice Activity Detection is switched on\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000453#else
Peter Kasting248b0b02015-06-03 12:32:41 -0700454 printf(
455 "useVAD : 0 Voice Activity Detection switched off (on not "
456 "supported)\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000457#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700458 printf(
459 "bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000460
Peter Kasting248b0b02015-06-03 12:32:41 -0700461 return (0);
462 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000463
Peter Kasting248b0b02015-06-03 12:32:41 -0700464 FILE* in_file = fopen(argv[1], "rb");
465 CHECK_NOT_NULL(in_file);
466 printf("Input file: %s\n", argv[1]);
467 FILE* out_file = fopen(argv[2], "wb");
468 CHECK_NOT_NULL(out_file);
469 printf("Output file: %s\n\n", argv[2]);
470 packet_size = atoi(argv[3]);
471 CHECK_NOT_NULL(packet_size);
472 printf("Packet size: %i\n", packet_size);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000473
Peter Kasting248b0b02015-06-03 12:32:41 -0700474 // check for stereo
475 if (argv[4][strlen(argv[4]) - 1] == '*') {
476 // use stereo
477 usingStereo = true;
478 numChannels = 2;
479 argv[4][strlen(argv[4]) - 1] = '\0';
480 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000481
Peter Kasting248b0b02015-06-03 12:32:41 -0700482 NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs,
483 &bitrate, &useRed);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000484
Peter Kasting248b0b02015-06-03 12:32:41 -0700485 if (useRed) {
486 RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant
487 payload, except last one which is 1 byte */
488 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000489
Peter Kasting248b0b02015-06-03 12:32:41 -0700490 useVAD = atoi(argv[5]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000491#if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700492 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
493 if (useVAD != 0) {
494 printf("Error: this simulation does not support VAD/DTX/CNG\n");
495 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000496#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000497
Peter Kasting248b0b02015-06-03 12:32:41 -0700498 // check stereo type
499 if (usingStereo) {
500 switch (usedCodec) {
501 // sample based codecs
502 case webrtc::kDecoderPCMu:
503 case webrtc::kDecoderPCMa:
504 case webrtc::kDecoderG722: {
505 // 1 octet per sample
506 stereoMode = STEREO_MODE_SAMPLE_1;
507 break;
508 }
509 case webrtc::kDecoderPCM16B:
510 case webrtc::kDecoderPCM16Bwb:
511 case webrtc::kDecoderPCM16Bswb32kHz:
512 case webrtc::kDecoderPCM16Bswb48kHz: {
513 // 2 octets per sample
514 stereoMode = STEREO_MODE_SAMPLE_2;
515 break;
516 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000517
Peter Kasting248b0b02015-06-03 12:32:41 -0700518 // fixed-rate frame codecs (with internal VAD)
519 default: {
520 printf("Cannot use codec %s as stereo codec\n", argv[4]);
521 exit(0);
522 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000523 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700524 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000525
Peter Kasting248b0b02015-06-03 12:32:41 -0700526 if ((usedCodec == webrtc::kDecoderISAC) ||
527 (usedCodec == webrtc::kDecoderISACswb)) {
528 if (argc != 7) {
529 if (usedCodec == webrtc::kDecoderISAC) {
530 bitrate = 32000;
531 printf(
532 "Running iSAC at default bitrate of 32000 bps (to specify "
533 "explicitly add the bps as last parameter)\n");
534 } else // (usedCodec==webrtc::kDecoderISACswb)
535 {
536 bitrate = 56000;
537 printf(
538 "Running iSAC at default bitrate of 56000 bps (to specify "
539 "explicitly add the bps as last parameter)\n");
540 }
541 } else {
542 bitrate = atoi(argv[6]);
543 if (usedCodec == webrtc::kDecoderISAC) {
544 if ((bitrate < 10000) || (bitrate > 32000)) {
545 printf(
546 "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is "
547 "invalid)\n",
548 bitrate);
549 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000550 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700551 printf("Running iSAC at bitrate of %i bps\n", bitrate);
552 } else // (usedCodec==webrtc::kDecoderISACswb)
553 {
554 if ((bitrate < 32000) || (bitrate > 56000)) {
555 printf(
556 "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i "
557 "is invalid)\n",
558 bitrate);
559 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000560 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700561 }
562 }
563 } else {
564 if (argc == 7) {
565 printf(
566 "Error: Bitrate parameter can only be specified for iSAC, G.723, and "
567 "G.729.1\n");
568 exit(0);
569 }
570 }
571
572 if (useRed) {
573 printf("Redundancy engaged. ");
574 }
575 printf("Used codec: %i\n", usedCodec);
576 printf("Payload type: %i\n", payloadType);
577
578 NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD,
579 numChannels);
580
581 /* write file header */
582 // fprintf(out_file, "#!RTPencode%s\n", "1.0");
583 fprintf(out_file, "#!rtpplay%s \n",
584 "1.0"); // this is the string that rtpplay needs
585 uint32_t dummy_variable = 0; // should be converted to network endian format,
586 // but does not matter when 0
587 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
588 return -1;
589 }
590 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
591 return -1;
592 }
593 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
594 return -1;
595 }
596 if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
597 return -1;
598 }
599 if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
600 return -1;
601 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000602
603#ifdef TIMESTAMP_WRAPAROUND
Peter Kasting248b0b02015-06-03 12:32:41 -0700604 timestamp = 0xFFFFFFFF - fs * 10; /* should give wrap-around in 10 seconds */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000605#endif
606#if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
Peter Kasting248b0b02015-06-03 12:32:41 -0700607 srand(RANDOM_SEED);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000608#endif
609
Peter Kasting248b0b02015-06-03 12:32:41 -0700610 /* if redundancy is used, the first redundant payload is zero length */
611 red_len[0] = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000612
Peter Kasting248b0b02015-06-03 12:32:41 -0700613 /* read first frame */
614 len = fread(org_data, 2, packet_size * numChannels, in_file) / numChannels;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000615
Peter Kasting248b0b02015-06-03 12:32:41 -0700616 /* de-interleave if stereo */
617 if (usingStereo) {
618 stereoDeInterleave(org_data, len * numChannels);
619 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000620
Peter Kasting248b0b02015-06-03 12:32:41 -0700621 while (len == packet_size) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000622#ifdef INSERT_DTMF_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700623 dtmfSent = false;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000624
Peter Kasting248b0b02015-06-03 12:32:41 -0700625 if (sendtime >= NTone * DTMF_PACKET_INTERVAL) {
626 if (sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION) {
627 // tone has not ended
628 if (DTMFfirst == 1) {
629 DTMFtimestamp = timestamp; // save this timestamp
630 DTMFfirst = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000631 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700632 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
633 enc_len = makeDTMFpayload(
634 &rtp_data[12], NTone % 12, 0, 4,
635 (int)(sendtime - NTone * DTMF_PACKET_INTERVAL) * (fs / 1000) + len);
636 } else {
637 // tone has ended
638 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
639 enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4,
640 DTMF_DURATION * (fs / 1000));
641 NTone++;
642 DTMFfirst = 1;
643 }
644
645 /* write RTP packet to file */
646 length = htons(12 + enc_len + 8);
647 plen = htons(12 + enc_len);
648 offset = (uint32_t)sendtime; //(timestamp/(fs/1000));
649 offset = htonl(offset);
650 if (fwrite(&length, 2, 1, out_file) != 1) {
651 return -1;
652 }
653 if (fwrite(&plen, 2, 1, out_file) != 1) {
654 return -1;
655 }
656 if (fwrite(&offset, 4, 1, out_file) != 1) {
657 return -1;
658 }
659 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
660 return -1;
661 }
662
663 dtmfSent = true;
664 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000665#endif
666
667#ifdef NO_DTMF_OVERDUB
Peter Kasting248b0b02015-06-03 12:32:41 -0700668 /* If DTMF is sent, we should not send any speech packets during the same
669 * time */
670 if (dtmfSent) {
671 enc_len = 0;
672 } else {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000673#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700674 /* encode frame */
675 enc_len =
676 NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12], fs,
677 &vad, useVAD, bitrate, numChannels);
678 if (enc_len == -1) {
679 printf("Error encoding frame\n");
680 exit(0);
681 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000682
Peter Kasting248b0b02015-06-03 12:32:41 -0700683 if (usingStereo && stereoMode != STEREO_MODE_FRAME && vad == 1) {
684 // interleave the encoded payload for sample-based codecs (not for CNG)
685 stereoInterleave(&rtp_data[12], enc_len, stereoMode);
686 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000687#ifdef NO_DTMF_OVERDUB
Peter Kasting248b0b02015-06-03 12:32:41 -0700688 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000689#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000690
Peter Kasting248b0b02015-06-03 12:32:41 -0700691 if (enc_len > 0 &&
692 (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
693 if (useRed) {
694 if (red_len[0] > 0) {
695 memmove(&rtp_data[RTPheaderLen + red_len[0]], &rtp_data[12], enc_len);
696 memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000697
Peter Kasting248b0b02015-06-03 12:32:41 -0700698 red_len[1] = enc_len;
699 red_TS[1] = timestamp;
700 if (vad)
701 red_PT[1] = payloadType;
702 else
703 red_PT[1] = NETEQ_CODEC_CN_PT;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000704
Peter Kasting248b0b02015-06-03 12:32:41 -0700705 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++,
706 ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000707
Peter Kasting248b0b02015-06-03 12:32:41 -0700708 enc_len += red_len[0] + RTPheaderLen - 12;
709 } else { // do not use redundancy payload for this packet, i.e., only
710 // last payload
711 memmove(&rtp_data[RTPheaderLen - 4], &rtp_data[12], enc_len);
712 // memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000713
Peter Kasting248b0b02015-06-03 12:32:41 -0700714 red_len[1] = enc_len;
715 red_TS[1] = timestamp;
716 if (vad)
717 red_PT[1] = payloadType;
718 else
719 red_PT[1] = NETEQ_CODEC_CN_PT;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000720
Peter Kasting248b0b02015-06-03 12:32:41 -0700721 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++,
722 ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000723
Peter Kasting248b0b02015-06-03 12:32:41 -0700724 enc_len += red_len[0] + RTPheaderLen - 4 -
725 12; // 4 is length of redundancy header (not used)
726 }
727 } else {
728 /* make RTP header */
729 if (vad) // regular speech data
730 makeRTPheader(rtp_data, payloadType, seqNo++, timestamp, ssrc);
731 else // CNG data
732 makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++, timestamp, ssrc);
733 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000734#ifdef MULTIPLE_SAME_TIMESTAMP
Peter Kasting248b0b02015-06-03 12:32:41 -0700735 int mult_pack = 0;
736 do {
737#endif // MULTIPLE_SAME_TIMESTAMP
738 /* write RTP packet to file */
739 length = htons(12 + enc_len + 8);
740 plen = htons(12 + enc_len);
741 offset = (uint32_t)sendtime;
742 //(timestamp/(fs/1000));
743 offset = htonl(offset);
744 if (fwrite(&length, 2, 1, out_file) != 1) {
745 return -1;
746 }
747 if (fwrite(&plen, 2, 1, out_file) != 1) {
748 return -1;
749 }
750 if (fwrite(&offset, 4, 1, out_file) != 1) {
751 return -1;
752 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000753#ifdef RANDOM_DATA
Peter Kasting248b0b02015-06-03 12:32:41 -0700754 for (int k = 0; k < 12 + enc_len; k++) {
755 rtp_data[k] = rand() + rand();
756 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000757#endif
758#ifdef RANDOM_PAYLOAD_DATA
Peter Kasting248b0b02015-06-03 12:32:41 -0700759 for (int k = 12; k < 12 + enc_len; k++) {
760 rtp_data[k] = rand() + rand();
761 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000762#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700763 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
764 return -1;
765 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000766#ifdef MULTIPLE_SAME_TIMESTAMP
Peter Kasting248b0b02015-06-03 12:32:41 -0700767 } while ((seqNo % REPEAT_PACKET_DISTANCE == 0) &&
768 (mult_pack++ < REPEAT_PACKET_COUNT));
769#endif // MULTIPLE_SAME_TIMESTAMP
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000770
771#ifdef INSERT_OLD_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700772 if (packet_age >= OLD_PACKET * fs) {
773 if (!first_old_packet) {
774 // send the old packet
775 if (fwrite(&old_length, 2, 1, out_file) != 1) {
776 return -1;
777 }
778 if (fwrite(&old_plen, 2, 1, out_file) != 1) {
779 return -1;
780 }
781 if (fwrite(&offset, 4, 1, out_file) != 1) {
782 return -1;
783 }
784 if (fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file) != 1) {
785 return -1;
786 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000787 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700788 // store current packet as old
789 old_length = length;
790 old_plen = plen;
791 memcpy(old_rtp_data, rtp_data, 12 + enc_len);
792 old_enc_len = enc_len;
793 first_old_packet = 0;
794 packet_age = 0;
795 }
796 packet_age += packet_size;
797#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000798
Peter Kasting248b0b02015-06-03 12:32:41 -0700799 if (useRed) {
800/* move data to redundancy store */
801#ifdef CODEC_ISAC
802 if (usedCodec == webrtc::kDecoderISAC) {
803 assert(!usingStereo); // Cannot handle stereo yet
804 red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data);
805 } else {
806#endif
807 memcpy(red_data, &rtp_data[RTPheaderLen + red_len[0]], enc_len);
808 red_len[0] = red_len[1];
809#ifdef CODEC_ISAC
810 }
811#endif
812 red_TS[0] = red_TS[1];
813 red_PT[0] = red_PT[1];
814 }
815 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000816
Peter Kasting248b0b02015-06-03 12:32:41 -0700817 /* read next frame */
818 len = fread(org_data, 2, packet_size * numChannels, in_file) / numChannels;
819 /* de-interleave if stereo */
820 if (usingStereo) {
821 stereoDeInterleave(org_data, len * numChannels);
822 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000823
Peter Kasting248b0b02015-06-03 12:32:41 -0700824 if (payloadType == NETEQ_CODEC_G722_PT)
825 timestamp += len >> 1;
826 else
827 timestamp += len;
828
829 sendtime += (double)len / (fs / 1000);
830 }
831
832 NetEQTest_free_coders(usedCodec, numChannels);
833 fclose(in_file);
834 fclose(out_file);
835 printf("Done!\n");
836
837 return (0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000838}
839
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000840/****************/
841/* Subfunctions */
842/****************/
843
pkasting@chromium.orgd3245462015-02-23 21:28:22 +0000844void NetEQTest_GetCodec_and_PT(char* name,
845 webrtc::NetEqDecoder* codec,
846 int* PT,
847 int frameLen,
848 int* fs,
849 int* bitrate,
850 int* useRed) {
Peter Kasting248b0b02015-06-03 12:32:41 -0700851 *bitrate = 0; /* Default bitrate setting */
852 *useRed = 0; /* Default no redundancy */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000853
Peter Kasting248b0b02015-06-03 12:32:41 -0700854 if (!strcmp(name, "pcmu")) {
855 *codec = webrtc::kDecoderPCMu;
856 *PT = NETEQ_CODEC_PCMU_PT;
857 *fs = 8000;
858 } else if (!strcmp(name, "pcma")) {
859 *codec = webrtc::kDecoderPCMa;
860 *PT = NETEQ_CODEC_PCMA_PT;
861 *fs = 8000;
862 } else if (!strcmp(name, "pcm16b")) {
863 *codec = webrtc::kDecoderPCM16B;
864 *PT = NETEQ_CODEC_PCM16B_PT;
865 *fs = 8000;
866 } else if (!strcmp(name, "pcm16b_wb")) {
867 *codec = webrtc::kDecoderPCM16Bwb;
868 *PT = NETEQ_CODEC_PCM16B_WB_PT;
869 *fs = 16000;
870 } else if (!strcmp(name, "pcm16b_swb32")) {
871 *codec = webrtc::kDecoderPCM16Bswb32kHz;
872 *PT = NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
873 *fs = 32000;
874 } else if (!strcmp(name, "pcm16b_swb48")) {
875 *codec = webrtc::kDecoderPCM16Bswb48kHz;
876 *PT = NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
877 *fs = 48000;
878 } else if (!strcmp(name, "g722")) {
879 *codec = webrtc::kDecoderG722;
880 *PT = NETEQ_CODEC_G722_PT;
881 *fs = 16000;
882 } else if ((!strcmp(name, "ilbc")) &&
883 ((frameLen % 240 == 0) || (frameLen % 160 == 0))) {
884 *fs = 8000;
885 *codec = webrtc::kDecoderILBC;
886 *PT = NETEQ_CODEC_ILBC_PT;
887 } else if (!strcmp(name, "isac")) {
888 *fs = 16000;
889 *codec = webrtc::kDecoderISAC;
890 *PT = NETEQ_CODEC_ISAC_PT;
891 } else if (!strcmp(name, "isacswb")) {
892 *fs = 32000;
893 *codec = webrtc::kDecoderISACswb;
894 *PT = NETEQ_CODEC_ISACSWB_PT;
895 } else if (!strcmp(name, "red_pcm")) {
896 *codec = webrtc::kDecoderPCMa;
897 *PT = NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
898 *fs = 8000;
899 *useRed = 1;
900 } else if (!strcmp(name, "red_isac")) {
901 *codec = webrtc::kDecoderISAC;
902 *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
903 *fs = 16000;
904 *useRed = 1;
905 } else {
906 printf("Error: Not a supported codec (%s)\n", name);
907 exit(0);
908 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000909}
910
Peter Kasting248b0b02015-06-03 12:32:41 -0700911int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
912 int enc_frameSize,
913 int bitrate,
914 int sampfreq,
915 int vad,
916 int numChannels) {
917 int ok = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000918
Peter Kasting248b0b02015-06-03 12:32:41 -0700919 for (int k = 0; k < numChannels; k++) {
920 VAD_inst[k] = WebRtcVad_Create();
921 if (!VAD_inst[k]) {
922 printf("Error: Couldn't allocate memory for VAD instance\n");
923 exit(0);
924 }
925 ok = WebRtcVad_Init(VAD_inst[k]);
926 if (ok == -1) {
927 printf("Error: Initialization of VAD struct failed\n");
928 exit(0);
929 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000930
931#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700932 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
933 ok = WebRtcCng_CreateEnc(&CNGenc_inst[k]);
934 if (ok != 0) {
935 printf("Error: Couldn't allocate memory for CNG encoding instance\n");
936 exit(0);
937 }
938 if (sampfreq <= 16000) {
939 ok = WebRtcCng_InitEnc(CNGenc_inst[k], sampfreq, 200, 5);
940 if (ok == -1) {
941 printf("Error: Initialization of CNG struct failed. Error code %d\n",
942 WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
943 exit(0);
944 }
945 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000946#endif
947
Peter Kasting248b0b02015-06-03 12:32:41 -0700948 switch (coder) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000949#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -0700950 case webrtc::kDecoderPCM16B:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000951#endif
952#ifdef CODEC_PCM16B_WB
Peter Kasting248b0b02015-06-03 12:32:41 -0700953 case webrtc::kDecoderPCM16Bwb:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000954#endif
955#ifdef CODEC_PCM16B_32KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -0700956 case webrtc::kDecoderPCM16Bswb32kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000957#endif
958#ifdef CODEC_PCM16B_48KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -0700959 case webrtc::kDecoderPCM16Bswb48kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000960#endif
961#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700962 case webrtc::kDecoderPCMu:
963 case webrtc::kDecoderPCMa:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000964#endif
965 // do nothing
966 break;
967#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700968 case webrtc::kDecoderG729:
969 if (sampfreq == 8000) {
970 if ((enc_frameSize == 80) || (enc_frameSize == 160) ||
971 (enc_frameSize == 240) || (enc_frameSize == 320) ||
972 (enc_frameSize == 400) || (enc_frameSize == 480)) {
973 ok = WebRtcG729_CreateEnc(&G729enc_inst[k]);
974 if (ok != 0) {
975 printf(
976 "Error: Couldn't allocate memory for G729 encoding "
977 "instance\n");
978 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000979 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700980 } else {
981 printf(
982 "\nError: g729 only supports 10, 20, 30, 40, 50 or 60 "
983 "ms!!\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000984 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -0700985 }
986 WebRtcG729_EncoderInit(G729enc_inst[k], vad);
987 if ((vad == 1) && (enc_frameSize != 80)) {
988 printf(
989 "\nError - This simulation only supports VAD for G729 at 10ms "
990 "packets (not %dms)\n",
991 (enc_frameSize >> 3));
992 }
993 } else {
994 printf("\nError - g729 is only developed for 8kHz \n");
995 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000996 }
997 break;
998#endif
999#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -07001000 case webrtc::kDecoderG729_1:
1001 if (sampfreq == 16000) {
1002 if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
1003 (enc_frameSize == 960)) {
1004 ok = WebRtcG7291_Create(&G729_1_inst[k]);
1005 if (ok != 0) {
1006 printf(
1007 "Error: Couldn't allocate memory for G.729.1 codec "
1008 "instance\n");
1009 exit(0);
1010 }
1011 } else {
1012 printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001013 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001014 }
1015 if (!(((bitrate >= 12000) && (bitrate <= 32000) &&
1016 (bitrate % 2000 == 0)) ||
1017 (bitrate == 8000))) {
1018 /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
1019 printf(
1020 "\nError: G.729.1 bitrate must be 8000 or 12000--32000 in "
1021 "steps of 2000 bps\n");
1022 exit(0);
1023 }
1024 WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/,
1025 0 /*flagG729mode*/);
1026 } else {
1027 printf("\nError - G.729.1 input is always 16 kHz \n");
1028 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001029 }
1030 break;
1031#endif
1032#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -07001033 case webrtc::kDecoderSPEEX_8:
1034 if (sampfreq == 8000) {
1035 if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
1036 (enc_frameSize == 480)) {
1037 ok = WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
1038 if (ok != 0) {
1039 printf(
1040 "Error: Couldn't allocate memory for Speex encoding "
1041 "instance\n");
1042 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001043 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001044 } else {
1045 printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
1046 exit(0);
1047 }
1048 if ((vad == 1) && (enc_frameSize != 160)) {
1049 printf(
1050 "\nError - This simulation only supports VAD for Speex at 20ms "
1051 "packets (not %dms)\n",
1052 (enc_frameSize >> 3));
1053 vad = 0;
1054 }
1055 ok = WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0 /*vbr*/,
1056 3 /*complexity*/, vad);
1057 if (ok != 0)
1058 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001059 } else {
Peter Kasting248b0b02015-06-03 12:32:41 -07001060 printf(
1061 "\nError - Speex8 called with sample frequency other than 8 "
1062 "kHz.\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001063 }
1064 break;
1065#endif
1066#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001067 case webrtc::kDecoderSPEEX_16:
1068 if (sampfreq == 16000) {
1069 if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
1070 (enc_frameSize == 960)) {
1071 ok = WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
1072 if (ok != 0) {
1073 printf(
1074 "Error: Couldn't allocate memory for Speex encoding "
1075 "instance\n");
1076 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001077 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001078 } else {
1079 printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
1080 exit(0);
1081 }
1082 if ((vad == 1) && (enc_frameSize != 320)) {
1083 printf(
1084 "\nError - This simulation only supports VAD for Speex at 20ms "
1085 "packets (not %dms)\n",
1086 (enc_frameSize >> 4));
1087 vad = 0;
1088 }
1089 ok = WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0 /*vbr*/,
1090 3 /*complexity*/, vad);
1091 if (ok != 0)
1092 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001093 } else {
Peter Kasting248b0b02015-06-03 12:32:41 -07001094 printf(
1095 "\nError - Speex16 called with sample frequency other than 16 "
1096 "kHz.\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001097 }
1098 break;
1099#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001100
1101#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001102 case webrtc::kDecoderG722_1_16:
1103 if (sampfreq == 16000) {
1104 ok = WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
1105 if (ok != 0) {
1106 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001107 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001108 }
1109 if (enc_frameSize == 320) {
1110 } else {
1111 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1112 exit(0);
1113 }
1114 WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
1115 } else {
1116 printf("\nError - G722.1 is only developed for 16kHz \n");
1117 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001118 }
1119 break;
1120#endif
1121#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001122 case webrtc::kDecoderG722_1_24:
1123 if (sampfreq == 16000) {
1124 ok = WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
1125 if (ok != 0) {
1126 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001127 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001128 }
1129 if (enc_frameSize == 320) {
1130 } else {
1131 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1132 exit(0);
1133 }
1134 WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
1135 } else {
1136 printf("\nError - G722.1 is only developed for 16kHz \n");
1137 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001138 }
1139 break;
1140#endif
1141#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001142 case webrtc::kDecoderG722_1_32:
1143 if (sampfreq == 16000) {
1144 ok = WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
1145 if (ok != 0) {
1146 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001147 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001148 }
1149 if (enc_frameSize == 320) {
1150 } else {
1151 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1152 exit(0);
1153 }
1154 WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
1155 } else {
1156 printf("\nError - G722.1 is only developed for 16kHz \n");
1157 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001158 }
1159 break;
1160#endif
1161#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001162 case webrtc::kDecoderG722_1C_24:
1163 if (sampfreq == 32000) {
1164 ok = WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
1165 if (ok != 0) {
1166 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001167 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001168 }
1169 if (enc_frameSize == 640) {
1170 } else {
1171 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1172 exit(0);
1173 }
1174 WebRtcG7221C_EncoderInit24(
1175 (G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
1176 } else {
1177 printf("\nError - G722.1 C is only developed for 32kHz \n");
1178 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001179 }
1180 break;
1181#endif
1182#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001183 case webrtc::kDecoderG722_1C_32:
1184 if (sampfreq == 32000) {
1185 ok = WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
1186 if (ok != 0) {
1187 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001188 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001189 }
1190 if (enc_frameSize == 640) {
1191 } else {
1192 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1193 exit(0);
1194 }
1195 WebRtcG7221C_EncoderInit32(
1196 (G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
1197 } else {
1198 printf("\nError - G722.1 C is only developed for 32kHz \n");
1199 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001200 }
1201 break;
1202#endif
1203#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -07001204 case webrtc::kDecoderG722_1C_48:
1205 if (sampfreq == 32000) {
1206 ok = WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
1207 if (ok != 0) {
1208 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001209 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001210 }
1211 if (enc_frameSize == 640) {
1212 } else {
1213 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1214 exit(0);
1215 }
1216 WebRtcG7221C_EncoderInit48(
1217 (G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
1218 } else {
1219 printf("\nError - G722.1 C is only developed for 32kHz \n");
1220 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001221 }
1222 break;
1223#endif
1224#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -07001225 case webrtc::kDecoderG722:
1226 if (sampfreq == 16000) {
1227 if (enc_frameSize % 2 == 0) {
1228 } else {
1229 printf(
1230 "\nError - g722 frames must have an even number of "
1231 "enc_frameSize\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001232 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001233 }
1234 WebRtcG722_CreateEncoder(&g722EncState[k]);
1235 WebRtcG722_EncoderInit(g722EncState[k]);
1236 } else {
1237 printf("\nError - g722 is only developed for 16kHz \n");
1238 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001239 }
1240 break;
1241#endif
1242#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -07001243 case webrtc::kDecoderAMR:
1244 if (sampfreq == 8000) {
1245 ok = WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
1246 if (ok != 0) {
1247 printf(
1248 "Error: Couldn't allocate memory for AMR encoding instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001249 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001250 }
1251 if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
1252 (enc_frameSize == 480)) {
1253 } else {
1254 printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
1255 exit(0);
1256 }
1257 WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
1258 WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
1259 AMR_bitrate = bitrate;
1260 } else {
1261 printf("\nError - AMR is only developed for 8kHz \n");
1262 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001263 }
1264 break;
1265#endif
1266#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001267 case webrtc::kDecoderAMRWB:
1268 if (sampfreq == 16000) {
1269 ok = WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
1270 if (ok != 0) {
1271 printf(
1272 "Error: Couldn't allocate memory for AMRWB encoding "
1273 "instance\n");
1274 exit(0);
1275 }
1276 if (((enc_frameSize / 320) < 0) || ((enc_frameSize / 320) > 3) ||
1277 ((enc_frameSize % 320) != 0)) {
1278 printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
1279 exit(0);
1280 }
1281 WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
1282 if (bitrate == 7000) {
1283 AMRWB_bitrate = AMRWB_MODE_7k;
1284 } else if (bitrate == 9000) {
1285 AMRWB_bitrate = AMRWB_MODE_9k;
1286 } else if (bitrate == 12000) {
1287 AMRWB_bitrate = AMRWB_MODE_12k;
1288 } else if (bitrate == 14000) {
1289 AMRWB_bitrate = AMRWB_MODE_14k;
1290 } else if (bitrate == 16000) {
1291 AMRWB_bitrate = AMRWB_MODE_16k;
1292 } else if (bitrate == 18000) {
1293 AMRWB_bitrate = AMRWB_MODE_18k;
1294 } else if (bitrate == 20000) {
1295 AMRWB_bitrate = AMRWB_MODE_20k;
1296 } else if (bitrate == 23000) {
1297 AMRWB_bitrate = AMRWB_MODE_23k;
1298 } else if (bitrate == 24000) {
1299 AMRWB_bitrate = AMRWB_MODE_24k;
1300 }
1301 WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001302
1303 } else {
Peter Kasting248b0b02015-06-03 12:32:41 -07001304 printf("\nError - AMRwb is only developed for 16kHz \n");
1305 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001306 }
1307 break;
1308#endif
1309#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -07001310 case webrtc::kDecoderILBC:
1311 if (sampfreq == 8000) {
1312 ok = WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
1313 if (ok != 0) {
1314 printf(
1315 "Error: Couldn't allocate memory for iLBC encoding instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001316 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001317 }
1318 if ((enc_frameSize == 160) || (enc_frameSize == 240) ||
1319 (enc_frameSize == 320) || (enc_frameSize == 480)) {
1320 } else {
1321 printf(
1322 "\nError - iLBC only supports 160, 240, 320 and 480 "
1323 "enc_frameSize (20, 30, 40 and 60 ms)\n");
1324 exit(0);
1325 }
1326 if ((enc_frameSize == 160) || (enc_frameSize == 320)) {
1327 /* 20 ms version */
1328 WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
1329 } else {
1330 /* 30 ms version */
1331 WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
1332 }
1333 } else {
1334 printf("\nError - iLBC is only developed for 8kHz \n");
1335 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001336 }
1337 break;
1338#endif
1339#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -07001340 case webrtc::kDecoderISAC:
1341 if (sampfreq == 16000) {
1342 ok = WebRtcIsac_Create(&ISAC_inst[k]);
1343 if (ok != 0) {
1344 printf("Error: Couldn't allocate memory for iSAC instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001345 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001346 }
1347 if ((enc_frameSize == 480) || (enc_frameSize == 960)) {
1348 } else {
1349 printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
1350 exit(0);
1351 }
1352 WebRtcIsac_EncoderInit(ISAC_inst[k], 1);
1353 if ((bitrate < 10000) || (bitrate > 32000)) {
1354 printf(
1355 "\nError - iSAC bitrate has to be between 10000 and 32000 bps "
1356 "(not %i)\n",
1357 bitrate);
1358 exit(0);
1359 }
1360 WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize >> 4);
1361 } else {
1362 printf(
1363 "\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 "
1364 "ms)\n");
1365 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001366 }
1367 break;
1368#endif
1369#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -07001370 case webrtc::kDecoderISAC:
1371 if (sampfreq == 16000) {
1372 ok = WebRtcIsacfix_Create(&ISAC_inst[k]);
1373 if (ok != 0) {
1374 printf("Error: Couldn't allocate memory for iSAC instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001375 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001376 }
1377 if ((enc_frameSize == 480) || (enc_frameSize == 960)) {
1378 } else {
1379 printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
1380 exit(0);
1381 }
1382 WebRtcIsacfix_EncoderInit(ISAC_inst[k], 1);
1383 if ((bitrate < 10000) || (bitrate > 32000)) {
1384 printf(
1385 "\nError - iSAC bitrate has to be between 10000 and 32000 bps "
1386 "(not %i)\n",
1387 bitrate);
1388 exit(0);
1389 }
1390 WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize >> 4);
1391 } else {
1392 printf(
1393 "\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 "
1394 "ms)\n");
1395 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001396 }
1397 break;
1398#endif
1399#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001400 case webrtc::kDecoderISACswb:
1401 if (sampfreq == 32000) {
1402 ok = WebRtcIsac_Create(&ISACSWB_inst[k]);
1403 if (ok != 0) {
1404 printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001405 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001406 }
1407 if (enc_frameSize == 960) {
1408 } else {
1409 printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
1410 exit(0);
1411 }
1412 ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
1413 if (ok != 0) {
1414 printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
1415 exit(0);
1416 }
1417 WebRtcIsac_EncoderInit(ISACSWB_inst[k], 1);
1418 if ((bitrate < 32000) || (bitrate > 56000)) {
1419 printf(
1420 "\nError - iSAC SWB bitrate has to be between 32000 and 56000 "
1421 "bps (not %i)\n",
1422 bitrate);
1423 exit(0);
1424 }
1425 WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize >> 5);
1426 } else {
1427 printf(
1428 "\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n");
1429 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001430 }
1431 break;
1432#endif
1433#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -07001434 case webrtc::kDecoderGSMFR:
1435 if (sampfreq == 8000) {
1436 ok = WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
1437 if (ok != 0) {
1438 printf(
1439 "Error: Couldn't allocate memory for GSM FR encoding "
1440 "instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001441 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001442 }
1443 if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
1444 (enc_frameSize == 480)) {
1445 } else {
1446 printf(
1447 "\nError - GSM FR must have a multiple of 160 enc_frameSize\n");
1448 exit(0);
1449 }
1450 WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
1451 } else {
1452 printf("\nError - GSM FR is only developed for 8kHz \n");
1453 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001454 }
1455 break;
1456#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001457 default:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001458 printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
1459 exit(0);
1460 break;
Peter Kasting248b0b02015-06-03 12:32:41 -07001461 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001462
Peter Kasting248b0b02015-06-03 12:32:41 -07001463 if (ok != 0) {
1464 return (ok);
1465 }
1466 } // end for
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001467
Peter Kasting248b0b02015-06-03 12:32:41 -07001468 return (0);
1469}
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001470
1471int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001472 for (int k = 0; k < numChannels; k++) {
1473 WebRtcVad_Free(VAD_inst[k]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001474#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -07001475 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
1476 WebRtcCng_FreeEnc(CNGenc_inst[k]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001477#endif
1478
Peter Kasting248b0b02015-06-03 12:32:41 -07001479 switch (coder) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001480#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -07001481 case webrtc::kDecoderPCM16B:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001482#endif
1483#ifdef CODEC_PCM16B_WB
Peter Kasting248b0b02015-06-03 12:32:41 -07001484 case webrtc::kDecoderPCM16Bwb:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001485#endif
1486#ifdef CODEC_PCM16B_32KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -07001487 case webrtc::kDecoderPCM16Bswb32kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001488#endif
1489#ifdef CODEC_PCM16B_48KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -07001490 case webrtc::kDecoderPCM16Bswb48kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001491#endif
1492#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -07001493 case webrtc::kDecoderPCMu:
1494 case webrtc::kDecoderPCMa:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001495#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001496 // do nothing
1497 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001498#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -07001499 case webrtc::kDecoderG729:
1500 WebRtcG729_FreeEnc(G729enc_inst[k]);
1501 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001502#endif
1503#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -07001504 case webrtc::kDecoderG729_1:
1505 WebRtcG7291_Free(G729_1_inst[k]);
1506 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001507#endif
1508#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -07001509 case webrtc::kDecoderSPEEX_8:
1510 WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
1511 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001512#endif
1513#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001514 case webrtc::kDecoderSPEEX_16:
1515 WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
1516 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001517#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001518
1519#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001520 case webrtc::kDecoderG722_1_16:
1521 WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
1522 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001523#endif
1524#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001525 case webrtc::kDecoderG722_1_24:
1526 WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
1527 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001528#endif
1529#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001530 case webrtc::kDecoderG722_1_32:
1531 WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
1532 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001533#endif
1534#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001535 case webrtc::kDecoderG722_1C_24:
1536 WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
1537 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001538#endif
1539#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001540 case webrtc::kDecoderG722_1C_32:
1541 WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
1542 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001543#endif
1544#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -07001545 case webrtc::kDecoderG722_1C_48:
1546 WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
1547 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001548#endif
1549#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -07001550 case webrtc::kDecoderG722:
1551 WebRtcG722_FreeEncoder(g722EncState[k]);
1552 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001553#endif
1554#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -07001555 case webrtc::kDecoderAMR:
1556 WebRtcAmr_FreeEnc(AMRenc_inst[k]);
1557 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001558#endif
1559#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001560 case webrtc::kDecoderAMRWB:
1561 WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
1562 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001563#endif
1564#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -07001565 case webrtc::kDecoderILBC:
1566 WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
1567 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001568#endif
1569#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -07001570 case webrtc::kDecoderISAC:
1571 WebRtcIsac_Free(ISAC_inst[k]);
1572 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001573#endif
1574#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -07001575 case webrtc::kDecoderISAC:
1576 WebRtcIsacfix_Free(ISAC_inst[k]);
1577 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001578#endif
1579#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001580 case webrtc::kDecoderISACswb:
1581 WebRtcIsac_Free(ISACSWB_inst[k]);
1582 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001583#endif
1584#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -07001585 case webrtc::kDecoderGSMFR:
1586 WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
1587 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001588#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001589 default:
1590 printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
1591 exit(0);
1592 break;
1593 }
1594 }
1595
1596 return (0);
1597}
1598
1599int NetEQTest_encode(int coder,
1600 int16_t* indata,
1601 int frameLen,
1602 unsigned char* encoded,
1603 int sampleRate,
1604 int* vad,
1605 int useVAD,
1606 int bitrate,
1607 int numChannels) {
Peter Kasting83ad33a2015-06-09 17:19:57 -07001608 int cdlen = 0;
Peter Kasting248b0b02015-06-03 12:32:41 -07001609 int16_t* tempdata;
1610 static int first_cng = 1;
1611 int16_t tempLen;
1612
1613 *vad = 1;
1614
1615 // check VAD first
1616 if (useVAD) {
1617 *vad = 0;
1618
1619 for (int k = 0; k < numChannels; k++) {
1620 tempLen = frameLen;
1621 tempdata = &indata[k * frameLen];
1622 int localVad = 0;
1623 /* Partition the signal and test each chunk for VAD.
1624 All chunks must be VAD=0 to produce a total VAD=0. */
1625 while (tempLen >= 10 * sampleRate / 1000) {
1626 if ((tempLen % 30 * sampleRate / 1000) ==
1627 0) { // tempLen is multiple of 30ms
1628 localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
1629 30 * sampleRate / 1000);
1630 tempdata += 30 * sampleRate / 1000;
1631 tempLen -= 30 * sampleRate / 1000;
1632 } else if (tempLen >= 20 * sampleRate / 1000) { // tempLen >= 20ms
1633 localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
1634 20 * sampleRate / 1000);
1635 tempdata += 20 * sampleRate / 1000;
1636 tempLen -= 20 * sampleRate / 1000;
1637 } else { // use 10ms
1638 localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
1639 10 * sampleRate / 1000);
1640 tempdata += 10 * sampleRate / 1000;
1641 tempLen -= 10 * sampleRate / 1000;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001642 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001643 }
1644
1645 // aggregate all VAD decisions over all channels
1646 *vad |= localVad;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001647 }
1648
Peter Kasting248b0b02015-06-03 12:32:41 -07001649 if (!*vad) {
1650 // all channels are silent
1651 cdlen = 0;
1652 for (int k = 0; k < numChannels; k++) {
1653 WebRtcCng_Encode(CNGenc_inst[k], &indata[k * frameLen],
1654 (frameLen <= 640 ? frameLen : 640) /* max 640 */,
1655 encoded, &tempLen, first_cng);
1656 encoded += tempLen;
1657 cdlen += tempLen;
1658 }
1659 *vad = 0;
1660 first_cng = 0;
1661 return (cdlen);
1662 }
1663 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001664
Peter Kasting248b0b02015-06-03 12:32:41 -07001665 // loop over all channels
1666 int totalLen = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001667
Peter Kasting248b0b02015-06-03 12:32:41 -07001668 for (int k = 0; k < numChannels; k++) {
1669 /* Encode with the selected coder type */
1670 if (coder == webrtc::kDecoderPCMu) { /*g711 u-law */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001671#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -07001672 cdlen = WebRtcG711_EncodeU(indata, frameLen, encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001673#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001674 } else if (coder == webrtc::kDecoderPCMa) { /*g711 A-law */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001675#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -07001676 cdlen = WebRtcG711_EncodeA(indata, frameLen, encoded);
1677 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001678#endif
1679#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -07001680 else if ((coder == webrtc::kDecoderPCM16B) ||
1681 (coder == webrtc::kDecoderPCM16Bwb) ||
1682 (coder == webrtc::kDecoderPCM16Bswb32kHz) ||
1683 (coder == webrtc::
1684 kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz,
1685 32kHz or 48kHz) */
1686 cdlen = WebRtcPcm16b_Encode(indata, frameLen, encoded);
1687 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001688#endif
1689#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -07001690 else if (coder == webrtc::kDecoderG722) { /*g722 */
1691 cdlen = WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded);
1692 assert(cdlen == frameLen >> 1);
1693 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001694#endif
1695#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -07001696 else if (coder == webrtc::kDecoderILBC) { /*iLBC */
1697 cdlen = WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata, frameLen, encoded);
1698 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001699#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001700#if (defined(CODEC_ISAC) || \
1701 defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all
1702 // NETEQ_ISACFIX_CODEC
1703 else if (coder == webrtc::kDecoderISAC) { /*iSAC */
1704 int noOfCalls = 0;
1705 cdlen = 0;
1706 while (cdlen <= 0) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001707#ifdef CODEC_ISAC /* floating point */
Peter Kasting248b0b02015-06-03 12:32:41 -07001708 cdlen =
1709 WebRtcIsac_Encode(ISAC_inst[k], &indata[noOfCalls * 160], encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001710#else /* fixed point */
Peter Kasting248b0b02015-06-03 12:32:41 -07001711 cdlen = WebRtcIsacfix_Encode(ISAC_inst[k], &indata[noOfCalls * 160],
1712 encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001713#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001714 noOfCalls++;
1715 }
1716 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001717#endif
1718#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001719 else if (coder == webrtc::kDecoderISACswb) { /* iSAC SWB */
1720 int noOfCalls = 0;
1721 cdlen = 0;
1722 while (cdlen <= 0) {
1723 cdlen = WebRtcIsac_Encode(ISACSWB_inst[k], &indata[noOfCalls * 320],
1724 encoded);
1725 noOfCalls++;
1726 }
1727 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001728#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001729 indata += frameLen;
1730 encoded += cdlen;
1731 totalLen += cdlen;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001732
Peter Kasting248b0b02015-06-03 12:32:41 -07001733 } // end for
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001734
Peter Kasting248b0b02015-06-03 12:32:41 -07001735 first_cng = 1;
1736 return (totalLen);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001737}
1738
pkasting@chromium.orgd3245462015-02-23 21:28:22 +00001739void makeRTPheader(unsigned char* rtp_data,
1740 int payloadType,
1741 int seqNo,
1742 uint32_t timestamp,
1743 uint32_t ssrc) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001744 rtp_data[0] = 0x80;
1745 rtp_data[1] = payloadType & 0xFF;
1746 rtp_data[2] = (seqNo >> 8) & 0xFF;
1747 rtp_data[3] = seqNo & 0xFF;
1748 rtp_data[4] = timestamp >> 24;
1749 rtp_data[5] = (timestamp >> 16) & 0xFF;
1750 rtp_data[6] = (timestamp >> 8) & 0xFF;
1751 rtp_data[7] = timestamp & 0xFF;
1752 rtp_data[8] = ssrc >> 24;
1753 rtp_data[9] = (ssrc >> 16) & 0xFF;
1754 rtp_data[10] = (ssrc >> 8) & 0xFF;
1755 rtp_data[11] = ssrc & 0xFF;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001756}
1757
pkasting@chromium.orgd3245462015-02-23 21:28:22 +00001758int makeRedundantHeader(unsigned char* rtp_data,
1759 int* payloadType,
1760 int numPayloads,
1761 uint32_t* timestamp,
1762 uint16_t* blockLen,
1763 int seqNo,
Peter Kasting248b0b02015-06-03 12:32:41 -07001764 uint32_t ssrc) {
1765 int i;
1766 unsigned char* rtpPointer;
1767 uint16_t offset;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001768
Peter Kasting248b0b02015-06-03 12:32:41 -07001769 /* first create "standard" RTP header */
1770 makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads - 1],
1771 ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001772
Peter Kasting248b0b02015-06-03 12:32:41 -07001773 rtpPointer = &rtp_data[12];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001774
Peter Kasting248b0b02015-06-03 12:32:41 -07001775 /* add one sub-header for each redundant payload (not the primary) */
1776 for (i = 0; i < numPayloads - 1; i++) {
1777 if (blockLen[i] > 0) {
1778 offset = static_cast<uint16_t>(timestamp[numPayloads - 1] - timestamp[i]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001779
Peter Kasting248b0b02015-06-03 12:32:41 -07001780 // Byte |0| |1 2 | 3 |
1781 // Bit |0|1234567|01234567012345|6701234567|
1782 // |F|payload| timestamp | block |
1783 // | | type | offset | length |
1784 rtpPointer[0] = (payloadType[i] & 0x7F) | 0x80;
1785 rtpPointer[1] = (offset >> 6) & 0xFF;
1786 rtpPointer[2] = ((offset & 0x3F) << 2) | ((blockLen[i] >> 8) & 0x03);
1787 rtpPointer[3] = blockLen[i] & 0xFF;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001788
Peter Kasting248b0b02015-06-03 12:32:41 -07001789 rtpPointer += 4;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001790 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001791 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001792
Peter Kasting248b0b02015-06-03 12:32:41 -07001793 // Bit |0|1234567|
1794 // |0|payload|
1795 // | | type |
1796 rtpPointer[0] = payloadType[numPayloads - 1] & 0x7F;
1797 ++rtpPointer;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001798
Peter Kasting248b0b02015-06-03 12:32:41 -07001799 return rtpPointer - rtp_data; // length of header in bytes
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001800}
1801
Peter Kasting248b0b02015-06-03 12:32:41 -07001802int makeDTMFpayload(unsigned char* payload_data,
1803 int Event,
1804 int End,
1805 int Volume,
1806 int Duration) {
1807 unsigned char E, R, V;
1808 R = 0;
1809 V = (unsigned char)Volume;
1810 if (End == 0) {
1811 E = 0x00;
1812 } else {
1813 E = 0x80;
1814 }
1815 payload_data[0] = (unsigned char)Event;
1816 payload_data[1] = (unsigned char)(E | R | V);
1817 // Duration equals 8 times time_ms, default is 8000 Hz.
1818 payload_data[2] = (unsigned char)((Duration >> 8) & 0xFF);
1819 payload_data[3] = (unsigned char)(Duration & 0xFF);
1820 return (4);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001821}
1822
Peter Kasting248b0b02015-06-03 12:32:41 -07001823void stereoDeInterleave(int16_t* audioSamples, int numSamples) {
1824 int16_t* tempVec;
1825 int16_t* readPtr, *writeL, *writeR;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001826
Peter Kasting248b0b02015-06-03 12:32:41 -07001827 if (numSamples <= 0)
1828 return;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001829
Peter Kasting248b0b02015-06-03 12:32:41 -07001830 tempVec = (int16_t*)malloc(sizeof(int16_t) * numSamples);
1831 if (tempVec == NULL) {
1832 printf("Error allocating memory\n");
1833 exit(0);
1834 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001835
Peter Kasting248b0b02015-06-03 12:32:41 -07001836 memcpy(tempVec, audioSamples, numSamples * sizeof(int16_t));
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001837
Peter Kasting248b0b02015-06-03 12:32:41 -07001838 writeL = audioSamples;
1839 writeR = &audioSamples[numSamples / 2];
1840 readPtr = tempVec;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001841
Peter Kasting248b0b02015-06-03 12:32:41 -07001842 for (int k = 0; k < numSamples; k += 2) {
1843 *writeL = *readPtr;
1844 readPtr++;
1845 *writeR = *readPtr;
1846 readPtr++;
1847 writeL++;
1848 writeR++;
1849 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001850
Peter Kasting248b0b02015-06-03 12:32:41 -07001851 free(tempVec);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001852}
1853
Peter Kasting248b0b02015-06-03 12:32:41 -07001854void stereoInterleave(unsigned char* data, int dataLen, int stride) {
1855 unsigned char* ptrL, *ptrR;
1856 unsigned char temp[10];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001857
Peter Kasting248b0b02015-06-03 12:32:41 -07001858 if (stride > 10) {
1859 exit(0);
1860 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001861
Peter Kasting248b0b02015-06-03 12:32:41 -07001862 if (dataLen % 1 != 0) {
1863 // must be even number of samples
1864 printf("Error: cannot interleave odd sample number\n");
1865 exit(0);
1866 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001867
Peter Kasting248b0b02015-06-03 12:32:41 -07001868 ptrL = data + stride;
1869 ptrR = &data[dataLen / 2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001870
Peter Kasting248b0b02015-06-03 12:32:41 -07001871 while (ptrL < ptrR) {
1872 // copy from right pointer to temp
1873 memcpy(temp, ptrR, stride);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001874
Peter Kasting248b0b02015-06-03 12:32:41 -07001875 // shift data between pointers
1876 memmove(ptrL + stride, ptrL, ptrR - ptrL);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001877
Peter Kasting248b0b02015-06-03 12:32:41 -07001878 // copy from temp to left pointer
1879 memcpy(ptrL, temp, stride);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001880
Peter Kasting248b0b02015-06-03 12:32:41 -07001881 // advance pointers
1882 ptrL += stride * 2;
1883 ptrR += stride;
1884 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001885}