blob: b2df07aa4e97751207a640dfb72d99fa900201ca [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
Peter Kastingdce40cf2015-08-24 14:52:23 -070026#include <algorithm>
27
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000028#include "webrtc/typedefs.h"
29// needed for NetEqDecoder
kwiberg@webrtc.orge04a93b2014-12-09 10:12:53 +000030#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000031#include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000032
33/************************/
34/* Define payload types */
35/************************/
36
37#include "PayloadTypes.h"
38
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000039/*********************/
40/* Misc. definitions */
41/*********************/
42
43#define STOPSENDTIME 3000
Peter Kasting248b0b02015-06-03 12:32:41 -070044#define RESTARTSENDTIME 0 // 162500
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000045#define FIRSTLINELEN 40
Peter Kasting248b0b02015-06-03 12:32:41 -070046#define CHECK_NOT_NULL(a) \
47 if ((a) == 0) { \
48 printf("\n %s \n line: %d \nerror at %s\n", __FILE__, __LINE__, #a); \
49 return (-1); \
50 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000051
52//#define MULTIPLE_SAME_TIMESTAMP
53#define REPEAT_PACKET_DISTANCE 17
54#define REPEAT_PACKET_COUNT 1 // number of extra packets to send
55
56//#define INSERT_OLD_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -070057#define OLD_PACKET 5 // how many seconds too old should the packet be?
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000058
59//#define TIMESTAMP_WRAPAROUND
60
61//#define RANDOM_DATA
62//#define RANDOM_PAYLOAD_DATA
63#define RANDOM_SEED 10
64
65//#define INSERT_DTMF_PACKETS
66//#define NO_DTMF_OVERDUB
67#define DTMF_PACKET_INTERVAL 2000
68#define DTMF_DURATION 500
69
70#define STEREO_MODE_FRAME 0
Peter Kasting248b0b02015-06-03 12:32:41 -070071#define STEREO_MODE_SAMPLE_1 1 // 1 octet per sample
72#define STEREO_MODE_SAMPLE_2 2 // 2 octets per sample
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000073
74/*************************/
75/* Function declarations */
76/*************************/
77
pkasting@chromium.orgd3245462015-02-23 21:28:22 +000078void NetEQTest_GetCodec_and_PT(char* name,
79 webrtc::NetEqDecoder* codec,
80 int* PT,
Peter Kastingdce40cf2015-08-24 14:52:23 -070081 size_t frameLen,
pkasting@chromium.orgd3245462015-02-23 21:28:22 +000082 int* fs,
83 int* bitrate,
84 int* useRed);
85int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
Peter Kastingdce40cf2015-08-24 14:52:23 -070086 size_t enc_frameSize,
pkasting@chromium.orgd3245462015-02-23 21:28:22 +000087 int bitrate,
88 int sampfreq,
89 int vad,
Peter Kastingdce40cf2015-08-24 14:52:23 -070090 size_t numChannels);
pkasting@chromium.orgd3245462015-02-23 21:28:22 +000091void defineCodecs(webrtc::NetEqDecoder* usedCodec, int* noOfCodecs);
Peter Kastingdce40cf2015-08-24 14:52:23 -070092int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels);
93size_t NetEQTest_encode(int coder,
94 int16_t* indata,
95 size_t frameLen,
96 unsigned char* encoded,
97 int sampleRate,
98 int* vad,
99 int useVAD,
100 int bitrate,
101 size_t numChannels);
pkasting@chromium.orgd3245462015-02-23 21:28:22 +0000102void makeRTPheader(unsigned char* rtp_data,
103 int payloadType,
104 int seqNo,
105 uint32_t timestamp,
106 uint32_t ssrc);
107int makeRedundantHeader(unsigned char* rtp_data,
108 int* payloadType,
109 int numPayloads,
110 uint32_t* timestamp,
111 uint16_t* blockLen,
112 int seqNo,
113 uint32_t ssrc);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700114size_t makeDTMFpayload(unsigned char* payload_data,
115 int Event,
116 int End,
117 int Volume,
118 int Duration);
119void stereoDeInterleave(int16_t* audioSamples, size_t numSamples);
120void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000121
122/*********************/
123/* Codec definitions */
124/*********************/
125
126#include "webrtc_vad.h"
127
Peter Kasting248b0b02015-06-03 12:32:41 -0700128#if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
129#include "pcm16b.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000130#endif
131#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700132#include "g711_interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000133#endif
134#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700135#include "G729Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000136#endif
137#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -0700138#include "G729_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000139#endif
140#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -0700141#include "AMRInterface.h"
142#include "AMRCreation.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000143#endif
144#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700145#include "AMRWBInterface.h"
146#include "AMRWBCreation.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000147#endif
148#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -0700149#include "ilbc.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000150#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700151#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
152#include "isac.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000153#endif
154#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -0700155#include "isacfix.h"
156#ifdef CODEC_ISAC
Peter Kasting728d9032015-06-11 14:31:38 -0700157#error Cannot have both ISAC and ISACfix defined. Please de-select one.
Peter Kasting248b0b02015-06-03 12:32:41 -0700158#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000159#endif
160#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -0700161#include "g722_interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000162#endif
163#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700164#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000165#endif
166#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700167#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000168#endif
169#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700170#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000171#endif
172#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700173#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000174#endif
175#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700176#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000177#endif
178#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -0700179#include "G722_1Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000180#endif
181#ifdef CODEC_G726
Peter Kasting248b0b02015-06-03 12:32:41 -0700182#include "G726Creation.h"
183#include "G726Interface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000184#endif
185#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -0700186#include "GSMFRInterface.h"
187#include "GSMFRCreation.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000188#endif
189#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700190 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
191#include "webrtc_cng.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000192#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700193#if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
194#include "SpeexInterface.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000195#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000196
197/***********************************/
198/* Global codec instance variables */
199/***********************************/
200
Peter Kasting248b0b02015-06-03 12:32:41 -0700201WebRtcVadInst* VAD_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000202
203#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -0700204G722EncInst* g722EncState[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000205#endif
206
207#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700208G722_1_24_encinst_t* G722_1_24enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000209#endif
210#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700211G722_1_32_encinst_t* G722_1_32enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000212#endif
213#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700214G722_1_16_encinst_t* G722_1_16enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000215#endif
216#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -0700217G722_1C_24_encinst_t* G722_1C_24enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000218#endif
219#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -0700220G722_1C_32_encinst_t* G722_1C_32enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000221#endif
222#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -0700223G722_1C_48_encinst_t* G722_1C_48enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000224#endif
225#ifdef CODEC_G726
Peter Kasting248b0b02015-06-03 12:32:41 -0700226G726_encinst_t* G726enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000227#endif
228#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700229G729_encinst_t* G729enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000230#endif
231#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -0700232G729_1_inst_t* G729_1_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000233#endif
234#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -0700235AMR_encinst_t* AMRenc_inst[2];
236int16_t AMR_bitrate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000237#endif
238#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700239AMRWB_encinst_t* AMRWBenc_inst[2];
240int16_t AMRWB_bitrate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000241#endif
242#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -0700243IlbcEncoderInstance* iLBCenc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000244#endif
245#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -0700246ISACStruct* ISAC_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000247#endif
248#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -0700249ISACFIX_MainStruct* ISAC_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000250#endif
251#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -0700252ISACStruct* ISACSWB_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000253#endif
254#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -0700255GSMFR_encinst_t* GSMFRenc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000256#endif
257#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700258 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
259CNG_enc_inst* CNGenc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000260#endif
261#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -0700262SPEEX_encinst_t* SPEEX8enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000263#endif
264#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700265SPEEX_encinst_t* SPEEX16enc_inst[2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000266#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000267
Peter Kasting248b0b02015-06-03 12:32:41 -0700268int main(int argc, char* argv[]) {
Peter Kastingdce40cf2015-08-24 14:52:23 -0700269 size_t packet_size;
270 int fs;
Peter Kasting248b0b02015-06-03 12:32:41 -0700271 webrtc::NetEqDecoder usedCodec;
272 int payloadType;
273 int bitrate = 0;
274 int useVAD, vad;
275 int useRed = 0;
Peter Kastingdce40cf2015-08-24 14:52:23 -0700276 size_t len, enc_len;
Peter Kasting248b0b02015-06-03 12:32:41 -0700277 int16_t org_data[4000];
278 unsigned char rtp_data[8000];
279 int16_t seqNo = 0xFFF;
280 uint32_t ssrc = 1235412312;
281 uint32_t timestamp = 0xAC1245;
282 uint16_t length, plen;
283 uint32_t offset;
284 double sendtime = 0;
285 int red_PT[2] = {0};
286 uint32_t red_TS[2] = {0};
287 uint16_t red_len[2] = {0};
Peter Kastingdce40cf2015-08-24 14:52:23 -0700288 size_t RTPheaderLen = 12;
Peter Kasting248b0b02015-06-03 12:32:41 -0700289 uint8_t red_data[8000];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000290#ifdef INSERT_OLD_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700291 uint16_t old_length, old_plen;
Peter Kastingdce40cf2015-08-24 14:52:23 -0700292 size_t old_enc_len;
Peter Kasting248b0b02015-06-03 12:32:41 -0700293 int first_old_packet = 1;
294 unsigned char old_rtp_data[8000];
Peter Kastingdce40cf2015-08-24 14:52:23 -0700295 size_t packet_age = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000296#endif
297#ifdef INSERT_DTMF_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700298 int NTone = 1;
299 int DTMFfirst = 1;
300 uint32_t DTMFtimestamp;
301 bool dtmfSent = false;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000302#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700303 bool usingStereo = false;
Peter Kastingdce40cf2015-08-24 14:52:23 -0700304 size_t stereoMode = 0;
305 size_t numChannels = 1;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000306
Peter Kasting248b0b02015-06-03 12:32:41 -0700307 /* check number of parameters */
308 if ((argc != 6) && (argc != 7)) {
309 /* print help text and exit */
310 printf("Application to encode speech into an RTP stream.\n");
Peter Kasting2a100872015-06-09 17:26:40 -0700311 printf("The program reads a PCM file and encodes is using the specified "
312 "codec.\n");
313 printf("The coded speech is packetized in RTP packest and written to the "
314 "output file.\n");
315 printf("The format of the RTP stream file is simlilar to that of "
316 "rtpplay,\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700317 printf("but with the receive time euqal to 0 for all packets.\n");
318 printf("Usage:\n\n");
319 printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
320 printf("where:\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000321
Peter Kasting248b0b02015-06-03 12:32:41 -0700322 printf("PCMfile : PCM speech input file\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000323
Peter Kasting248b0b02015-06-03 12:32:41 -0700324 printf("RTPfile : RTP stream output file\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000325
Peter Kasting2a100872015-06-09 17:26:40 -0700326 printf("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 Kasting2a100872015-06-09 17:26:40 -0700349 printf(" : g729 G729 (8kHz and 8kbps) CELP (One-Three "
350 "frame(s)/packet)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000351#endif
352#ifdef CODEC_G729_1
Peter Kasting2a100872015-06-09 17:26:40 -0700353 printf(" : g729.1 G729.1 (16kHz) variable rate (8--32 "
354 "kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000355#endif
356#ifdef CODEC_G722_1_16
Peter Kasting2a100872015-06-09 17:26:40 -0700357 printf(" : g722.1_16 G722.1 coder (16kHz) (g722.1 with "
358 "16kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000359#endif
360#ifdef CODEC_G722_1_24
Peter Kasting2a100872015-06-09 17:26:40 -0700361 printf(" : g722.1_24 G722.1 coder (16kHz) (the 24kbps "
362 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000363#endif
364#ifdef CODEC_G722_1_32
Peter Kasting2a100872015-06-09 17:26:40 -0700365 printf(" : g722.1_32 G722.1 coder (16kHz) (the 32kbps "
366 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000367#endif
368#ifdef CODEC_G722_1C_24
Peter Kasting2a100872015-06-09 17:26:40 -0700369 printf(" : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps "
370 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000371#endif
372#ifdef CODEC_G722_1C_32
Peter Kasting2a100872015-06-09 17:26:40 -0700373 printf(" : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps "
374 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000375#endif
376#ifdef CODEC_G722_1C_48
Peter Kasting2a100872015-06-09 17:26:40 -0700377 printf(" : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps "
378 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000379#endif
380
381#ifdef CODEC_G726
Peter Kasting248b0b02015-06-03 12:32:41 -0700382 printf(" : g726_16 G726 coder (8kHz) 16kbps\n");
383 printf(" : g726_24 G726 coder (8kHz) 24kbps\n");
384 printf(" : g726_32 G726 coder (8kHz) 32kbps\n");
385 printf(" : g726_40 G726 coder (8kHz) 40kbps\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000386#endif
387#ifdef CODEC_AMR
Peter Kasting2a100872015-06-09 17:26:40 -0700388 printf(" : AMRXk Adaptive Multi Rate CELP codec "
389 "(8kHz)\n");
390 printf(" X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, "
391 "10.2 or 12.2\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000392#endif
393#ifdef CODEC_AMRWB
Peter Kasting2a100872015-06-09 17:26:40 -0700394 printf(" : AMRwbXk Adaptive Multi Rate Wideband CELP "
395 "codec (16kHz)\n");
396 printf(" X = 7, 9, 12, 14, 16, 18, 20, 23 or "
397 "24\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000398#endif
399#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -0700400 printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000401#endif
402#ifdef CODEC_ISAC
Peter Kasting2a100872015-06-09 17:26:40 -0700403 printf(" : isac iSAC (16kHz and 32.0 kbps). To set "
404 "rate specify a rate parameter as last parameter\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000405#endif
406#ifdef CODEC_ISAC_SWB
Peter Kasting2a100872015-06-09 17:26:40 -0700407 printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). "
408 "To set rate specify a rate parameter as last parameter\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000409#endif
410#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -0700411 printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000412#endif
413#ifdef CODEC_G722
Peter Kasting2a100872015-06-09 17:26:40 -0700414 printf(" : g722 g722 coder (16kHz) (the 64kbps "
415 "version)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000416#endif
417#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -0700418 printf(" : speex8 speex coder (8 kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000419#endif
420#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -0700421 printf(" : speex16 speex coder (16 kHz)\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000422#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000423#ifdef CODEC_RED
424#ifdef CODEC_G711
Peter Kasting2a100872015-06-09 17:26:40 -0700425 printf(" : red_pcm Redundancy RTP packet with 2*G711A "
426 "frames\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000427#endif
428#ifdef CODEC_ISAC
Peter Kasting2a100872015-06-09 17:26:40 -0700429 printf(" : red_isac Redundancy RTP packet with 2*iSAC "
430 "frames\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000431#endif
432#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700433 printf("\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000434
435#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700436 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
437 printf("useVAD : 0 Voice Activity Detection is switched off\n");
438 printf(" : 1 Voice Activity Detection is switched on\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000439#else
Peter Kasting2a100872015-06-09 17:26:40 -0700440 printf("useVAD : 0 Voice Activity Detection switched off (on not "
441 "supported)\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000442#endif
Peter Kasting2a100872015-06-09 17:26:40 -0700443 printf("bitrate : Codec bitrate in bps (only applies to vbr "
444 "codecs)\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000445
Peter Kasting248b0b02015-06-03 12:32:41 -0700446 return (0);
447 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000448
Peter Kasting248b0b02015-06-03 12:32:41 -0700449 FILE* in_file = fopen(argv[1], "rb");
450 CHECK_NOT_NULL(in_file);
451 printf("Input file: %s\n", argv[1]);
452 FILE* out_file = fopen(argv[2], "wb");
453 CHECK_NOT_NULL(out_file);
454 printf("Output file: %s\n\n", argv[2]);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700455 int packet_size_int = atoi(argv[3]);
456 if (packet_size_int <= 0) {
457 printf("Packet size %d must be positive", packet_size_int);
Peter Kastingf045e4d2015-06-10 21:15:38 -0700458 return -1;
459 }
Peter Kastingdce40cf2015-08-24 14:52:23 -0700460 printf("Packet size: %d\n", packet_size_int);
461 packet_size = static_cast<size_t>(packet_size_int);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000462
Peter Kasting248b0b02015-06-03 12:32:41 -0700463 // check for stereo
464 if (argv[4][strlen(argv[4]) - 1] == '*') {
465 // use stereo
466 usingStereo = true;
467 numChannels = 2;
468 argv[4][strlen(argv[4]) - 1] = '\0';
469 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000470
Peter Kasting248b0b02015-06-03 12:32:41 -0700471 NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs,
472 &bitrate, &useRed);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000473
Peter Kasting248b0b02015-06-03 12:32:41 -0700474 if (useRed) {
475 RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant
476 payload, except last one which is 1 byte */
477 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000478
Peter Kasting248b0b02015-06-03 12:32:41 -0700479 useVAD = atoi(argv[5]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000480#if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700481 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
482 if (useVAD != 0) {
483 printf("Error: this simulation does not support VAD/DTX/CNG\n");
484 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000485#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000486
Peter Kasting248b0b02015-06-03 12:32:41 -0700487 // check stereo type
488 if (usingStereo) {
489 switch (usedCodec) {
490 // sample based codecs
491 case webrtc::kDecoderPCMu:
492 case webrtc::kDecoderPCMa:
493 case webrtc::kDecoderG722: {
494 // 1 octet per sample
495 stereoMode = STEREO_MODE_SAMPLE_1;
496 break;
497 }
498 case webrtc::kDecoderPCM16B:
499 case webrtc::kDecoderPCM16Bwb:
500 case webrtc::kDecoderPCM16Bswb32kHz:
501 case webrtc::kDecoderPCM16Bswb48kHz: {
502 // 2 octets per sample
503 stereoMode = STEREO_MODE_SAMPLE_2;
504 break;
505 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000506
Peter Kasting248b0b02015-06-03 12:32:41 -0700507 // fixed-rate frame codecs (with internal VAD)
508 default: {
509 printf("Cannot use codec %s as stereo codec\n", argv[4]);
510 exit(0);
511 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000512 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700513 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000514
Peter Kasting248b0b02015-06-03 12:32:41 -0700515 if ((usedCodec == webrtc::kDecoderISAC) ||
516 (usedCodec == webrtc::kDecoderISACswb)) {
517 if (argc != 7) {
518 if (usedCodec == webrtc::kDecoderISAC) {
519 bitrate = 32000;
Peter Kasting2a100872015-06-09 17:26:40 -0700520 printf("Running iSAC at default bitrate of 32000 bps (to specify "
521 "explicitly add the bps as last parameter)\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700522 } else // (usedCodec==webrtc::kDecoderISACswb)
523 {
524 bitrate = 56000;
Peter Kasting2a100872015-06-09 17:26:40 -0700525 printf("Running iSAC at default bitrate of 56000 bps (to specify "
526 "explicitly add the bps as last parameter)\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700527 }
528 } else {
529 bitrate = atoi(argv[6]);
530 if (usedCodec == webrtc::kDecoderISAC) {
531 if ((bitrate < 10000) || (bitrate > 32000)) {
Peter Kasting2a100872015-06-09 17:26:40 -0700532 printf("Error: iSAC bitrate must be between 10000 and 32000 bps (%i "
533 "is invalid)\n", bitrate);
Peter Kasting248b0b02015-06-03 12:32:41 -0700534 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000535 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700536 printf("Running iSAC at bitrate of %i bps\n", bitrate);
537 } else // (usedCodec==webrtc::kDecoderISACswb)
538 {
539 if ((bitrate < 32000) || (bitrate > 56000)) {
Peter Kasting2a100872015-06-09 17:26:40 -0700540 printf("Error: iSAC SWB bitrate must be between 32000 and 56000 bps "
541 "(%i is invalid)\n", bitrate);
Peter Kasting248b0b02015-06-03 12:32:41 -0700542 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000543 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700544 }
545 }
546 } else {
547 if (argc == 7) {
Peter Kasting2a100872015-06-09 17:26:40 -0700548 printf("Error: Bitrate parameter can only be specified for iSAC, G.723, "
549 "and G.729.1\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700550 exit(0);
551 }
552 }
553
554 if (useRed) {
555 printf("Redundancy engaged. ");
556 }
557 printf("Used codec: %i\n", usedCodec);
558 printf("Payload type: %i\n", payloadType);
559
560 NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD,
561 numChannels);
562
563 /* write file header */
564 // fprintf(out_file, "#!RTPencode%s\n", "1.0");
565 fprintf(out_file, "#!rtpplay%s \n",
566 "1.0"); // this is the string that rtpplay needs
567 uint32_t dummy_variable = 0; // should be converted to network endian format,
568 // but does not matter when 0
569 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
570 return -1;
571 }
572 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
573 return -1;
574 }
575 if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
576 return -1;
577 }
578 if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
579 return -1;
580 }
581 if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
582 return -1;
583 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000584
585#ifdef TIMESTAMP_WRAPAROUND
Peter Kasting248b0b02015-06-03 12:32:41 -0700586 timestamp = 0xFFFFFFFF - fs * 10; /* should give wrap-around in 10 seconds */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000587#endif
588#if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
Peter Kasting248b0b02015-06-03 12:32:41 -0700589 srand(RANDOM_SEED);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000590#endif
591
Peter Kasting248b0b02015-06-03 12:32:41 -0700592 /* if redundancy is used, the first redundant payload is zero length */
593 red_len[0] = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000594
Peter Kasting248b0b02015-06-03 12:32:41 -0700595 /* read first frame */
596 len = fread(org_data, 2, packet_size * numChannels, in_file) / numChannels;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000597
Peter Kasting248b0b02015-06-03 12:32:41 -0700598 /* de-interleave if stereo */
599 if (usingStereo) {
600 stereoDeInterleave(org_data, len * numChannels);
601 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000602
Peter Kasting248b0b02015-06-03 12:32:41 -0700603 while (len == packet_size) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000604#ifdef INSERT_DTMF_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700605 dtmfSent = false;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000606
Peter Kasting248b0b02015-06-03 12:32:41 -0700607 if (sendtime >= NTone * DTMF_PACKET_INTERVAL) {
608 if (sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION) {
609 // tone has not ended
610 if (DTMFfirst == 1) {
611 DTMFtimestamp = timestamp; // save this timestamp
612 DTMFfirst = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000613 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700614 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
615 enc_len = makeDTMFpayload(
616 &rtp_data[12], NTone % 12, 0, 4,
617 (int)(sendtime - NTone * DTMF_PACKET_INTERVAL) * (fs / 1000) + len);
618 } else {
619 // tone has ended
620 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
621 enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4,
622 DTMF_DURATION * (fs / 1000));
623 NTone++;
624 DTMFfirst = 1;
625 }
626
627 /* write RTP packet to file */
Peter Kastingb7e50542015-06-11 12:55:50 -0700628 length = htons(static_cast<unsigned short>(12 + enc_len + 8));
629 plen = htons(static_cast<unsigned short>(12 + enc_len));
Peter Kasting248b0b02015-06-03 12:32:41 -0700630 offset = (uint32_t)sendtime; //(timestamp/(fs/1000));
631 offset = htonl(offset);
632 if (fwrite(&length, 2, 1, out_file) != 1) {
633 return -1;
634 }
635 if (fwrite(&plen, 2, 1, out_file) != 1) {
636 return -1;
637 }
638 if (fwrite(&offset, 4, 1, out_file) != 1) {
639 return -1;
640 }
641 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
642 return -1;
643 }
644
645 dtmfSent = true;
646 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000647#endif
648
649#ifdef NO_DTMF_OVERDUB
Peter Kasting248b0b02015-06-03 12:32:41 -0700650 /* If DTMF is sent, we should not send any speech packets during the same
651 * time */
652 if (dtmfSent) {
653 enc_len = 0;
654 } else {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000655#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700656 /* encode frame */
657 enc_len =
658 NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12], fs,
659 &vad, useVAD, bitrate, numChannels);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000660
Peter Kasting248b0b02015-06-03 12:32:41 -0700661 if (usingStereo && stereoMode != STEREO_MODE_FRAME && vad == 1) {
662 // interleave the encoded payload for sample-based codecs (not for CNG)
663 stereoInterleave(&rtp_data[12], enc_len, stereoMode);
664 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000665#ifdef NO_DTMF_OVERDUB
Peter Kasting248b0b02015-06-03 12:32:41 -0700666 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000667#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000668
Peter Kasting248b0b02015-06-03 12:32:41 -0700669 if (enc_len > 0 &&
670 (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
671 if (useRed) {
672 if (red_len[0] > 0) {
673 memmove(&rtp_data[RTPheaderLen + red_len[0]], &rtp_data[12], enc_len);
674 memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000675
Peter Kastingb7e50542015-06-11 12:55:50 -0700676 red_len[1] = static_cast<uint16_t>(enc_len);
Peter Kasting248b0b02015-06-03 12:32:41 -0700677 red_TS[1] = timestamp;
678 if (vad)
679 red_PT[1] = payloadType;
680 else
681 red_PT[1] = NETEQ_CODEC_CN_PT;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000682
Peter Kasting248b0b02015-06-03 12:32:41 -0700683 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++,
684 ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000685
Peter Kasting248b0b02015-06-03 12:32:41 -0700686 enc_len += red_len[0] + RTPheaderLen - 12;
687 } else { // do not use redundancy payload for this packet, i.e., only
688 // last payload
689 memmove(&rtp_data[RTPheaderLen - 4], &rtp_data[12], enc_len);
690 // memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000691
Peter Kastingb7e50542015-06-11 12:55:50 -0700692 red_len[1] = static_cast<uint16_t>(enc_len);
Peter Kasting248b0b02015-06-03 12:32:41 -0700693 red_TS[1] = timestamp;
694 if (vad)
695 red_PT[1] = payloadType;
696 else
697 red_PT[1] = NETEQ_CODEC_CN_PT;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000698
Peter Kasting248b0b02015-06-03 12:32:41 -0700699 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++,
700 ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000701
Peter Kasting248b0b02015-06-03 12:32:41 -0700702 enc_len += red_len[0] + RTPheaderLen - 4 -
703 12; // 4 is length of redundancy header (not used)
704 }
705 } else {
706 /* make RTP header */
707 if (vad) // regular speech data
708 makeRTPheader(rtp_data, payloadType, seqNo++, timestamp, ssrc);
709 else // CNG data
710 makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++, timestamp, ssrc);
711 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000712#ifdef MULTIPLE_SAME_TIMESTAMP
Peter Kasting248b0b02015-06-03 12:32:41 -0700713 int mult_pack = 0;
714 do {
715#endif // MULTIPLE_SAME_TIMESTAMP
716 /* write RTP packet to file */
Peter Kastingb7e50542015-06-11 12:55:50 -0700717 length = htons(static_cast<unsigned short>(12 + enc_len + 8));
718 plen = htons(static_cast<unsigned short>(12 + enc_len));
Peter Kasting248b0b02015-06-03 12:32:41 -0700719 offset = (uint32_t)sendtime;
720 //(timestamp/(fs/1000));
721 offset = htonl(offset);
722 if (fwrite(&length, 2, 1, out_file) != 1) {
723 return -1;
724 }
725 if (fwrite(&plen, 2, 1, out_file) != 1) {
726 return -1;
727 }
728 if (fwrite(&offset, 4, 1, out_file) != 1) {
729 return -1;
730 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000731#ifdef RANDOM_DATA
Peter Kastingdce40cf2015-08-24 14:52:23 -0700732 for (size_t k = 0; k < 12 + enc_len; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -0700733 rtp_data[k] = rand() + rand();
734 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000735#endif
736#ifdef RANDOM_PAYLOAD_DATA
Peter Kastingdce40cf2015-08-24 14:52:23 -0700737 for (size_t k = 12; k < 12 + enc_len; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -0700738 rtp_data[k] = rand() + rand();
739 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000740#endif
Peter Kasting248b0b02015-06-03 12:32:41 -0700741 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
742 return -1;
743 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000744#ifdef MULTIPLE_SAME_TIMESTAMP
Peter Kasting248b0b02015-06-03 12:32:41 -0700745 } while ((seqNo % REPEAT_PACKET_DISTANCE == 0) &&
746 (mult_pack++ < REPEAT_PACKET_COUNT));
747#endif // MULTIPLE_SAME_TIMESTAMP
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000748
749#ifdef INSERT_OLD_PACKETS
Peter Kasting248b0b02015-06-03 12:32:41 -0700750 if (packet_age >= OLD_PACKET * fs) {
751 if (!first_old_packet) {
752 // send the old packet
753 if (fwrite(&old_length, 2, 1, out_file) != 1) {
754 return -1;
755 }
756 if (fwrite(&old_plen, 2, 1, out_file) != 1) {
757 return -1;
758 }
759 if (fwrite(&offset, 4, 1, out_file) != 1) {
760 return -1;
761 }
762 if (fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file) != 1) {
763 return -1;
764 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000765 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700766 // store current packet as old
767 old_length = length;
768 old_plen = plen;
769 memcpy(old_rtp_data, rtp_data, 12 + enc_len);
770 old_enc_len = enc_len;
771 first_old_packet = 0;
772 packet_age = 0;
773 }
774 packet_age += packet_size;
775#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000776
Peter Kasting248b0b02015-06-03 12:32:41 -0700777 if (useRed) {
778/* move data to redundancy store */
779#ifdef CODEC_ISAC
780 if (usedCodec == webrtc::kDecoderISAC) {
781 assert(!usingStereo); // Cannot handle stereo yet
782 red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data);
783 } else {
784#endif
785 memcpy(red_data, &rtp_data[RTPheaderLen + red_len[0]], enc_len);
786 red_len[0] = red_len[1];
787#ifdef CODEC_ISAC
788 }
789#endif
790 red_TS[0] = red_TS[1];
791 red_PT[0] = red_PT[1];
792 }
793 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000794
Peter Kasting248b0b02015-06-03 12:32:41 -0700795 /* read next frame */
796 len = fread(org_data, 2, packet_size * numChannels, in_file) / numChannels;
797 /* de-interleave if stereo */
798 if (usingStereo) {
799 stereoDeInterleave(org_data, len * numChannels);
800 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000801
Peter Kasting248b0b02015-06-03 12:32:41 -0700802 if (payloadType == NETEQ_CODEC_G722_PT)
803 timestamp += len >> 1;
804 else
805 timestamp += len;
806
807 sendtime += (double)len / (fs / 1000);
808 }
809
810 NetEQTest_free_coders(usedCodec, numChannels);
811 fclose(in_file);
812 fclose(out_file);
813 printf("Done!\n");
814
815 return (0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000816}
817
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000818/****************/
819/* Subfunctions */
820/****************/
821
pkasting@chromium.orgd3245462015-02-23 21:28:22 +0000822void NetEQTest_GetCodec_and_PT(char* name,
823 webrtc::NetEqDecoder* codec,
824 int* PT,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700825 size_t frameLen,
pkasting@chromium.orgd3245462015-02-23 21:28:22 +0000826 int* fs,
827 int* bitrate,
828 int* useRed) {
Peter Kasting248b0b02015-06-03 12:32:41 -0700829 *bitrate = 0; /* Default bitrate setting */
830 *useRed = 0; /* Default no redundancy */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000831
Peter Kasting248b0b02015-06-03 12:32:41 -0700832 if (!strcmp(name, "pcmu")) {
833 *codec = webrtc::kDecoderPCMu;
834 *PT = NETEQ_CODEC_PCMU_PT;
835 *fs = 8000;
836 } else if (!strcmp(name, "pcma")) {
837 *codec = webrtc::kDecoderPCMa;
838 *PT = NETEQ_CODEC_PCMA_PT;
839 *fs = 8000;
840 } else if (!strcmp(name, "pcm16b")) {
841 *codec = webrtc::kDecoderPCM16B;
842 *PT = NETEQ_CODEC_PCM16B_PT;
843 *fs = 8000;
844 } else if (!strcmp(name, "pcm16b_wb")) {
845 *codec = webrtc::kDecoderPCM16Bwb;
846 *PT = NETEQ_CODEC_PCM16B_WB_PT;
847 *fs = 16000;
848 } else if (!strcmp(name, "pcm16b_swb32")) {
849 *codec = webrtc::kDecoderPCM16Bswb32kHz;
850 *PT = NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
851 *fs = 32000;
852 } else if (!strcmp(name, "pcm16b_swb48")) {
853 *codec = webrtc::kDecoderPCM16Bswb48kHz;
854 *PT = NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
855 *fs = 48000;
856 } else if (!strcmp(name, "g722")) {
857 *codec = webrtc::kDecoderG722;
858 *PT = NETEQ_CODEC_G722_PT;
859 *fs = 16000;
860 } else if ((!strcmp(name, "ilbc")) &&
861 ((frameLen % 240 == 0) || (frameLen % 160 == 0))) {
862 *fs = 8000;
863 *codec = webrtc::kDecoderILBC;
864 *PT = NETEQ_CODEC_ILBC_PT;
865 } else if (!strcmp(name, "isac")) {
866 *fs = 16000;
867 *codec = webrtc::kDecoderISAC;
868 *PT = NETEQ_CODEC_ISAC_PT;
869 } else if (!strcmp(name, "isacswb")) {
870 *fs = 32000;
871 *codec = webrtc::kDecoderISACswb;
872 *PT = NETEQ_CODEC_ISACSWB_PT;
873 } else if (!strcmp(name, "red_pcm")) {
874 *codec = webrtc::kDecoderPCMa;
875 *PT = NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
876 *fs = 8000;
877 *useRed = 1;
878 } else if (!strcmp(name, "red_isac")) {
879 *codec = webrtc::kDecoderISAC;
880 *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
881 *fs = 16000;
882 *useRed = 1;
883 } else {
884 printf("Error: Not a supported codec (%s)\n", name);
885 exit(0);
886 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000887}
888
Peter Kasting248b0b02015-06-03 12:32:41 -0700889int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700890 size_t enc_frameSize,
Peter Kasting248b0b02015-06-03 12:32:41 -0700891 int bitrate,
892 int sampfreq,
893 int vad,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700894 size_t numChannels) {
Peter Kasting248b0b02015-06-03 12:32:41 -0700895 int ok = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000896
Peter Kastingdce40cf2015-08-24 14:52:23 -0700897 for (size_t k = 0; k < numChannels; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -0700898 VAD_inst[k] = WebRtcVad_Create();
899 if (!VAD_inst[k]) {
900 printf("Error: Couldn't allocate memory for VAD instance\n");
901 exit(0);
902 }
903 ok = WebRtcVad_Init(VAD_inst[k]);
904 if (ok == -1) {
905 printf("Error: Initialization of VAD struct failed\n");
906 exit(0);
907 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000908
909#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -0700910 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
911 ok = WebRtcCng_CreateEnc(&CNGenc_inst[k]);
912 if (ok != 0) {
913 printf("Error: Couldn't allocate memory for CNG encoding instance\n");
914 exit(0);
915 }
916 if (sampfreq <= 16000) {
917 ok = WebRtcCng_InitEnc(CNGenc_inst[k], sampfreq, 200, 5);
918 if (ok == -1) {
919 printf("Error: Initialization of CNG struct failed. Error code %d\n",
920 WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
921 exit(0);
922 }
923 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000924#endif
925
Peter Kasting248b0b02015-06-03 12:32:41 -0700926 switch (coder) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000927#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -0700928 case webrtc::kDecoderPCM16B:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000929#endif
930#ifdef CODEC_PCM16B_WB
Peter Kasting248b0b02015-06-03 12:32:41 -0700931 case webrtc::kDecoderPCM16Bwb:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000932#endif
933#ifdef CODEC_PCM16B_32KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -0700934 case webrtc::kDecoderPCM16Bswb32kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000935#endif
936#ifdef CODEC_PCM16B_48KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -0700937 case webrtc::kDecoderPCM16Bswb48kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000938#endif
939#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -0700940 case webrtc::kDecoderPCMu:
941 case webrtc::kDecoderPCMa:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000942#endif
943 // do nothing
944 break;
945#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -0700946 case webrtc::kDecoderG729:
947 if (sampfreq == 8000) {
948 if ((enc_frameSize == 80) || (enc_frameSize == 160) ||
949 (enc_frameSize == 240) || (enc_frameSize == 320) ||
950 (enc_frameSize == 400) || (enc_frameSize == 480)) {
951 ok = WebRtcG729_CreateEnc(&G729enc_inst[k]);
952 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -0700953 printf("Error: Couldn't allocate memory for G729 encoding "
954 "instance\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700955 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000956 }
Peter Kasting248b0b02015-06-03 12:32:41 -0700957 } else {
Peter Kasting2a100872015-06-09 17:26:40 -0700958 printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 "
959 "ms!!\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000960 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -0700961 }
962 WebRtcG729_EncoderInit(G729enc_inst[k], vad);
963 if ((vad == 1) && (enc_frameSize != 80)) {
Peter Kasting2a100872015-06-09 17:26:40 -0700964 printf("\nError - This simulation only supports VAD for G729 at "
Peter Kastingdce40cf2015-08-24 14:52:23 -0700965 "10ms packets (not %" PRIuS "ms)\n", (enc_frameSize >> 3));
Peter Kasting248b0b02015-06-03 12:32:41 -0700966 }
967 } else {
968 printf("\nError - g729 is only developed for 8kHz \n");
969 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000970 }
971 break;
972#endif
973#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -0700974 case webrtc::kDecoderG729_1:
975 if (sampfreq == 16000) {
976 if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
977 (enc_frameSize == 960)) {
978 ok = WebRtcG7291_Create(&G729_1_inst[k]);
979 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -0700980 printf("Error: Couldn't allocate memory for G.729.1 codec "
981 "instance\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700982 exit(0);
983 }
984 } else {
985 printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000986 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -0700987 }
988 if (!(((bitrate >= 12000) && (bitrate <= 32000) &&
989 (bitrate % 2000 == 0)) ||
990 (bitrate == 8000))) {
991 /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
Peter Kasting2a100872015-06-09 17:26:40 -0700992 printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in "
993 "steps of 2000 bps\n");
Peter Kasting248b0b02015-06-03 12:32:41 -0700994 exit(0);
995 }
996 WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/,
997 0 /*flagG729mode*/);
998 } else {
999 printf("\nError - G.729.1 input is always 16 kHz \n");
1000 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001001 }
1002 break;
1003#endif
1004#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -07001005 case webrtc::kDecoderSPEEX_8:
1006 if (sampfreq == 8000) {
1007 if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
1008 (enc_frameSize == 480)) {
1009 ok = WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
1010 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -07001011 printf("Error: Couldn't allocate memory for Speex encoding "
1012 "instance\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001013 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001014 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001015 } else {
1016 printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
1017 exit(0);
1018 }
1019 if ((vad == 1) && (enc_frameSize != 160)) {
Peter Kasting2a100872015-06-09 17:26:40 -07001020 printf("\nError - This simulation only supports VAD for Speex at "
Peter Kastingdce40cf2015-08-24 14:52:23 -07001021 "20ms packets (not %" PRIuS "ms)\n",
Peter Kasting248b0b02015-06-03 12:32:41 -07001022 (enc_frameSize >> 3));
1023 vad = 0;
1024 }
1025 ok = WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0 /*vbr*/,
1026 3 /*complexity*/, vad);
1027 if (ok != 0)
1028 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001029 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001030 printf("\nError - Speex8 called with sample frequency other than 8 "
1031 "kHz.\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001032 }
1033 break;
1034#endif
1035#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001036 case webrtc::kDecoderSPEEX_16:
1037 if (sampfreq == 16000) {
1038 if ((enc_frameSize == 320) || (enc_frameSize == 640) ||
1039 (enc_frameSize == 960)) {
1040 ok = WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
1041 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -07001042 printf("Error: Couldn't allocate memory for Speex encoding "
1043 "instance\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001044 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001045 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001046 } else {
1047 printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
1048 exit(0);
1049 }
1050 if ((vad == 1) && (enc_frameSize != 320)) {
Peter Kasting2a100872015-06-09 17:26:40 -07001051 printf("\nError - This simulation only supports VAD for Speex at "
Peter Kastingdce40cf2015-08-24 14:52:23 -07001052 "20ms packets (not %" PRIuS "ms)\n",
Peter Kasting248b0b02015-06-03 12:32:41 -07001053 (enc_frameSize >> 4));
1054 vad = 0;
1055 }
1056 ok = WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0 /*vbr*/,
1057 3 /*complexity*/, vad);
1058 if (ok != 0)
1059 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001060 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001061 printf("\nError - Speex16 called with sample frequency other than 16 "
1062 "kHz.\n\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001063 }
1064 break;
1065#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001066
1067#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001068 case webrtc::kDecoderG722_1_16:
1069 if (sampfreq == 16000) {
1070 ok = WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
1071 if (ok != 0) {
1072 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001073 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001074 }
1075 if (enc_frameSize == 320) {
1076 } else {
1077 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1078 exit(0);
1079 }
1080 WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
1081 } else {
1082 printf("\nError - G722.1 is only developed for 16kHz \n");
1083 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001084 }
1085 break;
1086#endif
1087#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001088 case webrtc::kDecoderG722_1_24:
1089 if (sampfreq == 16000) {
1090 ok = WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
1091 if (ok != 0) {
1092 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001093 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001094 }
1095 if (enc_frameSize == 320) {
1096 } else {
1097 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1098 exit(0);
1099 }
1100 WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
1101 } else {
1102 printf("\nError - G722.1 is only developed for 16kHz \n");
1103 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001104 }
1105 break;
1106#endif
1107#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001108 case webrtc::kDecoderG722_1_32:
1109 if (sampfreq == 16000) {
1110 ok = WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
1111 if (ok != 0) {
1112 printf("Error: Couldn't allocate memory for G.722.1 instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001113 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001114 }
1115 if (enc_frameSize == 320) {
1116 } else {
1117 printf("\nError: G722.1 only supports 20 ms!!\n\n");
1118 exit(0);
1119 }
1120 WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
1121 } else {
1122 printf("\nError - G722.1 is only developed for 16kHz \n");
1123 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001124 }
1125 break;
1126#endif
1127#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001128 case webrtc::kDecoderG722_1C_24:
1129 if (sampfreq == 32000) {
1130 ok = WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
1131 if (ok != 0) {
1132 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001133 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001134 }
1135 if (enc_frameSize == 640) {
1136 } else {
1137 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1138 exit(0);
1139 }
1140 WebRtcG7221C_EncoderInit24(
1141 (G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
1142 } else {
1143 printf("\nError - G722.1 C is only developed for 32kHz \n");
1144 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001145 }
1146 break;
1147#endif
1148#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001149 case webrtc::kDecoderG722_1C_32:
1150 if (sampfreq == 32000) {
1151 ok = WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
1152 if (ok != 0) {
1153 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001154 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001155 }
1156 if (enc_frameSize == 640) {
1157 } else {
1158 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1159 exit(0);
1160 }
1161 WebRtcG7221C_EncoderInit32(
1162 (G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
1163 } else {
1164 printf("\nError - G722.1 C is only developed for 32kHz \n");
1165 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001166 }
1167 break;
1168#endif
1169#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -07001170 case webrtc::kDecoderG722_1C_48:
1171 if (sampfreq == 32000) {
1172 ok = WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
1173 if (ok != 0) {
1174 printf("Error: Couldn't allocate memory for G.722.1C instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001175 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001176 }
1177 if (enc_frameSize == 640) {
1178 } else {
1179 printf("\nError: G722.1 C only supports 20 ms!!\n\n");
1180 exit(0);
1181 }
1182 WebRtcG7221C_EncoderInit48(
1183 (G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
1184 } else {
1185 printf("\nError - G722.1 C is only developed for 32kHz \n");
1186 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001187 }
1188 break;
1189#endif
1190#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -07001191 case webrtc::kDecoderG722:
1192 if (sampfreq == 16000) {
1193 if (enc_frameSize % 2 == 0) {
1194 } else {
1195 printf(
1196 "\nError - g722 frames must have an even number of "
1197 "enc_frameSize\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001198 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001199 }
1200 WebRtcG722_CreateEncoder(&g722EncState[k]);
1201 WebRtcG722_EncoderInit(g722EncState[k]);
1202 } else {
1203 printf("\nError - g722 is only developed for 16kHz \n");
1204 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001205 }
1206 break;
1207#endif
1208#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -07001209 case webrtc::kDecoderAMR:
1210 if (sampfreq == 8000) {
1211 ok = WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
1212 if (ok != 0) {
1213 printf(
1214 "Error: Couldn't allocate memory for AMR encoding instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001215 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001216 }
1217 if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
1218 (enc_frameSize == 480)) {
1219 } else {
1220 printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
1221 exit(0);
1222 }
1223 WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
1224 WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
1225 AMR_bitrate = bitrate;
1226 } else {
1227 printf("\nError - AMR is only developed for 8kHz \n");
1228 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001229 }
1230 break;
1231#endif
1232#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001233 case webrtc::kDecoderAMRWB:
1234 if (sampfreq == 16000) {
1235 ok = WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
1236 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -07001237 printf("Error: Couldn't allocate memory for AMRWB encoding "
1238 "instance\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001239 exit(0);
1240 }
Peter Kastingdce40cf2015-08-24 14:52:23 -07001241 if (((enc_frameSize / 320) > 3) || ((enc_frameSize % 320) != 0)) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001242 printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
1243 exit(0);
1244 }
1245 WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
1246 if (bitrate == 7000) {
1247 AMRWB_bitrate = AMRWB_MODE_7k;
1248 } else if (bitrate == 9000) {
1249 AMRWB_bitrate = AMRWB_MODE_9k;
1250 } else if (bitrate == 12000) {
1251 AMRWB_bitrate = AMRWB_MODE_12k;
1252 } else if (bitrate == 14000) {
1253 AMRWB_bitrate = AMRWB_MODE_14k;
1254 } else if (bitrate == 16000) {
1255 AMRWB_bitrate = AMRWB_MODE_16k;
1256 } else if (bitrate == 18000) {
1257 AMRWB_bitrate = AMRWB_MODE_18k;
1258 } else if (bitrate == 20000) {
1259 AMRWB_bitrate = AMRWB_MODE_20k;
1260 } else if (bitrate == 23000) {
1261 AMRWB_bitrate = AMRWB_MODE_23k;
1262 } else if (bitrate == 24000) {
1263 AMRWB_bitrate = AMRWB_MODE_24k;
1264 }
1265 WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001266
1267 } else {
Peter Kasting248b0b02015-06-03 12:32:41 -07001268 printf("\nError - AMRwb is only developed for 16kHz \n");
1269 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001270 }
1271 break;
1272#endif
1273#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -07001274 case webrtc::kDecoderILBC:
1275 if (sampfreq == 8000) {
1276 ok = WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
1277 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -07001278 printf("Error: Couldn't allocate memory for iLBC encoding "
1279 "instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001280 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001281 }
1282 if ((enc_frameSize == 160) || (enc_frameSize == 240) ||
1283 (enc_frameSize == 320) || (enc_frameSize == 480)) {
1284 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001285 printf("\nError - iLBC only supports 160, 240, 320 and 480 "
1286 "enc_frameSize (20, 30, 40 and 60 ms)\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001287 exit(0);
1288 }
1289 if ((enc_frameSize == 160) || (enc_frameSize == 320)) {
1290 /* 20 ms version */
1291 WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
1292 } else {
1293 /* 30 ms version */
1294 WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
1295 }
1296 } else {
1297 printf("\nError - iLBC is only developed for 8kHz \n");
1298 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001299 }
1300 break;
1301#endif
1302#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -07001303 case webrtc::kDecoderISAC:
1304 if (sampfreq == 16000) {
1305 ok = WebRtcIsac_Create(&ISAC_inst[k]);
1306 if (ok != 0) {
1307 printf("Error: Couldn't allocate memory for iSAC instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001308 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001309 }
1310 if ((enc_frameSize == 480) || (enc_frameSize == 960)) {
1311 } else {
1312 printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
1313 exit(0);
1314 }
1315 WebRtcIsac_EncoderInit(ISAC_inst[k], 1);
1316 if ((bitrate < 10000) || (bitrate > 32000)) {
Peter Kasting2a100872015-06-09 17:26:40 -07001317 printf("\nError - iSAC bitrate has to be between 10000 and 32000 "
1318 "bps (not %i)\n",
Peter Kasting248b0b02015-06-03 12:32:41 -07001319 bitrate);
1320 exit(0);
1321 }
Peter Kastingdce40cf2015-08-24 14:52:23 -07001322 WebRtcIsac_Control(ISAC_inst[k], bitrate,
1323 static_cast<int>(enc_frameSize >> 4));
Peter Kasting248b0b02015-06-03 12:32:41 -07001324 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001325 printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or "
1326 "60 ms)\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001327 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001328 }
1329 break;
1330#endif
1331#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -07001332 case webrtc::kDecoderISAC:
1333 if (sampfreq == 16000) {
1334 ok = WebRtcIsacfix_Create(&ISAC_inst[k]);
1335 if (ok != 0) {
1336 printf("Error: Couldn't allocate memory for iSAC instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001337 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001338 }
1339 if ((enc_frameSize == 480) || (enc_frameSize == 960)) {
1340 } else {
1341 printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
1342 exit(0);
1343 }
1344 WebRtcIsacfix_EncoderInit(ISAC_inst[k], 1);
1345 if ((bitrate < 10000) || (bitrate > 32000)) {
Peter Kasting2a100872015-06-09 17:26:40 -07001346 printf("\nError - iSAC bitrate has to be between 10000 and 32000 "
1347 "bps (not %i)\n", bitrate);
Peter Kasting248b0b02015-06-03 12:32:41 -07001348 exit(0);
1349 }
1350 WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize >> 4);
1351 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001352 printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or "
1353 "60 ms)\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001354 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001355 }
1356 break;
1357#endif
1358#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001359 case webrtc::kDecoderISACswb:
1360 if (sampfreq == 32000) {
1361 ok = WebRtcIsac_Create(&ISACSWB_inst[k]);
1362 if (ok != 0) {
1363 printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001364 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001365 }
1366 if (enc_frameSize == 960) {
1367 } else {
1368 printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
1369 exit(0);
1370 }
1371 ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
1372 if (ok != 0) {
1373 printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
1374 exit(0);
1375 }
1376 WebRtcIsac_EncoderInit(ISACSWB_inst[k], 1);
1377 if ((bitrate < 32000) || (bitrate > 56000)) {
Peter Kasting2a100872015-06-09 17:26:40 -07001378 printf("\nError - iSAC SWB bitrate has to be between 32000 and "
1379 "56000 bps (not %i)\n", bitrate);
Peter Kasting248b0b02015-06-03 12:32:41 -07001380 exit(0);
1381 }
Peter Kastingdce40cf2015-08-24 14:52:23 -07001382 WebRtcIsac_Control(ISACSWB_inst[k], bitrate,
1383 static_cast<int>(enc_frameSize >> 5));
Peter Kasting248b0b02015-06-03 12:32:41 -07001384 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001385 printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 "
1386 "ms)\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001387 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001388 }
1389 break;
1390#endif
1391#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -07001392 case webrtc::kDecoderGSMFR:
1393 if (sampfreq == 8000) {
1394 ok = WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
1395 if (ok != 0) {
Peter Kasting2a100872015-06-09 17:26:40 -07001396 printf("Error: Couldn't allocate memory for GSM FR encoding "
1397 "instance\n");
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001398 exit(0);
Peter Kasting248b0b02015-06-03 12:32:41 -07001399 }
1400 if ((enc_frameSize == 160) || (enc_frameSize == 320) ||
1401 (enc_frameSize == 480)) {
1402 } else {
Peter Kasting2a100872015-06-09 17:26:40 -07001403 printf("\nError - GSM FR must have a multiple of 160 "
1404 "enc_frameSize\n");
Peter Kasting248b0b02015-06-03 12:32:41 -07001405 exit(0);
1406 }
1407 WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
1408 } else {
1409 printf("\nError - GSM FR is only developed for 8kHz \n");
1410 exit(0);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001411 }
1412 break;
1413#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001414 default:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001415 printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
1416 exit(0);
1417 break;
Peter Kasting248b0b02015-06-03 12:32:41 -07001418 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001419
Peter Kasting248b0b02015-06-03 12:32:41 -07001420 if (ok != 0) {
1421 return (ok);
1422 }
1423 } // end for
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001424
Peter Kasting248b0b02015-06-03 12:32:41 -07001425 return (0);
1426}
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001427
Peter Kastingdce40cf2015-08-24 14:52:23 -07001428int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
1429 for (size_t k = 0; k < numChannels; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001430 WebRtcVad_Free(VAD_inst[k]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001431#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
Peter Kasting248b0b02015-06-03 12:32:41 -07001432 defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
1433 WebRtcCng_FreeEnc(CNGenc_inst[k]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001434#endif
1435
Peter Kasting248b0b02015-06-03 12:32:41 -07001436 switch (coder) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001437#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -07001438 case webrtc::kDecoderPCM16B:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001439#endif
1440#ifdef CODEC_PCM16B_WB
Peter Kasting248b0b02015-06-03 12:32:41 -07001441 case webrtc::kDecoderPCM16Bwb:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001442#endif
1443#ifdef CODEC_PCM16B_32KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -07001444 case webrtc::kDecoderPCM16Bswb32kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001445#endif
1446#ifdef CODEC_PCM16B_48KHZ
Peter Kasting248b0b02015-06-03 12:32:41 -07001447 case webrtc::kDecoderPCM16Bswb48kHz:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001448#endif
1449#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -07001450 case webrtc::kDecoderPCMu:
1451 case webrtc::kDecoderPCMa:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001452#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001453 // do nothing
1454 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001455#ifdef CODEC_G729
Peter Kasting248b0b02015-06-03 12:32:41 -07001456 case webrtc::kDecoderG729:
1457 WebRtcG729_FreeEnc(G729enc_inst[k]);
1458 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001459#endif
1460#ifdef CODEC_G729_1
Peter Kasting248b0b02015-06-03 12:32:41 -07001461 case webrtc::kDecoderG729_1:
1462 WebRtcG7291_Free(G729_1_inst[k]);
1463 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001464#endif
1465#ifdef CODEC_SPEEX_8
Peter Kasting248b0b02015-06-03 12:32:41 -07001466 case webrtc::kDecoderSPEEX_8:
1467 WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
1468 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001469#endif
1470#ifdef CODEC_SPEEX_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001471 case webrtc::kDecoderSPEEX_16:
1472 WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
1473 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001474#endif
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001475
1476#ifdef CODEC_G722_1_16
Peter Kasting248b0b02015-06-03 12:32:41 -07001477 case webrtc::kDecoderG722_1_16:
1478 WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
1479 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001480#endif
1481#ifdef CODEC_G722_1_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001482 case webrtc::kDecoderG722_1_24:
1483 WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
1484 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001485#endif
1486#ifdef CODEC_G722_1_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001487 case webrtc::kDecoderG722_1_32:
1488 WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
1489 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001490#endif
1491#ifdef CODEC_G722_1C_24
Peter Kasting248b0b02015-06-03 12:32:41 -07001492 case webrtc::kDecoderG722_1C_24:
1493 WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
1494 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001495#endif
1496#ifdef CODEC_G722_1C_32
Peter Kasting248b0b02015-06-03 12:32:41 -07001497 case webrtc::kDecoderG722_1C_32:
1498 WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
1499 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001500#endif
1501#ifdef CODEC_G722_1C_48
Peter Kasting248b0b02015-06-03 12:32:41 -07001502 case webrtc::kDecoderG722_1C_48:
1503 WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
1504 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001505#endif
1506#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -07001507 case webrtc::kDecoderG722:
1508 WebRtcG722_FreeEncoder(g722EncState[k]);
1509 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001510#endif
1511#ifdef CODEC_AMR
Peter Kasting248b0b02015-06-03 12:32:41 -07001512 case webrtc::kDecoderAMR:
1513 WebRtcAmr_FreeEnc(AMRenc_inst[k]);
1514 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001515#endif
1516#ifdef CODEC_AMRWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001517 case webrtc::kDecoderAMRWB:
1518 WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
1519 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001520#endif
1521#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -07001522 case webrtc::kDecoderILBC:
1523 WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
1524 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001525#endif
1526#ifdef CODEC_ISAC
Peter Kasting248b0b02015-06-03 12:32:41 -07001527 case webrtc::kDecoderISAC:
1528 WebRtcIsac_Free(ISAC_inst[k]);
1529 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001530#endif
1531#ifdef NETEQ_ISACFIX_CODEC
Peter Kasting248b0b02015-06-03 12:32:41 -07001532 case webrtc::kDecoderISAC:
1533 WebRtcIsacfix_Free(ISAC_inst[k]);
1534 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001535#endif
1536#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001537 case webrtc::kDecoderISACswb:
1538 WebRtcIsac_Free(ISACSWB_inst[k]);
1539 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001540#endif
1541#ifdef CODEC_GSMFR
Peter Kasting248b0b02015-06-03 12:32:41 -07001542 case webrtc::kDecoderGSMFR:
1543 WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
1544 break;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001545#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001546 default:
1547 printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
1548 exit(0);
1549 break;
1550 }
1551 }
1552
1553 return (0);
1554}
1555
Peter Kastingdce40cf2015-08-24 14:52:23 -07001556size_t NetEQTest_encode(int coder,
1557 int16_t* indata,
1558 size_t frameLen,
1559 unsigned char* encoded,
1560 int sampleRate,
1561 int* vad,
1562 int useVAD,
1563 int bitrate,
1564 size_t numChannels) {
1565 size_t cdlen = 0;
Peter Kasting248b0b02015-06-03 12:32:41 -07001566 int16_t* tempdata;
1567 static int first_cng = 1;
Peter Kastingdce40cf2015-08-24 14:52:23 -07001568 size_t tempLen;
Peter Kasting248b0b02015-06-03 12:32:41 -07001569 *vad = 1;
1570
1571 // check VAD first
1572 if (useVAD) {
1573 *vad = 0;
1574
Peter Kastingdce40cf2015-08-24 14:52:23 -07001575 size_t sampleRate_10 = static_cast<size_t>(10 * sampleRate / 1000);
1576 size_t sampleRate_20 = static_cast<size_t>(20 * sampleRate / 1000);
1577 size_t sampleRate_30 = static_cast<size_t>(30 * sampleRate / 1000);
1578 for (size_t k = 0; k < numChannels; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001579 tempLen = frameLen;
1580 tempdata = &indata[k * frameLen];
1581 int localVad = 0;
1582 /* Partition the signal and test each chunk for VAD.
Peter Kastingdce40cf2015-08-24 14:52:23 -07001583 All chunks must be VAD=0 to produce a total VAD=0. */
pkastingb297c5a2015-07-22 15:17:22 -07001584 while (tempLen >= sampleRate_10) {
1585 if ((tempLen % sampleRate_30) == 0) { // tempLen is multiple of 30ms
Peter Kasting248b0b02015-06-03 12:32:41 -07001586 localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
pkastingb297c5a2015-07-22 15:17:22 -07001587 sampleRate_30);
1588 tempdata += sampleRate_30;
1589 tempLen -= sampleRate_30;
1590 } else if (tempLen >= sampleRate_20) { // tempLen >= 20ms
Peter Kasting248b0b02015-06-03 12:32:41 -07001591 localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
pkastingb297c5a2015-07-22 15:17:22 -07001592 sampleRate_20);
1593 tempdata += sampleRate_20;
1594 tempLen -= sampleRate_20;
Peter Kasting248b0b02015-06-03 12:32:41 -07001595 } else { // use 10ms
1596 localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata,
pkastingb297c5a2015-07-22 15:17:22 -07001597 sampleRate_10);
1598 tempdata += sampleRate_10;
1599 tempLen -= sampleRate_10;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001600 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001601 }
1602
1603 // aggregate all VAD decisions over all channels
1604 *vad |= localVad;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001605 }
1606
Peter Kasting248b0b02015-06-03 12:32:41 -07001607 if (!*vad) {
1608 // all channels are silent
1609 cdlen = 0;
Peter Kastingdce40cf2015-08-24 14:52:23 -07001610 for (size_t k = 0; k < numChannels; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001611 WebRtcCng_Encode(CNGenc_inst[k], &indata[k * frameLen],
1612 (frameLen <= 640 ? frameLen : 640) /* max 640 */,
1613 encoded, &tempLen, first_cng);
1614 encoded += tempLen;
1615 cdlen += tempLen;
1616 }
1617 *vad = 0;
1618 first_cng = 0;
1619 return (cdlen);
1620 }
1621 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001622
Peter Kasting248b0b02015-06-03 12:32:41 -07001623 // loop over all channels
Peter Kastingdce40cf2015-08-24 14:52:23 -07001624 size_t totalLen = 0;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001625
Peter Kastingdce40cf2015-08-24 14:52:23 -07001626 for (size_t k = 0; k < numChannels; k++) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001627 /* Encode with the selected coder type */
1628 if (coder == webrtc::kDecoderPCMu) { /*g711 u-law */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001629#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -07001630 cdlen = WebRtcG711_EncodeU(indata, frameLen, encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001631#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001632 } else if (coder == webrtc::kDecoderPCMa) { /*g711 A-law */
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001633#ifdef CODEC_G711
Peter Kasting248b0b02015-06-03 12:32:41 -07001634 cdlen = WebRtcG711_EncodeA(indata, frameLen, encoded);
1635 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001636#endif
1637#ifdef CODEC_PCM16B
Peter Kasting248b0b02015-06-03 12:32:41 -07001638 else if ((coder == webrtc::kDecoderPCM16B) ||
1639 (coder == webrtc::kDecoderPCM16Bwb) ||
1640 (coder == webrtc::kDecoderPCM16Bswb32kHz) ||
1641 (coder == webrtc::
1642 kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz,
1643 32kHz or 48kHz) */
1644 cdlen = WebRtcPcm16b_Encode(indata, frameLen, encoded);
1645 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001646#endif
1647#ifdef CODEC_G722
Peter Kasting248b0b02015-06-03 12:32:41 -07001648 else if (coder == webrtc::kDecoderG722) { /*g722 */
1649 cdlen = WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded);
1650 assert(cdlen == frameLen >> 1);
1651 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001652#endif
1653#ifdef CODEC_ILBC
Peter Kasting248b0b02015-06-03 12:32:41 -07001654 else if (coder == webrtc::kDecoderILBC) { /*iLBC */
Peter Kastingdce40cf2015-08-24 14:52:23 -07001655 cdlen = static_cast<size_t>(std::max(
1656 WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata, frameLen, encoded), 0));
Peter Kasting248b0b02015-06-03 12:32:41 -07001657 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001658#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001659#if (defined(CODEC_ISAC) || \
1660 defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all
1661 // NETEQ_ISACFIX_CODEC
1662 else if (coder == webrtc::kDecoderISAC) { /*iSAC */
1663 int noOfCalls = 0;
Peter Kastingdce40cf2015-08-24 14:52:23 -07001664 int res = 0;
1665 while (res <= 0) {
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001666#ifdef CODEC_ISAC /* floating point */
Peter Kastingdce40cf2015-08-24 14:52:23 -07001667 res =
Peter Kasting248b0b02015-06-03 12:32:41 -07001668 WebRtcIsac_Encode(ISAC_inst[k], &indata[noOfCalls * 160], encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001669#else /* fixed point */
Peter Kastingdce40cf2015-08-24 14:52:23 -07001670 res = WebRtcIsacfix_Encode(ISAC_inst[k], &indata[noOfCalls * 160],
1671 encoded);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001672#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001673 noOfCalls++;
1674 }
Peter Kastingdce40cf2015-08-24 14:52:23 -07001675 cdlen = static_cast<size_t>(res);
Peter Kasting248b0b02015-06-03 12:32:41 -07001676 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001677#endif
1678#ifdef CODEC_ISAC_SWB
Peter Kasting248b0b02015-06-03 12:32:41 -07001679 else if (coder == webrtc::kDecoderISACswb) { /* iSAC SWB */
1680 int noOfCalls = 0;
Peter Kastingdce40cf2015-08-24 14:52:23 -07001681 int res = 0;
1682 while (res <= 0) {
1683 res = WebRtcIsac_Encode(ISACSWB_inst[k], &indata[noOfCalls * 320],
1684 encoded);
Peter Kasting248b0b02015-06-03 12:32:41 -07001685 noOfCalls++;
1686 }
Peter Kastingdce40cf2015-08-24 14:52:23 -07001687 cdlen = static_cast<size_t>(res);
Peter Kasting248b0b02015-06-03 12:32:41 -07001688 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001689#endif
Peter Kasting248b0b02015-06-03 12:32:41 -07001690 indata += frameLen;
1691 encoded += cdlen;
1692 totalLen += cdlen;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001693
Peter Kasting248b0b02015-06-03 12:32:41 -07001694 } // end for
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001695
Peter Kasting248b0b02015-06-03 12:32:41 -07001696 first_cng = 1;
1697 return (totalLen);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001698}
1699
pkasting@chromium.orgd3245462015-02-23 21:28:22 +00001700void makeRTPheader(unsigned char* rtp_data,
1701 int payloadType,
1702 int seqNo,
1703 uint32_t timestamp,
1704 uint32_t ssrc) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001705 rtp_data[0] = 0x80;
1706 rtp_data[1] = payloadType & 0xFF;
1707 rtp_data[2] = (seqNo >> 8) & 0xFF;
1708 rtp_data[3] = seqNo & 0xFF;
1709 rtp_data[4] = timestamp >> 24;
1710 rtp_data[5] = (timestamp >> 16) & 0xFF;
1711 rtp_data[6] = (timestamp >> 8) & 0xFF;
1712 rtp_data[7] = timestamp & 0xFF;
1713 rtp_data[8] = ssrc >> 24;
1714 rtp_data[9] = (ssrc >> 16) & 0xFF;
1715 rtp_data[10] = (ssrc >> 8) & 0xFF;
1716 rtp_data[11] = ssrc & 0xFF;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001717}
1718
pkasting@chromium.orgd3245462015-02-23 21:28:22 +00001719int makeRedundantHeader(unsigned char* rtp_data,
1720 int* payloadType,
1721 int numPayloads,
1722 uint32_t* timestamp,
1723 uint16_t* blockLen,
1724 int seqNo,
Peter Kasting248b0b02015-06-03 12:32:41 -07001725 uint32_t ssrc) {
1726 int i;
1727 unsigned char* rtpPointer;
1728 uint16_t offset;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001729
Peter Kasting248b0b02015-06-03 12:32:41 -07001730 /* first create "standard" RTP header */
1731 makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads - 1],
1732 ssrc);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001733
Peter Kasting248b0b02015-06-03 12:32:41 -07001734 rtpPointer = &rtp_data[12];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001735
Peter Kasting248b0b02015-06-03 12:32:41 -07001736 /* add one sub-header for each redundant payload (not the primary) */
1737 for (i = 0; i < numPayloads - 1; i++) {
1738 if (blockLen[i] > 0) {
1739 offset = static_cast<uint16_t>(timestamp[numPayloads - 1] - timestamp[i]);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001740
Peter Kasting248b0b02015-06-03 12:32:41 -07001741 // Byte |0| |1 2 | 3 |
1742 // Bit |0|1234567|01234567012345|6701234567|
1743 // |F|payload| timestamp | block |
1744 // | | type | offset | length |
1745 rtpPointer[0] = (payloadType[i] & 0x7F) | 0x80;
1746 rtpPointer[1] = (offset >> 6) & 0xFF;
1747 rtpPointer[2] = ((offset & 0x3F) << 2) | ((blockLen[i] >> 8) & 0x03);
1748 rtpPointer[3] = blockLen[i] & 0xFF;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001749
Peter Kasting248b0b02015-06-03 12:32:41 -07001750 rtpPointer += 4;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001751 }
Peter Kasting248b0b02015-06-03 12:32:41 -07001752 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001753
Peter Kasting248b0b02015-06-03 12:32:41 -07001754 // Bit |0|1234567|
1755 // |0|payload|
1756 // | | type |
1757 rtpPointer[0] = payloadType[numPayloads - 1] & 0x7F;
1758 ++rtpPointer;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001759
Peter Kasting248b0b02015-06-03 12:32:41 -07001760 return rtpPointer - rtp_data; // length of header in bytes
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001761}
1762
Peter Kastingdce40cf2015-08-24 14:52:23 -07001763size_t makeDTMFpayload(unsigned char* payload_data,
1764 int Event,
1765 int End,
1766 int Volume,
1767 int Duration) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001768 unsigned char E, R, V;
1769 R = 0;
1770 V = (unsigned char)Volume;
1771 if (End == 0) {
1772 E = 0x00;
1773 } else {
1774 E = 0x80;
1775 }
1776 payload_data[0] = (unsigned char)Event;
1777 payload_data[1] = (unsigned char)(E | R | V);
1778 // Duration equals 8 times time_ms, default is 8000 Hz.
1779 payload_data[2] = (unsigned char)((Duration >> 8) & 0xFF);
1780 payload_data[3] = (unsigned char)(Duration & 0xFF);
1781 return (4);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001782}
1783
Peter Kastingdce40cf2015-08-24 14:52:23 -07001784void stereoDeInterleave(int16_t* audioSamples, size_t numSamples) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001785 int16_t* tempVec;
1786 int16_t* readPtr, *writeL, *writeR;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001787
Peter Kastingdce40cf2015-08-24 14:52:23 -07001788 if (numSamples == 0)
Peter Kasting248b0b02015-06-03 12:32:41 -07001789 return;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001790
Peter Kasting248b0b02015-06-03 12:32:41 -07001791 tempVec = (int16_t*)malloc(sizeof(int16_t) * numSamples);
1792 if (tempVec == NULL) {
1793 printf("Error allocating memory\n");
1794 exit(0);
1795 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001796
Peter Kasting248b0b02015-06-03 12:32:41 -07001797 memcpy(tempVec, audioSamples, numSamples * sizeof(int16_t));
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001798
Peter Kasting248b0b02015-06-03 12:32:41 -07001799 writeL = audioSamples;
1800 writeR = &audioSamples[numSamples / 2];
1801 readPtr = tempVec;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001802
Peter Kastingdce40cf2015-08-24 14:52:23 -07001803 for (size_t k = 0; k < numSamples; k += 2) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001804 *writeL = *readPtr;
1805 readPtr++;
1806 *writeR = *readPtr;
1807 readPtr++;
1808 writeL++;
1809 writeR++;
1810 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001811
Peter Kasting248b0b02015-06-03 12:32:41 -07001812 free(tempVec);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001813}
1814
Peter Kastingdce40cf2015-08-24 14:52:23 -07001815void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride) {
Peter Kasting248b0b02015-06-03 12:32:41 -07001816 unsigned char* ptrL, *ptrR;
1817 unsigned char temp[10];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001818
Peter Kasting248b0b02015-06-03 12:32:41 -07001819 if (stride > 10) {
1820 exit(0);
1821 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001822
Peter Kasting248b0b02015-06-03 12:32:41 -07001823 if (dataLen % 1 != 0) {
1824 // must be even number of samples
1825 printf("Error: cannot interleave odd sample number\n");
1826 exit(0);
1827 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001828
Peter Kasting248b0b02015-06-03 12:32:41 -07001829 ptrL = data + stride;
1830 ptrR = &data[dataLen / 2];
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001831
Peter Kasting248b0b02015-06-03 12:32:41 -07001832 while (ptrL < ptrR) {
1833 // copy from right pointer to temp
1834 memcpy(temp, ptrR, stride);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001835
Peter Kasting248b0b02015-06-03 12:32:41 -07001836 // shift data between pointers
1837 memmove(ptrL + stride, ptrL, ptrR - ptrL);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001838
Peter Kasting248b0b02015-06-03 12:32:41 -07001839 // copy from temp to left pointer
1840 memcpy(ptrL, temp, stride);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001841
Peter Kasting248b0b02015-06-03 12:32:41 -07001842 // advance pointers
1843 ptrL += stride * 2;
1844 ptrR += stride;
1845 }
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001846}