Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.
It has been through extensive internal review during the course of
the project.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1073005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc
new file mode 100644
index 0000000..c79d5db
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/test/RTPencode.cc
@@ -0,0 +1,1826 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+//TODO(hlundin): Reformat file to meet style guide.
+
+/* header includes */
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#ifdef WIN32
+#include <winsock2.h>
+#endif
+#ifdef WEBRTC_LINUX
+#include <netinet/in.h>
+#endif
+
+#include <cassert>
+
+#include "webrtc/typedefs.h"
+// needed for NetEqDecoder
+#include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
+
+/************************/
+/* Define payload types */
+/************************/
+
+#include "PayloadTypes.h"
+
+
+
+/*********************/
+/* Misc. definitions */
+/*********************/
+
+#define STOPSENDTIME 3000
+#define RESTARTSENDTIME 0 //162500
+#define FIRSTLINELEN 40
+#define CHECK_NOT_NULL(a) if((a)==0){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
+
+//#define MULTIPLE_SAME_TIMESTAMP
+#define REPEAT_PACKET_DISTANCE 17
+#define REPEAT_PACKET_COUNT 1 // number of extra packets to send
+
+//#define INSERT_OLD_PACKETS
+#define OLD_PACKET 5 // how many seconds too old should the packet be?
+
+//#define TIMESTAMP_WRAPAROUND
+
+//#define RANDOM_DATA
+//#define RANDOM_PAYLOAD_DATA
+#define RANDOM_SEED 10
+
+//#define INSERT_DTMF_PACKETS
+//#define NO_DTMF_OVERDUB
+#define DTMF_PACKET_INTERVAL 2000
+#define DTMF_DURATION 500
+
+#define STEREO_MODE_FRAME 0
+#define STEREO_MODE_SAMPLE_1 1 //1 octet per sample
+#define STEREO_MODE_SAMPLE_2 2 //2 octets per sample
+
+/*************************/
+/* Function declarations */
+/*************************/
+
+void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed);
+int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels);
+void defineCodecs(webrtc::NetEqDecoder *usedCodec, int *noOfCodecs );
+int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels);
+int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc);
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
+ int seqNo, WebRtc_UWord32 ssrc);
+int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration);
+void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples);
+void stereoInterleave(unsigned char* data, int dataLen, int stride);
+
+/*********************/
+/* Codec definitions */
+/*********************/
+
+#include "webrtc_vad.h"
+
+#if ((defined CODEC_PCM16B)||(defined NETEQ_ARBITRARY_CODEC))
+ #include "pcm16b.h"
+#endif
+#ifdef CODEC_G711
+ #include "g711_interface.h"
+#endif
+#ifdef CODEC_G729
+ #include "G729Interface.h"
+#endif
+#ifdef CODEC_G729_1
+ #include "G729_1Interface.h"
+#endif
+#ifdef CODEC_AMR
+ #include "AMRInterface.h"
+ #include "AMRCreation.h"
+#endif
+#ifdef CODEC_AMRWB
+ #include "AMRWBInterface.h"
+ #include "AMRWBCreation.h"
+#endif
+#ifdef CODEC_ILBC
+ #include "ilbc.h"
+#endif
+#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
+ #include "isac.h"
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ #include "isacfix.h"
+ #ifdef CODEC_ISAC
+ #error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp
+ #endif
+#endif
+#ifdef CODEC_G722
+ #include "g722_interface.h"
+#endif
+#ifdef CODEC_G722_1_24
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1_32
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1_16
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1C_24
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1C_32
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G722_1C_48
+ #include "G722_1Interface.h"
+#endif
+#ifdef CODEC_G726
+ #include "G726Creation.h"
+ #include "G726Interface.h"
+#endif
+#ifdef CODEC_GSMFR
+ #include "GSMFRInterface.h"
+ #include "GSMFRCreation.h"
+#endif
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ #include "webrtc_cng.h"
+#endif
+#if ((defined CODEC_SPEEX_8)||(defined CODEC_SPEEX_16))
+ #include "SpeexInterface.h"
+#endif
+#ifdef CODEC_CELT_32
+#include "celt_interface.h"
+#endif
+
+
+/***********************************/
+/* Global codec instance variables */
+/***********************************/
+
+WebRtcVadInst *VAD_inst[2];
+
+#ifdef CODEC_G722
+ G722EncInst *g722EncState[2];
+#endif
+
+#ifdef CODEC_G722_1_24
+ G722_1_24_encinst_t *G722_1_24enc_inst[2];
+#endif
+#ifdef CODEC_G722_1_32
+ G722_1_32_encinst_t *G722_1_32enc_inst[2];
+#endif
+#ifdef CODEC_G722_1_16
+ G722_1_16_encinst_t *G722_1_16enc_inst[2];
+#endif
+#ifdef CODEC_G722_1C_24
+ G722_1C_24_encinst_t *G722_1C_24enc_inst[2];
+#endif
+#ifdef CODEC_G722_1C_32
+ G722_1C_32_encinst_t *G722_1C_32enc_inst[2];
+#endif
+#ifdef CODEC_G722_1C_48
+ G722_1C_48_encinst_t *G722_1C_48enc_inst[2];
+#endif
+#ifdef CODEC_G726
+ G726_encinst_t *G726enc_inst[2];
+#endif
+#ifdef CODEC_G729
+ G729_encinst_t *G729enc_inst[2];
+#endif
+#ifdef CODEC_G729_1
+ G729_1_inst_t *G729_1_inst[2];
+#endif
+#ifdef CODEC_AMR
+ AMR_encinst_t *AMRenc_inst[2];
+ WebRtc_Word16 AMR_bitrate;
+#endif
+#ifdef CODEC_AMRWB
+ AMRWB_encinst_t *AMRWBenc_inst[2];
+ WebRtc_Word16 AMRWB_bitrate;
+#endif
+#ifdef CODEC_ILBC
+ iLBC_encinst_t *iLBCenc_inst[2];
+#endif
+#ifdef CODEC_ISAC
+ ISACStruct *ISAC_inst[2];
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ ISACFIX_MainStruct *ISAC_inst[2];
+#endif
+#ifdef CODEC_ISAC_SWB
+ ISACStruct *ISACSWB_inst[2];
+#endif
+#ifdef CODEC_GSMFR
+ GSMFR_encinst_t *GSMFRenc_inst[2];
+#endif
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ CNG_enc_inst *CNGenc_inst[2];
+#endif
+#ifdef CODEC_SPEEX_8
+ SPEEX_encinst_t *SPEEX8enc_inst[2];
+#endif
+#ifdef CODEC_SPEEX_16
+ SPEEX_encinst_t *SPEEX16enc_inst[2];
+#endif
+#ifdef CODEC_CELT_32
+ CELT_encinst_t *CELT32enc_inst[2];
+#endif
+#ifdef CODEC_G711
+ void *G711state[2]={NULL, NULL};
+#endif
+
+
+int main(int argc, char* argv[])
+{
+ int packet_size, fs;
+ webrtc::NetEqDecoder usedCodec;
+ int payloadType;
+ int bitrate = 0;
+ int useVAD, vad;
+ int useRed=0;
+ int len, enc_len;
+ WebRtc_Word16 org_data[4000];
+ unsigned char rtp_data[8000];
+ WebRtc_Word16 seqNo=0xFFF;
+ WebRtc_UWord32 ssrc=1235412312;
+ WebRtc_UWord32 timestamp=0xAC1245;
+ WebRtc_UWord16 length, plen;
+ WebRtc_UWord32 offset;
+ double sendtime = 0;
+ int red_PT[2] = {0};
+ WebRtc_UWord32 red_TS[2] = {0};
+ WebRtc_UWord16 red_len[2] = {0};
+ int RTPheaderLen=12;
+ unsigned char red_data[8000];
+#ifdef INSERT_OLD_PACKETS
+ WebRtc_UWord16 old_length, old_plen;
+ int old_enc_len;
+ int first_old_packet=1;
+ unsigned char old_rtp_data[8000];
+ int packet_age=0;
+#endif
+#ifdef INSERT_DTMF_PACKETS
+ int NTone = 1;
+ int DTMFfirst = 1;
+ WebRtc_UWord32 DTMFtimestamp;
+ bool dtmfSent = false;
+#endif
+ bool usingStereo = false;
+ int stereoMode = 0;
+ int numChannels = 1;
+
+ /* check number of parameters */
+ if ((argc != 6) && (argc != 7)) {
+ /* print help text and exit */
+ printf("Application to encode speech into an RTP stream.\n");
+ printf("The program reads a PCM file and encodes is using the specified codec.\n");
+ printf("The coded speech is packetized in RTP packest and written to the output file.\n");
+ printf("The format of the RTP stream file is simlilar to that of rtpplay,\n");
+ printf("but with the receive time euqal to 0 for all packets.\n");
+ printf("Usage:\n\n");
+ printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
+ printf("where:\n");
+
+ printf("PCMfile : PCM speech input file\n\n");
+
+ printf("RTPfile : RTP stream output file\n\n");
+
+ printf("frameLen : 80...960... Number of samples per packet (limit depends on codec)\n\n");
+
+ printf("codecName\n");
+#ifdef CODEC_PCM16B
+ printf(" : pcm16b 16 bit PCM (8kHz)\n");
+#endif
+#ifdef CODEC_PCM16B_WB
+ printf(" : pcm16b_wb 16 bit PCM (16kHz)\n");
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ printf(" : pcm16b_swb32 16 bit PCM (32kHz)\n");
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ printf(" : pcm16b_swb48 16 bit PCM (48kHz)\n");
+#endif
+#ifdef CODEC_G711
+ printf(" : pcma g711 A-law (8kHz)\n");
+#endif
+#ifdef CODEC_G711
+ printf(" : pcmu g711 u-law (8kHz)\n");
+#endif
+#ifdef CODEC_G729
+ printf(" : g729 G729 (8kHz and 8kbps) CELP (One-Three frame(s)/packet)\n");
+#endif
+#ifdef CODEC_G729_1
+ printf(" : g729.1 G729.1 (16kHz) variable rate (8--32 kbps)\n");
+#endif
+#ifdef CODEC_G722_1_16
+ printf(" : g722.1_16 G722.1 coder (16kHz) (g722.1 with 16kbps)\n");
+#endif
+#ifdef CODEC_G722_1_24
+ printf(" : g722.1_24 G722.1 coder (16kHz) (the 24kbps version)\n");
+#endif
+#ifdef CODEC_G722_1_32
+ printf(" : g722.1_32 G722.1 coder (16kHz) (the 32kbps version)\n");
+#endif
+#ifdef CODEC_G722_1C_24
+ printf(" : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps version)\n");
+#endif
+#ifdef CODEC_G722_1C_32
+ printf(" : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps version)\n");
+#endif
+#ifdef CODEC_G722_1C_48
+ printf(" : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps)\n");
+#endif
+
+#ifdef CODEC_G726
+ printf(" : g726_16 G726 coder (8kHz) 16kbps\n");
+ printf(" : g726_24 G726 coder (8kHz) 24kbps\n");
+ printf(" : g726_32 G726 coder (8kHz) 32kbps\n");
+ printf(" : g726_40 G726 coder (8kHz) 40kbps\n");
+#endif
+#ifdef CODEC_AMR
+ printf(" : AMRXk Adaptive Multi Rate CELP codec (8kHz)\n");
+ printf(" X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 or 12.2\n");
+#endif
+#ifdef CODEC_AMRWB
+ printf(" : AMRwbXk Adaptive Multi Rate Wideband CELP codec (16kHz)\n");
+ printf(" X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n");
+#endif
+#ifdef CODEC_ILBC
+ printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n");
+#endif
+#ifdef CODEC_ISAC
+ printf(" : isac iSAC (16kHz and 32.0 kbps). To set rate specify a rate parameter as last parameter\n");
+#endif
+#ifdef CODEC_ISAC_SWB
+ printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
+#endif
+#ifdef CODEC_GSMFR
+ printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
+#endif
+#ifdef CODEC_G722
+ printf(" : g722 g722 coder (16kHz) (the 64kbps version)\n");
+#endif
+#ifdef CODEC_SPEEX_8
+ printf(" : speex8 speex coder (8 kHz)\n");
+#endif
+#ifdef CODEC_SPEEX_16
+ printf(" : speex16 speex coder (16 kHz)\n");
+#endif
+#ifdef CODEC_CELT_32
+ printf(" : celt32 celt coder (32 kHz)\n");
+#endif
+#ifdef CODEC_RED
+#ifdef CODEC_G711
+ printf(" : red_pcm Redundancy RTP packet with 2*G711A frames\n");
+#endif
+#ifdef CODEC_ISAC
+ printf(" : red_isac Redundancy RTP packet with 2*iSAC frames\n");
+#endif
+#endif
+ printf("\n");
+
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ printf("useVAD : 0 Voice Activity Detection is switched off\n");
+ printf(" : 1 Voice Activity Detection is switched on\n\n");
+#else
+ printf("useVAD : 0 Voice Activity Detection switched off (on not supported)\n\n");
+#endif
+ printf("bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n");
+
+ return(0);
+ }
+
+ FILE* in_file=fopen(argv[1],"rb");
+ CHECK_NOT_NULL(in_file);
+ printf("Input file: %s\n",argv[1]);
+ FILE* out_file=fopen(argv[2],"wb");
+ CHECK_NOT_NULL(out_file);
+ printf("Output file: %s\n\n",argv[2]);
+ packet_size=atoi(argv[3]);
+ CHECK_NOT_NULL(packet_size);
+ printf("Packet size: %i\n",packet_size);
+
+ // check for stereo
+ if(argv[4][strlen(argv[4])-1] == '*') {
+ // use stereo
+ usingStereo = true;
+ numChannels = 2;
+ argv[4][strlen(argv[4])-1] = '\0';
+ }
+
+ NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed);
+
+ if(useRed) {
+ RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant payload, except last one which is 1 byte */
+ }
+
+ useVAD=atoi(argv[5]);
+#if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ if (useVAD!=0) {
+ printf("Error: this simulation does not support VAD/DTX/CNG\n");
+ }
+#endif
+
+ // check stereo type
+ if(usingStereo)
+ {
+ switch(usedCodec)
+ {
+ // sample based codecs
+ case webrtc::kDecoderPCMu:
+ case webrtc::kDecoderPCMa:
+ case webrtc::kDecoderG722:
+ {
+ // 1 octet per sample
+ stereoMode = STEREO_MODE_SAMPLE_1;
+ break;
+ }
+ case webrtc::kDecoderPCM16B:
+ case webrtc::kDecoderPCM16Bwb:
+ case webrtc::kDecoderPCM16Bswb32kHz:
+ case webrtc::kDecoderPCM16Bswb48kHz:
+ {
+ // 2 octets per sample
+ stereoMode = STEREO_MODE_SAMPLE_2;
+ break;
+ }
+
+ // fixed-rate frame codecs (with internal VAD)
+ default:
+ {
+ printf("Cannot use codec %s as stereo codec\n", argv[4]);
+ exit(0);
+ }
+ }
+ }
+
+ if ((usedCodec == webrtc::kDecoderISAC) || (usedCodec == webrtc::kDecoderISACswb))
+ {
+ if (argc != 7)
+ {
+ if (usedCodec == webrtc::kDecoderISAC)
+ {
+ bitrate = 32000;
+ printf(
+ "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n");
+ }
+ else // (usedCodec==webrtc::kDecoderISACswb)
+ {
+ bitrate = 56000;
+ printf(
+ "Running iSAC at default bitrate of 56000 bps (to specify explicitly add the bps as last parameter)\n");
+ }
+ }
+ else
+ {
+ bitrate = atoi(argv[6]);
+ if (usedCodec == webrtc::kDecoderISAC)
+ {
+ if ((bitrate < 10000) || (bitrate > 32000))
+ {
+ printf(
+ "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is invalid)\n",
+ bitrate);
+ exit(0);
+ }
+ printf("Running iSAC at bitrate of %i bps\n", bitrate);
+ }
+ else // (usedCodec==webrtc::kDecoderISACswb)
+ {
+ if ((bitrate < 32000) || (bitrate > 56000))
+ {
+ printf(
+ "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
+ bitrate);
+ exit(0);
+ }
+ }
+ }
+ }
+ else
+ {
+ if (argc == 7)
+ {
+ printf(
+ "Error: Bitrate parameter can only be specified for iSAC, G.723, and G.729.1\n");
+ exit(0);
+ }
+ }
+
+ if(useRed) {
+ printf("Redundancy engaged. ");
+ }
+ printf("Used codec: %i\n",usedCodec);
+ printf("Payload type: %i\n",payloadType);
+
+ NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels);
+
+ /* write file header */
+ //fprintf(out_file, "#!RTPencode%s\n", "1.0");
+ fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
+ WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
+ if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
+ return -1;
+ }
+
+#ifdef TIMESTAMP_WRAPAROUND
+ timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */
+#endif
+#if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
+ srand(RANDOM_SEED);
+#endif
+
+ /* if redundancy is used, the first redundant payload is zero length */
+ red_len[0] = 0;
+
+ /* read first frame */
+ len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
+
+ /* de-interleave if stereo */
+ if ( usingStereo )
+ {
+ stereoDeInterleave(org_data, len * numChannels);
+ }
+
+ while (len==packet_size) {
+
+#ifdef INSERT_DTMF_PACKETS
+ dtmfSent = false;
+
+ if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) {
+ if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) {
+ // tone has not ended
+ if (DTMFfirst==1) {
+ DTMFtimestamp = timestamp; // save this timestamp
+ DTMFfirst=0;
+ }
+ makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
+ enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len);
+ }
+ else {
+ // tone has ended
+ makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
+ enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4, DTMF_DURATION*(fs/1000));
+ NTone++;
+ DTMFfirst=1;
+ }
+
+ /* write RTP packet to file */
+ length = htons(12 + enc_len + 8);
+ plen = htons(12 + enc_len);
+ offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
+ offset = htonl(offset);
+ if (fwrite(&length, 2, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&plen, 2, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&offset, 4, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
+ return -1;
+ }
+
+ dtmfSent = true;
+ }
+#endif
+
+#ifdef NO_DTMF_OVERDUB
+ /* If DTMF is sent, we should not send any speech packets during the same time */
+ if (dtmfSent) {
+ enc_len = 0;
+ }
+ else {
+#endif
+ /* encode frame */
+ enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels);
+ if (enc_len==-1) {
+ printf("Error encoding frame\n");
+ exit(0);
+ }
+
+ if ( usingStereo &&
+ stereoMode != STEREO_MODE_FRAME &&
+ vad == 1 )
+ {
+ // interleave the encoded payload for sample-based codecs (not for CNG)
+ stereoInterleave(&rtp_data[12], enc_len, stereoMode);
+ }
+#ifdef NO_DTMF_OVERDUB
+ }
+#endif
+
+ if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
+ if(useRed) {
+ if(red_len[0] > 0) {
+ memmove(&rtp_data[RTPheaderLen+red_len[0]], &rtp_data[12], enc_len);
+ memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
+
+ red_len[1] = enc_len;
+ red_TS[1] = timestamp;
+ if(vad)
+ red_PT[1] = payloadType;
+ else
+ red_PT[1] = NETEQ_CODEC_CN_PT;
+
+ makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
+
+
+ enc_len += red_len[0] + RTPheaderLen - 12;
+ }
+ else { // do not use redundancy payload for this packet, i.e., only last payload
+ memmove(&rtp_data[RTPheaderLen-4], &rtp_data[12], enc_len);
+ //memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
+
+ red_len[1] = enc_len;
+ red_TS[1] = timestamp;
+ if(vad)
+ red_PT[1] = payloadType;
+ else
+ red_PT[1] = NETEQ_CODEC_CN_PT;
+
+ makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
+
+
+ enc_len += red_len[0] + RTPheaderLen - 4 - 12; // 4 is length of redundancy header (not used)
+ }
+ }
+ else {
+
+ /* make RTP header */
+ if (vad) // regular speech data
+ makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc);
+ else // CNG data
+ makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc);
+
+ }
+#ifdef MULTIPLE_SAME_TIMESTAMP
+ int mult_pack=0;
+ do {
+#endif //MULTIPLE_SAME_TIMESTAMP
+ /* write RTP packet to file */
+ length = htons(12 + enc_len + 8);
+ plen = htons(12 + enc_len);
+ offset = (WebRtc_UWord32) sendtime;
+ //(timestamp/(fs/1000));
+ offset = htonl(offset);
+ if (fwrite(&length, 2, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&plen, 2, 1, out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&offset, 4, 1, out_file) != 1) {
+ return -1;
+ }
+#ifdef RANDOM_DATA
+ for (int k=0; k<12+enc_len; k++) {
+ rtp_data[k] = rand() + rand();
+ }
+#endif
+#ifdef RANDOM_PAYLOAD_DATA
+ for (int k=12; k<12+enc_len; k++) {
+ rtp_data[k] = rand() + rand();
+ }
+#endif
+ if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
+ return -1;
+ }
+#ifdef MULTIPLE_SAME_TIMESTAMP
+ } while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) );
+#endif //MULTIPLE_SAME_TIMESTAMP
+
+#ifdef INSERT_OLD_PACKETS
+ if (packet_age >= OLD_PACKET*fs) {
+ if (!first_old_packet) {
+ // send the old packet
+ if (fwrite(&old_length, 2, 1,
+ out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&old_plen, 2, 1,
+ out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(&offset, 4, 1,
+ out_file) != 1) {
+ return -1;
+ }
+ if (fwrite(old_rtp_data, 12 + old_enc_len,
+ 1, out_file) != 1) {
+ return -1;
+ }
+ }
+ // store current packet as old
+ old_length=length;
+ old_plen=plen;
+ memcpy(old_rtp_data,rtp_data,12+enc_len);
+ old_enc_len=enc_len;
+ first_old_packet=0;
+ packet_age=0;
+
+ }
+ packet_age += packet_size;
+#endif
+
+ if(useRed) {
+ /* move data to redundancy store */
+#ifdef CODEC_ISAC
+ if(usedCodec==webrtc::kDecoderISAC)
+ {
+ assert(!usingStereo); // Cannot handle stereo yet
+ red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (WebRtc_Word16*)red_data);
+ }
+ else
+ {
+#endif
+ memcpy(red_data, &rtp_data[RTPheaderLen+red_len[0]], enc_len);
+ red_len[0]=red_len[1];
+#ifdef CODEC_ISAC
+ }
+#endif
+ red_TS[0]=red_TS[1];
+ red_PT[0]=red_PT[1];
+ }
+
+ }
+
+ /* read next frame */
+ len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
+ /* de-interleave if stereo */
+ if ( usingStereo )
+ {
+ stereoDeInterleave(org_data, len * numChannels);
+ }
+
+ if (payloadType==NETEQ_CODEC_G722_PT)
+ timestamp+=len>>1;
+ else
+ timestamp+=len;
+
+ sendtime += (double) len/(fs/1000);
+ }
+
+ NetEQTest_free_coders(usedCodec, numChannels);
+ fclose(in_file);
+ fclose(out_file);
+ printf("Done!\n");
+
+ return(0);
+}
+
+
+
+
+/****************/
+/* Subfunctions */
+/****************/
+
+void NetEQTest_GetCodec_and_PT(char * name, webrtc::NetEqDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed) {
+
+ *bitrate = 0; /* Default bitrate setting */
+ *useRed = 0; /* Default no redundancy */
+
+ if(!strcmp(name,"pcmu")){
+ *codec=webrtc::kDecoderPCMu;
+ *PT=NETEQ_CODEC_PCMU_PT;
+ *fs=8000;
+ }
+ else if(!strcmp(name,"pcma")){
+ *codec=webrtc::kDecoderPCMa;
+ *PT=NETEQ_CODEC_PCMA_PT;
+ *fs=8000;
+ }
+ else if(!strcmp(name,"pcm16b")){
+ *codec=webrtc::kDecoderPCM16B;
+ *PT=NETEQ_CODEC_PCM16B_PT;
+ *fs=8000;
+ }
+ else if(!strcmp(name,"pcm16b_wb")){
+ *codec=webrtc::kDecoderPCM16Bwb;
+ *PT=NETEQ_CODEC_PCM16B_WB_PT;
+ *fs=16000;
+ }
+ else if(!strcmp(name,"pcm16b_swb32")){
+ *codec=webrtc::kDecoderPCM16Bswb32kHz;
+ *PT=NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
+ *fs=32000;
+ }
+ else if(!strcmp(name,"pcm16b_swb48")){
+ *codec=webrtc::kDecoderPCM16Bswb48kHz;
+ *PT=NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
+ *fs=48000;
+ }
+ else if(!strcmp(name,"g722")){
+ *codec=webrtc::kDecoderG722;
+ *PT=NETEQ_CODEC_G722_PT;
+ *fs=16000;
+ }
+ else if((!strcmp(name,"ilbc"))&&((frameLen%240==0)||(frameLen%160==0))){
+ *fs=8000;
+ *codec=webrtc::kDecoderILBC;
+ *PT=NETEQ_CODEC_ILBC_PT;
+ }
+ else if(!strcmp(name,"isac")){
+ *fs=16000;
+ *codec=webrtc::kDecoderISAC;
+ *PT=NETEQ_CODEC_ISAC_PT;
+ }
+ else if(!strcmp(name,"isacswb")){
+ *fs=32000;
+ *codec=webrtc::kDecoderISACswb;
+ *PT=NETEQ_CODEC_ISACSWB_PT;
+ }
+ else if(!strcmp(name,"celt32")){
+ *fs=32000;
+ *codec=webrtc::kDecoderCELT_32;
+ *PT=NETEQ_CODEC_CELT32_PT;
+ }
+ else if(!strcmp(name,"red_pcm")){
+ *codec=webrtc::kDecoderPCMa;
+ *PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
+ *fs=8000;
+ *useRed = 1;
+ } else if(!strcmp(name,"red_isac")){
+ *codec=webrtc::kDecoderISAC;
+ *PT=NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
+ *fs=16000;
+ *useRed = 1;
+ } else {
+ printf("Error: Not a supported codec (%s)\n", name);
+ exit(0);
+ }
+
+}
+
+
+
+
+int NetEQTest_init_coders(webrtc::NetEqDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels){
+
+ int ok=0;
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ ok=WebRtcVad_Create(&VAD_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for VAD instance\n");
+ exit(0);
+ }
+ ok=WebRtcVad_Init(VAD_inst[k]);
+ if (ok==-1) {
+ printf("Error: Initialization of VAD struct failed\n");
+ exit(0);
+ }
+
+
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ ok=WebRtcCng_CreateEnc(&CNGenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for CNG encoding instance\n");
+ exit(0);
+ }
+ if(sampfreq <= 16000) {
+ ok=WebRtcCng_InitEnc(CNGenc_inst[k],sampfreq, 200, 5);
+ if (ok==-1) {
+ printf("Error: Initialization of CNG struct failed. Error code %d\n",
+ WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
+ exit(0);
+ }
+ }
+#endif
+
+ switch (coder) {
+#ifdef CODEC_PCM16B
+ case webrtc::kDecoderPCM16B :
+#endif
+#ifdef CODEC_PCM16B_WB
+ case webrtc::kDecoderPCM16Bwb :
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ case webrtc::kDecoderPCM16Bswb32kHz :
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ case webrtc::kDecoderPCM16Bswb48kHz :
+#endif
+#ifdef CODEC_G711
+ case webrtc::kDecoderPCMu :
+ case webrtc::kDecoderPCMa :
+#endif
+ // do nothing
+ break;
+#ifdef CODEC_G729
+ case webrtc::kDecoderG729:
+ if (sampfreq==8000) {
+ if ((enc_frameSize==80)||(enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==400)||(enc_frameSize==480)) {
+ ok=WebRtcG729_CreateEnc(&G729enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G729 encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG729_EncoderInit(G729enc_inst[k], vad);
+ if ((vad==1)&&(enc_frameSize!=80)) {
+ printf("\nError - This simulation only supports VAD for G729 at 10ms packets (not %dms)\n", (enc_frameSize>>3));
+ }
+ } else {
+ printf("\nError - g729 is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G729_1
+ case webrtc::kDecoderG729_1:
+ if (sampfreq==16000) {
+ if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)
+ ) {
+ ok=WebRtcG7291_Create(&G729_1_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.729.1 codec instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
+ exit(0);
+ }
+ if (!(((bitrate >= 12000) && (bitrate <= 32000) && (bitrate%2000 == 0)) || (bitrate == 8000))) {
+ /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
+ printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in steps of 2000 bps\n");
+ exit(0);
+ }
+ WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/, 0 /*flagG729mode*/);
+ } else {
+ printf("\nError - G.729.1 input is always 16 kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_SPEEX_8
+ case webrtc::kDecoderSPEEX_8 :
+ if (sampfreq==8000) {
+ if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ ok=WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for Speex encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
+ exit(0);
+ }
+ if ((vad==1)&&(enc_frameSize!=160)) {
+ printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>3));
+ vad=0;
+ }
+ ok=WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
+ if (ok!=0) exit(0);
+ } else {
+ printf("\nError - Speex8 called with sample frequency other than 8 kHz.\n\n");
+ }
+ break;
+#endif
+#ifdef CODEC_SPEEX_16
+ case webrtc::kDecoderSPEEX_16 :
+ if (sampfreq==16000) {
+ if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)) {
+ ok=WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for Speex encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
+ exit(0);
+ }
+ if ((vad==1)&&(enc_frameSize!=320)) {
+ printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>4));
+ vad=0;
+ }
+ ok=WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
+ if (ok!=0) exit(0);
+ } else {
+ printf("\nError - Speex16 called with sample frequency other than 16 kHz.\n\n");
+ }
+ break;
+#endif
+#ifdef CODEC_CELT_32
+ case webrtc::kDecoderCELT_32 :
+ if (sampfreq==32000) {
+ if (enc_frameSize==320) {
+ ok=WebRtcCelt_CreateEnc(&CELT32enc_inst[k], 1 /*mono*/);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for Celt encoding instance\n");
+ exit(0);
+ }
+ } else {
+ printf("\nError: Celt only supports 10 ms!!\n\n");
+ exit(0);
+ }
+ ok=WebRtcCelt_EncoderInit(CELT32enc_inst[k], 1 /*mono*/, 48000 /*bitrate*/);
+ if (ok!=0) exit(0);
+ } else {
+ printf("\nError - Celt32 called with sample frequency other than 32 kHz.\n\n");
+ }
+ break;
+#endif
+
+#ifdef CODEC_G722_1_16
+ case webrtc::kDecoderG722_1_16 :
+ if (sampfreq==16000) {
+ ok=WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1 instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==320) {
+ } else {
+ printf("\nError: G722.1 only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1_24
+ case webrtc::kDecoderG722_1_24 :
+ if (sampfreq==16000) {
+ ok=WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1 instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==320) {
+ } else {
+ printf("\nError: G722.1 only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1_32
+ case webrtc::kDecoderG722_1_32 :
+ if (sampfreq==16000) {
+ ok=WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1 instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==320) {
+ } else {
+ printf("\nError: G722.1 only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1C_24
+ case webrtc::kDecoderG722_1C_24 :
+ if (sampfreq==32000) {
+ ok=WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1C instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==640) {
+ } else {
+ printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221C_EncoderInit24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 C is only developed for 32kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1C_32
+ case webrtc::kDecoderG722_1C_32 :
+ if (sampfreq==32000) {
+ ok=WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1C instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==640) {
+ } else {
+ printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221C_EncoderInit32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 C is only developed for 32kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722_1C_48
+ case webrtc::kDecoderG722_1C_48 :
+ if (sampfreq==32000) {
+ ok=WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for G.722.1C instance\n");
+ exit(0);
+ }
+ if (enc_frameSize==640) {
+ } else {
+ printf("\nError: G722.1 C only supports 20 ms!!\n\n");
+ exit(0);
+ }
+ WebRtcG7221C_EncoderInit48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
+ } else {
+ printf("\nError - G722.1 C is only developed for 32kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_G722
+ case webrtc::kDecoderG722 :
+ if (sampfreq==16000) {
+ if (enc_frameSize%2==0) {
+ } else {
+ printf("\nError - g722 frames must have an even number of enc_frameSize\n");
+ exit(0);
+ }
+ WebRtcG722_CreateEncoder(&g722EncState[k]);
+ WebRtcG722_EncoderInit(g722EncState[k]);
+ } else {
+ printf("\nError - g722 is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_AMR
+ case webrtc::kDecoderAMR :
+ if (sampfreq==8000) {
+ ok=WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for AMR encoding instance\n");
+ exit(0);
+ }if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ } else {
+ printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
+ exit(0);
+ }
+ WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
+ WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
+ AMR_bitrate = bitrate;
+ } else {
+ printf("\nError - AMR is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_AMRWB
+ case webrtc::kDecoderAMRWB :
+ if (sampfreq==16000) {
+ ok=WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for AMRWB encoding instance\n");
+ exit(0);
+ }
+ if (((enc_frameSize/320)<0)||((enc_frameSize/320)>3)||((enc_frameSize%320)!=0)) {
+ printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
+ exit(0);
+ }
+ WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
+ if (bitrate==7000) {
+ AMRWB_bitrate = AMRWB_MODE_7k;
+ } else if (bitrate==9000) {
+ AMRWB_bitrate = AMRWB_MODE_9k;
+ } else if (bitrate==12000) {
+ AMRWB_bitrate = AMRWB_MODE_12k;
+ } else if (bitrate==14000) {
+ AMRWB_bitrate = AMRWB_MODE_14k;
+ } else if (bitrate==16000) {
+ AMRWB_bitrate = AMRWB_MODE_16k;
+ } else if (bitrate==18000) {
+ AMRWB_bitrate = AMRWB_MODE_18k;
+ } else if (bitrate==20000) {
+ AMRWB_bitrate = AMRWB_MODE_20k;
+ } else if (bitrate==23000) {
+ AMRWB_bitrate = AMRWB_MODE_23k;
+ } else if (bitrate==24000) {
+ AMRWB_bitrate = AMRWB_MODE_24k;
+ }
+ WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
+
+ } else {
+ printf("\nError - AMRwb is only developed for 16kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_ILBC
+ case webrtc::kDecoderILBC :
+ if (sampfreq==8000) {
+ ok=WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iLBC encoding instance\n");
+ exit(0);
+ }
+ if ((enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ } else {
+ printf("\nError - iLBC only supports 160, 240, 320 and 480 enc_frameSize (20, 30, 40 and 60 ms)\n");
+ exit(0);
+ }
+ if ((enc_frameSize==160)||(enc_frameSize==320)) {
+ /* 20 ms version */
+ WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
+ } else {
+ /* 30 ms version */
+ WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
+ }
+ } else {
+ printf("\nError - iLBC is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_ISAC
+ case webrtc::kDecoderISAC:
+ if (sampfreq==16000) {
+ ok=WebRtcIsac_Create(&ISAC_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iSAC instance\n");
+ exit(0);
+ }if ((enc_frameSize==480)||(enc_frameSize==960)) {
+ } else {
+ printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
+ exit(0);
+ }
+ WebRtcIsac_EncoderInit(ISAC_inst[k],1);
+ if ((bitrate<10000)||(bitrate>32000)) {
+ printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
+ exit(0);
+ }
+ WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
+ } else {
+ printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ case webrtc::kDecoderISAC:
+ if (sampfreq==16000) {
+ ok=WebRtcIsacfix_Create(&ISAC_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iSAC instance\n");
+ exit(0);
+ }if ((enc_frameSize==480)||(enc_frameSize==960)) {
+ } else {
+ printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
+ exit(0);
+ }
+ WebRtcIsacfix_EncoderInit(ISAC_inst[k],1);
+ if ((bitrate<10000)||(bitrate>32000)) {
+ printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
+ exit(0);
+ }
+ WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
+ } else {
+ printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_ISAC_SWB
+ case webrtc::kDecoderISACswb:
+ if (sampfreq==32000) {
+ ok=WebRtcIsac_Create(&ISACSWB_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
+ exit(0);
+ }if (enc_frameSize==960) {
+ } else {
+ printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
+ exit(0);
+ }
+ ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], 32000);
+ if (ok!=0) {
+ printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
+ exit(0);
+ }
+ WebRtcIsac_EncoderInit(ISACSWB_inst[k],1);
+ if ((bitrate<32000)||(bitrate>56000)) {
+ printf("\nError - iSAC SWB bitrate has to be between 32000 and 56000 bps (not %i)\n", bitrate);
+ exit(0);
+ }
+ WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize>>5);
+ } else {
+ printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n");
+ exit(0);
+ }
+ break;
+#endif
+#ifdef CODEC_GSMFR
+ case webrtc::kDecoderGSMFR:
+ if (sampfreq==8000) {
+ ok=WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
+ if (ok!=0) {
+ printf("Error: Couldn't allocate memory for GSM FR encoding instance\n");
+ exit(0);
+ }
+ if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
+ } else {
+ printf("\nError - GSM FR must have a multiple of 160 enc_frameSize\n");
+ exit(0);
+ }
+ WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
+ } else {
+ printf("\nError - GSM FR is only developed for 8kHz \n");
+ exit(0);
+ }
+ break;
+#endif
+ default :
+ printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
+ exit(0);
+ break;
+ }
+
+ if (ok != 0) {
+ return(ok);
+ }
+ } // end for
+
+ return(0);
+}
+
+
+
+
+int NetEQTest_free_coders(webrtc::NetEqDecoder coder, int numChannels) {
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ WebRtcVad_Free(VAD_inst[k]);
+#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
+ defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
+ WebRtcCng_FreeEnc(CNGenc_inst[k]);
+#endif
+
+ switch (coder)
+ {
+#ifdef CODEC_PCM16B
+ case webrtc::kDecoderPCM16B :
+#endif
+#ifdef CODEC_PCM16B_WB
+ case webrtc::kDecoderPCM16Bwb :
+#endif
+#ifdef CODEC_PCM16B_32KHZ
+ case webrtc::kDecoderPCM16Bswb32kHz :
+#endif
+#ifdef CODEC_PCM16B_48KHZ
+ case webrtc::kDecoderPCM16Bswb48kHz :
+#endif
+#ifdef CODEC_G711
+ case webrtc::kDecoderPCMu :
+ case webrtc::kDecoderPCMa :
+#endif
+ // do nothing
+ break;
+#ifdef CODEC_G729
+ case webrtc::kDecoderG729:
+ WebRtcG729_FreeEnc(G729enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G729_1
+ case webrtc::kDecoderG729_1:
+ WebRtcG7291_Free(G729_1_inst[k]);
+ break;
+#endif
+#ifdef CODEC_SPEEX_8
+ case webrtc::kDecoderSPEEX_8 :
+ WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_SPEEX_16
+ case webrtc::kDecoderSPEEX_16 :
+ WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_CELT_32
+ case webrtc::kDecoderCELT_32 :
+ WebRtcCelt_FreeEnc(CELT32enc_inst[k]);
+ break;
+#endif
+
+#ifdef CODEC_G722_1_16
+ case webrtc::kDecoderG722_1_16 :
+ WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1_24
+ case webrtc::kDecoderG722_1_24 :
+ WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1_32
+ case webrtc::kDecoderG722_1_32 :
+ WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1C_24
+ case webrtc::kDecoderG722_1C_24 :
+ WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1C_32
+ case webrtc::kDecoderG722_1C_32 :
+ WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722_1C_48
+ case webrtc::kDecoderG722_1C_48 :
+ WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_G722
+ case webrtc::kDecoderG722 :
+ WebRtcG722_FreeEncoder(g722EncState[k]);
+ break;
+#endif
+#ifdef CODEC_AMR
+ case webrtc::kDecoderAMR :
+ WebRtcAmr_FreeEnc(AMRenc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_AMRWB
+ case webrtc::kDecoderAMRWB :
+ WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_ILBC
+ case webrtc::kDecoderILBC :
+ WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
+ break;
+#endif
+#ifdef CODEC_ISAC
+ case webrtc::kDecoderISAC:
+ WebRtcIsac_Free(ISAC_inst[k]);
+ break;
+#endif
+#ifdef NETEQ_ISACFIX_CODEC
+ case webrtc::kDecoderISAC:
+ WebRtcIsacfix_Free(ISAC_inst[k]);
+ break;
+#endif
+#ifdef CODEC_ISAC_SWB
+ case webrtc::kDecoderISACswb:
+ WebRtcIsac_Free(ISACSWB_inst[k]);
+ break;
+#endif
+#ifdef CODEC_GSMFR
+ case webrtc::kDecoderGSMFR:
+ WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
+ break;
+#endif
+ default :
+ printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
+ exit(0);
+ break;
+ }
+ }
+
+ return(0);
+}
+
+
+
+
+
+
+int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate ,
+ int * vad, int useVAD, int bitrate, int numChannels){
+
+ short cdlen = 0;
+ WebRtc_Word16 *tempdata;
+ static int first_cng=1;
+ WebRtc_Word16 tempLen;
+
+ *vad =1;
+
+ // check VAD first
+ if(useVAD)
+ {
+ *vad = 0;
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ tempLen = frameLen;
+ tempdata = &indata[k*frameLen];
+ int localVad=0;
+ /* Partition the signal and test each chunk for VAD.
+ All chunks must be VAD=0 to produce a total VAD=0. */
+ while (tempLen >= 10*sampleRate/1000) {
+ if ((tempLen % 30*sampleRate/1000) == 0) { // tempLen is multiple of 30ms
+ localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 30*sampleRate/1000);
+ tempdata += 30*sampleRate/1000;
+ tempLen -= 30*sampleRate/1000;
+ }
+ else if (tempLen >= 20*sampleRate/1000) { // tempLen >= 20ms
+ localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 20*sampleRate/1000);
+ tempdata += 20*sampleRate/1000;
+ tempLen -= 20*sampleRate/1000;
+ }
+ else { // use 10ms
+ localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 10*sampleRate/1000);
+ tempdata += 10*sampleRate/1000;
+ tempLen -= 10*sampleRate/1000;
+ }
+ }
+
+ // aggregate all VAD decisions over all channels
+ *vad |= localVad;
+ }
+
+ if(!*vad){
+ // all channels are silent
+ cdlen = 0;
+ for (int k = 0; k < numChannels; k++)
+ {
+ WebRtcCng_Encode(CNGenc_inst[k],&indata[k*frameLen], (frameLen <= 640 ? frameLen : 640) /* max 640 */,
+ encoded,&tempLen,first_cng);
+ encoded += tempLen;
+ cdlen += tempLen;
+ }
+ *vad=0;
+ first_cng=0;
+ return(cdlen);
+ }
+ }
+
+
+ // loop over all channels
+ int totalLen = 0;
+
+ for (int k = 0; k < numChannels; k++)
+ {
+ /* Encode with the selected coder type */
+ if (coder==webrtc::kDecoderPCMu) { /*g711 u-law */
+#ifdef CODEC_G711
+ cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+#endif
+ }
+ else if (coder==webrtc::kDecoderPCMa) { /*g711 A-law */
+#ifdef CODEC_G711
+ cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
+ }
+#endif
+#ifdef CODEC_PCM16B
+ else if ((coder==webrtc::kDecoderPCM16B)||(coder==webrtc::kDecoderPCM16Bwb)||
+ (coder==webrtc::kDecoderPCM16Bswb32kHz)||(coder==webrtc::kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
+ cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (WebRtc_Word16*) encoded);
+ }
+#endif
+#ifdef CODEC_G722
+ else if (coder==webrtc::kDecoderG722) { /*g722 */
+ cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (WebRtc_Word16*)encoded);
+ cdlen=frameLen>>1;
+ }
+#endif
+#ifdef CODEC_ILBC
+ else if (coder==webrtc::kDecoderILBC) { /*iLBC */
+ cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(WebRtc_Word16*)encoded);
+ }
+#endif
+#if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
+ else if (coder==webrtc::kDecoderISAC) { /*iSAC */
+ int noOfCalls=0;
+ cdlen=0;
+ while (cdlen<=0) {
+#ifdef CODEC_ISAC /* floating point */
+ cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+#else /* fixed point */
+ cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
+#endif
+ noOfCalls++;
+ }
+ }
+#endif
+#ifdef CODEC_ISAC_SWB
+ else if (coder==webrtc::kDecoderISACswb) { /* iSAC SWB */
+ int noOfCalls=0;
+ cdlen=0;
+ while (cdlen<=0) {
+ cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(WebRtc_Word16*)encoded);
+ noOfCalls++;
+ }
+ }
+#endif
+#ifdef CODEC_CELT_32
+ else if (coder==webrtc::kDecoderCELT_32) { /* Celt */
+ int encodedLen = 0;
+ cdlen = 0;
+ while (cdlen <= 0) {
+ cdlen = WebRtcCelt_Encode(CELT32enc_inst[k], &indata[encodedLen], encoded);
+ encodedLen += 10*32; /* 10 ms */
+ }
+ if( (encodedLen != frameLen) || cdlen < 0) {
+ printf("Error encoding Celt frame!\n");
+ exit(0);
+ }
+ }
+#endif
+
+ indata += frameLen;
+ encoded += cdlen;
+ totalLen += cdlen;
+
+ } // end for
+
+ first_cng=1;
+ return(totalLen);
+}
+
+
+
+void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc){
+
+ rtp_data[0]=(unsigned char)0x80;
+ rtp_data[1]=(unsigned char)(payloadType & 0xFF);
+ rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF);
+ rtp_data[3]=(unsigned char)((seqNo)&0xFF);
+ rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF);
+ rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF);
+
+ rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF);
+ rtp_data[7]=(unsigned char)(timestamp & 0xFF);
+
+ rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF);
+ rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF);
+
+ rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF);
+ rtp_data[11]=(unsigned char)(ssrc & 0xFF);
+}
+
+
+int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
+ int seqNo, WebRtc_UWord32 ssrc)
+{
+
+ int i;
+ unsigned char *rtpPointer;
+ WebRtc_UWord16 offset;
+
+ /* first create "standard" RTP header */
+ makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc);
+
+ rtpPointer = &rtp_data[12];
+
+ /* add one sub-header for each redundant payload (not the primary) */
+ for(i=0; i<numPayloads-1; i++) { /* |0 1 2 3 4 5 6 7| */
+ if(blockLen[i] > 0) {
+ offset = (WebRtc_UWord16) (timestamp[numPayloads-1] - timestamp[i]);
+
+ rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F| block PT | */
+ rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF); /* | timestamp- | */
+ rtpPointer[2] = (unsigned char) ( ((offset & 0x3F)<<2) |
+ ( (blockLen[i]>>8) & 0x03 ) ); /* | -offset |bl-| */
+ rtpPointer[3] = (unsigned char) ( blockLen[i] & 0xFF ); /* | -ock length | */
+
+ rtpPointer += 4;
+ }
+ }
+
+ /* last sub-header */
+ rtpPointer[0]= (unsigned char) (0x00 | (0x7F&payloadType[numPayloads-1]));/* |F| block PT | */
+ rtpPointer += 1;
+
+ return(rtpPointer - rtp_data); /* length of header in bytes */
+}
+
+
+
+int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) {
+ unsigned char E,R,V;
+ R=0;
+ V=(unsigned char)Volume;
+ if (End==0) {
+ E = 0x00;
+ } else {
+ E = 0x80;
+ }
+ payload_data[0]=(unsigned char)Event;
+ payload_data[1]=(unsigned char)(E|R|V);
+ //Duration equals 8 times time_ms, default is 8000 Hz.
+ payload_data[2]=(unsigned char)((Duration>>8)&0xFF);
+ payload_data[3]=(unsigned char)(Duration&0xFF);
+ return(4);
+}
+
+void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples)
+{
+
+ WebRtc_Word16 *tempVec;
+ WebRtc_Word16 *readPtr, *writeL, *writeR;
+
+ if (numSamples <= 0)
+ return;
+
+ tempVec = (WebRtc_Word16 *) malloc(sizeof(WebRtc_Word16) * numSamples);
+ if (tempVec == NULL) {
+ printf("Error allocating memory\n");
+ exit(0);
+ }
+
+ memcpy(tempVec, audioSamples, numSamples*sizeof(WebRtc_Word16));
+
+ writeL = audioSamples;
+ writeR = &audioSamples[numSamples/2];
+ readPtr = tempVec;
+
+ for (int k = 0; k < numSamples; k += 2)
+ {
+ *writeL = *readPtr;
+ readPtr++;
+ *writeR = *readPtr;
+ readPtr++;
+ writeL++;
+ writeR++;
+ }
+
+ free(tempVec);
+
+}
+
+
+void stereoInterleave(unsigned char* data, int dataLen, int stride)
+{
+
+ unsigned char *ptrL, *ptrR;
+ unsigned char temp[10];
+
+ if (stride > 10)
+ {
+ exit(0);
+ }
+
+ if (dataLen%1 != 0)
+ {
+ // must be even number of samples
+ printf("Error: cannot interleave odd sample number\n");
+ exit(0);
+ }
+
+ ptrL = data + stride;
+ ptrR = &data[dataLen/2];
+
+ while (ptrL < ptrR) {
+ // copy from right pointer to temp
+ memcpy(temp, ptrR, stride);
+
+ // shift data between pointers
+ memmove(ptrL + stride, ptrL, ptrR - ptrL);
+
+ // copy from temp to left pointer
+ memcpy(ptrL, temp, stride);
+
+ // advance pointers
+ ptrL += stride*2;
+ ptrR += stride;
+ }
+
+}