| /* |
| * 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 <assert.h> |
| |
| #include "webrtc/typedefs.h" |
| // needed for NetEqDecoder |
| #include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h" |
| #include "webrtc/modules/audio_coding/neteq/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, int16_t *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, uint32_t timestamp, uint32_t ssrc); |
| int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, uint32_t *timestamp, uint16_t *blockLen, |
| int seqNo, uint32_t ssrc); |
| int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration); |
| void stereoDeInterleave(int16_t* 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]; |
| int16_t AMR_bitrate; |
| #endif |
| #ifdef CODEC_AMRWB |
| AMRWB_encinst_t *AMRWBenc_inst[2]; |
| int16_t 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 |
| |
| |
| 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; |
| int16_t org_data[4000]; |
| unsigned char rtp_data[8000]; |
| int16_t seqNo=0xFFF; |
| uint32_t ssrc=1235412312; |
| uint32_t timestamp=0xAC1245; |
| uint16_t length, plen; |
| uint32_t offset; |
| double sendtime = 0; |
| int red_PT[2] = {0}; |
| uint32_t red_TS[2] = {0}; |
| uint16_t red_len[2] = {0}; |
| int RTPheaderLen=12; |
| uint8_t red_data[8000]; |
| #ifdef INSERT_OLD_PACKETS |
| uint16_t 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; |
| uint32_t 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 |
| uint32_t 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 = (uint32_t) 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 = (uint32_t) 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], 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, int16_t *indata, int frameLen, unsigned char * encoded,int sampleRate , |
| int * vad, int useVAD, int bitrate, int numChannels){ |
| |
| short cdlen = 0; |
| int16_t *tempdata; |
| static int first_cng=1; |
| int16_t 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(indata, frameLen, (int16_t*) encoded); |
| #endif |
| } |
| else if (coder==webrtc::kDecoderPCMa) { /*g711 A-law */ |
| #ifdef CODEC_G711 |
| cdlen = WebRtcG711_EncodeA(indata, frameLen, (int16_t*) 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, (int16_t*) encoded); |
| } |
| #endif |
| #ifdef CODEC_G722 |
| else if (coder==webrtc::kDecoderG722) { /*g722 */ |
| cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (int16_t*)encoded); |
| assert(cdlen == frameLen>>1); |
| } |
| #endif |
| #ifdef CODEC_ILBC |
| else if (coder==webrtc::kDecoderILBC) { /*iLBC */ |
| cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(int16_t*)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], |
| encoded); |
| #else /* fixed point */ |
| cdlen = WebRtcIsacfix_Encode(ISAC_inst[k], |
| &indata[noOfCalls * 160], |
| 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], |
| 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, uint32_t timestamp, uint32_t 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, uint32_t *timestamp, uint16_t *blockLen, |
| int seqNo, uint32_t ssrc) |
| { |
| |
| int i; |
| unsigned char *rtpPointer; |
| uint16_t 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 = (uint16_t) (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(int16_t* audioSamples, int numSamples) |
| { |
| |
| int16_t *tempVec; |
| int16_t *readPtr, *writeL, *writeR; |
| |
| if (numSamples <= 0) |
| return; |
| |
| tempVec = (int16_t *) malloc(sizeof(int16_t) * numSamples); |
| if (tempVec == NULL) { |
| printf("Error allocating memory\n"); |
| exit(0); |
| } |
| |
| memcpy(tempVec, audioSamples, numSamples*sizeof(int16_t)); |
| |
| 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; |
| } |
| |
| } |