| /* |
| * 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(henrik.lundin): Refactor or replace all of this application. |
| |
| /* 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 <algorithm> |
| |
| #include "rtc_base/checks.h" |
| #include "typedefs.h" // NOLINT(build/include) |
| |
| // needed for NetEqDecoder |
| #include "modules/audio_coding/neteq/include/neteq.h" |
| |
| /************************/ |
| /* Define payload types */ |
| /************************/ |
| |
| #include "PayloadTypes.h" |
| |
| namespace { |
| const size_t kRtpDataSize = 8000; |
| } |
| |
| /*********************/ |
| /* 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, |
| size_t frameLen, |
| int* fs, |
| int* bitrate, |
| int* useRed); |
| int NetEQTest_init_coders(webrtc::NetEqDecoder coder, |
| size_t enc_frameSize, |
| int bitrate, |
| int sampfreq, |
| int vad, |
| size_t numChannels); |
| void defineCodecs(webrtc::NetEqDecoder* usedCodec, int* noOfCodecs); |
| int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels); |
| size_t NetEQTest_encode(webrtc::NetEqDecoder coder, |
| int16_t* indata, |
| size_t frameLen, |
| unsigned char* encoded, |
| int sampleRate, |
| int* vad, |
| int useVAD, |
| int bitrate, |
| size_t 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); |
| size_t makeDTMFpayload(unsigned char* payload_data, |
| int Event, |
| int End, |
| int Volume, |
| int Duration); |
| void stereoDeInterleave(int16_t* audioSamples, size_t numSamples); |
| void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride); |
| |
| /*********************/ |
| /* Codec definitions */ |
| /*********************/ |
| |
| #include "webrtc_vad.h" |
| |
| #if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC)) |
| #include "modules/audio_coding/codecs/pcm16b/pcm16b.h" |
| #endif |
| #ifdef CODEC_G711 |
| #include "modules/audio_coding/codecs/g711/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 "modules/audio_coding/codecs/ilbc/ilbc.h" |
| #endif |
| #if (defined CODEC_ISAC || defined CODEC_ISAC_SWB) |
| #include "modules/audio_coding/codecs/isac/main/include/isac.h" |
| #endif |
| #ifdef NETEQ_ISACFIX_CODEC |
| #include "modules/audio_coding/codecs/isac/fix/include/isacfix.h" |
| #ifdef CODEC_ISAC |
| #error Cannot have both ISAC and ISACfix defined. Please de-select one. |
| #endif |
| #endif |
| #ifdef CODEC_G722 |
| #include "modules/audio_coding/codecs/g722/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 "modules/audio_coding/codecs/cng/webrtc_cng.h" |
| #endif |
| #ifdef CODEC_OPUS |
| #include "modules/audio_coding/codecs/opus/opus_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 |
| IlbcEncoderInstance* 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)) |
| webrtc::ComfortNoiseEncoder *CNG_encoder[2]; |
| #endif |
| #ifdef CODEC_OPUS |
| OpusEncInst* opus_inst[2]; |
| #endif |
| |
| int main(int argc, char* argv[]) { |
| size_t packet_size; |
| int fs; |
| webrtc::NetEqDecoder usedCodec; |
| int payloadType; |
| int bitrate = 0; |
| int useVAD, vad; |
| int useRed = 0; |
| size_t len, enc_len; |
| int16_t org_data[4000]; |
| unsigned char rtp_data[kRtpDataSize]; |
| 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}; |
| size_t RTPheaderLen = 12; |
| uint8_t red_data[kRtpDataSize]; |
| #ifdef INSERT_OLD_PACKETS |
| uint16_t old_length, old_plen; |
| size_t old_enc_len; |
| int first_old_packet = 1; |
| unsigned char old_rtp_data[kRtpDataSize]; |
| size_t packet_age = 0; |
| #endif |
| #ifdef INSERT_DTMF_PACKETS |
| int NTone = 1; |
| int DTMFfirst = 1; |
| uint32_t DTMFtimestamp; |
| bool dtmfSent = false; |
| #endif |
| bool usingStereo = false; |
| size_t stereoMode = 0; |
| size_t 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 packets 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 " |
| "version)\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_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 // CODEC_RED |
| #ifdef CODEC_OPUS |
| printf(" : opus Opus codec with FEC (48kHz, 32kbps, FEC" |
| " on and tuned for 5%% packet losses)\n"); |
| #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]); |
| int packet_size_int = atoi(argv[3]); |
| if (packet_size_int <= 0) { |
| printf("Packet size %d must be positive", packet_size_int); |
| return -1; |
| } |
| printf("Packet size: %d\n", packet_size_int); |
| packet_size = static_cast<size_t>(packet_size_int); |
| |
| // 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::NetEqDecoder::kDecoderPCMu: |
| case webrtc::NetEqDecoder::kDecoderPCMa: |
| case webrtc::NetEqDecoder::kDecoderG722: { |
| // 1 octet per sample |
| stereoMode = STEREO_MODE_SAMPLE_1; |
| break; |
| } |
| case webrtc::NetEqDecoder::kDecoderPCM16B: |
| case webrtc::NetEqDecoder::kDecoderPCM16Bwb: |
| case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz: |
| case webrtc::NetEqDecoder::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::NetEqDecoder::kDecoderISAC) || |
| (usedCodec == webrtc::NetEqDecoder::kDecoderISACswb)) { |
| if (argc != 7) { |
| if (usedCodec == webrtc::NetEqDecoder::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::NetEqDecoder::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", static_cast<int>(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(static_cast<unsigned short>(12 + enc_len + 8)); |
| plen = htons(static_cast<unsigned short>(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 (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] = static_cast<uint16_t>(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] = static_cast<uint16_t>(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(static_cast<unsigned short>(12 + enc_len + 8)); |
| plen = htons(static_cast<unsigned short>(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 (size_t k = 0; k < 12 + enc_len; k++) { |
| rtp_data[k] = rand() + rand(); |
| } |
| #endif |
| #ifdef RANDOM_PAYLOAD_DATA |
| for (size_t 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::NetEqDecoder::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, |
| size_t frameLen, |
| int* fs, |
| int* bitrate, |
| int* useRed) { |
| *bitrate = 0; /* Default bitrate setting */ |
| *useRed = 0; /* Default no redundancy */ |
| |
| if (!strcmp(name, "pcmu")) { |
| *codec = webrtc::NetEqDecoder::kDecoderPCMu; |
| *PT = NETEQ_CODEC_PCMU_PT; |
| *fs = 8000; |
| } else if (!strcmp(name, "pcma")) { |
| *codec = webrtc::NetEqDecoder::kDecoderPCMa; |
| *PT = NETEQ_CODEC_PCMA_PT; |
| *fs = 8000; |
| } else if (!strcmp(name, "pcm16b")) { |
| *codec = webrtc::NetEqDecoder::kDecoderPCM16B; |
| *PT = NETEQ_CODEC_PCM16B_PT; |
| *fs = 8000; |
| } else if (!strcmp(name, "pcm16b_wb")) { |
| *codec = webrtc::NetEqDecoder::kDecoderPCM16Bwb; |
| *PT = NETEQ_CODEC_PCM16B_WB_PT; |
| *fs = 16000; |
| } else if (!strcmp(name, "pcm16b_swb32")) { |
| *codec = webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz; |
| *PT = NETEQ_CODEC_PCM16B_SWB32KHZ_PT; |
| *fs = 32000; |
| } else if (!strcmp(name, "pcm16b_swb48")) { |
| *codec = webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz; |
| *PT = NETEQ_CODEC_PCM16B_SWB48KHZ_PT; |
| *fs = 48000; |
| } else if (!strcmp(name, "g722")) { |
| *codec = webrtc::NetEqDecoder::kDecoderG722; |
| *PT = NETEQ_CODEC_G722_PT; |
| *fs = 16000; |
| } else if ((!strcmp(name, "ilbc")) && |
| ((frameLen % 240 == 0) || (frameLen % 160 == 0))) { |
| *fs = 8000; |
| *codec = webrtc::NetEqDecoder::kDecoderILBC; |
| *PT = NETEQ_CODEC_ILBC_PT; |
| } else if (!strcmp(name, "isac")) { |
| *fs = 16000; |
| *codec = webrtc::NetEqDecoder::kDecoderISAC; |
| *PT = NETEQ_CODEC_ISAC_PT; |
| } else if (!strcmp(name, "isacswb")) { |
| *fs = 32000; |
| *codec = webrtc::NetEqDecoder::kDecoderISACswb; |
| *PT = NETEQ_CODEC_ISACSWB_PT; |
| } else if (!strcmp(name, "red_pcm")) { |
| *codec = webrtc::NetEqDecoder::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::NetEqDecoder::kDecoderISAC; |
| *PT = NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */ |
| *fs = 16000; |
| *useRed = 1; |
| } else if (!strcmp(name, "opus")) { |
| *codec = webrtc::NetEqDecoder::kDecoderOpus; |
| *PT = NETEQ_CODEC_OPUS_PT; /* this will be the PT for the sub-headers */ |
| *fs = 48000; |
| } else { |
| printf("Error: Not a supported codec (%s)\n", name); |
| exit(0); |
| } |
| } |
| |
| int NetEQTest_init_coders(webrtc::NetEqDecoder coder, |
| size_t enc_frameSize, |
| int bitrate, |
| int sampfreq, |
| int vad, |
| size_t numChannels) { |
| int ok = 0; |
| |
| for (size_t k = 0; k < numChannels; k++) { |
| VAD_inst[k] = WebRtcVad_Create(); |
| if (!VAD_inst[k]) { |
| 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)) |
| if (sampfreq <= 16000) { |
| CNG_encoder[k] = new webrtc::ComfortNoiseEncoder(sampfreq, 200, 5); |
| } |
| #endif |
| |
| switch (coder) { |
| #ifdef CODEC_PCM16B |
| case webrtc::NetEqDecoder::kDecoderPCM16B: |
| #endif |
| #ifdef CODEC_PCM16B_WB |
| case webrtc::NetEqDecoder::kDecoderPCM16Bwb: |
| #endif |
| #ifdef CODEC_PCM16B_32KHZ |
| case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz: |
| #endif |
| #ifdef CODEC_PCM16B_48KHZ |
| case webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz: |
| #endif |
| #ifdef CODEC_G711 |
| case webrtc::NetEqDecoder::kDecoderPCMu: |
| case webrtc::NetEqDecoder::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 %" PRIuS "ms)\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_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::NetEqDecoder::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) > 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::NetEqDecoder::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::NetEqDecoder::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, |
| static_cast<int>(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::NetEqDecoder::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, |
| static_cast<int>(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 |
| #ifdef CODEC_OPUS |
| case webrtc::NetEqDecoder::kDecoderOpus: |
| ok = WebRtcOpus_EncoderCreate(&opus_inst[k], 1, 0); |
| if (ok != 0) { |
| printf("Error: Couldn't allocate memory for Opus encoding " |
| "instance\n"); |
| exit(0); |
| } |
| WebRtcOpus_EnableFec(opus_inst[k]); |
| WebRtcOpus_SetPacketLossRate(opus_inst[k], 5); |
| 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, size_t numChannels) { |
| for (size_t k = 0; k < numChannels; k++) { |
| WebRtcVad_Free(VAD_inst[k]); |
| #if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \ |
| defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48)) |
| delete CNG_encoder[k]; |
| CNG_encoder[k] = nullptr; |
| #endif |
| |
| switch (coder) { |
| #ifdef CODEC_PCM16B |
| case webrtc::NetEqDecoder::kDecoderPCM16B: |
| #endif |
| #ifdef CODEC_PCM16B_WB |
| case webrtc::NetEqDecoder::kDecoderPCM16Bwb: |
| #endif |
| #ifdef CODEC_PCM16B_32KHZ |
| case webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz: |
| #endif |
| #ifdef CODEC_PCM16B_48KHZ |
| case webrtc::NetEqDecoder::kDecoderPCM16Bswb48kHz: |
| #endif |
| #ifdef CODEC_G711 |
| case webrtc::NetEqDecoder::kDecoderPCMu: |
| case webrtc::NetEqDecoder::kDecoderPCMa: |
| #endif |
| // do nothing |
| break; |
| #ifdef CODEC_G729 |
| case webrtc::NetEqDecoder::kDecoderG729: |
| WebRtcG729_FreeEnc(G729enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G729_1 |
| case webrtc::NetEqDecoder::kDecoderG729_1: |
| WebRtcG7291_Free(G729_1_inst[k]); |
| break; |
| #endif |
| |
| #ifdef CODEC_G722_1_16 |
| case webrtc::NetEqDecoder::kDecoderG722_1_16: |
| WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G722_1_24 |
| case webrtc::NetEqDecoder::kDecoderG722_1_24: |
| WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G722_1_32 |
| case webrtc::NetEqDecoder::kDecoderG722_1_32: |
| WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G722_1C_24 |
| case webrtc::NetEqDecoder::kDecoderG722_1C_24: |
| WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G722_1C_32 |
| case webrtc::NetEqDecoder::kDecoderG722_1C_32: |
| WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G722_1C_48 |
| case webrtc::NetEqDecoder::kDecoderG722_1C_48: |
| WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_G722 |
| case webrtc::NetEqDecoder::kDecoderG722: |
| WebRtcG722_FreeEncoder(g722EncState[k]); |
| break; |
| #endif |
| #ifdef CODEC_AMR |
| case webrtc::NetEqDecoder::kDecoderAMR: |
| WebRtcAmr_FreeEnc(AMRenc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_AMRWB |
| case webrtc::NetEqDecoder::kDecoderAMRWB: |
| WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_ILBC |
| case webrtc::NetEqDecoder::kDecoderILBC: |
| WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_ISAC |
| case webrtc::NetEqDecoder::kDecoderISAC: |
| WebRtcIsac_Free(ISAC_inst[k]); |
| break; |
| #endif |
| #ifdef NETEQ_ISACFIX_CODEC |
| case webrtc::NetEqDecoder::kDecoderISAC: |
| WebRtcIsacfix_Free(ISAC_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_ISAC_SWB |
| case webrtc::NetEqDecoder::kDecoderISACswb: |
| WebRtcIsac_Free(ISACSWB_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_GSMFR |
| case webrtc::NetEqDecoder::kDecoderGSMFR: |
| WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]); |
| break; |
| #endif |
| #ifdef CODEC_OPUS |
| case webrtc::NetEqDecoder::kDecoderOpus: |
| WebRtcOpus_EncoderFree(opus_inst[k]); |
| break; |
| #endif |
| default: |
| printf("Error: unknown codec in call to NetEQTest_init_coders.\n"); |
| exit(0); |
| break; |
| } |
| } |
| |
| return (0); |
| } |
| |
| size_t NetEQTest_encode(webrtc::NetEqDecoder coder, |
| int16_t* indata, |
| size_t frameLen, |
| unsigned char* encoded, |
| int sampleRate, |
| int* vad, |
| int useVAD, |
| int bitrate, |
| size_t numChannels) { |
| size_t cdlen = 0; |
| int16_t* tempdata; |
| static bool first_cng = true; |
| size_t tempLen; |
| *vad = 1; |
| |
| // check VAD first |
| if (useVAD) { |
| *vad = 0; |
| |
| const size_t sampleRate_10 = static_cast<size_t>(10 * sampleRate / 1000); |
| const size_t sampleRate_20 = static_cast<size_t>(20 * sampleRate / 1000); |
| const size_t sampleRate_30 = static_cast<size_t>(30 * sampleRate / 1000); |
| for (size_t 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 >= sampleRate_10) { |
| if ((tempLen % sampleRate_30) == 0) { // tempLen is multiple of 30ms |
| localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata, |
| sampleRate_30); |
| tempdata += sampleRate_30; |
| tempLen -= sampleRate_30; |
| } else if (tempLen >= sampleRate_20) { // tempLen >= 20ms |
| localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata, |
| sampleRate_20); |
| tempdata += sampleRate_20; |
| tempLen -= sampleRate_20; |
| } else { // use 10ms |
| localVad |= WebRtcVad_Process(VAD_inst[k], sampleRate, tempdata, |
| sampleRate_10); |
| tempdata += sampleRate_10; |
| tempLen -= sampleRate_10; |
| } |
| } |
| |
| // aggregate all VAD decisions over all channels |
| *vad |= localVad; |
| } |
| |
| if (!*vad) { |
| // all channels are silent |
| rtc::Buffer workaround; |
| cdlen = 0; |
| for (size_t k = 0; k < numChannels; k++) { |
| workaround.Clear(); |
| tempLen = CNG_encoder[k]->Encode( |
| rtc::ArrayView<const int16_t>( |
| &indata[k * frameLen], |
| (frameLen <= 640 ? frameLen : 640) /* max 640 */), |
| first_cng, |
| &workaround); |
| memcpy(encoded, workaround.data(), tempLen); |
| encoded += tempLen; |
| cdlen += tempLen; |
| } |
| *vad = 0; |
| first_cng = false; |
| return (cdlen); |
| } |
| } |
| |
| // loop over all channels |
| size_t totalLen = 0; |
| |
| for (size_t k = 0; k < numChannels; k++) { |
| /* Encode with the selected coder type */ |
| if (coder == webrtc::NetEqDecoder::kDecoderPCMu) { /*g711 u-law */ |
| #ifdef CODEC_G711 |
| cdlen = WebRtcG711_EncodeU(indata, frameLen, encoded); |
| #endif |
| } else if (coder == webrtc::NetEqDecoder::kDecoderPCMa) { /*g711 A-law */ |
| #ifdef CODEC_G711 |
| cdlen = WebRtcG711_EncodeA(indata, frameLen, encoded); |
| } |
| #endif |
| #ifdef CODEC_PCM16B |
| else if ((coder == webrtc::NetEqDecoder::kDecoderPCM16B) || |
| (coder == webrtc::NetEqDecoder::kDecoderPCM16Bwb) || |
| (coder == webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz) || |
| (coder == webrtc::NetEqDecoder:: |
| kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, |
| 32kHz or 48kHz) */ |
| cdlen = WebRtcPcm16b_Encode(indata, frameLen, encoded); |
| } |
| #endif |
| #ifdef CODEC_G722 |
| else if (coder == webrtc::NetEqDecoder::kDecoderG722) { /*g722 */ |
| cdlen = WebRtcG722_Encode(g722EncState[k], indata, frameLen, encoded); |
| assert(cdlen == frameLen >> 1); |
| } |
| #endif |
| #ifdef CODEC_ILBC |
| else if (coder == webrtc::NetEqDecoder::kDecoderILBC) { /*iLBC */ |
| cdlen = static_cast<size_t>(std::max( |
| WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata, frameLen, encoded), 0)); |
| } |
| #endif |
| #if (defined(CODEC_ISAC) || \ |
| defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all |
| // NETEQ_ISACFIX_CODEC |
| else if (coder == webrtc::NetEqDecoder::kDecoderISAC) { /*iSAC */ |
| int noOfCalls = 0; |
| int res = 0; |
| while (res <= 0) { |
| #ifdef CODEC_ISAC /* floating point */ |
| res = |
| WebRtcIsac_Encode(ISAC_inst[k], &indata[noOfCalls * 160], encoded); |
| #else /* fixed point */ |
| res = WebRtcIsacfix_Encode(ISAC_inst[k], &indata[noOfCalls * 160], |
| encoded); |
| #endif |
| noOfCalls++; |
| } |
| cdlen = static_cast<size_t>(res); |
| } |
| #endif |
| #ifdef CODEC_ISAC_SWB |
| else if (coder == webrtc::NetEqDecoder::kDecoderISACswb) { /* iSAC SWB */ |
| int noOfCalls = 0; |
| int res = 0; |
| while (res <= 0) { |
| res = WebRtcIsac_Encode(ISACSWB_inst[k], &indata[noOfCalls * 320], |
| encoded); |
| noOfCalls++; |
| } |
| cdlen = static_cast<size_t>(res); |
| } |
| #endif |
| #ifdef CODEC_OPUS |
| cdlen = WebRtcOpus_Encode(opus_inst[k], indata, frameLen, kRtpDataSize - 12, |
| encoded); |
| RTC_CHECK_GT(cdlen, 0); |
| #endif |
| indata += frameLen; |
| encoded += cdlen; |
| totalLen += cdlen; |
| |
| } // end for |
| |
| first_cng = true; |
| return (totalLen); |
| } |
| |
| void makeRTPheader(unsigned char* rtp_data, |
| int payloadType, |
| int seqNo, |
| uint32_t timestamp, |
| uint32_t ssrc) { |
| rtp_data[0] = 0x80; |
| rtp_data[1] = payloadType & 0xFF; |
| rtp_data[2] = (seqNo >> 8) & 0xFF; |
| rtp_data[3] = seqNo & 0xFF; |
| rtp_data[4] = timestamp >> 24; |
| rtp_data[5] = (timestamp >> 16) & 0xFF; |
| rtp_data[6] = (timestamp >> 8) & 0xFF; |
| rtp_data[7] = timestamp & 0xFF; |
| rtp_data[8] = ssrc >> 24; |
| rtp_data[9] = (ssrc >> 16) & 0xFF; |
| rtp_data[10] = (ssrc >> 8) & 0xFF; |
| rtp_data[11] = 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++) { |
| if (blockLen[i] > 0) { |
| offset = static_cast<uint16_t>(timestamp[numPayloads - 1] - timestamp[i]); |
| |
| // Byte |0| |1 2 | 3 | |
| // Bit |0|1234567|01234567012345|6701234567| |
| // |F|payload| timestamp | block | |
| // | | type | offset | length | |
| rtpPointer[0] = (payloadType[i] & 0x7F) | 0x80; |
| rtpPointer[1] = (offset >> 6) & 0xFF; |
| rtpPointer[2] = ((offset & 0x3F) << 2) | ((blockLen[i] >> 8) & 0x03); |
| rtpPointer[3] = blockLen[i] & 0xFF; |
| |
| rtpPointer += 4; |
| } |
| } |
| |
| // Bit |0|1234567| |
| // |0|payload| |
| // | | type | |
| rtpPointer[0] = payloadType[numPayloads - 1] & 0x7F; |
| ++rtpPointer; |
| |
| return rtpPointer - rtp_data; // length of header in bytes |
| } |
| |
| size_t 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, size_t 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 (size_t k = 0; k < numSamples; k += 2) { |
| *writeL = *readPtr; |
| readPtr++; |
| *writeR = *readPtr; |
| readPtr++; |
| writeL++; |
| writeR++; |
| } |
| |
| free(tempVec); |
| } |
| |
| void stereoInterleave(unsigned char* data, size_t dataLen, size_t 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; |
| } |
| } |
