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webrtc/modules/audio_coding/neteq/test/RTPencode.cc
Karl Wiberg 31fbb5425e NetEq: Drop unnecessary dependency on the audio decoder implementations 
BUG=webrtc:8396

Change-Id: I7524dae93b43b656a13fdd535e48373bc29b405e
Reviewed-on: https://webrtc-review.googlesource.com/10804
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20310}
2017-10-16 12:57:47 +00:00

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/*
* 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;
}
}
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