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Adding Opus to RTPencode.

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As a step toward fixing webrtc:3987, here we update the RTPencode to allow Opus RTP payloads.

BUG=webrtc:3987, webrtc:2692

Review URL: https://codereview.webrtc.org/1516653003

Cr-Commit-Position: refs/heads/master@{#10987}
This commit is contained in:
minyue
2015-12-11 01:58:26 -08:00
committed by Commit bot
parent 71f5a9a377
commit cb23c0d984
3 changed files with 48 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/neteq/neteq_tests.gypi
View File

@ -56,6 +56,7 @@
'isac',
'neteq_test_tools', # Test helpers
'pcm16b',
'webrtc_opus',
],
'defines': [
'CODEC_ILBC',
@ -72,6 +73,7 @@
'CODEC_CNGCODEC32',
'CODEC_ATEVENT_DECODE',
'CODEC_RED',
'CODEC_OPUS',
],
'include_dirs': [
'include',

2
webrtc/modules/audio_coding/neteq/test/PayloadTypes.h
View File

@ -39,7 +39,7 @@
#define NETEQ_CODEC_G722_1_16_PT 108
#define NETEQ_CODEC_G722_1_24_PT 109
#define NETEQ_CODEC_G722_1_32_PT 110
#define NETEQ_CODEC_SC3_PT 111
#define NETEQ_CODEC_OPUS_PT 111
#define NETEQ_CODEC_AMR_PT 112
#define NETEQ_CODEC_GSMEFR_PT 113
//#define NETEQ_CODEC_ILBCRCU_PT 114

49
webrtc/modules/audio_coding/neteq/test/RTPencode.cc
View File

@ -25,7 +25,9 @@
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/typedefs.h"
// needed for NetEqDecoder
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
@ -36,6 +38,10 @@
#include "PayloadTypes.h"
namespace {
const size_t kRtpDataSize = 8000;
}
/*********************/
/* Misc. definitions */
/*********************/
@ -193,6 +199,9 @@ void stereoInterleave(unsigned char* data, size_t dataLen, size_t stride);
#if ((defined CODEC_SPEEX_8) || (defined CODEC_SPEEX_16))
#include "SpeexInterface.h"
#endif
#ifdef CODEC_OPUS
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#endif
/***********************************/
/* Global codec instance variables */
@ -264,6 +273,9 @@ SPEEX_encinst_t* SPEEX8enc_inst[2];
#ifdef CODEC_SPEEX_16
SPEEX_encinst_t* SPEEX16enc_inst[2];
#endif
#ifdef CODEC_OPUS
OpusEncInst* opus_inst[2];
#endif
int main(int argc, char* argv[]) {
size_t packet_size;
@ -275,7 +287,7 @@ int main(int argc, char* argv[]) {
int useRed = 0;
size_t len, enc_len;
int16_t org_data[4000];
unsigned char rtp_data[8000];
unsigned char rtp_data[kRtpDataSize];
int16_t seqNo = 0xFFF;
uint32_t ssrc = 1235412312;
uint32_t timestamp = 0xAC1245;
@ -286,12 +298,12 @@ int main(int argc, char* argv[]) {
uint32_t red_TS[2] = {0};
uint16_t red_len[2] = {0};
size_t RTPheaderLen = 12;
uint8_t red_data[8000];
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[8000];
unsigned char old_rtp_data[kRtpDataSize];
size_t packet_age = 0;
#endif
#ifdef INSERT_DTMF_PACKETS
@ -429,6 +441,10 @@ int main(int argc, char* argv[]) {
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");
@ -880,6 +896,10 @@ void NetEQTest_GetCodec_and_PT(char* name,
*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);
@ -1410,13 +1430,24 @@ int NetEQTest_init_coders(webrtc::NetEqDecoder coder,
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);
}
@ -1542,6 +1573,11 @@ int NetEQTest_free_coders(webrtc::NetEqDecoder coder, size_t numChannels) {
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");
@ -1686,6 +1722,11 @@ size_t NetEQTest_encode(webrtc::NetEqDecoder coder,
}
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, 0u);
#endif
indata += frameLen;
encoded += cdlen;