Cleaning up all use of RTP_PAYLOAD_NAME_SIZE and RTCP_CNAME_SIZE also fixed the char handing in trace.
Review URL: https://webrtc-codereview.appspot.com/358001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1535 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/modules/interface/module.h b/src/modules/interface/module.h
index a274d95..37e5027 100644
--- a/src/modules/interface/module.h
+++ b/src/modules/interface/module.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -41,7 +41,7 @@
// TODO(perkj): Make this pure virtual when Chromium have implemented
// reference counting ADM and Video capture module.
virtual int32_t AddRef() {
- assert(!"Not implemented.");
+ assert(false && "Not implemented.");
return 1;
}
@@ -52,7 +52,7 @@
// TODO(perkj): Make this pure virtual when Chromium have implemented
// reference counting ADM and Video capture module.
virtual int32_t Release() {
- assert(!"Not implemented.");
+ assert(false && "Not implemented.");
return 1;
}
diff --git a/src/modules/rtp_rtcp/interface/rtp_rtcp.h b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
index 0032c4b..08a92b3 100644
--- a/src/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/src/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -621,14 +621,14 @@
*
* return -1 on failure else 0
*/
- virtual WebRtc_Word32 SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0;
+ virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0;
/*
* Get RTCP CName (i.e unique identifier)
*
* return -1 on failure else 0
*/
- virtual WebRtc_Word32 CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0;
+ virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]) = 0;
/*
* Get remote CName
@@ -637,7 +637,7 @@
*/
virtual WebRtc_Word32 RemoteCNAME(
const WebRtc_UWord32 remoteSSRC,
- WebRtc_Word8 cName[RTCP_CNAME_SIZE]) const = 0;
+ char cName[RTCP_CNAME_SIZE]) const = 0;
/*
* Get remote NTP
@@ -657,7 +657,7 @@
*/
virtual WebRtc_Word32 AddMixedCNAME(
const WebRtc_UWord32 SSRC,
- const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) = 0;
+ const char cName[RTCP_CNAME_SIZE]) = 0;
/*
* RemoveMixedCNAME
diff --git a/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index d154cfd..fbbb032 100644
--- a/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -189,7 +189,7 @@
virtual WebRtc_Word32 OnInitializeDecoder(
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) = 0;
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.cc b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
index 468d16b..f7d08f0 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -564,6 +564,7 @@
return it->second;
}
RTCPCnameInformation* cnameInfo = new RTCPCnameInformation;
+ memset(cnameInfo->name, 0, RTCP_CNAME_SIZE);
_receivedCnameMap[remoteSSRC] = cnameInfo;
return cnameInfo;
}
@@ -707,17 +708,14 @@
}
// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
+ const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
+ RTCPCnameInformation* cnameInfo =
+ CreateCnameInformation(rtcpPacket.CName.SenderSSRC);
+ assert(cnameInfo);
- RTCPCnameInformation* cnameInfo = CreateCnameInformation(rtcpPacket.CName.SenderSSRC);
- if (cnameInfo)
- {
- memcpy(cnameInfo->name, rtcpPacket.CName.CName, rtcpPacket.CName.CNameLength);
- cnameInfo->length = rtcpPacket.CName.CNameLength;
- }
+ cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1);
}
// no need for critsect we have _criticalSectionRTCPReceiver
@@ -1383,27 +1381,20 @@
}
-WebRtc_Word32
-RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC,
- WebRtc_Word8 cName[RTCP_CNAME_SIZE]) const
-{
- if(cName == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -1;
- }
+WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const {
+ if (cName == NULL) {
+ WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
+ "%s invalid argument", __FUNCTION__);
+ return -1;
+ }
+ CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
+ RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
+ assert(cnameInfo);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
- if(cnameInfo == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "\tfailed to GetCnameInformation(%d)", remoteSSRC);
- return -1;
- }
- memcpy(cName, cnameInfo->name, cnameInfo->length);
- cName[cnameInfo->length] = 0;
- return 0;
+ cName[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(cName, cnameInfo->name, RTCP_CNAME_SIZE - 1);
+ return 0;
}
// no callbacks allowed inside this function
diff --git a/src/modules/rtp_rtcp/source/rtcp_receiver.h b/src/modules/rtp_rtcp/source/rtcp_receiver.h
index 738b030..580b1c1 100644
--- a/src/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/src/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -54,7 +54,7 @@
// get received cname
WebRtc_Word32 CNAME(const WebRtc_UWord32 remoteSSRC,
- WebRtc_Word8 cName[RTCP_CNAME_SIZE]) const;
+ char cName[RTCP_CNAME_SIZE]) const;
// get received NTP
WebRtc_Word32 NTP(WebRtc_UWord32 *ReceivedNTPsecs,
diff --git a/src/modules/rtp_rtcp/source/rtcp_sender.cc b/src/modules/rtp_rtcp/source/rtcp_sender.cc
index f8ad39f..c8a9cdd 100644
--- a/src/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -345,54 +345,38 @@
return 0;
}
-WebRtc_Word32
-RTCPSender::CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE])
-{
- if(cName == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- memcpy(cName, _CNAME, RTCP_CNAME_SIZE);
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) {
- size_t length = strlen(cName);
- if (length >= RTCP_CNAME_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument, too long cName", __FUNCTION__);
- return -1;
- }
+WebRtc_Word32 RTCPSender::CNAME(char cName[RTCP_CNAME_SIZE]) {
+ assert(cName);
CriticalSectionScoped lock(_criticalSectionRTCPSender);
- memcpy(_CNAME, cName, length + 1);
+ cName[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(cName, _CNAME, RTCP_CNAME_SIZE - 1);
return 0;
}
-WebRtc_Word32 RTCPSender::AddMixedCNAME(
- const WebRtc_UWord32 SSRC,
- const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) {
- size_t length = strlen(cName);
- if (length >= RTCP_CNAME_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument, too long cName", __FUNCTION__);
- return -1;
- }
+WebRtc_Word32 RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
+ assert(cName);
+ CriticalSectionScoped lock(_criticalSectionRTCPSender);
+ _CNAME[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1);
+ return 0;
+}
+
+WebRtc_Word32 RTCPSender::AddMixedCNAME(const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]) {
+ assert(cName);
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if (_csrcCNAMEs.size() >= kRtpCsrcSize) {
return -1;
}
RTCPCnameInformation* ptr = new RTCPCnameInformation();
- memcpy(ptr->name, cName, length + 1);
- ptr->length = static_cast<WebRtc_UWord8>(length);
+ ptr->name[RTCP_CNAME_SIZE - 1] = 0;
+ strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1);
_csrcCNAMEs[SSRC] = ptr;
return 0;
}
WebRtc_Word32 RTCPSender::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
_csrcCNAMEs.find(SSRC);
@@ -747,14 +731,16 @@
// CNAME = 1
rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1);
- rtcpbuffer[pos++]= cname->length;
+ size_t length = strlen(cname->name);
+ assert(length < RTCP_CNAME_SIZE);
+ rtcpbuffer[pos++]= static_cast<WebRtc_UWord8>(length);
SDESLength += 6;
- memcpy(&rtcpbuffer[pos], cname->name, cname->length);
- pos += cname->length;
- SDESLength += cname->length;
+ memcpy(&rtcpbuffer[pos],cname->name, length);
+ pos += length;
+ SDESLength += length;
WebRtc_UWord16 padding = 0;
// We must have a zero field even if we have an even multiple of 4 bytes
@@ -769,8 +755,9 @@
SDESLength += padding;
}
// in 32-bit words minus one and we don't count the header
- WebRtc_UWord16 length = (SDESLength / 4) - 1;
- ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, length);
+ WebRtc_UWord16 buffer_length = (SDESLength / 4) - 1;
+ ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos,
+ buffer_length);
return 0;
}
diff --git a/src/modules/rtp_rtcp/source/rtcp_sender.h b/src/modules/rtp_rtcp/source/rtcp_sender.h
index c7c163f..fec0b28 100644
--- a/src/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtcp_sender.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -51,11 +51,11 @@
WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
- WebRtc_Word32 CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE]);
- WebRtc_Word32 SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE]);
+ WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
+ WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
- const WebRtc_Word8 cName[RTCP_CNAME_SIZE]);
+ const char cName[RTCP_CNAME_SIZE]);
WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
diff --git a/src/modules/rtp_rtcp/source/rtcp_utility.cc b/src/modules/rtp_rtcp/source/rtcp_utility.cc
index 0cba9a8..f421672 100644
--- a/src/modules/rtp_rtcp/source/rtcp_utility.cc
+++ b/src/modules/rtp_rtcp/source/rtcp_utility.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -16,17 +16,6 @@
namespace webrtc {
-RTCPUtility::RTCPCnameInformation::RTCPCnameInformation() :
- length(0)
-{
- memset(name, 0, sizeof(name));
-}
-
-RTCPUtility::RTCPCnameInformation::~RTCPCnameInformation()
-{
-}
-
-///////////
// RTCPParserV2 : currently read only
RTCPUtility::RTCPParserV2::RTCPParserV2(const WebRtc_UWord8* rtcpData,
@@ -759,8 +748,8 @@
EndCurrentBlock();
return false;
}
-
- for (unsigned int i = 0; i < len; ++i)
+ WebRtc_UWord8 i = 0;
+ for (; i < len; ++i)
{
const WebRtc_UWord8 c = _ptrRTCPData[i];
if ((c < ' ') || (c > '{') || (c == '%') || (c == '\\'))
@@ -771,16 +760,14 @@
EndCurrentBlock();
return false;
}
-
_packet.CName.CName[i] = c;
}
-
- _packetType = kRtcpSdesChunkCode;
- _packet.CName.CNameLength = len;
+ // Make sure we are null terminated.
+ _packet.CName.CName[i] = 0;
+ _packetType = kRtcpSdesChunkCode;
foundCName = true;
}
-
_ptrRTCPData += len;
itemOctetsRead += len;
}
diff --git a/src/modules/rtp_rtcp/source/rtcp_utility.h b/src/modules/rtp_rtcp/source/rtcp_utility.h
index 542ec0d..faba2ef 100644
--- a/src/modules/rtp_rtcp/source/rtcp_utility.h
+++ b/src/modules/rtp_rtcp/source/rtcp_utility.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -20,16 +20,10 @@
namespace webrtc {
namespace RTCPUtility {
// CNAME
- class RTCPCnameInformation
+ struct RTCPCnameInformation
{
- public:
- RTCPCnameInformation();
- ~RTCPCnameInformation();
-
- WebRtc_UWord8 name[RTCP_CNAME_SIZE];
- WebRtc_UWord8 length;
+ char name[RTCP_CNAME_SIZE];
};
-
struct RTCPPacketRR
{
WebRtc_UWord32 SenderSSRC;
@@ -58,13 +52,11 @@
WebRtc_UWord32 LastSR;
WebRtc_UWord32 DelayLastSR;
};
-
struct RTCPPacketSDESCName
{
// RFC3550
WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord8 CName[RTCP_CNAME_SIZE];
- WebRtc_UWord8 CNameLength;
+ char CName[RTCP_CNAME_SIZE];
};
struct RTCPPacketExtendedJitterReportItem
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver.cc b/src/modules/rtp_rtcp/source/rtp_receiver.cc
index 0bbcb49..e943cfb 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver.cc
@@ -348,7 +348,7 @@
}
WebRtc_Word32 RTPReceiver::RegisterReceivePayload(
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
@@ -450,7 +450,8 @@
_redPayloadType = payloadType;
payload = new Payload;
payload->audio = false;
- memcpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE);
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
} else {
if (_audio) {
payload = RegisterReceiveAudioPayload(payloadName, payloadType,
@@ -493,7 +494,7 @@
}
WebRtc_Word32 RTPReceiver::ReceivePayloadType(
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate,
@@ -549,7 +550,7 @@
WebRtc_Word32 RTPReceiver::ReceivePayload(
const WebRtc_Word8 payloadType,
- WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ char payloadName[RTP_PAYLOAD_NAME_SIZE],
WebRtc_UWord32* frequency,
WebRtc_UWord8* channels,
WebRtc_UWord32* rate) const {
@@ -587,18 +588,20 @@
}
}
if (payloadName) {
- memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE);
+ payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
}
return 0;
}
WebRtc_Word32 RTPReceiver::RemotePayload(
- WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ char payloadName[RTP_PAYLOAD_NAME_SIZE],
WebRtc_Word8* payloadType,
WebRtc_UWord32* frequency,
WebRtc_UWord8* channels) const {
if(_lastReceivedPayloadType == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
+ WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
+ "%s invalid state", __FUNCTION__);
return -1;
}
std::map<WebRtc_Word8, Payload*>::const_iterator it =
@@ -610,7 +613,7 @@
Payload* payload = it->second;
assert(payload);
payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if (payloadType) {
*payloadType = _lastReceivedPayloadType;
@@ -1134,11 +1137,10 @@
void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader) {
bool newSSRC = false;
bool reInitializeDecoder = false;
- WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE];
+ char payloadName[RTP_PAYLOAD_NAME_SIZE];
WebRtc_UWord32 frequency = 90000; // default video freq
WebRtc_UWord8 channels = 1;
WebRtc_UWord32 rate = 0;
- memset(payloadName, 0, sizeof(payloadName));
{
CriticalSectionScoped lock(_criticalSectionRTPReceiver);
@@ -1170,7 +1172,7 @@
Payload* payload = it->second;
assert(payload);
payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
if(payload->audio) {
frequency = payload->typeSpecific.Audio.frequency;
channels = payload->typeSpecific.Audio.channels;
@@ -1214,7 +1216,7 @@
VideoPayload& videoSpecificPayload) {
bool reInitializeDecoder = false;
- WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE];
+ char payloadName[RTP_PAYLOAD_NAME_SIZE];
WebRtc_Word8 payloadType = rtpHeader->header.payloadType;
{
@@ -1262,7 +1264,7 @@
Payload* payload = it->second;
assert(payload);
payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- memcpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
+ strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
_lastReceivedPayloadType = payloadType;
reInitializeDecoder = true;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver.h b/src/modules/rtp_rtcp/source/rtp_receiver.h
index 142c6cf..4a14c25 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -56,31 +56,32 @@
WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
- WebRtc_Word32 RegisterReceivePayload( const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
+ WebRtc_Word32 RegisterReceivePayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payloadType);
WebRtc_Word32 ReceivePayloadType(
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate,
WebRtc_Word8* payloadType) const;
WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType,
- WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ char payloadName[RTP_PAYLOAD_NAME_SIZE],
WebRtc_UWord32* frequency,
WebRtc_UWord8* channels,
WebRtc_UWord32* rate) const;
- WebRtc_Word32 RemotePayload(WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- WebRtc_Word8* payloadType,
- WebRtc_UWord32* frequency,
- WebRtc_UWord8* channels) const;
+ WebRtc_Word32 RemotePayload(char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ WebRtc_Word8* payloadType,
+ WebRtc_UWord32* frequency,
+ WebRtc_UWord8* channels) const;
WebRtc_Word32 IncomingRTPPacket(WebRtcRTPHeader* rtpheader,
const WebRtc_UWord8* incomingRtpPacket,
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index b7eda91..80da505 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -202,92 +202,64 @@
G7221 frame N/A
*/
-ModuleRTPUtility::Payload*
-RTPReceiverAudio::RegisterReceiveAudioPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate)
-{
- WebRtc_Word32 length = (WebRtc_Word32)strlen(payloadName);
- if(length > RTP_PAYLOAD_NAME_SIZE)
- {
- assert(false);
- return NULL;
+ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate) {
+ if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) {
+ _telephoneEventPayloadType = payloadType;
+ }
+ if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) {
+ // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
+ if(frequency == 8000){
+ _cngNBPayloadType = payloadType;
+ } else if(frequency == 16000) {
+ _cngWBPayloadType = payloadType;
+ } else if(frequency == 32000) {
+ _cngSWBPayloadType = payloadType;
+ } else {
+ assert(false);
+ return NULL;
}
-
- if (ModuleRTPUtility::StringCompare(payloadName,"telephone-event",15))
- {
- _telephoneEventPayloadType = payloadType;
+ }
+ WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based
+ if (ModuleRTPUtility::StringCompare(payloadName, "DVI4", 4)) {
+ bitsPerSample = 4;
+ } else if(ModuleRTPUtility::StringCompare(payloadName, "G722", 4)) {
+ if(ModuleRTPUtility::StringCompare(payloadName, "G7221", 5)) {
+ // frame based
+ } else {
+ _G722PayloadType = payloadType;
+ bitsPerSample = 4;
}
- if (ModuleRTPUtility::StringCompare(payloadName,"cn",2))
- {
- // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
- if(frequency == 8000)
- {
- _cngNBPayloadType = payloadType;
-
- } else if(frequency == 16000)
- {
- _cngWBPayloadType = payloadType;
-
- } else if(frequency == 32000)
- {
- _cngSWBPayloadType = payloadType;
- }else
- {
- assert(false);
- return NULL;
- }
- }
- WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based
- if (ModuleRTPUtility::StringCompare(payloadName,"DVI4",4))
- {
- bitsPerSample = 4;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G722",4))
- {
- if(ModuleRTPUtility::StringCompare(payloadName,"G7221",5))
- {
- // frame based
- } else
- {
- _G722PayloadType = payloadType;
- bitsPerSample = 4;
- }
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-40",7))
- {
- bitsPerSample = 5;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-32",7))
- {
- bitsPerSample = 4;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-24",7))
- {
- bitsPerSample = 3;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-16",7))
- {
- bitsPerSample = 2;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"L8",2))
- {
- bitsPerSample = 8;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"L16",3))
- {
- bitsPerSample = 16;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMU",4))
- {
- bitsPerSample = 8;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4))
- {
- bitsPerSample = 8;
- }
-
- ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
- memcpy(payload->name, payloadName, length+1);
- payload->typeSpecific.Audio.frequency = frequency;
- payload->typeSpecific.Audio.channels = channels;
- payload->typeSpecific.Audio.bitsPerSample = bitsPerSample;
- payload->typeSpecific.Audio.rate = rate;
- payload->audio = true;
- return payload;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-40",7)) {
+ bitsPerSample = 5;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-32",7)) {
+ bitsPerSample = 4;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-24",7)) {
+ bitsPerSample = 3;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-16",7)) {
+ bitsPerSample = 2;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"L8",2)) {
+ bitsPerSample = 8;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"L16",3)) {
+ bitsPerSample = 16;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMU",4)) {
+ bitsPerSample = 8;
+ } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4)) {
+ bitsPerSample = 8;
+ }
+ ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ payload->typeSpecific.Audio.frequency = frequency;
+ payload->typeSpecific.Audio.channels = channels;
+ payload->typeSpecific.Audio.bitsPerSample = bitsPerSample;
+ payload->typeSpecific.Audio.rate = rate;
+ payload->audio = true;
+ return payload;
}
// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_audio.h b/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
index e3a096a..89cd062 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -33,11 +33,12 @@
WebRtc_Word32 RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback);
- ModuleRTPUtility::Payload* RegisterReceiveAudioPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
+ ModuleRTPUtility::Payload* RegisterReceiveAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate);
WebRtc_UWord32 AudioFrequency() const;
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 4bc83bb..354ee9f 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -99,38 +99,33 @@
}
}
-ModuleRTPUtility::Payload*
-RTPReceiverVideo::RegisterReceiveVideoPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 maxRate)
-{
- RtpVideoCodecTypes videoType = kRtpNoVideo;
- if (ModuleRTPUtility::StringCompare(payloadName, "VP8",3))
- {
- videoType = kRtpVp8Video;
- } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4))
- {
- videoType = kRtpNoVideo;
- } else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6))
- {
- // store this
- if(_receiveFEC == NULL)
- {
- _receiveFEC = new ReceiverFEC(_id, this);
- }
- _receiveFEC->SetPayloadTypeFEC(payloadType);
- videoType = kRtpFecVideo;
- } else
- {
- return NULL;
+ModuleRTPUtility::Payload* RTPReceiverVideo::RegisterReceiveVideoPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 maxRate) {
+ RtpVideoCodecTypes videoType = kRtpNoVideo;
+ if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) {
+ videoType = kRtpVp8Video;
+ } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
+ videoType = kRtpNoVideo;
+ } else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6)) {
+ // store this
+ if (_receiveFEC == NULL) {
+ _receiveFEC = new ReceiverFEC(_id, this);
}
+ _receiveFEC->SetPayloadTypeFEC(payloadType);
+ videoType = kRtpFecVideo;
+ } else {
+ return NULL;
+ }
+ ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
- ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE);
- payload->typeSpecific.Video.videoCodecType = videoType;
- payload->typeSpecific.Video.maxRate = maxRate;
- payload->audio = false;
- return payload;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ payload->typeSpecific.Video.videoCodecType = videoType;
+ payload->typeSpecific.Video.maxRate = maxRate;
+ payload->audio = false;
+ return payload;
}
void RTPReceiverVideo::ResetOverUseDetector()
@@ -406,56 +401,43 @@
payload->typeSpecific.Video.videoCodecType);
}
-WebRtc_Word32
-RTPReceiverVideo::SetCodecType(const RtpVideoCodecTypes videoType,
- WebRtcRTPHeader* rtpHeader) const
-{
- switch (videoType)
- {
+WebRtc_Word32 RTPReceiverVideo::SetCodecType(const RtpVideoCodecTypes videoType,
+ WebRtcRTPHeader* rtpHeader) const {
+ switch (videoType) {
case kRtpNoVideo:
- rtpHeader->type.Video.codec = kRTPVideoGeneric;
- break;
+ rtpHeader->type.Video.codec = kRTPVideoGeneric;
+ break;
case kRtpVp8Video:
- rtpHeader->type.Video.codec = kRTPVideoVP8;
- break;
+ rtpHeader->type.Video.codec = kRTPVideoVP8;
+ break;
case kRtpFecVideo:
- rtpHeader->type.Video.codec = kRTPVideoFEC;
- break;
- default:
- assert(((void)"ParseCodecSpecific videoType can not be unknown here!", false));
- return -1;
- }
- return 0;
+ rtpHeader->type.Video.codec = kRTPVideoFEC;
+ break;
+ }
+ return 0;
}
-
-WebRtc_Word32
-RTPReceiverVideo::ParseVideoCodecSpecificSwitch(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength,
- const RtpVideoCodecTypes videoType)
-{
- WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
- if(retVal != 0)
- {
- return retVal;
- }
-
- // all receive functions release _criticalSectionReceiverVideo before returning
- switch (videoType)
- {
- case kRtpNoVideo:
- retVal = ReceiveGenericCodec(rtpHeader, payloadData, payloadDataLength);
- break;
- case kRtpVp8Video:
- retVal = ReceiveVp8Codec(rtpHeader, payloadData, payloadDataLength);
- break;
- default:
- _criticalSectionReceiverVideo->Leave();
- assert(((void)"ParseCodecSpecific videoType can not be unknown here!", false));
- return -1;
- }
+WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecificSwitch(
+ WebRtcRTPHeader* rtpHeader,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord16 payloadDataLength,
+ const RtpVideoCodecTypes videoType) {
+ WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
+ if (retVal != 0) {
return retVal;
+ }
+ // All receive functions release _criticalSectionReceiverVideo before
+ // returning.
+ switch (videoType) {
+ case kRtpNoVideo:
+ return ReceiveGenericCodec(rtpHeader, payloadData, payloadDataLength);
+ case kRtpVp8Video:
+ return ReceiveVp8Codec(rtpHeader, payloadData, payloadDataLength);
+ case kRtpFecVideo:
+ break;
+ }
+ _criticalSectionReceiverVideo->Leave();
+ return -1;
}
WebRtc_Word32
diff --git a/src/modules/rtp_rtcp/source/rtp_receiver_video.h b/src/modules/rtp_rtcp/source/rtp_receiver_video.h
index bbfa8a8..feb4092 100644
--- a/src/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/src/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -44,7 +44,7 @@
const WebRtc_UWord16 roundTripTimeMs);
ModuleRTPUtility::Payload* RegisterReceiveVideoPayload(
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 maxRate);
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4fc675d..98495b5 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -1523,46 +1523,38 @@
return _rtcpSender.LastSendReport(lastRTCPTime);
}
-WebRtc_Word32
-ModuleRtpRtcpImpl::SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE])
-{
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetCNAME(%s)", cName);
-
- return _rtcpSender.SetCNAME(cName);
+WebRtc_Word32 ModuleRtpRtcpImpl::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetCNAME(%s)", cName);
+ return _rtcpSender.SetCNAME(cName);
}
-WebRtc_Word32
-ModuleRtpRtcpImpl::CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE])
-{
+WebRtc_Word32 ModuleRtpRtcpImpl::CNAME(char cName[RTCP_CNAME_SIZE]) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "CNAME()");
-
return _rtcpSender.CNAME(cName);
}
-WebRtc_Word32
-ModuleRtpRtcpImpl::AddMixedCNAME(const WebRtc_UWord32 SSRC,
- const WebRtc_Word8 cName[RTCP_CNAME_SIZE])
-{
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "AddMixedCNAME(SSRC:%u)", SSRC);
+WebRtc_Word32 ModuleRtpRtcpImpl::AddMixedCNAME(
+ const WebRtc_UWord32 SSRC,
+ const char cName[RTCP_CNAME_SIZE]) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "AddMixedCNAME(SSRC:%u)", SSRC);
- return _rtcpSender.AddMixedCNAME(SSRC, cName);
+ return _rtcpSender.AddMixedCNAME(SSRC, cName);
}
-WebRtc_Word32
-ModuleRtpRtcpImpl::RemoveMixedCNAME(const WebRtc_UWord32 SSRC)
-{
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoveMixedCNAME(SSRC:%u)", SSRC);
-
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "RemoveMixedCNAME(SSRC:%u)", SSRC);
return _rtcpSender.RemoveMixedCNAME(SSRC);
}
-WebRtc_Word32
-ModuleRtpRtcpImpl::RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
- WebRtc_Word8 cName[RTCP_CNAME_SIZE]) const
-{
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteCNAME(SSRC:%u)", remoteSSRC);
+WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCNAME(
+ const WebRtc_UWord32 remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const {
+ WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
+ "RemoteCNAME(SSRC:%u)", remoteSSRC);
- return _rtcpReceiver.CNAME(remoteSSRC, cName);
+ return _rtcpReceiver.CNAME(remoteSSRC, cName);
}
WebRtc_UWord16 ModuleRtpRtcpImpl::RemoteSequenceNumber() const
diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index d5f6948..b39e9fd 100644
--- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -162,13 +162,6 @@
virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec);
- // set codec name and payload type
- WebRtc_Word32 RegisterSendPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
-
virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
virtual WebRtc_Word8 SendPayloadType() const;
@@ -255,14 +248,14 @@
virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
// Set RTCP CName
- virtual WebRtc_Word32 SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE]);
+ virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
// Get RTCP CName
- virtual WebRtc_Word32 CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE]);
+ virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
// Get remote CName
virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
- WebRtc_Word8 cName[RTCP_CNAME_SIZE]) const;
+ char cName[RTCP_CNAME_SIZE]) const;
// Get remote NTP
virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs,
@@ -271,7 +264,7 @@
WebRtc_UWord32 *RTCPArrivalTimeFrac) const ;
virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
- const WebRtc_Word8 cName[RTCP_CNAME_SIZE]);
+ const char cName[RTCP_CNAME_SIZE]);
virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index f9e3ec1..374ce53 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -272,16 +272,12 @@
//can be called multiple times
WebRtc_Word32 RTPSender::RegisterPayload(
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadNumber,
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) {
- if (!payloadName) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument",
- __FUNCTION__);
- return -1;
- }
+ assert(payloadName);
CriticalSectionScoped cs(_sendCritsect);
if (payloadNumber == _keepAlivePayloadType) {
@@ -298,11 +294,8 @@
assert(payload);
// check if it's the same as we already have
- WebRtc_Word32 payloadNameLength = (WebRtc_Word32)strlen(payloadName);
- WebRtc_Word32 nameLength = (WebRtc_Word32)strlen(payload->name);
- if (payloadNameLength == nameLength &&
- ModuleRTPUtility::StringCompare(payload->name, payloadName,
- nameLength)) {
+ if (ModuleRTPUtility::StringCompare(payload->name, payloadName,
+ RTP_PAYLOAD_NAME_SIZE - 1)) {
if (_audioConfigured && payload->audio &&
payload->typeSpecific.Audio.frequency == frequency &&
(payload->typeSpecific.Audio.rate == rate ||
diff --git a/src/modules/rtp_rtcp/source/rtp_sender.h b/src/modules/rtp_rtcp/source/rtp_sender.h
index 6e1e762..00012e1 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -90,7 +90,7 @@
WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
WebRtc_Word32 RegisterPayload(
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 frequency,
const WebRtc_UWord8 channels,
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 7d83eb6..ee3c414 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -106,56 +106,44 @@
return 0;
}
-WebRtc_Word32
-RTPSenderAudio::RegisterAudioPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate,
- ModuleRTPUtility::Payload*& payload)
-{
- WebRtc_Word32 length = (WebRtc_Word32)strlen(payloadName);
- if(length > RTP_PAYLOAD_NAME_SIZE)
- {
- return -1;
+WebRtc_Word32 RTPSenderAudio::RegisterAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate,
+ ModuleRTPUtility::Payload*& payload) {
+ CriticalSectionScoped cs(_sendAudioCritsect);
+
+ if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) {
+ // we can have multiple CNG payload types
+ if (frequency == 8000) {
+ _cngNBPayloadType = payloadType;
+
+ } else if (frequency == 16000) {
+ _cngWBPayloadType = payloadType;
+
+ } else if (frequency == 32000) {
+ _cngSWBPayloadType = payloadType;
+ } else {
+ return -1;
}
-
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- if (ModuleRTPUtility::StringCompare(payloadName,"cn",2))
- {
- // we can have multiple CNG payload types
- if(frequency == 8000)
- {
- _cngNBPayloadType = payloadType;
-
- } else if(frequency == 16000)
- {
- _cngWBPayloadType = payloadType;
-
- } else if(frequency == 32000)
- {
- _cngSWBPayloadType = payloadType;
- }else
- {
- return -1;
- }
- }
- if (ModuleRTPUtility::StringCompare(payloadName,"telephone-event",15))
- {
- // Don't add it to the list
- // we dont want to allow send with a DTMF payloadtype
- _dtmfPayloadType = payloadType;
- return 0;
- // The default timestamp rate is 8000 Hz, but other rates may be defined.
- }
- payload = new ModuleRTPUtility::Payload;
- payload->typeSpecific.Audio.frequency = frequency;
- payload->typeSpecific.Audio.channels = channels;
- payload->typeSpecific.Audio.rate = rate;
- payload->audio = true;
- memcpy(payload->name, payloadName, length+1);
+ }
+ if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) {
+ // Don't add it to the list
+ // we dont want to allow send with a DTMF payloadtype
+ _dtmfPayloadType = payloadType;
return 0;
+ // The default timestamp rate is 8000 Hz, but other rates may be defined.
+ }
+ payload = new ModuleRTPUtility::Payload;
+ payload->typeSpecific.Audio.frequency = frequency;
+ payload->typeSpecific.Audio.channels = channels;
+ payload->typeSpecific.Audio.rate = rate;
+ payload->audio = true;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
+ strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
+ return 0;
}
bool
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.h b/src/modules/rtp_rtcp/source/rtp_sender_audio.h
index 905458f..5fda2ef 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -32,19 +32,20 @@
WebRtc_Word32 Init();
- WebRtc_Word32 RegisterAudioPayload(const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate,
- ModuleRTPUtility::Payload*& payload);
+ WebRtc_Word32 RegisterAudioPayload(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 frequency,
+ const WebRtc_UWord8 channels,
+ const WebRtc_UWord32 rate,
+ ModuleRTPUtility::Payload*& payload);
WebRtc_Word32 SendAudio(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation);
+ const WebRtc_Word8 payloadType,
+ const WebRtc_UWord32 captureTimeStamp,
+ const WebRtc_UWord8* payloadData,
+ const WebRtc_UWord32 payloadSize,
+ const RTPFragmentationHeader* fragmentation);
// set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
diff --git a/src/modules/rtp_rtcp/source/rtp_sender_video.cc b/src/modules/rtp_rtcp/source/rtp_sender_video.cc
index dc33e4b..79e55d7 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -113,12 +113,11 @@
} else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
videoType = kRtpNoVideo;
} else {
- videoType = kRtpNoVideo;
return -1;
}
payload = new ModuleRTPUtility::Payload;
+ payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0; // Null terminate string.
payload->typeSpecific.Video.videoCodecType = videoType;
payload->typeSpecific.Video.maxRate = maxBitRate;
payload->audio = false;
diff --git a/src/modules/rtp_rtcp/source/rtp_utility.cc b/src/modules/rtp_rtcp/source/rtp_utility.cc
index 228de49..bddcdbd 100644
--- a/src/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/src/modules/rtp_rtcp/source/rtp_utility.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -322,15 +322,15 @@
*/
#if defined(_WIN32)
-bool ModuleRTPUtility::StringCompare(const WebRtc_Word8* str1,
- const WebRtc_Word8* str2,
+bool ModuleRTPUtility::StringCompare(const char* str1,
+ const char* str2,
const WebRtc_UWord32 length)
{
return (_strnicmp(str1, str2, length) == 0)?true: false;
}
#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-bool ModuleRTPUtility::StringCompare(const WebRtc_Word8* str1,
- const WebRtc_Word8* str2,
+bool ModuleRTPUtility::StringCompare(const char* str1,
+ const char* str2,
const WebRtc_UWord32 length)
{
return (strncasecmp(str1, str2, length) == 0)?true: false;
diff --git a/src/modules/rtp_rtcp/source/rtp_utility.h b/src/modules/rtp_rtcp/source/rtp_utility.h
index a0eee1d..8901223 100644
--- a/src/modules/rtp_rtcp/source/rtp_utility.h
+++ b/src/modules/rtp_rtcp/source/rtp_utility.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -55,8 +55,8 @@
};
struct Payload
{
- WebRtc_Word8 name[RTP_PAYLOAD_NAME_SIZE];
- bool audio;
+ char name[RTP_PAYLOAD_NAME_SIZE];
+ bool audio;
PayloadUnion typeSpecific;
};
@@ -89,8 +89,8 @@
uint32_t existingTimestamp,
bool* wrapped);
- bool StringCompare(const WebRtc_Word8* str1,
- const WebRtc_Word8* str2,
+ bool StringCompare(const char* str1,
+ const char* str2,
const WebRtc_UWord32 length);
void AssignUWord32ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value);
diff --git a/src/system_wrappers/interface/trace.h b/src/system_wrappers/interface/trace.h
index 8330f7c..0333e76 100644
--- a/src/system_wrappers/interface/trace.h
+++ b/src/system_wrappers/interface/trace.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -45,11 +45,11 @@
// Sets the file name. If addFileCounter is false the same file will be
// reused when it fills up. If it's true a new file with incremented name
// will be used.
- static WebRtc_Word32 SetTraceFile(const WebRtc_Word8* fileName,
+ static WebRtc_Word32 SetTraceFile(const char* fileName,
const bool addFileCounter = false);
// Returns the name of the file that the trace is currently writing to.
- static WebRtc_Word32 TraceFile(WebRtc_Word8 fileName[1024]);
+ static WebRtc_Word32 TraceFile(char fileName[1024]);
// Registers callback to receive trace messages. TODO (hellner)
// why not use OutStream instead? Why is TraceCallback not defined in this
diff --git a/src/system_wrappers/source/trace_impl.cc b/src/system_wrappers/source/trace_impl.cc
index adf6fcf..b5094c1 100644
--- a/src/system_wrappers/source/trace_impl.cc
+++ b/src/system_wrappers/source/trace_impl.cc
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -94,7 +94,7 @@
for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
{
_messageQueue[m][n] = new
- WebRtc_Word8[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
+ char[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
}
}
}
@@ -350,7 +350,7 @@
return 25;
}
-WebRtc_Word32 TraceImpl::SetTraceFileImpl(const WebRtc_Word8* fileNameUTF8,
+WebRtc_Word32 TraceImpl::SetTraceFileImpl(const char* fileNameUTF8,
const bool addFileCounter)
{
CriticalSectionScoped lock(_critsectInterface);
@@ -364,7 +364,7 @@
{
_fileCountText = 1;
- WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize];
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize];
CreateFileName(fileNameUTF8, fileNameWithCounterUTF8,
_fileCountText);
if(_traceFile.OpenFile(fileNameWithCounterUTF8, false, false,
@@ -385,7 +385,7 @@
}
WebRtc_Word32 TraceImpl::TraceFileImpl(
- WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize])
+ char fileNameUTF8[FileWrapper::kMaxFileNameSize])
{
CriticalSectionScoped lock(_critsectInterface);
return _traceFile.FileName(fileNameUTF8, FileWrapper::kMaxFileNameSize);
@@ -556,8 +556,8 @@
_traceFile.Rewind();
} else
{
- WebRtc_Word8 oldFileName[FileWrapper::kMaxFileNameSize];
- WebRtc_Word8 newFileName[FileWrapper::kMaxFileNameSize];
+ char oldFileName[FileWrapper::kMaxFileNameSize];
+ char newFileName[FileWrapper::kMaxFileNameSize];
// get current name
_traceFile.FileName(oldFileName,
@@ -577,7 +577,7 @@
}
if(_rowCountText == 0)
{
- WebRtc_Word8 message[WEBRTC_TRACE_MAX_MESSAGE_SIZE + 1];
+ char message[WEBRTC_TRACE_MAX_MESSAGE_SIZE + 1];
WebRtc_Word32 length = AddDateTimeInfo(message);
if(length != -1)
{
@@ -669,8 +669,8 @@
}
bool TraceImpl::UpdateFileName(
- const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
- WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
const WebRtc_UWord32 newCount) const
{
WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
@@ -712,8 +712,8 @@
}
bool TraceImpl::CreateFileName(
- const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
- WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
const WebRtc_UWord32 newCount) const
{
WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
@@ -766,7 +766,7 @@
return 0;
}
-WebRtc_Word32 Trace::TraceFile(WebRtc_Word8 fileName[FileWrapper::kMaxFileNameSize])
+WebRtc_Word32 Trace::TraceFile(char fileName[FileWrapper::kMaxFileNameSize])
{
TraceImpl* trace = TraceImpl::GetTrace();
if(trace)
@@ -778,7 +778,7 @@
return -1;
}
-WebRtc_Word32 Trace::SetTraceFile(const WebRtc_Word8* fileName,
+WebRtc_Word32 Trace::SetTraceFile(const char* fileName,
const bool addFileCounter)
{
TraceImpl* trace = TraceImpl::GetTrace();
diff --git a/src/system_wrappers/source/trace_impl.h b/src/system_wrappers/source/trace_impl.h
index 7e99048..2b85813 100644
--- a/src/system_wrappers/source/trace_impl.h
+++ b/src/system_wrappers/source/trace_impl.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ * 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
@@ -49,10 +49,10 @@
static TraceImpl* CreateInstance();
static TraceImpl* GetTrace(const TraceLevel level = kTraceAll);
- WebRtc_Word32 SetTraceFileImpl(const WebRtc_Word8* fileName,
+ WebRtc_Word32 SetTraceFileImpl(const char* fileName,
const bool addFileCounter);
WebRtc_Word32 TraceFileImpl(
- WebRtc_Word8 fileName[FileWrapper::kMaxFileNameSize]);
+ char fileName[FileWrapper::kMaxFileNameSize]);
WebRtc_Word32 SetTraceCallbackImpl(TraceCallback* callback);
@@ -99,13 +99,13 @@
const TraceLevel level);
bool UpdateFileName(
- const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
- WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
const WebRtc_UWord32 newCount) const;
bool CreateFileName(
- const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
- WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
+ const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
+ char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
const WebRtc_UWord32 newCount) const;
void WriteToFile();
@@ -124,7 +124,7 @@
WebRtc_UWord16 _nextFreeIdx[WEBRTC_TRACE_NUM_ARRAY];
TraceLevel _level[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
WebRtc_UWord16 _length[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
- WebRtc_Word8* _messageQueue[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
+ char* _messageQueue[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
WebRtc_UWord8 _activeQueue;
};
} // namespace webrtc
