Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 222ce4b..dc27e96 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -858,7 +858,7 @@
return 0;
}
WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- unsigned int length)) {
+ size_t length)) {
WEBRTC_CHECK_CHANNEL(channel);
if (!channels_[channel]->external_transport) return -1;
channels_[channel]->packets.push_back(
@@ -866,7 +866,7 @@
return 0;
}
WEBRTC_FUNC(ReceivedRTPPacket, (int channel, const void* data,
- unsigned int length,
+ size_t length,
const webrtc::PacketTime& packet_time)) {
WEBRTC_CHECK_CHANNEL(channel);
if (ReceivedRTPPacket(channel, data, length) == -1) {
@@ -877,7 +877,7 @@
}
WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
- unsigned int length));
+ size_t length));
// webrtc::VoERTP_RTCP
WEBRTC_STUB(RegisterRTPObserver, (int channel,