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Unified Diff: webrtc/p2p/base/asyncstuntcpsocket.cc

Issue 2912523003: Fix issue with send-side bandwidth estimation over TURN TCP connections. (Closed)
Patch Set: Created 3 years, 7 months ago
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Index: webrtc/p2p/base/asyncstuntcpsocket.cc
diff --git a/webrtc/p2p/base/asyncstuntcpsocket.cc b/webrtc/p2p/base/asyncstuntcpsocket.cc
index 296a9d413a12c3b48ffe53a2c98c9dcbcf13e82b..3369bc8b48dc3e9f37f4284ec6e6895bb05f4371 100644
--- a/webrtc/p2p/base/asyncstuntcpsocket.cc
+++ b/webrtc/p2p/base/asyncstuntcpsocket.cc
@@ -79,6 +79,9 @@ int AsyncStunTCPSocket::Send(const void *pv, size_t cb,
return res;
}
+ rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
juberti1 2017/05/27 23:17:21 This signal should probably only be fired when the
holmer 2017/06/02 07:08:00 Yes, I agree. If I correctly understand how the TC
Taylor Brandstetter 2017/06/02 17:42:43 I guess I don't see how filling a buffer in usersp
+ SignalSentPacket(this, sent_packet);
+
// We claim to have sent the whole thing, even if we only sent partial
return static_cast<int>(cb);
}
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