Webrtc deps roll.

content/browser/renderer_host/p2p/socket_host_unittest.cc

Index: content/browser/renderer_host/p2p/socket_host_unittest.cc
diff --git a/content/browser/renderer_host/p2p/socket_host_unittest.cc b/content/browser/renderer_host/p2p/socket_host_unittest.cc
index 1404ced5bc69bf8b1234e42e47b09d6804b071c2..fc96e4a158a7af3c43e665a3bd7f11c138dcd6e1 100644
--- a/content/browser/renderer_host/p2p/socket_host_unittest.cc
+++ b/content/browser/renderer_host/p2p/socket_host_unittest.cc
@@ -297,7 +297,7 @@ TEST(P2PSocketHostTest, TestUpdateAbsSendTimeExtensionInTurnSendIndication) {
 // without HMAC value in the packet.
 TEST(P2PSocketHostTest, TestApplyPacketOptionsWithDefaultValues) {
   unsigned char fake_tag[4] = { 0xba, 0xdd, 0xba, 0xdd };
-  talk_base::PacketOptions options;
+  rtc::PacketOptions options;
   std::vector<char> rtp_packet;
   rtp_packet.resize(sizeof(kRtpMsgWithAbsSendTimeExtension) + 4);  // tag length
   memcpy(&rtp_packet[0], kRtpMsgWithAbsSendTimeExtension,
@@ -317,7 +317,7 @@ TEST(P2PSocketHostTest, TestApplyPacketOptionsWithDefaultValues) {
 
 // Veirfy HMAC is updated when packet option parameters are set.
 TEST(P2PSocketHostTest, TestApplyPacketOptionsWithAuthParams) {
-  talk_base::PacketOptions options;
+  rtc::PacketOptions options;
   options.packet_time_params.srtp_auth_key.resize(20);
   options.packet_time_params.srtp_auth_key.assign(
       kTestKey, kTestKey + sizeof(kTestKey));
@@ -348,7 +348,7 @@ TEST(P2PSocketHostTest, TestUpdateAbsSendTimeExtensionInRtpPacket) {
 
 // Verify we update both AbsSendTime extension header and HMAC.
 TEST(P2PSocketHostTest, TestApplyPacketOptionsWithAuthParamsAndAbsSendTime) {
-  talk_base::PacketOptions options;
+  rtc::PacketOptions options;
   options.packet_time_params.srtp_auth_key.resize(20);
   options.packet_time_params.srtp_auth_key.assign(
       kTestKey, kTestKey + sizeof(kTestKey));