[rtp_rtcp] Lint errors cleared from rtp_rtcp/test

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  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
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 147 matching lines @@
   uint16_t test_sequence_number;
   uint32_t test_CSRC[webrtc::kRtpCsrcSize];
   SimulatedClock fake_clock;
 };
 
 TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetSSRC(test_ssrc);
   module1->SetStartTimestamp(test_timestamp);
 
   // Test detection at the end of a DTMF tone.
-  //EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
+  // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
 
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
   // Start basic RTP test.
 
   // Send an empty RTP packet.
   // Should fail since we have not registered the payload type.
   EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
                                           96, 0, -1, NULL, 0));
 
@@ skipping 148 matching lines @@
   // Send a DTMF tone using RFC 2833 (4733).
   for (int i = 0; i < 16; i++) {
     EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
   }
   timeStamp += 160;  // Prepare for next packet.
 
   const uint8_t test[9] = "test";
 
   // Send RTP packets for 16 tones a 160 ms  100ms
   // pause between = 2560ms + 1600ms = 4160ms
-  for (;timeStamp <= 250 * 160; timeStamp += 160) {
+  for (; timeStamp <= 250 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
 
-  for (;timeStamp <= 740 * 160; timeStamp += 160) {
+  for (; timeStamp <= 740 * 160; timeStamp += 160) {
     EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
                                            timeStamp, -1, test, 4));
     fake_clock.AdvanceTimeMilliseconds(20);
     module1->Process();
   }
 }
 
 }  // namespace
 }  // namespace webrtc