NetEq: only update current_rtp_payload_type_ when validated

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 1503 matching lines @@
   ++seq_no;
   timestamp += kCngPeriodSamples;
 
   // Pull audio once and make sure CNG is played.
   bool muted;
   ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
 
   // Insert some speech packets.
-  for (int i = 0; i < 3; ++i) {
+  const uint32_t first_speech_timestamp = timestamp;
+  int timeout_counter = 0;
+  do {
+    ASSERT_LT(timeout_counter++, 20) << "Test timed out";
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
-  }
+  } while (!IsNewerTimestamp(out_frame_.timestamp_, first_speech_timestamp));
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
 
 class NetEqDecodingTestWithMutedState : public NetEqDecodingTest {
  public:
   NetEqDecodingTestWithMutedState() : NetEqDecodingTest() {
     config_.enable_muted_state = true;
   }
 
@@ skipping 235 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc