Revert of Change NetEq::InsertPacket to take an RTPHeader

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 306 matching lines @@
 void NetEqDecodingTest::Process() {
   // Check if time to receive.
   while (packet_ && sim_clock_ >= packet_->time_ms()) {
     if (packet_->payload_length_bytes() > 0) {
       WebRtcRTPHeader rtp_header;
       packet_->ConvertHeader(&rtp_header);
 #ifndef WEBRTC_CODEC_ISAC
       // Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
       if (rtp_header.header.payloadType != 104)
 #endif
-        ASSERT_EQ(0,
-                  neteq_->InsertPacket(
-                      rtp_header.header,
-                      rtc::ArrayView<const uint8_t>(
-                          packet_->payload(), packet_->payload_length_bytes()),
-                      static_cast<uint32_t>(packet_->time_ms() *
-                                            (output_sample_rate_ / 1000))));
+      ASSERT_EQ(0, neteq_->InsertPacket(
+                       rtp_header,
+                       rtc::ArrayView<const uint8_t>(
+                           packet_->payload(), packet_->payload_length_bytes()),
+                       static_cast<uint32_t>(packet_->time_ms() *
+                                             (output_sample_rate_ / 1000))));
     }
     // Get next packet.
     packet_ = rtp_source_->NextPacket();
   }
 
   // Get audio from NetEq.
   bool muted;
   ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_FALSE(muted);
   ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
@@ skipping 176 matching lines @@
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   for (size_t i = 0; i < num_frames; ++i) {
     const uint8_t payload[kPayloadBytes] = {0};
     WebRtcRTPHeader rtp_info;
     rtp_info.header.sequenceNumber = i;
     rtp_info.header.timestamp = i * kSamples;
     rtp_info.header.ssrc = 0x1234;  // Just an arbitrary SSRC.
     rtp_info.header.payloadType = 94;  // PCM16b WB codec.
     rtp_info.header.markerBit = 0;
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   }
   // Pull out all data.
   for (size_t i = 0; i < num_frames; ++i) {
     bool muted;
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics stats;
   EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
@@ skipping 20 matching lines @@
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   while (frame_index < kNumFrames) {
     // Insert one packet each time, except every 10th time where we insert two
     // packets at once. This will create a negative clock-drift of approx. 10%.
     int num_packets = (frame_index % 10 == 0 ? 2 : 1);
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     bool muted;
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
   EXPECT_EQ(-103192, network_stats.clockdrift_ppm);
 }
 
 TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
   const int kNumFrames = 5000;  // Needed for convergence.
   int frame_index = 0;
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   for (int i = 0; i < kNumFrames; ++i) {
     // Insert one packet each time, except every 10th time where we don't insert
     // any packet. This will create a positive clock-drift of approx. 11%.
     int num_packets = (i % 10 == 9 ? 0 : 1);
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     bool muted;
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics network_stats;
@@ skipping 17 matching lines @@
 
   // Insert speech for 5 seconds.
   const int kSpeechDurationMs = 5000;
   for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one 30 ms speech frame.
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
     }
     // Pull out data once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
   rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
   ASSERT_TRUE(playout_timestamp);
   int32_t delay_before = timestamp - *playout_timestamp;
 
   // Insert CNG for 1 minute (= 60000 ms).
   const int kCngPeriodMs = 100;
   const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
   const int kCngDurationMs = 60000;
   for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one CNG frame each 100 ms.
       uint8_t payload[kPayloadBytes];
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
       ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_info.header,
+                       rtp_info,
                        rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
     }
     // Pull out data once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
@@ skipping 22 matching lines @@
         ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
         EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
         t_ms += 10;
       }
       // Insert one CNG frame each 100 ms.
       uint8_t payload[kPayloadBytes];
       size_t payload_len;
       WebRtcRTPHeader rtp_info;
       PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
       ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_info.header,
+                       rtp_info,
                        rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
       ++seq_no;
       timestamp += kCngPeriodSamples;
       next_input_time_ms += kCngPeriodMs * drift_factor;
     }
   }
 
   // Insert speech again until output type is speech.
   double speech_restart_time_ms = t_ms;
   while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one 30 ms speech frame.
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++seq_no;
       timestamp += kSamples;
       next_input_time_ms += kFrameSizeMs * drift_factor;
     }
     // Pull out data once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
     // Increase clock.
     t_ms += 10;
   }
@@ skipping 91 matching lines @@
                         kDelayToleranceMs,
                         kMaxTimeToSpeechMs);
 }
 
 TEST_F(NetEqDecodingTest, UnknownPayloadType) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 1;  // Not registered as a decoder.
-  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info.header, payload, 0));
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
 #define MAYBE_DecoderError DecoderError
 #else
 #define MAYBE_DecoderError DISABLED_DecoderError
 #endif
 
 TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
     out_frame_.data_[i] = 1;
   }
   bool muted;
   EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_FALSE(muted);
   // Verify that there is a decoder error to check.
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
@@ skipping 86 matching lines @@
 
     uint32_t receive_timestamp = 0;
     bool muted;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
       auto block = input.GetNextBlock();
       ASSERT_EQ(expected_samples_per_channel, block.size());
       size_t enc_len_bytes =
           WebRtcPcm16b_Encode(block.data(), block.size(), payload);
       ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
 
-      ASSERT_EQ(0, neteq_->InsertPacket(
-                       rtp_info.header,
-                       rtc::ArrayView<const uint8_t>(payload, enc_len_bytes),
-                       receive_timestamp));
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, enc_len_bytes),
+                                        receive_timestamp));
       output.Reset();
       ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
       ASSERT_EQ(1u, output.num_channels_);
       ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
       ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
 
       // Next packet.
       rtp_info.header.timestamp += expected_samples_per_channel;
       rtp_info.header.sequenceNumber++;
       receive_timestamp += expected_samples_per_channel;
@@ skipping 113 matching lines @@
   bool seq_no_wrapped = false;
   for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one 30 ms speech frame.
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(seq_no, timestamp, &rtp_info);
       if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
         // This sequence number was not in the set to drop. Insert it.
-        ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload,
-                                          receive_timestamp));
+        ASSERT_EQ(0,
+                  neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
         ++packets_inserted;
       }
       NetEqNetworkStatistics network_stats;
       ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
 
       // Due to internal NetEq logic, preferred buffer-size is about 4 times the
       // packet size for first few packets. Therefore we refrain from checking
       // the criteria.
       if (packets_inserted > 4) {
         // Expect preferred and actual buffer size to be no more than 2 frames.
@@ skipping 67 matching lines @@
 
   const int algorithmic_delay_samples = std::max(
       algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
   // Insert three speech packets. Three are needed to get the frame length
   // correct.
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   bool muted;
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+    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_);
   }
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 
   // Insert same CNG packet twice.
   const int kCngPeriodMs = 100;
   const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
   size_t payload_len;
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   // This is the first time this CNG packet is inserted.
-  ASSERT_EQ(0, neteq_->InsertPacket(
-                   rtp_info.header,
-                   rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+  ASSERT_EQ(
+      0, neteq_->InsertPacket(
+             rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio once and make sure CNG is played.
   ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
   EXPECT_FALSE(PlayoutTimestamp());  // Returns empty value during CNG.
   EXPECT_EQ(timestamp - algorithmic_delay_samples,
             out_frame_.timestamp_ + out_frame_.samples_per_channel_);
 
   // Insert the same CNG packet again. Note that at this point it is old, since
   // we have already decoded the first copy of it.
-  ASSERT_EQ(0, neteq_->InsertPacket(
-                   rtp_info.header,
-                   rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+  ASSERT_EQ(
+      0, neteq_->InsertPacket(
+             rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
 
   // Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
   // we have already pulled out CNG once.
   for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
     EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
     EXPECT_FALSE(PlayoutTimestamp());  // Returns empty value during CNG.
     EXPECT_EQ(timestamp - algorithmic_delay_samples,
               out_frame_.timestamp_ + out_frame_.samples_per_channel_);
   }
 
   // Insert speech again.
   ++seq_no;
   timestamp += kCngPeriodSamples;
   PopulateRtpInfo(seq_no, timestamp, &rtp_info);
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
 
   // Pull audio once and verify that the output is speech again.
   ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
   rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
   ASSERT_TRUE(playout_timestamp);
   EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
             *playout_timestamp);
 }
@@ skipping 12 matching lines @@
   const int kSamples = kFrameSizeMs * kSampleRateKhz;
   const int kPayloadBytes = kSamples * 2;
   const int kCngPeriodMs = 100;
   const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
   size_t payload_len;
 
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
 
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
-  ASSERT_EQ(NetEq::kOK,
-            neteq_->InsertPacket(
-                rtp_info.header,
-                rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+  ASSERT_EQ(
+      NetEq::kOK,
+      neteq_->InsertPacket(
+          rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
   ++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.
   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.header, payload, 0));
+    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;
   }
 
  protected:
   static constexpr size_t kSamples = 10 * 16;
   static constexpr size_t kPayloadBytes = kSamples * 2;
 
   void InsertPacket(uint32_t rtp_timestamp) {
     uint8_t payload[kPayloadBytes] = {0};
     WebRtcRTPHeader rtp_info;
     PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
-    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   }
 
   void InsertCngPacket(uint32_t rtp_timestamp) {
     uint8_t payload[kPayloadBytes] = {0};
     WebRtcRTPHeader rtp_info;
     size_t payload_len;
     PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
-    EXPECT_EQ(NetEq::kOK,
-              neteq_->InsertPacket(
-                  rtp_info.header,
-                  rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+    EXPECT_EQ(
+        NetEq::kOK,
+        neteq_->InsertPacket(
+            rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
   }
 
   bool GetAudioReturnMuted() {
     bool muted;
     EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     return muted;
   }
 
   void GetAudioUntilMuted() {
     while (!GetAudioReturnMuted()) {
@@ skipping 204 matching lines @@
   ASSERT_FALSE(config_.enable_muted_state);
   config2_.enable_muted_state = true;
   CreateSecondInstance();
 
   // Insert one speech packet into both NetEqs.
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
-  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info.header, payload, 0));
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
 
   AudioFrame out_frame1, out_frame2;
   bool muted;
   for (int i = 0; i < 1000; ++i) {
     std::ostringstream ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
     EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
     EXPECT_FALSE(muted);
     EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_TRUE(muted);
 
   // Insert new data. Timestamp is corrected for the time elapsed since the last
   // packet.
   PopulateRtpInfo(0, kSamples * 1000, &rtp_info);
-  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
-  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info.header, payload, 0));
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
 
   int counter = 0;
   while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
     ASSERT_LT(counter++, 1000) << "Test timed out";
     std::ostringstream ss;
     ss << "counter = " << counter;
     SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
     EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
     EXPECT_FALSE(muted);
     EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc