Let PacketSource::NextPacket() return an std::unique_ptr

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 331 matching lines @@
       if (rtp_header.header.payloadType != 104)
 #endif
       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_.reset(rtp_source_->NextPacket());
+    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) ||
               (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
               (out_frame_.samples_per_channel_ == kBlockSize48kHz));
@@ skipping 17 matching lines @@
   ResultSink output(ref_out_file);
 
   std::string stat_out_file =
       gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
   ResultSink network_stats(stat_out_file);
 
   std::string rtcp_out_file =
       gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : "";
   ResultSink rtcp_stats(rtcp_out_file);
 
-  packet_.reset(rtp_source_->NextPacket());
+  packet_ = rtp_source_->NextPacket();
   int i = 0;
   while (packet_) {
     std::ostringstream ss;
     ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     ASSERT_NO_FATAL_FAILURE(Process());
     ASSERT_NO_FATAL_FAILURE(output.AddResult(
         out_frame_.data_, out_frame_.samples_per_channel_));
 
     // Query the network statistics API once per second
@@ skipping 1368 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc