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

webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

 /*
  *  Copyright (c) 2014 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 136 matching lines @@
     AudioDecoder* external_decoder,
     int sample_rate_hz,
     int num_channels,
     const std::string& name) {
   return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
                                             sample_rate_hz, num_channels, name);
 }
 
 void AcmReceiveTestOldApi::Run() {
   for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
-       packet.reset(packet_source_->NextPacket())) {
+       packet = packet_source_->NextPacket()) {
     // Pull audio until time to insert packet.
     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
       AudioFrame output_frame;
       bool muted;
       EXPECT_EQ(0,
                 acm_->PlayoutData10Ms(output_freq_hz_, &output_frame, &muted));
       ASSERT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
       ASSERT_FALSE(muted);
       const size_t samples_per_block =
           static_cast<size_t>(output_freq_hz_ * 10 / 1000);
@@ skipping 50 matching lines @@
   if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
     output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
                           ? output_freq_hz_2_
                           : output_freq_hz_1_;
     last_toggle_time_ms_ = clock_.TimeInMilliseconds();
   }
 }
 
 }  // namespace test
 }  // namespace webrtc