Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/

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.
  */
 
 #include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
 
+#include <memory>

[hlundin-webrtc, 2016/02/15 11:18:15]

You do not have to repeat the same include in both .h and .cc files.

https://google.github.io/styleguide/cppguide.html#Names_and_Order_of_Includes
"any includes present in the related header do not need to be included again in
the related cc (i.e., foo.cc can rely on foo.h's includes)."

[kwiberg-webrtc, 2016/02/15 11:42:50]

Yeah, the script I whipped up for automating this simply ensures that we
#include <memory> in all files that mention unique_ptr, without taking
subtleties like this one into account. I'll see if I can make it handle this
case without spending too much time on it.

+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
@@ skipping 118 matching lines @@
     int rtp_payload_type,
     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 (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
+  for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
        packet.reset(packet_source_->NextPacket())) {
     // Pull audio until time to insert packet.
     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
       AudioFrame output_frame;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
       EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
       const size_t samples_per_block =
           static_cast<size_t>(output_freq_hz_ * 10 / 1000);
       EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
       if (exptected_output_channels_ != kArbitraryChannels) {
@@ skipping 48 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