Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

webrtc/common_audio/audio_converter.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/common_audio/audio_converter.h"
 
 #include <cstring>
+#include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/resampler/push_sinc_resampler.h"
 #include "webrtc/system_wrappers/include/scoped_vector.h"
 
 using rtc::checked_cast;
 
 namespace webrtc {
@@ skipping 74 matching lines @@
 
  private:
   ScopedVector<PushSincResampler> resamplers_;
 };
 
 // Apply a vector of converters in serial, in the order given. At least two
 // converters must be provided.
 class CompositionConverter : public AudioConverter {
  public:
   CompositionConverter(ScopedVector<AudioConverter> converters)
-      : converters_(converters.Pass()) {
+      : converters_(std::move(converters)) {
     RTC_CHECK_GE(converters_.size(), 2u);
     // We need an intermediate buffer after every converter.
     for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
       buffers_.push_back(new ChannelBuffer<float>((*it)->dst_frames(),
                                                   (*it)->dst_channels()));
   }
   ~CompositionConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
@@ skipping 21 matching lines @@
                                                        int dst_channels,
                                                        size_t dst_frames) {
   rtc::scoped_ptr<AudioConverter> sp;
   if (src_channels > dst_channels) {
     if (src_frames != dst_frames) {
       ScopedVector<AudioConverter> converters;
       converters.push_back(new DownmixConverter(src_channels, src_frames,
                                                 dst_channels, src_frames));
       converters.push_back(new ResampleConverter(dst_channels, src_frames,
                                                  dst_channels, dst_frames));
-      sp.reset(new CompositionConverter(converters.Pass()));
+      sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
                                     dst_frames));
     }
   } else if (src_channels < dst_channels) {
     if (src_frames != dst_frames) {
       ScopedVector<AudioConverter> converters;
       converters.push_back(new ResampleConverter(src_channels, src_frames,
                                                  src_channels, dst_frames));
       converters.push_back(new UpmixConverter(src_channels, dst_frames,
                                               dst_channels, dst_frames));
-      sp.reset(new CompositionConverter(converters.Pass()));
+      sp.reset(new CompositionConverter(std::move(converters)));
     } else {
       sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
                                   dst_frames));
     }
   } else if (src_frames != dst_frames) {
     sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
                                    dst_frames));
   } else {
     sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
                                dst_frames));
   }
 
-  return sp.Pass();
+  return sp;
 }
 
 // For CompositionConverter.
 AudioConverter::AudioConverter()
     : src_channels_(0),
       src_frames_(0),
       dst_channels_(0),
       dst_frames_(0) {}
 
 AudioConverter::AudioConverter(int src_channels, size_t src_frames,
                                int dst_channels, size_t dst_frames)
     : src_channels_(src_channels),
       src_frames_(src_frames),
       dst_channels_(dst_channels),
       dst_frames_(dst_frames) {
   RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
             src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
   RTC_CHECK_EQ(src_size, src_channels() * src_frames());
   RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc