| Index: webrtc/modules/audio_processing/test/audio_file_processor.cc
|
| diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
|
| index ca244d550fed05248c5f650d93c09afb36dadc25..4c773566c4525f6f1da6c19c765c242c8954eac6 100644
|
| --- a/webrtc/modules/audio_processing/test/audio_file_processor.cc
|
| +++ b/webrtc/modules/audio_processing/test/audio_file_processor.cc
|
| @@ -11,6 +11,7 @@
|
| #include "webrtc/modules/audio_processing/test/audio_file_processor.h"
|
|
|
| #include <algorithm>
|
| +#include <utility>
|
|
|
| #include "webrtc/base/checks.h"
|
| #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
|
| @@ -43,13 +44,13 @@ ChannelBuffer<float> GetChannelBuffer(const WavFile& file) {
|
| WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
|
| scoped_ptr<WavReader> in_file,
|
| scoped_ptr<WavWriter> out_file)
|
| - : ap_(ap.Pass()),
|
| + : ap_(std::move(ap)),
|
| in_buf_(GetChannelBuffer(*in_file)),
|
| out_buf_(GetChannelBuffer(*out_file)),
|
| input_config_(GetStreamConfig(*in_file)),
|
| output_config_(GetStreamConfig(*out_file)),
|
| - buffer_reader_(in_file.Pass()),
|
| - buffer_writer_(out_file.Pass()) {}
|
| + buffer_reader_(std::move(in_file)),
|
| + buffer_writer_(std::move(out_file)) {}
|
|
|
| bool WavFileProcessor::ProcessChunk() {
|
| if (!buffer_reader_.Read(&in_buf_)) {
|
| @@ -68,11 +69,11 @@ bool WavFileProcessor::ProcessChunk() {
|
| AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,
|
| FILE* dump_file,
|
| scoped_ptr<WavWriter> out_file)
|
| - : ap_(ap.Pass()),
|
| + : ap_(std::move(ap)),
|
| dump_file_(dump_file),
|
| out_buf_(GetChannelBuffer(*out_file)),
|
| output_config_(GetStreamConfig(*out_file)),
|
| - buffer_writer_(out_file.Pass()) {
|
| + buffer_writer_(std::move(out_file)) {
|
| RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
|
| }
|
|
|
|
|