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."; |
} |