Replace scoped_ptr with unique_ptr in webrtc/modules/audio_processing/test/
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1694423002
Cr-Commit-Position: refs/heads/master@{#11653}
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 0bd7012..0d50b0c 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -33,7 +33,7 @@
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}
-ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
+ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
: file_(std::move(file)) {}
bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
@@ -50,7 +50,7 @@
return true;
}
-ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
: file_(std::move(file)) {}
void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
@@ -80,7 +80,7 @@
WavWriter* wav_file,
RawFile* raw_file) {
size_t length = num_channels * samples_per_channel;
- rtc::scoped_ptr<float[]> buffer(new float[length]);
+ std::unique_ptr<float[]> buffer(new float[length]);
Interleave(data, samples_per_channel, num_channels, buffer.get());
if (raw_file) {
raw_file->WriteSamples(buffer.get(), length);