Index: webrtc/modules/audio_processing/test/audio_file_processor.h |
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.h b/webrtc/modules/audio_processing/test/audio_file_processor.h |
index 3ba20340d275f6ff5202284a0095306c07e5bf11..f3db86dc848522f3afb96e37b08e20290a4cbb09 100644 |
--- a/webrtc/modules/audio_processing/test/audio_file_processor.h |
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.h |
@@ -13,9 +13,9 @@ |
#include <algorithm> |
#include <limits> |
+#include <memory> |
#include <vector> |
-#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/common_audio/channel_buffer.h" |
#include "webrtc/common_audio/wav_file.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
@@ -84,16 +84,16 @@ class AudioFileProcessor { |
class WavFileProcessor final : public AudioFileProcessor { |
public: |
// Takes ownership of all parameters. |
- WavFileProcessor(rtc::scoped_ptr<AudioProcessing> ap, |
- rtc::scoped_ptr<WavReader> in_file, |
- rtc::scoped_ptr<WavWriter> out_file); |
+ WavFileProcessor(std::unique_ptr<AudioProcessing> ap, |
+ std::unique_ptr<WavReader> in_file, |
+ std::unique_ptr<WavWriter> out_file); |
virtual ~WavFileProcessor() {} |
// Processes one chunk from the WAV input and writes to the WAV output. |
bool ProcessChunk() override; |
private: |
- rtc::scoped_ptr<AudioProcessing> ap_; |
+ std::unique_ptr<AudioProcessing> ap_; |
ChannelBuffer<float> in_buf_; |
ChannelBuffer<float> out_buf_; |
@@ -107,9 +107,9 @@ class WavFileProcessor final : public AudioFileProcessor { |
class AecDumpFileProcessor final : public AudioFileProcessor { |
public: |
// Takes ownership of all parameters. |
- AecDumpFileProcessor(rtc::scoped_ptr<AudioProcessing> ap, |
+ AecDumpFileProcessor(std::unique_ptr<AudioProcessing> ap, |
FILE* dump_file, |
- rtc::scoped_ptr<WavWriter> out_file); |
+ std::unique_ptr<WavWriter> out_file); |
virtual ~AecDumpFileProcessor(); |
@@ -122,11 +122,11 @@ class AecDumpFileProcessor final : public AudioFileProcessor { |
void HandleMessage(const webrtc::audioproc::Stream& msg); |
void HandleMessage(const webrtc::audioproc::ReverseStream& msg); |
- rtc::scoped_ptr<AudioProcessing> ap_; |
+ std::unique_ptr<AudioProcessing> ap_; |
FILE* dump_file_; |
- rtc::scoped_ptr<ChannelBuffer<float>> in_buf_; |
- rtc::scoped_ptr<ChannelBuffer<float>> reverse_buf_; |
+ std::unique_ptr<ChannelBuffer<float>> in_buf_; |
+ std::unique_ptr<ChannelBuffer<float>> reverse_buf_; |
ChannelBuffer<float> out_buf_; |
StreamConfig input_config_; |
StreamConfig reverse_config_; |