Replace scoped_ptr with unique_ptr in webrtc/common_audio/

webrtc/modules/audio_processing/transient/transient_suppressor.cc

Index: webrtc/modules/audio_processing/transient/transient_suppressor.cc
diff --git a/webrtc/modules/audio_processing/transient/transient_suppressor.cc b/webrtc/modules/audio_processing/transient/transient_suppressor.cc
index 25909b90080de6742fcc4a69622f9bd2f428f340..46bb574c0be2f40aa4d80252c73c83543955f693 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppressor.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppressor.cc
@@ -17,6 +17,7 @@
 #include <deque>
 #include <set>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/fft4g.h"
 #include "webrtc/common_audio/include/audio_util.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@@ -100,13 +101,13 @@ int TransientSuppressor::Initialize(int sample_rate_hz,
   detector_.reset(new TransientDetector(detection_rate_hz));
   data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000;
   if (data_length_ > analysis_length_) {
-    assert(false);
+    RTC_NOTREACHED();
     return -1;
   }
   buffer_delay_ = analysis_length_ - data_length_;
 
   complex_analysis_length_ = analysis_length_ / 2 + 1;
-  assert(complex_analysis_length_ >= kMaxVoiceBin);
+  RTC_DCHECK_GE(complex_analysis_length_, kMaxVoiceBin);
   num_channels_ = num_channels;
   in_buffer_.reset(new float[analysis_length_ * num_channels_]);
   memset(in_buffer_.get(),