webrtc/modules/audio_processing: Use RTC_DCHECK() instead of assert()

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: webrtc/modules/audio_processing/audio_processing_impl.cc
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 72f154a66d94c84034e078897ccf40e39a4d1a2b..2e0c52b8154de34f854384a6e8ad60704082b2a2 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -10,7 +10,6 @@
 
 #include "webrtc/modules/audio_processing/audio_processing_impl.h"
 
-#include <assert.h>
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
@@ -92,7 +91,7 @@ static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
       return true;
   }
 
-  assert(false);
+  RTC_NOTREACHED();
   return false;
 }
 
@@ -698,8 +697,8 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
         MaybeInitializeCapture(processing_config, reinitialization_required));
   }
   rtc::CritScope cs_capture(&crit_capture_);
-  assert(processing_config.input_stream().num_frames() ==
-         formats_.api_format.input_stream().num_frames());
+  RTC_DCHECK_EQ(processing_config.input_stream().num_frames(),
+                formats_.api_format.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {
@@ -1015,8 +1014,8 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
   processing_config.reverse_output_stream() = reverse_output_config;
 
   RETURN_ON_ERR(MaybeInitializeRender(processing_config));
-  assert(reverse_input_config.num_frames() ==
-         formats_.api_format.reverse_input_stream().num_frames());
+  RTC_DCHECK_EQ(reverse_input_config.num_frames(),
+                formats_.api_format.reverse_input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->is_open()) {