commit 9e2be5f292c574ca88e145ebb15b0ffd63f9b87c
author kwiberg <kwiberg@webrtc.org> Wed Sep 14 05:23:22 2016 -0700
committer Commit bot <commit-bot@chromium.org> Wed Sep 14 12:23:29 2016 +0000
tree 32de1167cfc7ced9fe4c5321e5e7271e780e1972
parent 3a7f35b1c4abb7d08a41d79098e7f54a24a70783

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

Review-Url: https://codereview.webrtc.org/2320053003
Cr-Commit-Position: refs/heads/master@{#14211}

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 5478456..9b7b953 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"
@@ -84,7 +83,7 @@
       return true;
   }
 
-  assert(false);
+  RTC_NOTREACHED();
   return false;
 }
 
@@ -693,8 +692,8 @@
         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()) {
@@ -1010,8 +1009,8 @@
   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()) {