Replace calls to assert() with RTC_DCHECK_*() in .c code

webrtc/common_audio/signal_processing/splitting_filter.c

Index: webrtc/common_audio/signal_processing/splitting_filter.c
diff --git a/webrtc/common_audio/signal_processing/splitting_filter.c b/webrtc/common_audio/signal_processing/splitting_filter.c
index ba6e77dc6f9fde0a01ed4e1e23df39bc30f357a0..1400623a726591f66d67183eaaa89f97b91309eb 100644
--- a/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/webrtc/common_audio/signal_processing/splitting_filter.c
@@ -13,10 +13,9 @@
  *
  */
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 
-#include <assert.h>
-
 // Maximum number of samples in a low/high-band frame.
 enum
 {
@@ -136,8 +135,8 @@ void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
     const size_t band_length = in_data_length / 2;
-    assert(in_data_length % 2 == 0);
-    assert(band_length <= kMaxBandFrameLength);
+    RTC_DCHECK_EQ(0, in_data_length % 2);
+    RTC_DCHECK_LE(band_length, kMaxBandFrameLength);
 
     // Split even and odd samples. Also shift them to Q10.
     for (i = 0, k = 0; i < band_length; i++, k += 2)
@@ -175,7 +174,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
     int32_t filter2[kMaxBandFrameLength];
     size_t i;
     int16_t k;
-    assert(band_length <= kMaxBandFrameLength);
+    RTC_DCHECK_LE(band_length, kMaxBandFrameLength);
 
     // Obtain the sum and difference channels out of upper and lower-band channels.
     // Also shift to Q10 domain.