Index: webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c |
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c |
index 757c0b85c839fd56bc750771a79b3a54db3e3517..248511fec068f8b80adc2e08906e1fbff0404ef8 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c |
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c |
@@ -15,9 +15,9 @@ |
* |
*/ |
+#include "webrtc/base/checks.h" |
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h" |
-#include <assert.h> |
#include <stdio.h> |
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h" |
@@ -455,7 +455,7 @@ int WebRtcIsacfix_EncodeImpl(int16_t *in, |
while (stream_length < MinBytes) |
{ |
- assert(stream_length >= 0); |
+ RTC_DCHECK_GE(stream_length, 0); |
if (stream_length & 0x0001){ |
ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); |
ISACenc_obj->bitstr_obj.stream[stream_length / 2] |= |