Index: webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
index db78e6de2e351ab12b7b8c0a16c47e048e5d5118..3492bfae004cc3de795913ccd16c9cad7c525e88 100644 |
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c |
@@ -504,7 +504,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, |
int16_t streamLenUB = 0; |
int16_t streamLen = 0; |
int16_t k = 0; |
- int garbageLen = 0; |
+ uint8_t garbageLen = 0; |
int32_t bottleneck = 0; |
int16_t bottleneckIdx = 0; |
int16_t jitterInfo = 0; |
@@ -645,7 +645,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, |
memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB); |
streamLen = streamLenLB; |
if (streamLenUB > 0) { |
- encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8; |
+ encoded[streamLenLB] = (uint8_t)(streamLenUB + 1 + LEN_CHECK_SUM_WORD8); |
memcpy(&encoded[streamLenLB + 1], |
instUB->ISACencUB_obj.bitstr_obj.stream, |
streamLenUB); |
@@ -703,7 +703,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, |
} |
minBytes = (minBytes > limit) ? limit : minBytes; |
- garbageLen = (minBytes > streamLen) ? (minBytes - streamLen) : 0; |
+ garbageLen = (minBytes > streamLen) ? (uint8_t)(minBytes - streamLen) : 0; |
/* Save data for creation of multiple bit-streams. */ |
/* If bit-stream too short then add garbage at the end. */ |