| 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. */
|
|
|