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Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]

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The affected functions are

  WebRtcIsacfix_ReadFrameLen
  WebRtcIsacfix_GetNewBitStream
  WebRtcIsacfix_ReadBwIndex

and

  WebRtcIsac_ReadFrameLen
  WebRtcIsac_GetNewBitStream
  WebRtcIsac_ReadBwIndex
  WebRtcIsac_GetRedPayload

BUG=909
R=aluebs@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7429 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-10-13 10:53:42 +00:00
parent 3c16d8bd1c
commit 1172988c79
10 changed files with 75 additions and 59 deletions
Download Patch File Download Diff File Expand all files Collapse all files

33
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -687,8 +687,12 @@ int main(int argc, char* argv[])
/************************* Main Transcoding stream *******************************/
WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC, &jitterInfoTC);
streamLenTransCoding = WebRtcIsac_GetNewBitStream(
ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
(int16_t*)streamDataTransCoding, false);
ISAC_main_inst,
bnIdxTC,
jitterInfoTC,
rateTransCoding,
reinterpret_cast<uint8_t*>(streamDataTransCoding),
false);
if(streamLenTransCoding < 0)
{
fprintf(stderr, "Error in trans-coding\n");
@ -714,9 +718,10 @@ int main(int argc, char* argv[])
return -1;
}
WebRtcIsac_ReadBwIndex((int16_t*)streamDataTransCoding, &indexStream);
if(indexStream != bnIdxTC)
{
WebRtcIsac_ReadBwIndex(reinterpret_cast<const uint8_t*>(
streamDataTransCoding),
&indexStream);
if (indexStream != bnIdxTC) {
fprintf(stderr, "Error in inserting Bandwidth index into transcoding stream.\n");
exit(0);
}
@ -780,14 +785,18 @@ int main(int argc, char* argv[])
// RED.
if(lostFrame)
{
stream_len = WebRtcIsac_GetRedPayload(ISAC_main_inst,
(int16_t*)streamdata);
stream_len = WebRtcIsac_GetRedPayload(
ISAC_main_inst, reinterpret_cast<uint8_t*>(streamdata));
if(doTransCoding)
{
streamLenTransCoding = WebRtcIsac_GetNewBitStream(
ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
(int16_t*)streamDataTransCoding, true);
ISAC_main_inst,
bnIdxTC,
jitterInfoTC,
rateTransCoding,
reinterpret_cast<uint8_t*>(streamDataTransCoding),
true);
if(streamLenTransCoding < 0)
{
fprintf(stderr, "Error in RED trans-coding\n");
@ -872,8 +881,10 @@ int main(int argc, char* argv[])
}
/* Call getFramelen, only used here for function test */
err = WebRtcIsac_ReadFrameLen(ISAC_main_inst,
(int16_t*)streamdata, &FL);
err = WebRtcIsac_ReadFrameLen(
ISAC_main_inst,
reinterpret_cast<const uint8_t*>(streamdata),
&FL);
if(err < 0)
{
/* exit if returned with error */

6
webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@ -287,8 +287,10 @@ int main(int argc, char* argv[])
(uint8_t*)bitStream);
int16_t ggg;
if (streamLen > 0) {
if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx],
(short *) bitStream, &ggg))<0)
if ((WebRtcIsac_ReadFrameLen(
codecInstance[receiverIdx],
reinterpret_cast<const uint8_t*>(bitStream),
&ggg)) < 0)
printf("ERROR\n");
}

3
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -395,7 +395,8 @@ valid values are 8 and 16.\n", sampFreqKHz);
break;
}
rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, (int16_t*)payloadRCU);
rcuStreamLen = WebRtcIsac_GetRedPayload(
ISAC_main_inst, (uint8_t*)payloadRCU);
get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
sampFreqKHz * 1000, sampFreqKHz * 1000);