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| author | kwiberg@webrtc.org <kwiberg@webrtc.org> | 2014-10-13 13:29:04 +0000 |
|---|---|---|
| committer | kwiberg@webrtc.org <kwiberg@webrtc.org> | 2014-10-13 13:29:04 +0000 |
| commit | 88f9c81aca9e9c7ef48eaa4e837ec4fa17572dab (patch) | |
| tree | 50d1e019eb1b63d4b4ab7da6cb3e116b9b40a1ed | |
| parent | cc05752091dd843c21de2f79db7616d9fbe0dfd5 (diff) | |
| download | webrtc-88f9c81aca9e9c7ef48eaa4e837ec4fa17572dab.tar.gz | |
iSAC tests: Type buffers as uint8_t[] to avoid casts
The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.
R=bjornv@webrtc.org, henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7433 4adac7df-926f-26a2-2b94-8c16560cd09d
| -rw-r--r-- | modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc | 15 | ||||
| -rw-r--r-- | modules/audio_coding/codecs/isac/main/test/simpleKenny.c | 14 |
2 files changed, 14 insertions, 15 deletions
diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc index b9fec4b7..567ec85b 100644 --- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc +++ b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc @@ -93,7 +93,7 @@ int main(int argc, char* argv[]) //FILE logFile; bool doTransCoding = false; int32_t rateTransCoding = 0; - uint16_t streamDataTransCoding[600]; + uint8_t streamDataTransCoding[1200]; int16_t streamLenTransCoding = 0; FILE* transCodingFile = NULL; FILE* transcodingBitstream = NULL; @@ -691,7 +691,7 @@ int main(int argc, char* argv[]) bnIdxTC, jitterInfoTC, rateTransCoding, - reinterpret_cast<uint8_t*>(streamDataTransCoding), + streamDataTransCoding, false); if(streamLenTransCoding < 0) { @@ -710,7 +710,7 @@ int main(int argc, char* argv[]) return -1; } - if (fwrite((uint8_t*)streamDataTransCoding, + if (fwrite(streamDataTransCoding, sizeof(uint8_t), streamLenTransCoding, transcodingBitstream) != @@ -718,8 +718,7 @@ int main(int argc, char* argv[]) return -1; } - WebRtcIsac_ReadBwIndex(reinterpret_cast<const uint8_t*>( - streamDataTransCoding), + WebRtcIsac_ReadBwIndex(streamDataTransCoding, &indexStream); if (indexStream != bnIdxTC) { fprintf(stderr, "Error in inserting Bandwidth index into transcoding stream.\n"); @@ -795,7 +794,7 @@ int main(int argc, char* argv[]) bnIdxTC, jitterInfoTC, rateTransCoding, - reinterpret_cast<uint8_t*>(streamDataTransCoding), + streamDataTransCoding, true); if(streamLenTransCoding < 0) { @@ -920,7 +919,7 @@ int main(int argc, char* argv[]) { declenTC = WebRtcIsac_DecodeRcu( decoderTransCoding, - reinterpret_cast<const uint8_t*>(streamDataTransCoding), + streamDataTransCoding, streamLenTransCoding, decodedTC, speechType); @@ -938,7 +937,7 @@ int main(int argc, char* argv[]) { declenTC = WebRtcIsac_Decode( decoderTransCoding, - reinterpret_cast<const uint8_t*>(streamDataTransCoding), + streamDataTransCoding, streamLenTransCoding, decodedTC, speechType); diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c index ce759a43..1e752a18 100644 --- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c +++ b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c @@ -102,8 +102,8 @@ int main(int argc, char* argv[]) unsigned int tmpSumStreamLen = 0; unsigned int packetCntr = 0; unsigned int lostPacketCntr = 0; - uint16_t payload[600]; - uint16_t payloadRCU[600]; + uint8_t payload[1200]; + uint8_t payloadRCU[1200]; uint16_t packetLossPercent = 0; int16_t rcuStreamLen = 0; int onlyEncode; @@ -376,7 +376,7 @@ valid values are 8 and 16.\n", sampFreqKHz); stream_len = WebRtcIsac_Encode( ISAC_main_inst, shortdata, - (uint8_t*)payload); + payload); if(stream_len < 0) { @@ -396,12 +396,12 @@ valid values are 8 and 16.\n", sampFreqKHz); } rcuStreamLen = WebRtcIsac_GetRedPayload( - ISAC_main_inst, (uint8_t*)payloadRCU); + ISAC_main_inst, payloadRCU); get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData, sampFreqKHz * 1000, sampFreqKHz * 1000); if(WebRtcIsac_UpdateBwEstimate(ISAC_main_inst, - (const uint8_t*)payload, + payload, stream_len, packetData.rtp_number, packetData.sample_count, @@ -460,7 +460,7 @@ valid values are 8 and 16.\n", sampFreqKHz); { declen = WebRtcIsac_DecodeRcu( ISAC_main_inst, - (const uint8_t*)payloadRCU, + payloadRCU, rcuStreamLen, decoded, speechType); @@ -470,7 +470,7 @@ valid values are 8 and 16.\n", sampFreqKHz); { declen = WebRtcIsac_Decode( ISAC_main_inst, - (const uint8_t*)payload, + payload, stream_len, decoded, speechType); |