Some WebRtcIsac_* and WebRtcIsacfix_* functions: type encoded stream as uint8[]
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
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
index c113b03..70fff13 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -357,7 +357,7 @@
*
*/
- int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
+ int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
int encoded_len_bytes,
int16_t* frameLength);
@@ -557,7 +557,7 @@
int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
int16_t bweIndex,
float scale,
- int16_t *encoded);
+ uint8_t* encoded);
/****************************************************************************
@@ -608,7 +608,7 @@
*
*/
- int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
+ int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
int encoded_len_bytes,
int16_t* rateIndex);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 44fff0e..9f3bdb6 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -536,7 +536,7 @@
int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
int16_t bweIndex,
float scale,
- int16_t *encoded)
+ uint8_t* encoded)
{
ISACFIX_SubStruct *ISAC_inst;
int16_t stream_len;
@@ -564,8 +564,9 @@
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k=0;k<(stream_len+1)>>1;k++) {
- encoded[k] = (int16_t)( ( (uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
- | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
+ ((int16_t*)encoded)[k] = (int16_t)(
+ ((uint16_t)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8) |
+ (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
}
#else
@@ -1315,7 +1316,7 @@
*
*/
-int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
+int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
int encoded_len_bytes,
int16_t* frameLength)
{
@@ -1334,7 +1335,8 @@
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < kRequiredEncodedLenBytes / 2; k++) {
- streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ uint16_t ek = ((uint16_t*)encoded)[k];
+ streamdata.stream[k] = (uint16_t)((ek >> 8) | ((ek & 0xff) << 8));
}
#else
memcpy(streamdata.stream, encoded, kRequiredEncodedLenBytes);
@@ -1363,7 +1365,7 @@
*
*/
-int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
+int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
int encoded_len_bytes,
int16_t* rateIndex)
{
@@ -1382,7 +1384,8 @@
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < kRequiredEncodedLenBytes / 2; k++) {
- streamdata.stream[k] = (uint16_t) (((uint16_t)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
+ uint16_t ek = ((uint16_t*)encoded)[k];
+ streamdata.stream[k] = (uint16_t)((ek >> 8) | ((ek & 0xff) << 8));
}
#else
memcpy(streamdata.stream, encoded, kRequiredEncodedLenBytes);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index 39a413b..6b044f8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -572,12 +572,14 @@
if (stream_len>0) {
if (testCE == 1) {
err = WebRtcIsacfix_ReadBwIndex(
- (int16_t*)streamdata, stream_len, &bwe);
+ reinterpret_cast<const uint8_t*>(streamdata),
+ stream_len,
+ &bwe);
stream_len = WebRtcIsacfix_GetNewBitStream(
ISAC_main_inst,
bwe,
scale,
- (int16_t*)streamdata);
+ reinterpret_cast<uint8_t*>(streamdata));
} else if (testCE == 2) {
/* transcode function not supported */
} else if (testCE == 3) {
@@ -742,7 +744,7 @@
short FL;
/* Call getFramelen, only used here for function test */
err = WebRtcIsacfix_ReadFrameLen(
- (int16_t*)streamdata, stream_len, &FL);
+ reinterpret_cast<const uint8_t*>(streamdata), stream_len, &FL);
declen = WebRtcIsacfix_Decode( ISAC_main_inst, streamdata, stream_len,
decoded, speechType );
/* Error check */
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
index 4067058..a3786bf 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -319,7 +319,7 @@
int16_t WebRtcIsac_ReadFrameLen(
ISACStruct* ISAC_main_inst,
- const int16_t* encoded,
+ const uint8_t* encoded,
int16_t* frameLength);
@@ -574,7 +574,7 @@
int16_t bweIndex,
int16_t jitterInfo,
int32_t rate,
- int16_t* encoded,
+ uint8_t* encoded,
int16_t isRCU);
@@ -631,7 +631,7 @@
*/
int16_t WebRtcIsac_ReadBwIndex(
- const int16_t* encoded,
+ const uint8_t* encoded,
int16_t* bweIndex);
@@ -679,7 +679,7 @@
*/
int16_t WebRtcIsac_GetRedPayload(
ISACStruct* ISAC_main_inst,
- int16_t* encoded);
+ uint8_t* encoded);
/****************************************************************************
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 13170a0..882712d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -786,7 +786,7 @@
int16_t bweIndex,
int16_t jitterInfo,
int32_t rate,
- int16_t* encoded,
+ uint8_t* encoded,
int16_t isRCU) {
Bitstr iSACBitStreamInst; /* Local struct for bitstream handling */
int16_t streamLenLB;
@@ -799,7 +799,6 @@
double rateLB;
double rateUB;
int32_t currentBN;
- uint8_t* encodedPtrUW8 = (uint8_t*)encoded;
uint32_t crc;
#ifndef WEBRTC_ARCH_BIG_ENDIAN
int16_t k;
@@ -885,20 +884,20 @@
}
totalStreamLen = streamLenLB + streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
- encodedPtrUW8[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
+ encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
- memcpy(&encodedPtrUW8[streamLenLB + 1], iSACBitStreamInst.stream,
+ memcpy(&encoded[streamLenLB + 1], iSACBitStreamInst.stream,
streamLenUB);
- WebRtcIsac_GetCrc((int16_t*)(&(encodedPtrUW8[streamLenLB + 1])),
+ WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])),
streamLenUB, &crc);
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
- encodedPtrUW8[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
+ encoded[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
(uint8_t)((crc >> (24 - k * 8)) & 0xFF);
}
#else
- memcpy(&encodedPtrUW8[streamLenLB + streamLenUB + 1], &crc,
+ memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
#endif
return totalStreamLen;
@@ -1734,7 +1733,7 @@
* - bweIndex : Bandwidth estimate in bit-stream
*
*/
-int16_t WebRtcIsac_ReadBwIndex(const int16_t* encoded,
+int16_t WebRtcIsac_ReadBwIndex(const uint8_t* encoded,
int16_t* bweIndex) {
Bitstr streamdata;
#ifndef WEBRTC_ARCH_BIG_ENDIAN
@@ -1746,8 +1745,8 @@
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < 10; k++) {
- streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
- ((k & 1) << 3)) & 0xFF);
+ int16_t ek2 = ((const int16_t*)encoded)[k >> 1];
+ streamdata.stream[k] = (uint8_t)((ek2 >> ((k & 1) << 3)) & 0xff);
}
#else
memcpy(streamdata.stream, encoded, 10);
@@ -1783,7 +1782,7 @@
*
*/
int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
- const int16_t* encoded,
+ const uint8_t* encoded,
int16_t* frameLength) {
Bitstr streamdata;
#ifndef WEBRTC_ARCH_BIG_ENDIAN
@@ -1796,8 +1795,8 @@
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < 10; k++) {
- streamdata.stream[k] = (uint8_t)((encoded[k >> 1] >>
- ((k & 1) << 3)) & 0xFF);
+ int16_t ek2 = ((const int16_t*)encoded)[k >> 1];
+ streamdata.stream[k] = (uint8_t)((ek2 >> ((k & 1) << 3)) & 0xff);
}
#else
memcpy(streamdata.stream, encoded, 10);
@@ -2096,13 +2095,12 @@
* : -1 - Error
*/
int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
- int16_t* encoded) {
+ uint8_t* encoded) {
Bitstr iSACBitStreamInst;
int16_t streamLenLB;
int16_t streamLenUB;
int16_t streamLen;
int16_t totalLenUB;
- uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst;
#ifndef WEBRTC_ARCH_BIG_ENDIAN
int k;
@@ -2125,7 +2123,7 @@
}
/* convert from bytes to int16_t. */
- memcpy(ptrEncodedUW8, iSACBitStreamInst.stream, streamLenLB);
+ memcpy(encoded, iSACBitStreamInst.stream, streamLenLB);
streamLen = streamLenLB;
if (instISAC->bandwidthKHz == isac8kHz) {
return streamLenLB;
@@ -2154,19 +2152,19 @@
(streamLenUB > 0)) {
uint32_t crc;
streamLen += totalLenUB;
- ptrEncodedUW8[streamLenLB] = (uint8_t)totalLenUB;
- memcpy(&ptrEncodedUW8[streamLenLB + 1], iSACBitStreamInst.stream,
+ encoded[streamLenLB] = (uint8_t)totalLenUB;
+ memcpy(&encoded[streamLenLB + 1], iSACBitStreamInst.stream,
streamLenUB);
- WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
+ WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])),
streamLenUB, &crc);
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
- ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
+ encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] =
(uint8_t)((crc >> (24 - k * 8)) & 0xFF);
}
#else
- memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
+ memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
#endif
}
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 8af4e6f..e436549 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -687,8 +687,12 @@
/************************* 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 @@
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 @@
// 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 @@
}
/* 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 */
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
index 72d3fe8..6253e00 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
@@ -287,8 +287,10 @@
(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");
}
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 2df5a84..96aee1f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -395,7 +395,8 @@
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);
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index f3682f1..e4f6195 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -211,7 +211,7 @@
int16_t bwe_index,
int16_t /* jitter_index */,
int32_t rate,
- int16_t* bitstream,
+ uint8_t* bitstream,
bool is_red) {
if (is_red) {
// RED not supported with iSACFIX
@@ -437,7 +437,7 @@
*bitstream_len_byte = ACM_ISAC_GETNEWBITSTREAM(
codec_inst_ptr_->inst, q_bwe, jitter_info, rate,
- reinterpret_cast<int16_t*>(bitstream), (is_red) ? 1 : 0);
+ bitstream, (is_red) ? 1 : 0);
if (*bitstream_len_byte < 0) {
// error happened
@@ -591,9 +591,7 @@
#else
uint8_t* red_payload, int16_t* payload_bytes) {
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
- int16_t bytes =
- WebRtcIsac_GetRedPayload(
- codec_inst_ptr_->inst, reinterpret_cast<int16_t*>(red_payload));
+ int16_t bytes = WebRtcIsac_GetRedPayload(codec_inst_ptr_->inst, red_payload);
if (bytes < 0) {
return -1;
}
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 92bccee..ab338a7 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -261,7 +261,7 @@
uint32_t red_TS[2] = {0};
uint16_t red_len[2] = {0};
int RTPheaderLen=12;
- unsigned char red_data[8000];
+ uint8_t red_data[8000];
#ifdef INSERT_OLD_PACKETS
uint16_t old_length, old_plen;
int old_enc_len;
@@ -755,7 +755,8 @@
if(usedCodec==webrtc::kDecoderISAC)
{
assert(!usingStereo); // Cannot handle stereo yet
- red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (int16_t*)red_data);
+ red_len[0] =
+ WebRtcIsac_GetRedPayload(ISAC_inst[0], red_data);
}
else
{