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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
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
View File

@ -357,7 +357,7 @@ extern "C" {
*
*/
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 @@ extern "C" {
int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
int16_t bweIndex,
float scale,
int16_t *encoded);
uint8_t* encoded);
/****************************************************************************
@ -608,7 +608,7 @@ extern "C" {
*
*/
int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
int encoded_len_bytes,
int16_t* rateIndex);

17
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
View File

@ -536,7 +536,7 @@ int16_t WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
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 @@ int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
#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_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
*
*/
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 @@ int16_t WebRtcIsacfix_ReadFrameLen(const int16_t* encoded,
#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_ReadFrameLen(const int16_t* encoded,
*
*/
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 @@ int16_t WebRtcIsacfix_ReadBwIndex(const int16_t* encoded,
#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);

8
webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
View File

@ -572,12 +572,14 @@ int main(int argc, char* argv[])
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 @@ int main(int argc, char* argv[])
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 */

8
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
View File

@ -319,7 +319,7 @@ extern "C" {
int16_t WebRtcIsac_ReadFrameLen(
ISACStruct* ISAC_main_inst,
const int16_t* encoded,
const uint8_t* encoded,
int16_t* frameLength);
@ -574,7 +574,7 @@ extern "C" {
int16_t bweIndex,
int16_t jitterInfo,
int32_t rate,
int16_t* encoded,
uint8_t* encoded,
int16_t isRCU);
@ -631,7 +631,7 @@ extern "C" {
*/
int16_t WebRtcIsac_ReadBwIndex(
const int16_t* encoded,
const uint8_t* encoded,
int16_t* bweIndex);
@ -679,7 +679,7 @@ extern "C" {
*/
int16_t WebRtcIsac_GetRedPayload(
ISACStruct* ISAC_main_inst,
int16_t* encoded);
uint8_t* encoded);
/****************************************************************************

40
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -786,7 +786,7 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst,
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 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst,
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 @@ int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst,
}
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 @@ int16_t WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst,
* - 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 @@ int16_t WebRtcIsac_ReadBwIndex(const int16_t* encoded,
#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_ReadBwIndex(const int16_t* encoded,
*
*/
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 @@ int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
#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 @@ int16_t WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst,
* : -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 @@ int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
}
/* 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 @@ int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst,
(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
}

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);

8
webrtc/modules/audio_coding/main/acm2/acm_isac.cc
View File

@ -211,7 +211,7 @@ static int16_t ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst,
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 @@ int16_t ACMISAC::Transcode(uint8_t* bitstream,
*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 @@ int32_t ACMISAC::GetRedPayloadSafe(
#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;
}

5
webrtc/modules/audio_coding/neteq/test/RTPencode.cc
View File

@ -261,7 +261,7 @@ int main(int argc, char* argv[])
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 @@ int main(int argc, char* argv[])
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
{