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Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."

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This was causing apparently legitimate failures on the following bots:
http://chromegw/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/2599
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29%28dbg%29/builds/2023
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29%28dbg%29/builds/1825
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28KK%20Nexus5%29/builds/2013
http://chromegw/i/client.webrtc/builders/Android%20Tests%20%28JB%20Nexus7.2%29/builds/1795

> WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t
> 
> We have to fix both at once, since there's a macro that calls one of
> them or the other.
> 
> BUG=909
> R=andrew@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, minyue@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/19229004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7267 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2014-09-23 01:32:57 +00:00
parent 8c5740b485
commit a3c4d4dd2c
15 changed files with 68 additions and 73 deletions
Download Patch File Download Diff File Expand all files Collapse all files

41
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -496,7 +496,7 @@ int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst,
*/
int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
uint8_t* encoded) {
int16_t* encoded) {
float inFrame[FRAMESAMPLES_10ms];
int16_t speechInLB[FRAMESAMPLES_10ms];
int16_t speechInUB[FRAMESAMPLES_10ms];
@ -504,6 +504,7 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
int16_t streamLenUB = 0;
int16_t streamLen = 0;
int16_t k = 0;
uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
int garbageLen = 0;
int32_t bottleneck = 0;
int16_t bottleneckIdx = 0;
@ -642,22 +643,23 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
streamLenUB = 0;
}
memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
streamLen = streamLenLB;
if (streamLenUB > 0) {
encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
memcpy(&encoded[streamLenLB + 1],
instUB->ISACencUB_obj.bitstr_obj.stream,
streamLenUB);
streamLen += encoded[streamLenLB];
ptrEncodedUW8[streamLenLB] = (uint8_t)(streamLenUB + 1 +
LEN_CHECK_SUM_WORD8);
memcpy(&ptrEncodedUW8[streamLenLB + 1],
instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB);
streamLen += ptrEncodedUW8[streamLenLB];
} else {
encoded[streamLenLB] = 0;
ptrEncodedUW8[streamLenLB] = 0;
}
} else {
if (streamLenLB == 0) {
return 0;
}
memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
streamLenLB);
streamLenUB = 0;
streamLen = streamLenLB;
}
@ -695,11 +697,11 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
* 255 is the max garbage length we can signal using 8 bits. */
if ((instISAC->bandwidthKHz == isac8kHz) ||
(streamLenUB == 0)) {
ptrGarbage = &encoded[streamLenLB];
ptrGarbage = &ptrEncodedUW8[streamLenLB];
limit = streamLen + 255;
} else {
ptrGarbage = &encoded[streamLenLB + 1 + streamLenUB];
limit = streamLen + (255 - encoded[streamLenLB]);
ptrGarbage = &ptrEncodedUW8[streamLenLB + 1 + streamLenUB];
limit = streamLen + (255 - ptrEncodedUW8[streamLenLB]);
}
minBytes = (minBytes > limit) ? limit : minBytes;
@ -716,12 +718,13 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
* That is the only way to preserve backward compatibility. */
if ((instISAC->bandwidthKHz == isac8kHz) ||
(streamLenUB == 0)) {
encoded[streamLenLB] = garbageLen;
ptrEncodedUW8[streamLenLB] = (uint8_t)garbageLen;
} else {
encoded[streamLenLB] += garbageLen;
ptrEncodedUW8[streamLenLB] += (uint8_t)garbageLen;
/* Write the length of the garbage at the end of the upper-band
* bit-stream, if exists. This helps for sanity check. */
encoded[streamLenLB + 1 + streamLenUB] = garbageLen;
ptrEncodedUW8[streamLenLB + 1 + streamLenUB] =
(uint8_t)garbageLen;
}
streamLen += garbageLen;
@ -738,14 +741,16 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
if ((instISAC->bandwidthKHz != isac8kHz) && (streamLenUB > 0)) {
uint32_t crc;
WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])),
WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
streamLenUB + garbageLen, &crc);
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = crc >> (24 - k * 8);
ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
(uint8_t)((crc >> (24 - k * 8)) & 0xFF);
}
#else
memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, LEN_CHECK_SUM_WORD8);
memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
#endif
}
return streamLen;

2
webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
View File

@ -31,7 +31,7 @@ class IsacTest : public ::testing::Test {
int16_t speech_data_[kIsacNumberOfSamples];
int16_t output_data_[kIsacNumberOfSamples];
uint8_t bitstream_[kMaxBytes];
int16_t bitstream_[kMaxBytes / 2];
uint8_t bitstream_small_[7]; // Simulate sync packets.
};