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WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[]

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This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.

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

Review URL: https://webrtc-codereview.appspot.com/25739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org 2014-10-13 11:23:24 +00:00
parent 3f7f899a15
commit 396a5e0001
11 changed files with 75 additions and 46 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
View File

@ -252,7 +252,7 @@ extern "C" {
*/
int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
const uint16_t *encoded,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType);

9
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
View File

@ -805,7 +805,7 @@ int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
const uint16_t *encoded,
const uint8_t* encoded,
int16_t len,
int16_t *decoded,
int16_t *speechType)
@ -844,10 +844,13 @@ int16_t WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
/* convert bitstream from int16_t to bytes */
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k=0; k<(len>>1); k++) {
(ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
uint16_t ek = ((const uint16_t*)encoded)[k];
ISAC_inst->ISACdec_obj.bitstr_obj.stream[k] =
(uint16_t)((ek >> 8) | ((ek & 0xff) << 8));
}
if (len & 0x0001)
(ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (uint16_t) ((encoded[k] & 0xFF)<<8);
ISAC_inst->ISACdec_obj.bitstr_obj.stream[k] =
(uint16_t)((((const uint16_t*)encoded)[k] & 0xff) << 8);
#else
memcpy(ISAC_inst->ISACdec_obj.bitstr_obj.stream, encoded, len);
#endif

2
webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
View File

@ -86,7 +86,7 @@ float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream, int encoded_bytes,
int16_t audio_type;
clock_t clocks = clock();
value = WebRtcIsacfix_Decode(ISACFIX_main_inst_,
reinterpret_cast<const uint16_t*>(bit_stream),
bit_stream,
encoded_bytes, out_data, &audio_type);
clocks = clock() - clocks;
EXPECT_EQ(output_length_sample_, value);

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

@ -746,8 +746,12 @@ int main(int argc, char* argv[])
/* Call getFramelen, only used here for function test */
err = WebRtcIsacfix_ReadFrameLen(
reinterpret_cast<const uint8_t*>(streamdata), stream_len, &FL);
declen = WebRtcIsacfix_Decode( ISAC_main_inst, streamdata, stream_len,
decoded, speechType );
declen = WebRtcIsacfix_Decode(
ISAC_main_inst,
reinterpret_cast<const uint8_t*>(streamdata),
stream_len,
decoded,
speechType);
/* Error check */
if (err<0 || declen<0 || FL!=declen) {
errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);

4
webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
View File

@ -216,7 +216,7 @@ extern "C" {
int16_t WebRtcIsac_Decode(
ISACStruct* ISAC_main_inst,
const uint16_t* encoded,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
@ -703,7 +703,7 @@ extern "C" {
*/
int16_t WebRtcIsac_DecodeRcu(
ISACStruct* ISAC_main_inst,
const uint16_t* encoded,
const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);

24
webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
View File

@ -1046,7 +1046,7 @@ int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
}
static int16_t Decode(ISACStruct* ISAC_main_inst,
const uint16_t* encoded,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType,
@ -1065,7 +1065,6 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
int16_t lenEncodedLBBytes;
int16_t validChecksum = 1;
int16_t k;
uint8_t* ptrEncodedUW8 = (uint8_t*)encoded;
uint16_t numLayer;
int16_t totSizeBytes;
int16_t err;
@ -1094,7 +1093,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
STREAM_SIZE_MAX : lenEncodedBytes;
/* Copy to lower-band bit-stream structure. */
memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, ptrEncodedUW8,
memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, encoded,
lenEncodedLBBytes);
/* Regardless of that the current codec is setup to work in
@ -1116,12 +1115,12 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
totSizeBytes = numDecodedBytesLB;
while (totSizeBytes != lenEncodedBytes) {
if ((totSizeBytes > lenEncodedBytes) ||
(ptrEncodedUW8[totSizeBytes] == 0) ||
(encoded[totSizeBytes] == 0) ||
(numLayer > MAX_NUM_LAYERS)) {
instISAC->errorCode = ISAC_LENGTH_MISMATCH;
return -1;
}
totSizeBytes += ptrEncodedUW8[totSizeBytes];
totSizeBytes += encoded[totSizeBytes];
numLayer++;
}
@ -1160,7 +1159,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
instISAC->resetFlag_8kHz = 2;
} else {
/* This includes the checksum and the bytes that stores the length. */
int16_t lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
int16_t lenNextStream = encoded[numDecodedBytesLB];
/* Is this garbage or valid super-wideband bit-stream?
* Check if checksum is valid. */
@ -1170,14 +1169,13 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
validChecksum = 0;
} else {
/* Run CRC to see if the checksum match. */
WebRtcIsac_GetCrc((int16_t*)(
&ptrEncodedUW8[numDecodedBytesLB + 1]),
WebRtcIsac_GetCrc((int16_t*)(&encoded[numDecodedBytesLB + 1]),
lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc);
validChecksum = 1;
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
validChecksum &= (((crc >> (24 - k * 8)) & 0xFF) ==
ptrEncodedUW8[numDecodedBytesLB + lenNextStream -
encoded[numDecodedBytesLB + lenNextStream -
LEN_CHECK_SUM_WORD8 + k]);
}
}
@ -1209,7 +1207,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
lenNextStream -= (LEN_CHECK_SUM_WORD8 + 1);
memcpy(decInstUB->bitstr_obj.stream,
&ptrEncodedUW8[numDecodedBytesLB + 1], lenNextStream);
&encoded[numDecodedBytesLB + 1], lenNextStream);
/* Reset bit-stream object, this is the first decoding. */
WebRtcIsac_ResetBitstream(&(decInstUB->bitstr_obj));
@ -1283,7 +1281,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
/* It might be less due to garbage. */
if ((numDecodedBytesUB != lenNextStream) &&
(numDecodedBytesUB != (lenNextStream -
ptrEncodedUW8[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
encoded[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
instISAC->errorCode = ISAC_LENGTH_MISMATCH;
return -1;
}
@ -1346,7 +1344,7 @@ static int16_t Decode(ISACStruct* ISAC_main_inst,
*/
int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
const uint16_t* encoded,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType) {
@ -1378,7 +1376,7 @@ int16_t WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
int16_t WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
const uint16_t* encoded,
const uint8_t* encoded,
int16_t lenEncodedBytes,
int16_t* decoded,
int16_t* speechType) {

35
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -909,26 +909,39 @@ int main(int argc, char* argv[])
if(lostFrame)
{
declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, streamdata,
stream_len, decoded, speechType);
declen = WebRtcIsac_DecodeRcu(
ISAC_main_inst,
reinterpret_cast<const uint8_t*>(streamdata),
stream_len,
decoded,
speechType);
if(doTransCoding)
{
declenTC = WebRtcIsac_DecodeRcu(decoderTransCoding,
streamDataTransCoding, streamLenTransCoding,
decodedTC, speechType);
declenTC = WebRtcIsac_DecodeRcu(
decoderTransCoding,
reinterpret_cast<const uint8_t*>(streamDataTransCoding),
streamLenTransCoding,
decodedTC,
speechType);
}
}
else
{
declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata,
stream_len, decoded, speechType);
declen = WebRtcIsac_Decode(
ISAC_main_inst,
reinterpret_cast<const uint8_t*>(streamdata),
stream_len,
decoded,
speechType);
if(doTransCoding)
{
declenTC = WebRtcIsac_Decode(decoderTransCoding,
streamDataTransCoding, streamLenTransCoding,
decodedTC, speechType);
declenTC = WebRtcIsac_Decode(
decoderTransCoding,
reinterpret_cast<const uint8_t*>(streamDataTransCoding),
streamLenTransCoding,
decodedTC,
speechType);
}
}

7
webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@ -424,8 +424,11 @@ int main(int argc, char* argv[])
/**/
// Decode
lenDecodedAudio = WebRtcIsac_Decode(
codecInstance[receiverIdx], bitStream, streamLen,
audioBuff60ms, speechType);
codecInstance[receiverIdx],
reinterpret_cast<const uint8_t*>(bitStream),
streamLen,
audioBuff60ms,
speechType);
if(lenDecodedAudio < 0)
{
printf(" Decoder error in client %d \n", receiverIdx + 1);

20
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@ -458,16 +458,24 @@ valid values are 8 and 16.\n", sampFreqKHz);
if((rand() % 100) < packetLossPercent)
{
declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, payloadRCU,
rcuStreamLen, decoded, speechType);
declen = WebRtcIsac_DecodeRcu(
ISAC_main_inst,
(const uint8_t*)payloadRCU,
rcuStreamLen,
decoded,
speechType);
lostPacketCntr++;
}
else
{
declen = WebRtcIsac_Decode(ISAC_main_inst, payload,
stream_len, decoded, speechType);
}
if(declen <= 0)
declen = WebRtcIsac_Decode(
ISAC_main_inst,
(const uint8_t*)payload,
stream_len,
decoded,
speechType);
}
if(declen <= 0)
{
//errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
fprintf(stderr,"\nError in decoder.\n");

4
webrtc/modules/audio_coding/main/acm2/acm_isac.cc
View File

@ -730,7 +730,7 @@ int ACMISAC::Decode(const uint8_t* encoded,
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
int ret =
ACM_ISAC_DECODE_B(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
reinterpret_cast<const uint16_t*>(encoded),
encoded,
static_cast<int16_t>(encoded_len),
decoded,
&temp_type);
@ -769,7 +769,7 @@ int ACMISAC::DecodeRedundant(const uint8_t* encoded,
CriticalSectionScoped lock(codec_inst_crit_sect_.get());
int16_t ret =
ACM_ISAC_DECODERCU(static_cast<ACM_ISAC_STRUCT*>(codec_inst_ptr_->inst),
reinterpret_cast<const uint16_t*>(encoded),
encoded,
static_cast<int16_t>(encoded_len),
decoded,
&temp_type);

6
webrtc/modules/audio_coding/neteq/audio_decoder_impl.cc
View File

@ -165,7 +165,7 @@ int AudioDecoderIsac::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIsac_Decode(static_cast<ISACStruct*>(state_),
reinterpret_cast<const uint16_t*>(encoded),
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
@ -177,7 +177,7 @@ int AudioDecoderIsac::DecodeRedundant(const uint8_t* encoded,
SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIsac_DecodeRcu(static_cast<ISACStruct*>(state_),
reinterpret_cast<const uint16_t*>(encoded),
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);
@ -236,7 +236,7 @@ int AudioDecoderIsacFix::Decode(const uint8_t* encoded, size_t encoded_len,
int16_t* decoded, SpeechType* speech_type) {
int16_t temp_type = 1; // Default is speech.
int16_t ret = WebRtcIsacfix_Decode(static_cast<ISACFIX_MainStruct*>(state_),
reinterpret_cast<const uint16_t*>(encoded),
encoded,
static_cast<int16_t>(encoded_len), decoded,
&temp_type);
*speech_type = ConvertSpeechType(temp_type);