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WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type encoded stream as uint8_t

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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

Committed: https://code.google.com/p/webrtc/source/detail?r=7266

Review URL: https://webrtc-codereview.appspot.com/19229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7285 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kwiberg@webrtc.org
2014-09-24 10:31:02 +00:00
parent d60d79a145
commit 7ee24a7906
15 changed files with 77 additions and 68 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
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@@ -130,7 +130,7 @@ extern "C" {
int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
int16_t *encoded);
uint8_t* encoded);

25
webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
View File

@@ -17,6 +17,7 @@
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
#include <assert.h>
#include <stdlib.h>
#include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
@@ -355,10 +356,10 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
int16_t *encoded)
uint8_t* encoded)
{
ISACFIX_SubStruct *ISAC_inst;
int16_t stream_len;
int16_t stream_len, stream_len_even;
#ifndef WEBRTC_ARCH_BIG_ENDIAN
int k;
#endif
@@ -382,16 +383,24 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
return -1;
}
/* One would think that only even stream lengths would make sense here. We do
in fact observe odd lengths, however, and in those cases we copy an extra
byte. */
stream_len_even = stream_len % 2 == 0 ? stream_len : stream_len + 1;
/* convert from bytes to int16_t */
#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));
/* The encoded data vector is supposesd to be big-endian, but our internal
representation is little-endian. So byteswap. */
for (k = 0; k < stream_len_even / 2; ++k) {
uint16_t s = ISAC_inst->ISACenc_obj.bitstr_obj.stream[k];
/* In big-endian, we have... */
encoded[2 * k] = s >> 8; /* ...most significant byte at low address... */
encoded[2 * k + 1] = s; /* ...least significant byte at high address. */
}
#else
WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
/* The encoded data vector and our internal representation are both
big-endian. */
memcpy(encoded, ISAC_inst->ISACenc_obj.bitstr_obj.stream, stream_len_even);
#endif

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

@@ -71,7 +71,7 @@ float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
size_t pointer = 0;
for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
reinterpret_cast<int16_t*>(bit_stream));
bit_stream);
}
clocks = clock() - clocks;
EXPECT_GT(value, 0);

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

@@ -565,7 +565,7 @@ int main(int argc, char* argv[])
/* Encode */
stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
shortdata,
(int16_t*)streamdata);
(uint8_t*)streamdata);
/* If packet is ready, and CE testing, call the different API
functions from the internal API. */

9
webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
View File

@@ -439,7 +439,9 @@ int main(int argc, char* argv[])
/* iSAC encoding */
if (mode==0 || mode ==1) {
stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata, streamdata);
stream_len = WebRtcIsac_Encode(ISAC_main_inst,
shortdata,
(uint8_t*)streamdata);
if (stream_len < 0) {
/* exit if returned with error */
errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
@@ -449,7 +451,10 @@ int main(int argc, char* argv[])
} else if (mode==2 || mode==3) {
/* iSAC encoding */
if (nbTest != 1)
stream_len = WebRtcIsacfix_Encode(ISACFIX_main_inst, shortdata, streamdata);
stream_len = WebRtcIsacfix_Encode(
ISACFIX_main_inst,
shortdata,
(uint8_t*)streamdata);
else
stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);

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

@@ -147,7 +147,7 @@ extern "C" {
int16_t WebRtcIsac_Encode(
ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
int16_t* encoded);
uint8_t* encoded);
/******************************************************************************

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,
int16_t* encoded) {
uint8_t* encoded) {
float inFrame[FRAMESAMPLES_10ms];
int16_t speechInLB[FRAMESAMPLES_10ms];
int16_t speechInUB[FRAMESAMPLES_10ms];
@@ -504,7 +504,6 @@ 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;
@@ -643,23 +642,22 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
streamLenUB = 0;
}
memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
streamLen = streamLenLB;
if (streamLenUB > 0) {
ptrEncodedUW8[streamLenLB] = (uint8_t)(streamLenUB + 1 +
LEN_CHECK_SUM_WORD8);
memcpy(&ptrEncodedUW8[streamLenLB + 1],
instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB);
streamLen += ptrEncodedUW8[streamLenLB];
encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
memcpy(&encoded[streamLenLB + 1],
instUB->ISACencUB_obj.bitstr_obj.stream,
streamLenUB);
streamLen += encoded[streamLenLB];
} else {
ptrEncodedUW8[streamLenLB] = 0;
encoded[streamLenLB] = 0;
}
} else {
if (streamLenLB == 0) {
return 0;
}
memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
streamLenLB);
memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
streamLenUB = 0;
streamLen = streamLenLB;
}
@@ -697,11 +695,11 @@ 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 = &ptrEncodedUW8[streamLenLB];
ptrGarbage = &encoded[streamLenLB];
limit = streamLen + 255;
} else {
ptrGarbage = &ptrEncodedUW8[streamLenLB + 1 + streamLenUB];
limit = streamLen + (255 - ptrEncodedUW8[streamLenLB]);
ptrGarbage = &encoded[streamLenLB + 1 + streamLenUB];
limit = streamLen + (255 - encoded[streamLenLB]);
}
minBytes = (minBytes > limit) ? limit : minBytes;
@@ -718,13 +716,12 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
* That is the only way to preserve backward compatibility. */
if ((instISAC->bandwidthKHz == isac8kHz) ||
(streamLenUB == 0)) {
ptrEncodedUW8[streamLenLB] = (uint8_t)garbageLen;
encoded[streamLenLB] = garbageLen;
} else {
ptrEncodedUW8[streamLenLB] += (uint8_t)garbageLen;
encoded[streamLenLB] += garbageLen;
/* Write the length of the garbage at the end of the upper-band
* bit-stream, if exists. This helps for sanity check. */
ptrEncodedUW8[streamLenLB + 1 + streamLenUB] =
(uint8_t)garbageLen;
encoded[streamLenLB + 1 + streamLenUB] = garbageLen;
}
streamLen += garbageLen;
@@ -741,16 +738,14 @@ int16_t WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
if ((instISAC->bandwidthKHz != isac8kHz) && (streamLenUB > 0)) {
uint32_t crc;
WebRtcIsac_GetCrc((int16_t*)(&(ptrEncodedUW8[streamLenLB + 1])),
WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])),
streamLenUB + garbageLen, &crc);
#ifndef WEBRTC_ARCH_BIG_ENDIAN
for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) {
ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
(uint8_t)((crc >> (24 - k * 8)) & 0xFF);
encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = crc >> (24 - k * 8);
}
#else
memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
LEN_CHECK_SUM_WORD8);
memcpy(&encoded[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];
int16_t bitstream_[kMaxBytes / 2];
uint8_t bitstream_[kMaxBytes];
uint8_t bitstream_small_[7]; // Simulate sync packets.
};

4
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@@ -662,8 +662,8 @@ int main(int argc, char* argv[])
if(!(testNum == 3 && framecnt == 0))
{
stream_len = WebRtcIsac_Encode(ISAC_main_inst,
shortdata,
(int16_t*)streamdata);
shortdata,
(uint8_t*)streamdata);
if((payloadSize != 0) && (stream_len > payloadSize))
{
if(testNum == 0)

3
webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@@ -283,7 +283,8 @@ int main(int argc, char* argv[])
streamLen = WebRtcIsac_Encode(codecInstance[senderIdx],
audioBuff10ms, (short*)bitStream);
audioBuff10ms,
(uint8_t*)bitStream);
int16_t ggg;
if (streamLen > 0) {
if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx],

6
webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
View File

@@ -373,8 +373,10 @@ valid values are 8 and 16.\n", sampFreqKHz);
cur_framesmpls += samplesIn10Ms;
//-------- iSAC encoding ---------
stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata,
(int16_t*)payload);
stream_len = WebRtcIsac_Encode(
ISAC_main_inst,
shortdata,
(uint8_t*)payload);
if(stream_len < 0)
{

5
webrtc/modules/audio_coding/main/acm2/acm_isac.cc
View File

@@ -347,8 +347,9 @@ int16_t ACMISAC::InternalEncode(uint8_t* bitstream,
return -1;
}
*bitstream_len_byte = ACM_ISAC_ENCODE(
codec_inst_ptr_->inst, &in_audio_[in_audio_ix_read_],
reinterpret_cast<int16_t*>(bitstream));
codec_inst_ptr_->inst,
&in_audio_[in_audio_ix_read_],
bitstream);
// increment the read index this tell the caller that how far
// we have gone forward in reading the audio buffer
in_audio_ix_read_ += samples_in_10ms_audio_;

27
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View File

@@ -348,14 +348,11 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
// Insert 3 * 10 ms. Expect non-zero output on third call.
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
reinterpret_cast<int16_t*>(output)));
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output));
input += input_size_;
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
reinterpret_cast<int16_t*>(output)));
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output));
input += input_size_;
int enc_len_bytes =
WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output));
int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output);
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}
@@ -388,14 +385,11 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
// Insert 3 * 10 ms. Expect non-zero output on third call.
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
reinterpret_cast<int16_t*>(output)));
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output));
input += input_size_;
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
reinterpret_cast<int16_t*>(output)));
EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output));
input += input_size_;
int enc_len_bytes =
WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output));
int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output);
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}
@@ -441,14 +435,11 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
virtual int EncodeFrame(const int16_t* input, size_t input_len_samples,
uint8_t* output) {
// Insert 3 * 10 ms. Expect non-zero output on third call.
EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input,
reinterpret_cast<int16_t*>(output)));
EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output));
input += input_size_;
EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input,
reinterpret_cast<int16_t*>(output)));
EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output));
input += input_size_;
int enc_len_bytes = WebRtcIsacfix_Encode(
encoder_, input, reinterpret_cast<int16_t*>(output));
int enc_len_bytes = WebRtcIsacfix_Encode(encoder_, input, output);
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}

12
webrtc/modules/audio_coding/neteq/test/RTPencode.cc
View File

@@ -1632,9 +1632,13 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e
cdlen=0;
while (cdlen<=0) {
#ifdef CODEC_ISAC /* floating point */
cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
cdlen = WebRtcIsac_Encode(ISAC_inst[k],
&indata[noOfCalls * 160],
encoded);
#else /* fixed point */
cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
cdlen = WebRtcIsacfix_Encode(ISAC_inst[k],
&indata[noOfCalls * 160],
encoded);
#endif
noOfCalls++;
}
@@ -1645,7 +1649,9 @@ int NetEQTest_encode(int coder, int16_t *indata, int frameLen, unsigned char * e
int noOfCalls=0;
cdlen=0;
while (cdlen<=0) {
cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(int16_t*)encoded);
cdlen = WebRtcIsac_Encode(ISACSWB_inst[k],
&indata[noOfCalls * 320],
encoded);
noOfCalls++;
}
}

3
webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
View File

@@ -138,8 +138,7 @@ int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
// The Isac encoder does not perform encoding (and returns 0) until it
// receives a sequence of sub-blocks that amount to the frame duration.
EXPECT_EQ(0, value);
value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer],
reinterpret_cast<int16_t*>(payload));
value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer], payload);
}
EXPECT_GT(value, 0);
return value;