Revert 7266 "WebRtcIsac_Encode and WebRtcIsacfix_Encode: Type en..."
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
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 942727a..7cd2954 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h
@@ -130,7 +130,7 @@
int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
- uint8_t* encoded);
+ int16_t *encoded);
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 48631b5..e855dae 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -17,7 +17,6 @@
#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"
@@ -356,7 +355,7 @@
int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
const int16_t *speechIn,
- uint8_t* encoded)
+ int16_t *encoded)
{
ISACFIX_SubStruct *ISAC_inst;
int16_t stream_len;
@@ -383,20 +382,16 @@
return -1;
}
- assert(stream_len % 2 == 0);
+
+ /* convert from bytes to int16_t */
#ifndef WEBRTC_ARCH_BIG_ENDIAN
- /* 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 / 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. */
+ 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));
}
+
#else
- /* The encoded data vector and our internal representation are both
- big-endian. */
- memcpy(encoded, ISAC_inst->ISACenc_obj.bitstr_obj.stream, stream_len);
+ WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
#endif
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index 207ee8c..3582751 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -71,7 +71,7 @@
size_t pointer = 0;
for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
- bit_stream);
+ reinterpret_cast<int16_t*>(bit_stream));
}
clocks = clock() - clocks;
EXPECT_GT(value, 0);
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 91c4d76..c3c6f13 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -565,7 +565,7 @@
/* Encode */
stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
shortdata,
- (uint8_t*)streamdata);
+ (int16_t*)streamdata);
/* If packet is ready, and CE testing, call the different API
functions from the internal API. */
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index 965f2bc..d5682b2 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -439,9 +439,7 @@
/* iSAC encoding */
if (mode==0 || mode ==1) {
- stream_len = WebRtcIsac_Encode(ISAC_main_inst,
- shortdata,
- (uint8_t*)streamdata);
+ stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata, streamdata);
if (stream_len < 0) {
/* exit if returned with error */
errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
@@ -451,10 +449,7 @@
} else if (mode==2 || mode==3) {
/* iSAC encoding */
if (nbTest != 1)
- stream_len = WebRtcIsacfix_Encode(
- ISACFIX_main_inst,
- shortdata,
- (uint8_t*)streamdata);
+ stream_len = WebRtcIsacfix_Encode(ISACFIX_main_inst, shortdata, streamdata);
else
stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);
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..76a61e6 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h
@@ -147,7 +147,7 @@
int16_t WebRtcIsac_Encode(
ISACStruct* ISAC_main_inst,
const int16_t* speechIn,
- uint8_t* encoded);
+ int16_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..d47eb80 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -496,7 +496,7 @@
*/
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 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 @@
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 @@
* 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 @@
* 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 @@
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;
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
index 1d65373..3c55bd3 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
@@ -31,7 +31,7 @@
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.
};
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..c5f9561 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
@@ -662,8 +662,8 @@
if(!(testNum == 3 && framecnt == 0))
{
stream_len = WebRtcIsac_Encode(ISAC_main_inst,
- shortdata,
- (uint8_t*)streamdata);
+ shortdata,
+ (int16_t*)streamdata);
if((payloadSize != 0) && (stream_len > payloadSize))
{
if(testNum == 0)
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..fd70eca 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
@@ -283,8 +283,7 @@
streamLen = WebRtcIsac_Encode(codecInstance[senderIdx],
- audioBuff10ms,
- (uint8_t*)bitStream);
+ audioBuff10ms, (short*)bitStream);
int16_t ggg;
if (streamLen > 0) {
if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx],
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..980465d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -373,10 +373,8 @@
cur_framesmpls += samplesIn10Ms;
//-------- iSAC encoding ---------
- stream_len = WebRtcIsac_Encode(
- ISAC_main_inst,
- shortdata,
- (uint8_t*)payload);
+ stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata,
+ (int16_t*)payload);
if(stream_len < 0)
{
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
index f3682f1..850204f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_isac.cc
@@ -347,9 +347,8 @@
return -1;
}
*bitstream_len_byte = ACM_ISAC_ENCODE(
- codec_inst_ptr_->inst,
- &in_audio_[in_audio_ix_read_],
- bitstream);
+ codec_inst_ptr_->inst, &in_audio_[in_audio_ix_read_],
+ reinterpret_cast<int16_t*>(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_;
diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
index 624e6a4..d33c8f2 100644
--- a/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -348,11 +348,14 @@
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, output));
+ EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
+ reinterpret_cast<int16_t*>(output)));
input += input_size_;
- EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output));
+ EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
+ reinterpret_cast<int16_t*>(output)));
input += input_size_;
- int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output);
+ int enc_len_bytes =
+ WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output));
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}
@@ -385,11 +388,14 @@
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, output));
+ EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
+ reinterpret_cast<int16_t*>(output)));
input += input_size_;
- EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output));
+ EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input,
+ reinterpret_cast<int16_t*>(output)));
input += input_size_;
- int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output);
+ int enc_len_bytes =
+ WebRtcIsac_Encode(encoder_, input, reinterpret_cast<int16_t*>(output));
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}
@@ -435,11 +441,14 @@
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, output));
+ EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input,
+ reinterpret_cast<int16_t*>(output)));
input += input_size_;
- EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output));
+ EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input,
+ reinterpret_cast<int16_t*>(output)));
input += input_size_;
- int enc_len_bytes = WebRtcIsacfix_Encode(encoder_, input, output);
+ int enc_len_bytes = WebRtcIsacfix_Encode(
+ encoder_, input, reinterpret_cast<int16_t*>(output));
EXPECT_GT(enc_len_bytes, 0);
return enc_len_bytes;
}
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index 92bccee..93b366b 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -1632,13 +1632,9 @@
cdlen=0;
while (cdlen<=0) {
#ifdef CODEC_ISAC /* floating point */
- cdlen = WebRtcIsac_Encode(ISAC_inst[k],
- &indata[noOfCalls * 160],
- encoded);
+ cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
#else /* fixed point */
- cdlen = WebRtcIsacfix_Encode(ISAC_inst[k],
- &indata[noOfCalls * 160],
- encoded);
+ cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(int16_t*)encoded);
#endif
noOfCalls++;
}
@@ -1649,9 +1645,7 @@
int noOfCalls=0;
cdlen=0;
while (cdlen<=0) {
- cdlen = WebRtcIsac_Encode(ISACSWB_inst[k],
- &indata[noOfCalls * 320],
- encoded);
+ cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(int16_t*)encoded);
noOfCalls++;
}
}
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index b672a0c..6b0f482 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -138,7 +138,8 @@
// 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], payload);
+ value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer],
+ reinterpret_cast<int16_t*>(payload));
}
EXPECT_GT(value, 0);
return value;
