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 942727ab4e..7cd2954408 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 @@ extern "C" { 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 48631b5090..e855daea35 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 #include #include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" @@ -356,7 +355,7 @@ int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst, 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 @@ int16_t WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst, 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 207ee8c301..358275111f 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 @@ 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], - bit_stream); + reinterpret_cast(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 91c4d76ab8..c3c6f13529 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 @@ int main(int argc, char* argv[]) /* 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 965f2bc1fe..d5682b2197 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 @@ int main(int argc, char* argv[]) /* 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 @@ 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, - (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 4067058028..76a61e6d33 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 @@ extern "C" { 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 13170a0844..d47eb80b9b 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_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; 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 1d653731af..3c55bd3dc2 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 @@ 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. }; 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 8af4e6f992..c5f9561b07 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 @@ int main(int argc, char* argv[]) 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 72d3fe861d..fd70eca714 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 @@ int main(int argc, char* argv[]) 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 2df5a84e47..980465d653 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 @@ valid values are 8 and 16.\n", sampFreqKHz); 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 f3682f1e3b..850204f31a 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 @@ 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_], - bitstream); + codec_inst_ptr_->inst, &in_audio_[in_audio_ix_read_], + reinterpret_cast(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 624e6a4d28..d33c8f23da 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 @@ 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, output)); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, + reinterpret_cast(output))); input += input_size_; - EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output)); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, + reinterpret_cast(output))); input += input_size_; - int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output); + int enc_len_bytes = + WebRtcIsac_Encode(encoder_, input, reinterpret_cast(output)); EXPECT_GT(enc_len_bytes, 0); return enc_len_bytes; } @@ -385,11 +388,14 @@ 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, output)); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, + reinterpret_cast(output))); input += input_size_; - EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, output)); + EXPECT_EQ(0, WebRtcIsac_Encode(encoder_, input, + reinterpret_cast(output))); input += input_size_; - int enc_len_bytes = WebRtcIsac_Encode(encoder_, input, output); + int enc_len_bytes = + WebRtcIsac_Encode(encoder_, input, reinterpret_cast(output)); EXPECT_GT(enc_len_bytes, 0); return enc_len_bytes; } @@ -435,11 +441,14 @@ 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, output)); + EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, + reinterpret_cast(output))); input += input_size_; - EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, output)); + EXPECT_EQ(0, WebRtcIsacfix_Encode(encoder_, input, + reinterpret_cast(output))); input += input_size_; - int enc_len_bytes = WebRtcIsacfix_Encode(encoder_, input, output); + int enc_len_bytes = WebRtcIsacfix_Encode( + encoder_, input, reinterpret_cast(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 92bccee2fa..93b366b9a2 100644 --- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc +++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc @@ -1632,13 +1632,9 @@ 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], - 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 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], - 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 b672a0c2b1..6b0f48286e 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 @@ 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], payload); + value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer], + reinterpret_cast(payload)); } EXPECT_GT(value, 0); return value;