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Check return result of fwrite [Audio Module]

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Description:
On ChromeOS/ARM, compiler enforces to check return result of a function.
Currently, we don't check return result of fwrite, it causes building errors.

The following files need to patch. The patch should be similar, before I patch all
of them, I will start with 2 files, please take a quick look, if the patch is OK,
I will continue and upload a new patch that covers all of them.
it to all of them.
Review URL: https://webrtc-codereview.appspot.com/566016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2345 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
leozwang@webrtc.org
2012-06-01 17:46:21 +00:00
parent c3b2683bf4
commit 354b0ed015
18 changed files with 393 additions and 171 deletions
Download Patch File Download Diff File Expand all files Collapse all files

19
src/modules/audio_coding/codecs/cng/test/CNG.cc
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -120,7 +120,6 @@ int main(int argc, char* argv[])
exit(EXIT_FAILURE);
}
static bool firstSilent=true;
int numSamp=0;
@@ -155,7 +154,6 @@ int main(int argc, char* argv[])
exit(EXIT_FAILURE);
}
firstSilent=false;
res=WebRtcCng_Encode(e_inst, &anaSpeech[frameLen/2], frameLen/2, SIDpkt,&size,1);
@@ -210,16 +208,17 @@ int main(int argc, char* argv[])
memset(state,0,frameLen*2);
}
fwrite(genSpeech,2,frameLen,outfile);
fwrite(state,2,frameLen,statefile);
if (fwrite(genSpeech, 2, frameLen,
outfile) != static_cast<size_t>(frameLen)) {
return -1;
}
if (fwrite(state, 2, frameLen,
statefile) != static_cast<size_t>(frameLen)) {
return -1;
}
}
fclose(infile);
fclose(outfile);
fclose(statefile);
return 0;
}

17
src/modules/audio_coding/codecs/g711/test/testG711.cc
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -130,7 +130,10 @@ int main(int argc, char* argv[])
stream_len = WebRtcG711_EncodeA(NULL, shortdata, framelength, streamdata);
if (argc==6){
/* Write bits to file */
fwrite(streamdata,sizeof(unsigned char),stream_len,bitp);
if (fwrite(streamdata, sizeof(unsigned char), stream_len,
bitp) != static_cast<size_t>(stream_len)) {
return -1;
}
}
err = WebRtcG711_DecodeA(NULL, streamdata, stream_len, decoded, speechType);
} else if (!strcmp(law,"u")){
@@ -138,7 +141,10 @@ int main(int argc, char* argv[])
stream_len = WebRtcG711_EncodeU(NULL, shortdata, framelength, streamdata);
if (argc==6){
/* Write bits to file */
fwrite(streamdata,sizeof(unsigned char),stream_len,bitp);
if (fwrite(streamdata, sizeof(unsigned char), stream_len,
bitp) != static_cast<size_t>(stream_len)) {
return -1;
}
}
err = WebRtcG711_DecodeU(NULL, streamdata, stream_len, decoded, speechType);
} else {
@@ -150,7 +156,10 @@ int main(int argc, char* argv[])
printf("Error in encoder/decoder\n");
} else {
/* Write coded speech to file */
fwrite(decoded,sizeof(short),framelength,outp);
if (fwrite(decoded, sizeof(short), framelength,
outp) != static_cast<size_t>(framelength)) {
return -1;
}
}
}

13
src/modules/audio_coding/codecs/g722/test/testG722.cc
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -134,9 +134,15 @@ int main(int argc, char* argv[])
printf("Error in encoder/decoder\n");
} else {
/* Write coded bits to file */
fwrite(streamdata,sizeof(short),stream_len/2,outbitp);
if (fwrite(streamdata, sizeof(short), stream_len/2,
outbitp) != static_cast<size_t>(stream_len/2)) {
return -1;
}
/* Write coded speech to file */
fwrite(decoded,sizeof(short),framelength,outp);
if (fwrite(decoded, sizeof(short), framelength,
outp) != static_cast<size_t>(framelength)) {
return -1;
}
}
}
@@ -154,4 +160,3 @@ int main(int argc, char* argv[])
return 0;
}

12
src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -628,7 +628,10 @@ int main(int argc, char* argv[])
errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
printf("\nError in encoder: %d.\n", errtype);
} else {
fwrite(streamdata, sizeof(char), stream_len, outbits);
if (fwrite(streamdata, sizeof(char),
stream_len, outbits) != (size_t)stream_len) {
return -1;
}
}
cur_framesmpls += FRAMESAMPLES_10ms;
@@ -777,7 +780,10 @@ int main(int argc, char* argv[])
}
/* Write decoded speech frame to file */
fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
if (fwrite(decoded, sizeof(WebRtc_Word16),
declen, outp) != (size_t)declen) {
return -1;
}
// fprintf( ratefile, "%f \n", stream_len / ( ((double)declen)/
// ((double)FS) ) * 8 );
} else {

33
src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -692,15 +692,24 @@ int main(int argc, char* argv[])
exit(0);
}
auxUW8 = (WebRtc_UWord8)(((streamLenTransCoding & 0xFF00) >> 8) & 0x00FF);
fwrite(&auxUW8, sizeof(WebRtc_UWord8),
1, transcodingBitstream);
if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1,
transcodingBitstream) != 1) {
return -1;
}
auxUW8 = (WebRtc_UWord8)(streamLenTransCoding & 0x00FF);
fwrite(&auxUW8, sizeof(WebRtc_UWord8),
1, transcodingBitstream);
if (fwrite(&auxUW8, sizeof(WebRtc_UWord8),
1, transcodingBitstream) != 1) {
return -1;
}
fwrite((WebRtc_UWord8*)streamDataTransCoding, sizeof(WebRtc_UWord8),
streamLenTransCoding, transcodingBitstream);
if (fwrite((WebRtc_UWord8*)streamDataTransCoding,
sizeof(WebRtc_UWord8),
streamLenTransCoding,
transcodingBitstream) !=
static_cast<size_t>(streamLenTransCoding)) {
return -1;
}
WebRtcIsac_ReadBwIndex((WebRtc_Word16*)streamDataTransCoding, &indexStream);
if(indexStream != bnIdxTC)
@@ -939,12 +948,18 @@ int main(int argc, char* argv[])
/* Write decoded speech frame to file */
if((declen > 0) && (numFileLoop == 0))
{
fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
if (fwrite(decoded, sizeof(WebRtc_Word16), declen,
outp) != static_cast<size_t>(declen)) {
return -1;
}
}
if((declenTC > 0) && (numFileLoop == 0))
{
fwrite(decodedTC, sizeof(WebRtc_Word16), declen, transCodingFile);
if (fwrite(decodedTC, sizeof(WebRtc_Word16), declen,
transCodingFile) != static_cast<size_t>(declen)) {
return -1;
}
}

30
src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -392,13 +392,19 @@ int main(int argc, char* argv[])
// Write the arrival time.
if(senderIdx == 0)
{
fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
arrivalTimeFile1);
if (fwrite(&(packetData[senderIdx]->arrival_time),
sizeof(unsigned int),
1, arrivalTimeFile1) != 1) {
return -1;
}
}
else
{
fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
arrivalTimeFile2);
if (fwrite(&(packetData[senderIdx]->arrival_time),
sizeof(unsigned int),
1, arrivalTimeFile2) != 1) {
return -1;
}
}
// BWE
@@ -426,13 +432,19 @@ int main(int argc, char* argv[])
{
WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, resampledAudio60ms,
resamplerState[receiverIdx]);
fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
outFile[receiverIdx]);
if (fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
outFile[receiverIdx]) !=
static_cast<size_t>(lenDecodedAudio << 1)) {
return -1;
}
}
else
{
fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
outFile[receiverIdx]);
if (fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
outFile[receiverIdx]) !=
static_cast<size_t>(lenDecodedAudio)) {
return -1;
}
}
num10ms[senderIdx] = 0;
}

20
src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -431,11 +431,18 @@ valid values are 8 and 16.\n", sampFreqKHz);
{
WebRtc_UWord8 auxUW8;
auxUW8 = (WebRtc_UWord8)(((stream_len & 0x7F00) >> 8) & 0xFF);
fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp);
if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp) != 1) {
return -1;
}
auxUW8 = (WebRtc_UWord8)(stream_len & 0xFF);
fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp);
fwrite(payload, 1, stream_len, outp);
if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp) != 1) {
return -1;
}
if (fwrite(payload, 1, stream_len,
outp) != (size_t)stream_len) {
return -1;
}
}
else
{
@@ -462,7 +469,10 @@ valid values are 8 and 16.\n", sampFreqKHz);
}
// Write decoded speech frame to file
fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
if (fwrite(decoded, sizeof(WebRtc_Word16),
declen, outp) != (size_t)declen) {
return -1;
}
cur_framesmpls = declen;
}
// Update Statistics

13
src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
View File

@@ -1,5 +1,5 @@
/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
@@ -168,7 +168,11 @@ int main(int argc, char* argv[])
/* write byte file */
fwrite(encoded_data, sizeof(WebRtc_Word16), ((len+1)/sizeof(WebRtc_Word16)), efileid);
if (fwrite(encoded_data, sizeof(WebRtc_Word16),
((len+1)/sizeof(WebRtc_Word16)), efileid) !=
(size_t)(((len+1)/sizeof(WebRtc_Word16)))) {
return -1;
}
/* get channel data if provided */
if (argc==6) {
@@ -204,7 +208,10 @@ int main(int argc, char* argv[])
/* write output file */
fwrite(decoded_data,sizeof(WebRtc_Word16),len,ofileid);
if (fwrite(decoded_data, sizeof(WebRtc_Word16), len,
ofileid) != (size_t)len) {
return -1;
}
}
/* close files */

30
src/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

@@ -431,9 +431,16 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
CriticalSectionScoped lock(*_callbackCritSect);
#ifdef ACM_QA_TEST
if(_outgoingPL != NULL) {
fwrite(&rtp_timestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL);
fwrite(&current_payload_type, sizeof(WebRtc_UWord8), 1, _outgoingPL);
fwrite(&length_bytes, sizeof(WebRtc_Word16), 1, _outgoingPL);
if (fwrite(&rtp_timestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL) != 1) {
return -1;
}
if (fwrite(&current_payload_type, sizeof(WebRtc_UWord8),
1, _outgoingPL) != 1) {
return -1;
}
if (fwrite(&length_bytes, sizeof(WebRtc_Word16), 1, _outgoingPL) != 1) {
return -1;
}
}
#endif
@@ -1489,11 +1496,18 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPacket(
CriticalSectionScoped lock(*_acmCritSect);
#ifdef ACM_QA_TEST
if(_incomingPL != NULL) {
fwrite(&rtp_info.header.timestamp, sizeof(WebRtc_UWord32), 1,
_incomingPL);
fwrite(&rtp_info.header.payloadType, sizeof(WebRtc_UWord8), 1,
_incomingPL);
fwrite(&payload_length, sizeof(WebRtc_Word16), 1, _incomingPL);
if (fwrite(&rtp_info.header.timestamp, sizeof(WebRtc_UWord32),
1, _incomingPL) != 1) {
return -1;
}
if (fwrite(&rtp_info.header.payloadType, sizeof(WebRtc_UWord8),
1, _incomingPL) != 1) {
return -1;
}
if (fwrite(&payload_length, sizeof(WebRtc_Word16),
1, _incomingPL) != 1) {
return -1;
}
}
#endif

26
src/modules/audio_coding/main/test/PCMFile.cc
View File

@@ -181,8 +181,11 @@ WebRtc_Word32 PCMFile::Read10MsData(AudioFrame& audio_frame) {
void PCMFile::Write10MsData(AudioFrame& audio_frame) {
if (audio_frame.num_channels_ == 1) {
if (!save_stereo_) {
fwrite(audio_frame.data_, sizeof(WebRtc_UWord16),
audio_frame.samples_per_channel_, pcm_file_);
if (fwrite(audio_frame.data_, sizeof(WebRtc_UWord16),
audio_frame.samples_per_channel_, pcm_file_) !=
static_cast<size_t>(audio_frame.samples_per_channel_)) {
return;
}
} else {
WebRtc_Word16* stereo_audio =
new WebRtc_Word16[2 * audio_frame.samples_per_channel_];
@@ -191,20 +194,29 @@ void PCMFile::Write10MsData(AudioFrame& audio_frame) {
stereo_audio[k << 1] = audio_frame.data_[k];
stereo_audio[(k << 1) + 1] = audio_frame.data_[k];
}
fwrite(stereo_audio, sizeof(WebRtc_Word16),
2 * audio_frame.samples_per_channel_, pcm_file_);
if (fwrite(stereo_audio, sizeof(WebRtc_Word16),
2 * audio_frame.samples_per_channel_, pcm_file_) !=
static_cast<size_t>(2 * audio_frame.samples_per_channel_)) {
return;
}
delete[] stereo_audio;
}
} else {
fwrite(audio_frame.data_, sizeof(WebRtc_Word16),
if (fwrite(audio_frame.data_, sizeof(WebRtc_Word16),
audio_frame.num_channels_ * audio_frame.samples_per_channel_,
pcm_file_);
pcm_file_) != static_cast<size_t>(
audio_frame.num_channels_ * audio_frame.samples_per_channel_)) {
return;
}
}
}
void PCMFile::Write10MsData(WebRtc_Word16* playout_buffer,
WebRtc_UWord16 length_smpls) {
fwrite(playout_buffer, sizeof(WebRtc_UWord16), length_smpls, pcm_file_);
if (fwrite(playout_buffer, sizeof(WebRtc_UWord16),
length_smpls, pcm_file_) != length_smpls) {
return;
}
}
void PCMFile::Close() {

43
src/modules/audio_coding/main/test/RTPFile.cc
View File

@@ -165,12 +165,23 @@ void RTPFile::WriteHeader()
{
// Write data in a format that NetEQ and RTP Play can parse
fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
fwrite(&dummy_variable, 4, 1, _rtpFile);
fwrite(&dummy_variable, 4, 1, _rtpFile);
fwrite(&dummy_variable, 4, 1, _rtpFile);
fwrite(&dummy_variable, 2, 1, _rtpFile);
fwrite(&dummy_variable, 2, 1, _rtpFile);
WebRtc_UWord32 dummy_variable = 0;
// should be converted to network endian format, but does not matter when 0
if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&dummy_variable, 2, 1, _rtpFile) != 1) {
return;
}
fflush(_rtpFile);
}
@@ -205,11 +216,21 @@ void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeSt
offsetMs = (timeStamp/(frequency/1000));
offsetMs = htonl(offsetMs);
fwrite(&lengthBytes, 2, 1, _rtpFile);
fwrite(&plen, 2, 1, _rtpFile);
fwrite(&offsetMs, 4, 1, _rtpFile);
fwrite(rtpHeader, 12, 1, _rtpFile);
fwrite(payloadData, 1, payloadSize, _rtpFile);
if (fwrite(&lengthBytes, 2, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&plen, 2, 1, _rtpFile) != 1) {
return;
}
if (fwrite(&offsetMs, 4, 1, _rtpFile) != 1) {
return;
}
if (fwrite(rtpHeader, 12, 1, _rtpFile) != 1) {
return;
}
if (fwrite(payloadData, 1, payloadSize, _rtpFile) != payloadSize) {
return;
}
}
WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,

8
src/modules/audio_coding/neteq/automode.c
View File

@@ -378,7 +378,9 @@ WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32
#ifdef NETEQ_DELAY_LOGGING
/* special code for offline delay logging */
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF;
fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
if (fwrite( &temp_var, sizeof(int), 1, delay_fid2 ) != 1) {
return -1;
}
temp_var = (int) (Bopt * inst->packetSpeechLenSamp);
#endif
@@ -518,7 +520,9 @@ WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32
#ifdef NETEQ_DELAY_LOGGING
/* special code for offline delay logging */
fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
if (fwrite( &temp_var, sizeof(int), 1, delay_fid2 ) != 1) {
return -1;
}
#endif
/* Sanity check: Bopt must be strictly positive */

21
src/modules/audio_coding/neteq/packet_buffer.c
View File

@@ -317,14 +317,23 @@ int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *R
if (*flushed)
{
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_FLUSH;
fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
}
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_RECIN;
fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
fwrite( &RTPpacket->timeStamp, sizeof(WebRtc_UWord32), 1, delay_fid2 );
fwrite( &RTPpacket->seqNumber, sizeof(WebRtc_UWord16), 1, delay_fid2 );
fwrite( &RTPpacket->payloadType, sizeof(int), 1, delay_fid2 );
fwrite( &RTPpacket->payloadLen, sizeof(WebRtc_Word16), 1, delay_fid2 );
if ((fwrite(&temp_var, sizeof(int),
1, delay_fid2) != 1) ||
(fwrite(&RTPpacket->timeStamp, sizeof(WebRtc_UWord32),
1, delay_fid2) != 1) ||
(fwrite(&RTPpacket->seqNumber, sizeof(WebRtc_UWord16),
1, delay_fid2) != 1) ||
(fwrite(&RTPpacket->payloadType, sizeof(int),
1, delay_fid2) != 1) ||
(fwrite(&RTPpacket->payloadLen, sizeof(WebRtc_Word16),
1, delay_fid2) != 1)) {
return -1;
}
tot_received_packets++;
#endif /* NETEQ_DELAY_LOGGING */

78
src/modules/audio_coding/neteq/recout.c
View File

@@ -318,8 +318,12 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
}
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
fwrite(&inst->fs, sizeof(WebRtc_UWord16), 1, delay_fid2);
if ((fwrite(&temp_var, sizeof(int),
1, delay_fid2) != 1) ||
(fwrite(&inst->fs, sizeof(WebRtc_UWord16),
1, delay_fid2) != 1)) {
return -1;
}
#endif
}
@@ -508,9 +512,17 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
pw16_decoded_buffer, &speechType);
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
fwrite(&inst->endTimestamp, sizeof(WebRtc_UWord32), 1, delay_fid2);
fwrite(&dspInfo->samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
if (fwrite(&inst->endTimestamp, sizeof(WebRtc_UWord32),
1, delay_fid2) != 1) {
return -1;
}
if (fwrite(&dspInfo->samplesLeft, sizeof(WebRtc_UWord16),
1, delay_fid2) != 1) {
return -1;
}
tot_received_packets++;
#endif
}
@@ -692,7 +704,9 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
case DSP_INSTR_MERGE:
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = -len;
#endif
/* Call Merge with history*/
@@ -710,7 +724,9 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
#ifdef NETEQ_DELAY_LOGGING
temp_var += len;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
/* If last packet was decoded as a inband CNG set mode to CNG instead */
if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
@@ -756,9 +772,13 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
inst->w16_concealedTS += len;
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = len;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
len = 0; /* already written the data, so do not write it again further down. */
}
@@ -812,9 +832,13 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
inst->curPosition += (borrowedSamples - len);
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = 3 * inst->timestampsPerCall - len;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
len = 0;
}
@@ -827,9 +851,13 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
(len-borrowedSamples));
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = 3 * inst->timestampsPerCall - len;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
len = len - borrowedSamples;
}
@@ -839,7 +867,9 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
{
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = len;
#endif
return_value = WebRtcNetEQ_Accelerate(inst,
@@ -856,7 +886,9 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
#ifdef NETEQ_DELAY_LOGGING
temp_var -= len;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
}
/* If last packet was decoded as a inband CNG set mode to CNG instead */
@@ -1147,9 +1179,13 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = len - w16_tmp1; /* number of samples added */
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
/* If last packet was decoded as inband CNG, set mode to CNG instead */
if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
@@ -1256,9 +1292,13 @@ int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
inst->w16_mode = MODE_FADE_TO_BGN;
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
temp_var = len;
fwrite(&temp_var, sizeof(int), 1, delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
#endif
break;

12
src/modules/audio_coding/neteq/signal_mcu.c
View File

@@ -646,9 +646,14 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE;
fwrite(&temp_var,sizeof(int),1,delay_fid2);
fwrite(&temp_pkt.timeStamp,sizeof(WebRtc_UWord32),1,delay_fid2);
fwrite(&dspInfo.samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
if ((fwrite(&temp_var, sizeof(int),
1, delay_fid2) != 1) ||
(fwrite(&temp_pkt.timeStamp, sizeof(WebRtc_UWord32),
1, delay_fid2) != 1) ||
(fwrite(&dspInfo.samplesLeft, sizeof(WebRtc_UWord16),
1, delay_fid2) != 1)) {
return -1;
}
#endif
*blockPtr = temp_pkt.payloadLen;
@@ -762,4 +767,3 @@ int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
return 0;
}

21
src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
View File

@@ -561,8 +561,12 @@ int main(int argc, char* argv[])
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CLOCK;
clock_float = (float) simClock;
fwrite(&temp_var,sizeof(int),1,delay_fid2);
fwrite(&clock_float, sizeof(float),1,delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
if (fwrite(&clock_float, sizeof(float), 1, delay_fid2) != 1) {
return -1;
}
#endif
/* time to set extra delay */
if (extraDelay > -1 && simClock >= nextExtraDelayTime) {
@@ -657,7 +661,9 @@ int main(int argc, char* argv[])
}
// write to file
fwrite(out_data,writeLen,2,out_file);
if (fwrite(out_data, writeLen, 2, out_file) != 2) {
return -1;
}
writtenSamples += writeLen;
@@ -678,8 +684,13 @@ int main(int argc, char* argv[])
#ifdef NETEQ_DELAY_LOGGING
temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EOF;
fwrite(&temp_var,sizeof(int),1,delay_fid2);
fwrite(&tot_received_packets,sizeof(WebRtc_UWord32),1,delay_fid2);
if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
return -1;
}
if (fwrite(&tot_received_packets, sizeof(WebRtc_UWord32),
1, delay_fid2) != 1) {
return -1;
}
fprintf(delay_fid2,"End of file\n");
fclose(delay_fid2);
#endif

75
src/modules/audio_coding/neteq/test/RTPencode.cc
View File

@@ -547,11 +547,21 @@ int main(int argc, char* argv[])
//fprintf(out_file, "#!RTPencode%s\n", "1.0");
fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
fwrite(&dummy_variable, 4, 1, out_file);
fwrite(&dummy_variable, 4, 1, out_file);
fwrite(&dummy_variable, 4, 1, out_file);
fwrite(&dummy_variable, 2, 1, out_file);
fwrite(&dummy_variable, 2, 1, out_file);
if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
return -1;
}
if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
return -1;
}
if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
return -1;
}
if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
return -1;
}
if (fwrite(&dummy_variable, 2, 1, out_file) != 1) {
return -1;
}
#ifdef TIMESTAMP_WRAPAROUND
timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */
@@ -600,10 +610,18 @@ int main(int argc, char* argv[])
plen = htons(12 + enc_len);
offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
offset = htonl(offset);
fwrite(&length, 2, 1, out_file);
fwrite(&plen, 2, 1, out_file);
fwrite(&offset, 4, 1, out_file);
fwrite(rtp_data, 12 + enc_len, 1, out_file);
if (fwrite(&length, 2, 1, out_file) != 1) {
return -1;
}
if (fwrite(&plen, 2, 1, out_file) != 1) {
return -1;
}
if (fwrite(&offset, 4, 1, out_file) != 1) {
return -1;
}
if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
return -1;
}
dtmfSent = true;
}
@@ -685,11 +703,18 @@ int main(int argc, char* argv[])
/* write RTP packet to file */
length = htons(12 + enc_len + 8);
plen = htons(12 + enc_len);
offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
offset = (WebRtc_UWord32) sendtime;
//(timestamp/(fs/1000));
offset = htonl(offset);
fwrite(&length, 2, 1, out_file);
fwrite(&plen, 2, 1, out_file);
fwrite(&offset, 4, 1, out_file);
if (fwrite(&length, 2, 1, out_file) != 1) {
return -1;
}
if (fwrite(&plen, 2, 1, out_file) != 1) {
return -1;
}
if (fwrite(&offset, 4, 1, out_file) != 1) {
return -1;
}
#ifdef RANDOM_DATA
for (int k=0; k<12+enc_len; k++) {
rtp_data[k] = rand() + rand();
@@ -700,7 +725,9 @@ int main(int argc, char* argv[])
rtp_data[k] = rand() + rand();
}
#endif
fwrite(rtp_data, 12 + enc_len, 1, out_file);
if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
return -1;
}
#ifdef MULTIPLE_SAME_TIMESTAMP
} while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) );
#endif //MULTIPLE_SAME_TIMESTAMP
@@ -709,10 +736,22 @@ int main(int argc, char* argv[])
if (packet_age >= OLD_PACKET*fs) {
if (!first_old_packet) {
// send the old packet
fwrite(&old_length, 2, 1, out_file);
fwrite(&old_plen, 2, 1, out_file);
fwrite(&offset, 4, 1, out_file);
fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file);
if (fwrite(&old_length, 2, 1,
out_file) != 1) {
return -1;
}
if (fwrite(&old_plen, 2, 1,
out_file) != 1) {
return -1;
}
if (fwrite(&offset, 4, 1,
out_file) != 1) {
return -1;
}
if (fwrite(old_rtp_data, 12 + old_enc_len,
1, out_file) != 1) {
return -1;
}
}
// store current packet as old
old_length=length;

7
src/modules/audio_coding/neteq/test/RTPjitter.cc
View File

@@ -167,7 +167,12 @@ int main(int argc, char* argv[])
}
// write packet to file
fwrite(temp_packet, sizeof(unsigned char), ntohs(*((WebRtc_UWord16*) temp_packet)), out_file);
if (fwrite(temp_packet, sizeof(unsigned char),
ntohs(*((WebRtc_UWord16*) temp_packet)),
out_file) !=
ntohs(*((WebRtc_UWord16*) temp_packet))) {
return -1;
}
}
}