Check return result of fwrite [Audio Module]

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

src/modules/audio_coding/codecs/cng/test/CNG.cc

@@ -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);
+        if (fwrite(genSpeech, 2, frameLen,
-        fwrite(state,2,frameLen,statefile);
+                   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;
 }
-
-
-

src/modules/audio_coding/codecs/g711/test/testG711.cc

@@ -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;
+          }
         }
     }
 

src/modules/audio_coding/codecs/g722/test/testG722.cc

@@ -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;
 }
-

src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c

@@ -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 {

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

@@ -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),
+                        if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1,
-                            1, transcodingBitstream);
+                                   transcodingBitstream) != 1) {
+                          return -1;
+                        }
 
                         auxUW8 = (WebRtc_UWord8)(streamLenTransCoding & 0x00FF);
-                        fwrite(&auxUW8, sizeof(WebRtc_UWord8),
+                        if (fwrite(&auxUW8, sizeof(WebRtc_UWord8),
-                            1, transcodingBitstream);
+                                   1, transcodingBitstream) != 1) {
+                          return -1;
+                        }
 
-                        fwrite((WebRtc_UWord8*)streamDataTransCoding, sizeof(WebRtc_UWord8),
+                        if (fwrite((WebRtc_UWord8*)streamDataTransCoding,
-                            streamLenTransCoding, transcodingBitstream);
+                                   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;
+          }
         }
 
 

src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc

@@ -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,
+          if (fwrite(&(packetData[senderIdx]->arrival_time),
-                 arrivalTimeFile1);
+                     sizeof(unsigned int),
+                     1, arrivalTimeFile1) != 1) {
+            return -1;
+          }
         }
         else
         {
-          fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
+          if (fwrite(&(packetData[senderIdx]->arrival_time),
-                 arrivalTimeFile2);
+                     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,
+          if (fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
-                 outFile[receiverIdx]);
+                     outFile[receiverIdx]) !=
+              static_cast<size_t>(lenDecodedAudio << 1)) {
+            return -1;
+          }
         }
         else
         {
-          fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
+          if (fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
-                 outFile[receiverIdx]);
+                     outFile[receiverIdx]) !=
+              static_cast<size_t>(lenDecodedAudio)) {
+            return -1;
+          }
         }
         num10ms[senderIdx] = 0;
       }

src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c

@@ -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);
+                  if (fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp) != 1) {
-			fwrite(payload, 1, stream_len, outp);
+                    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

src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c

@@ -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 */

src/modules/audio_coding/main/source/audio_coding_module_impl.cc

@@ -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);
+      if (fwrite(&rtp_timestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL) != 1) {
-      fwrite(&current_payload_type, sizeof(WebRtc_UWord8), 1, _outgoingPL);
+        return -1;
-      fwrite(&length_bytes, sizeof(WebRtc_Word16), 1, _outgoingPL);
+      }
+      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,
+      if (fwrite(&rtp_info.header.timestamp, sizeof(WebRtc_UWord32),
-             _incomingPL);
+                 1, _incomingPL) != 1) {
-      fwrite(&rtp_info.header.payloadType, sizeof(WebRtc_UWord8), 1,
+        return -1;
-             _incomingPL);
+      }
-      fwrite(&payload_length, sizeof(WebRtc_Word16), 1, _incomingPL);
+      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
 

src/modules/audio_coding/main/test/PCMFile.cc

@@ -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),
+      if (fwrite(audio_frame.data_, sizeof(WebRtc_UWord16),
-             audio_frame.samples_per_channel_, pcm_file_);
+                 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),
+      if (fwrite(stereo_audio, sizeof(WebRtc_Word16),
-             2 * audio_frame.samples_per_channel_, pcm_file_);
+                 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() {

src/modules/audio_coding/main/test/RTPFile.cc

@@ -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
+    WebRtc_UWord32 dummy_variable = 0;
-    fwrite(&dummy_variable, 4, 1, _rtpFile);
+    // should be converted to network endian format, but does not matter when 0
-    fwrite(&dummy_variable, 4, 1, _rtpFile);
+    if (fwrite(&dummy_variable, 4, 1, _rtpFile) != 1) {
-    fwrite(&dummy_variable, 4, 1, _rtpFile);
+      return;
-    fwrite(&dummy_variable, 2, 1, _rtpFile);
+    }
-    fwrite(&dummy_variable, 2, 1, _rtpFile);
+    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);
+    if (fwrite(&lengthBytes, 2, 1, _rtpFile) != 1) {
-    fwrite(&plen, 2, 1, _rtpFile);
+      return;
-    fwrite(&offsetMs, 4, 1, _rtpFile);
+    }
-    fwrite(rtpHeader, 12, 1, _rtpFile);
+    if (fwrite(&plen, 2, 1, _rtpFile) != 1) {
-    fwrite(payloadData, 1, payloadSize, _rtpFile);
+      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,

src/modules/audio_coding/neteq/automode.c

@@ -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 */

src/modules/audio_coding/neteq/packet_buffer.c

@@ -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 );
+    if ((fwrite(&temp_var, sizeof(int),
-    fwrite( &RTPpacket->timeStamp, sizeof(WebRtc_UWord32), 1, delay_fid2 );
+                1, delay_fid2) != 1) ||
-    fwrite( &RTPpacket->seqNumber, sizeof(WebRtc_UWord16), 1, delay_fid2 );
+        (fwrite(&RTPpacket->timeStamp, sizeof(WebRtc_UWord32),
-    fwrite( &RTPpacket->payloadType, sizeof(int), 1, delay_fid2 );
+                1, delay_fid2) != 1) ||
-    fwrite( &RTPpacket->payloadLen, sizeof(WebRtc_Word16), 1, delay_fid2 );
+        (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 */
 

src/modules/audio_coding/neteq/recout.c

@@ -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);
+            if ((fwrite(&temp_var, sizeof(int),
-            fwrite(&inst->fs, sizeof(WebRtc_UWord16), 1, delay_fid2);
+                        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);
+            if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
-            fwrite(&inst->endTimestamp, sizeof(WebRtc_UWord32), 1, delay_fid2);
+              return -1;
-            fwrite(&dspInfo->samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
+            }
+            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;

src/modules/audio_coding/neteq/signal_mcu.c

@@ -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);
+            if ((fwrite(&temp_var, sizeof(int),
-            fwrite(&temp_pkt.timeStamp,sizeof(WebRtc_UWord32),1,delay_fid2);
+                        1, delay_fid2) != 1) ||
-            fwrite(&dspInfo.samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
+                (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;
 
 }
-

src/modules/audio_coding/neteq/test/NetEqRTPplay.cc

@@ -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);
+        if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
-        fwrite(&clock_float, sizeof(float),1,delay_fid2);
+          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);
+    if (fwrite(&temp_var, sizeof(int), 1, delay_fid2) != 1) {
-    fwrite(&tot_received_packets,sizeof(WebRtc_UWord32),1,delay_fid2);
+      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

src/modules/audio_coding/neteq/test/RTPencode.cc

@@ -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);
+        if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
-	fwrite(&dummy_variable, 4, 1, out_file);
+          return -1;
-	fwrite(&dummy_variable, 4, 1, out_file);
+        }
-	fwrite(&dummy_variable, 2, 1, out_file);
+        if (fwrite(&dummy_variable, 4, 1, out_file) != 1) {
-	fwrite(&dummy_variable, 2, 1, out_file);
+          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);
+            if (fwrite(&length, 2, 1, out_file) != 1) {
-            fwrite(&plen, 2, 1, out_file);
+              return -1;
-            fwrite(&offset, 4, 1, out_file);
+            }
-            fwrite(rtp_data, 12 + enc_len, 1, out_file);
+            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);
+                          if (fwrite(&length, 2, 1, out_file) != 1) {
-			fwrite(&plen, 2, 1, out_file);
+                            return -1;
-			fwrite(&offset, 4, 1, out_file);
+                          }
+                          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);
+                                  if (fwrite(&old_length, 2, 1,
-					fwrite(&old_plen, 2, 1, out_file);
+                                             out_file) != 1) {
-					fwrite(&offset, 4, 1, out_file);
+                                    return -1;
-					fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file);
+                                  }
+                                  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;

src/modules/audio_coding/neteq/test/RTPjitter.cc

@@ -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;
+                        }
 		}
 	}