diff --git a/src/modules/audio_coding/main/test/EncodeDecodeTest.cc b/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
index 4afab37c8..02d1c126b 100644
--- a/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ b/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
@@ -198,14 +198,15 @@ bool Receiver::IncomingPacket() {
       _firstTime = false;
       _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
                                                _payloadSizeBytes, &_nextTime);
-      if (_realPayloadSizeBytes < 0) {
-        printf("Error in reading incoming payload.\n");
-        return false;
+      if (_realPayloadSizeBytes == 0) {
+        if (_rtpStream->EndOfFile()) {
+          _firstTime = true;
+          return true;
+        } else {
+          printf("Error in reading incoming payload.\n");
+          return false;
+        }
       }
-      if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
-        _firstTime = true;
-        return true;
-     }
    }
 
    WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
diff --git a/src/modules/audio_coding/main/test/RTPFile.cc b/src/modules/audio_coding/main/test/RTPFile.cc
index 4c71ed749..29d6a1d53 100644
--- a/src/modules/audio_coding/main/test/RTPFile.cc
+++ b/src/modules/audio_coding/main/test/RTPFile.cc
@@ -106,7 +106,7 @@ RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp
 
 WebRtc_UWord16
 RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
-                WebRtc_Word8* payloadData, 
+                WebRtc_Word8* payloadData,
                 WebRtc_UWord16 payloadSize,
                 WebRtc_UWord32* offset)
 {
@@ -125,7 +125,7 @@ RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
     }
     else
     {
-      return -1;
+        return 0;
     }
     *offset = (packet->timeStamp/(packet->frequency/1000));
 
@@ -248,11 +248,11 @@ WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
     {
       return -1;
     }
-    lengthBytes -= 20;
-    if (lengthBytes < 0)
+    if (lengthBytes < 20)
     {
       return -1;
     }
+    lengthBytes -= 20;
     EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
     return lengthBytes;
 }
diff --git a/src/modules/audio_coding/main/test/RTPFile.h b/src/modules/audio_coding/main/test/RTPFile.h
index 86c99a3fc..e11b16061 100644
--- a/src/modules/audio_coding/main/test/RTPFile.h
+++ b/src/modules/audio_coding/main/test/RTPFile.h
@@ -24,12 +24,15 @@ class RTPStream
 {
 public:
     virtual ~RTPStream(){}
-    
+
     virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
                                      const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
                                      const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency) = 0;
+
+    // Returns the packet's payload size. Zero should be treated as an
+    // end-of-stream (in the case that EndOfFile() is true) or an error.
     virtual WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
-                    WebRtc_Word8* payloadData, 
+                    WebRtc_Word8* payloadData,
                     WebRtc_UWord16 payloadSize,
                     WebRtc_UWord32* offset) = 0;
     virtual bool EndOfFile() const = 0;
@@ -65,7 +68,7 @@ public:
                                      const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
                                      const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
     WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
-                    WebRtc_Word8* payloadData, 
+                    WebRtc_Word8* payloadData,
                     WebRtc_UWord16 payloadSize,
                     WebRtc_UWord32* offset);
     virtual bool EndOfFile() const;