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Fix gcc warnings triggered by -Wextra.

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TEST=build and audio_coding_unittests and audio_coding_module_test on Linux

Review URL: https://webrtc-codereview.appspot.com/343010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1445 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2012-01-17 19:27:33 +00:00
parent 4259fd725c
commit 975e4a33c6
3 changed files with 18 additions and 14 deletions
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15
src/modules/audio_coding/main/test/EncodeDecodeTest.cc
View File

@@ -198,14 +198,15 @@ bool Receiver::IncomingPacket() {
_firstTime = false; _firstTime = false;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload, _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
_payloadSizeBytes, &_nextTime); _payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes < 0) { if (_realPayloadSizeBytes == 0) {
printf("Error in reading incoming payload.\n"); if (_rtpStream->EndOfFile()) {
return false; _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, WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,

8
src/modules/audio_coding/main/test/RTPFile.cc
View File

@@ -106,7 +106,7 @@ RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp
WebRtc_UWord16 WebRtc_UWord16
RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
WebRtc_Word8* payloadData, WebRtc_Word8* payloadData,
WebRtc_UWord16 payloadSize, WebRtc_UWord16 payloadSize,
WebRtc_UWord32* offset) WebRtc_UWord32* offset)
{ {
@@ -125,7 +125,7 @@ RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
} }
else else
{ {
return -1; return 0;
} }
*offset = (packet->timeStamp/(packet->frequency/1000)); *offset = (packet->timeStamp/(packet->frequency/1000));
@@ -248,11 +248,11 @@ WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
{ {
return -1; return -1;
} }
lengthBytes -= 20; if (lengthBytes < 20)
if (lengthBytes < 0)
{ {
return -1; return -1;
} }
lengthBytes -= 20;
EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile)); EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
return lengthBytes; return lengthBytes;
} }

9
src/modules/audio_coding/main/test/RTPFile.h
View File

@@ -24,12 +24,15 @@ class RTPStream
{ {
public: public:
virtual ~RTPStream(){} virtual ~RTPStream(){}
virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp, virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData, const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency) = 0; 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, virtual WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
WebRtc_Word8* payloadData, WebRtc_Word8* payloadData,
WebRtc_UWord16 payloadSize, WebRtc_UWord16 payloadSize,
WebRtc_UWord32* offset) = 0; WebRtc_UWord32* offset) = 0;
virtual bool EndOfFile() const = 0; virtual bool EndOfFile() const = 0;
@@ -65,7 +68,7 @@ public:
const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData, const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency); const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo, WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
WebRtc_Word8* payloadData, WebRtc_Word8* payloadData,
WebRtc_UWord16 payloadSize, WebRtc_UWord16 payloadSize,
WebRtc_UWord32* offset); WebRtc_UWord32* offset);
virtual bool EndOfFile() const; virtual bool EndOfFile() const;