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Fix VCM test build warnings on Mac with clang.

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BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/318008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1160 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2011-12-12 13:45:59 +00:00
parent 7889a9b49a
commit 1480f02faf
8 changed files with 34 additions and 30 deletions
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2
src/modules/video_coding/codecs/test_framework/normal_test.cc
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@ -235,7 +235,7 @@ NormalTest::Encode()
}
int
NormalTest::Decode()
NormalTest::Decode(int lossValue)
{
_encodedVideoBuffer.SetWidth(_inst.width);
_encodedVideoBuffer.SetHeight(_inst.height);

2
src/modules/video_coding/codecs/test_framework/normal_test.h
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@ -26,7 +26,7 @@ protected:
virtual void Setup();
virtual void Teardown();
virtual bool Encode();
virtual int Decode();
virtual int Decode(int lossValue = 0);
virtual void CodecSpecific_InitBitrate()=0;
virtual int DoPacketLoss() {return 0;};

5
src/modules/video_coding/main/test/generic_codec_test.h
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@ -26,6 +26,9 @@ Test consists of:
4. Decoder control test / General API functionality
*/
namespace webrtc {
int VCMGenericCodecTest(CmdArgs& args);
class GenericCodecTest
@ -97,4 +100,6 @@ private:
WebRtc_UWord32 _timeStamp;
}; // end of VCMEncodeCompleteCallback
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_

4
src/modules/video_coding/main/test/media_opt_test.h
View File

@ -20,8 +20,6 @@
#include "video_coding.h"
#include "video_source.h"
using namespace std;
// media optimization test
// This test simulates a complete encode-decode cycle via the RTP module.
// allows error resilience tests, packet loss tests, etc.
@ -76,7 +74,7 @@ private:
WebRtc_Word32 _frameCnt;
float _sumEncBytes;
WebRtc_Word32 _numFramesDropped;
string _codecName;
std::string _codecName;
webrtc::VideoCodecType _sendCodecType;
WebRtc_Word32 _numberOfCores;

3
src/modules/video_coding/main/test/mt_test_common.cc
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@ -14,6 +14,7 @@
#include "rtp_dump.h"
namespace webrtc {
TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
const char* filename):
@ -130,3 +131,5 @@ bool TransportThread(void *obj)
state->_transport.TransportPackets();
return true;
}
} // namespace webrtc

3
src/modules/video_coding/main/test/mt_test_common.h
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@ -20,7 +20,7 @@
#include "test_util.h"
#include "video_coding.h"
using namespace webrtc;
namespace webrtc {
class SendSharedState
{
@ -83,5 +83,6 @@ bool VCMProcessingThread(void* obj);
bool VCMDecodeThread(void* obj);
bool TransportThread(void *obj);
} // namespace webrtc
#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_

13
src/modules/video_coding/main/test/test_callbacks.cc
View File

@ -15,6 +15,8 @@
#include "rtp_dump.h"
#include "test_macros.h"
namespace webrtc {
/******************************
* VCMEncodeCompleteCallback
*****************************/
@ -503,14 +505,13 @@ VideoProtectionCallback::FECKeyUepProtection()
void
RTPFeedbackCallback::OnNetworkChanged(const WebRtc_Word32 id,
const WebRtc_UWord16 bitrateTargetKbit,
const WebRtc_UWord32 bitrateBps,
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTimeMs,
const WebRtc_UWord32 jitterMS,
const WebRtc_UWord16 bwEstimateKbitMin,
const WebRtc_UWord16 bwEstimateKbitMax)
const WebRtc_UWord16 roundTripTimeMs)
{
_vcm->SetChannelParameters(bitrateTargetKbit, fractionLost,
_vcm->SetChannelParameters(bitrateBps / 1000, fractionLost,
(WebRtc_UWord8)roundTripTimeMs);
}
} // namespace webrtc

32
src/modules/video_coding/main/test/test_callbacks.h
View File

@ -28,12 +28,9 @@
#include "trace.h"
#include "video_coding.h"
using namespace webrtc;
namespace webrtc
{
class RtpDump;
}
// Send Side - Packetization callback - send an encoded frame to the VCMReceiver
class VCMEncodeCompleteCallback: public VCMPacketizationCallback
@ -238,7 +235,7 @@ class VideoProtectionCallback: public VCMProtectionCallback
public:
VideoProtectionCallback();
virtual ~VideoProtectionCallback();
void RegisterRtpModule(RtpRtcp* rtp){_rtp = rtp;}
void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;}
WebRtc_Word32 ProtectionRequest(const WebRtc_UWord8 deltaFECRate,
const WebRtc_UWord8 keyFECRate,
const bool deltaUseUepProtection,
@ -259,23 +256,22 @@ private:
};
// Feed back from the RTP Module callback
class RTPFeedbackCallback: public RtpVideoFeedback
{
public:
RTPFeedbackCallback(VideoCodingModule* vcm) {_vcm = vcm;};
void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
const WebRtc_UWord8 message = 0){};
class RTPFeedbackCallback : public RtpVideoFeedback {
public:
RTPFeedbackCallback(VideoCodingModule* vcm) {_vcm = vcm;};
void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
const FrameType type,
const WebRtc_UWord8 streamIdx) {};
void OnNetworkChanged(const WebRtc_Word32 id,
const WebRtc_UWord16 bitrateTargetKbit,
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTimeMs,
const WebRtc_UWord32 jitterMS,
const WebRtc_UWord16 bwEstimateKbitMin,
const WebRtc_UWord16 bwEstimateKbitMax);
private:
VideoCodingModule* _vcm;
const WebRtc_UWord32 bitrateBps,
const WebRtc_UWord8 fractionLost,
const WebRtc_UWord16 roundTripTimeMs);
private:
VideoCodingModule* _vcm;
};
} // namespace webrtc
#endif