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Webrtc_Word32 => int32_t in video_coding/main/

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BUG=

Review URL: https://webrtc-codereview.appspot.com/1279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-02 15:54:38 +00:00
parent cfc07c943f
commit 7b859cc1e9
67 changed files with 1130 additions and 1130 deletions
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40
webrtc/modules/video_coding/main/test/rtp_player.h
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@ -30,7 +30,7 @@ struct PayloadCodecTuple;
struct RawRtpPacket
{
public:
RawRtpPacket(WebRtc_UWord8* rtp_data, WebRtc_UWord16 rtp_length);
RawRtpPacket(uint8_t* rtp_data, uint16_t rtp_length);
~RawRtpPacket();
uint8_t* data;
@ -66,10 +66,10 @@ class LostPackets {
struct PayloadCodecTuple
{
PayloadCodecTuple(WebRtc_UWord8 plType, std::string codecName, webrtc::VideoCodecType type) :
PayloadCodecTuple(uint8_t plType, std::string codecName, webrtc::VideoCodecType type) :
name(codecName), payloadType(plType), codecType(type) {};
const std::string name;
const WebRtc_UWord8 payloadType;
const uint8_t payloadType;
const webrtc::VideoCodecType codecType;
};
@ -81,37 +81,37 @@ public:
webrtc::Clock* clock);
virtual ~RTPPlayer();
WebRtc_Word32 Initialize(const PayloadTypeList* payloadList);
WebRtc_Word32 NextPacket(const WebRtc_Word64 timeNow);
WebRtc_UWord32 TimeUntilNextPacket() const;
WebRtc_Word32 SimulatePacketLoss(float lossRate, bool enableNack = false, WebRtc_UWord32 rttMs = 0);
WebRtc_Word32 SetReordering(bool enabled);
WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length);
int32_t Initialize(const PayloadTypeList* payloadList);
int32_t NextPacket(const int64_t timeNow);
uint32_t TimeUntilNextPacket() const;
int32_t SimulatePacketLoss(float lossRate, bool enableNack = false, uint32_t rttMs = 0);
int32_t SetReordering(bool enabled);
int32_t ResendPackets(const uint16_t* sequenceNumbers, uint16_t length);
void Print() const;
private:
WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
WebRtc_Word32 ReadHeader();
int32_t SendPacket(uint8_t* rtpData, uint16_t rtpLen);
int32_t ReadPacket(int16_t* rtpdata, uint32_t* offset);
int32_t ReadHeader();
webrtc::Clock* _clock;
FILE* _rtpFile;
webrtc::RtpRtcp* _rtpModule;
WebRtc_UWord32 _nextRtpTime;
uint32_t _nextRtpTime;
webrtc::RtpData* _dataCallback;
bool _firstPacket;
float _lossRate;
bool _nackEnabled;
LostPackets _lostPackets;
WebRtc_UWord32 _resendPacketCount;
WebRtc_Word32 _noLossStartup;
uint32_t _resendPacketCount;
int32_t _noLossStartup;
bool _endOfFile;
WebRtc_UWord32 _rttMs;
WebRtc_Word64 _firstPacketRtpTime;
WebRtc_Word64 _firstPacketTimeMs;
uint32_t _rttMs;
int64_t _firstPacketRtpTime;
int64_t _firstPacketTimeMs;
RawRtpPacket* _reorderBuffer;
bool _reordering;
WebRtc_Word16 _nextPacket[8000];
WebRtc_Word32 _nextPacketLength;
int16_t _nextPacket[8000];
int32_t _nextPacketLength;
int _randVec[RAND_VEC_LENGTH];
int _randVecPos;
};