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WebRtc_Word32 => int32_t for rtp_rtcp/

Browse Source 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2013-04-08 11:08:41 +00:00
parent 367804cce2
commit 2f44673d66
86 changed files with 3934 additions and 3956 deletions
Download Patch File Download Diff File Expand all files Collapse all files

126
webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
View File

@ -26,7 +26,7 @@
namespace webrtc{
const WebRtc_Word32 kDefaultVideoFrequency = 90000;
const int32_t kDefaultVideoFrequency = 90000;
enum RTCPMethod
{
@ -115,32 +115,32 @@ enum RtxMode {
struct RTCPSenderInfo
{
WebRtc_UWord32 NTPseconds;
WebRtc_UWord32 NTPfraction;
WebRtc_UWord32 RTPtimeStamp;
WebRtc_UWord32 sendPacketCount;
WebRtc_UWord32 sendOctetCount;
uint32_t NTPseconds;
uint32_t NTPfraction;
uint32_t RTPtimeStamp;
uint32_t sendPacketCount;
uint32_t sendOctetCount;
};
struct RTCPReportBlock
{
// Fields as described by RFC 3550 6.4.2.
WebRtc_UWord32 remoteSSRC; // SSRC of sender of this report.
WebRtc_UWord32 sourceSSRC; // SSRC of the RTP packet sender.
WebRtc_UWord8 fractionLost;
WebRtc_UWord32 cumulativeLost; // 24 bits valid
WebRtc_UWord32 extendedHighSeqNum;
WebRtc_UWord32 jitter;
WebRtc_UWord32 lastSR;
WebRtc_UWord32 delaySinceLastSR;
uint32_t remoteSSRC; // SSRC of sender of this report.
uint32_t sourceSSRC; // SSRC of the RTP packet sender.
uint8_t fractionLost;
uint32_t cumulativeLost; // 24 bits valid
uint32_t extendedHighSeqNum;
uint32_t jitter;
uint32_t lastSR;
uint32_t delaySinceLastSR;
};
class RtpData
{
public:
virtual WebRtc_Word32 OnReceivedPayloadData(
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
virtual int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const uint16_t payloadSize,
const WebRtcRTPHeader* rtpHeader) = 0;
protected:
virtual ~RtpData() {}
@ -149,29 +149,29 @@ protected:
class RtcpFeedback
{
public:
virtual void OnApplicationDataReceived(const WebRtc_Word32 /*id*/,
const WebRtc_UWord8 /*subType*/,
const WebRtc_UWord32 /*name*/,
const WebRtc_UWord16 /*length*/,
const WebRtc_UWord8* /*data*/) {};
virtual void OnApplicationDataReceived(const int32_t /*id*/,
const uint8_t /*subType*/,
const uint32_t /*name*/,
const uint16_t /*length*/,
const uint8_t* /*data*/) {};
virtual void OnXRVoIPMetricReceived(
const WebRtc_Word32 /*id*/,
const int32_t /*id*/,
const RTCPVoIPMetric* /*metric*/) {};
virtual void OnRTCPPacketTimeout(const WebRtc_Word32 /*id*/) {};
virtual void OnRTCPPacketTimeout(const int32_t /*id*/) {};
// |ntp_secs|, |ntp_frac| and |timestamp| are the NTP time and RTP timestamp
// parsed from the RTCP sender report from the sender with ssrc
// |senderSSRC|.
virtual void OnSendReportReceived(const WebRtc_Word32 id,
const WebRtc_UWord32 senderSSRC,
virtual void OnSendReportReceived(const int32_t id,
const uint32_t senderSSRC,
uint32_t ntp_secs,
uint32_t ntp_frac,
uint32_t timestamp) {};
virtual void OnReceiveReportReceived(const WebRtc_Word32 id,
const WebRtc_UWord32 senderSSRC) {};
virtual void OnReceiveReportReceived(const int32_t id,
const uint32_t senderSSRC) {};
protected:
virtual ~RtcpFeedback() {}
@ -184,27 +184,27 @@ public:
/*
* channels - number of channels in codec (1 = mono, 2 = stereo)
*/
virtual WebRtc_Word32 OnInitializeDecoder(
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
virtual int32_t OnInitializeDecoder(
const int32_t id,
const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) = 0;
const uint8_t channels,
const uint32_t rate) = 0;
virtual void OnPacketTimeout(const WebRtc_Word32 id) = 0;
virtual void OnPacketTimeout(const int32_t id) = 0;
virtual void OnReceivedPacket(const WebRtc_Word32 id,
virtual void OnReceivedPacket(const int32_t id,
const RtpRtcpPacketType packetType) = 0;
virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
virtual void OnPeriodicDeadOrAlive(const int32_t id,
const RTPAliveType alive) = 0;
virtual void OnIncomingSSRCChanged( const WebRtc_Word32 id,
const WebRtc_UWord32 SSRC) = 0;
virtual void OnIncomingSSRCChanged( const int32_t id,
const uint32_t SSRC) = 0;
virtual void OnIncomingCSRCChanged( const WebRtc_Word32 id,
const WebRtc_UWord32 CSRC,
virtual void OnIncomingCSRCChanged( const int32_t id,
const uint32_t CSRC,
const bool added) = 0;
protected:
@ -214,10 +214,10 @@ protected:
class RtpAudioFeedback {
public:
virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
const WebRtc_UWord8 event,
const WebRtc_UWord16 lengthMs,
const WebRtc_UWord8 volume) = 0;
virtual void OnPlayTelephoneEvent(const int32_t id,
const uint8_t event,
const uint16_t lengthMs,
const uint8_t volume) = 0;
protected:
virtual ~RtpAudioFeedback() {}
};
@ -264,29 +264,29 @@ class NullRtpFeedback : public RtpFeedback {
public:
virtual ~NullRtpFeedback() {}
virtual WebRtc_Word32 OnInitializeDecoder(
const WebRtc_Word32 id,
const WebRtc_Word8 payloadType,
virtual int32_t OnInitializeDecoder(
const int32_t id,
const int8_t payloadType,
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
const int frequency,
const WebRtc_UWord8 channels,
const WebRtc_UWord32 rate) {
const uint8_t channels,
const uint32_t rate) {
return 0;
}
virtual void OnPacketTimeout(const WebRtc_Word32 id) {}
virtual void OnPacketTimeout(const int32_t id) {}
virtual void OnReceivedPacket(const WebRtc_Word32 id,
virtual void OnReceivedPacket(const int32_t id,
const RtpRtcpPacketType packetType) {}
virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
virtual void OnPeriodicDeadOrAlive(const int32_t id,
const RTPAliveType alive) {}
virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
const WebRtc_UWord32 SSRC) {}
virtual void OnIncomingSSRCChanged(const int32_t id,
const uint32_t SSRC) {}
virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
const WebRtc_UWord32 CSRC,
virtual void OnIncomingCSRCChanged(const int32_t id,
const uint32_t CSRC,
const bool added) {}
};
@ -294,9 +294,9 @@ class NullRtpFeedback : public RtpFeedback {
class NullRtpData : public RtpData {
public:
virtual ~NullRtpData() {}
virtual WebRtc_Word32 OnReceivedPayloadData(
const WebRtc_UWord8* payloadData,
const WebRtc_UWord16 payloadSize,
virtual int32_t OnReceivedPayloadData(
const uint8_t* payloadData,
const uint16_t payloadSize,
const WebRtcRTPHeader* rtpHeader) {
return 0;
}
@ -307,10 +307,10 @@ class NullRtpAudioFeedback : public RtpAudioFeedback {
public:
virtual ~NullRtpAudioFeedback() {}
virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
const WebRtc_UWord8 event,
const WebRtc_UWord16 lengthMs,
const WebRtc_UWord8 volume) {}
virtual void OnPlayTelephoneEvent(const int32_t id,
const uint8_t event,
const uint16_t lengthMs,
const uint8_t volume) {}
};
} // namespace webrtc