git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/rtp_rtcp/interface/rtp_rtcp_defines.h

@@ -0,0 +1,222 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+
+#include "typedefs.h"
+#include "module_common_types.h"
+
+#ifndef NULL
+    #define NULL    0
+#endif
+
+#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500    // we assume ethernet
+#define RTP_PAYLOAD_NAME_SIZE 32
+#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
+#define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
+
+namespace webrtc{
+enum RTCPMethod
+{
+    kRtcpOff          = 0,
+    kRtcpCompound     = 1,
+    kRtcpNonCompound = 2
+};
+
+enum RTPAliveType
+{
+    kRtpDead   = 0,
+    kRtpNoRtp = 1,
+    kRtpAlive  = 2
+};
+
+enum RTCPAppSubTypes
+{
+    kAppSubtypeBwe     = 0x00
+};
+
+enum RTCPPacketType
+{
+    kRtcpReport         = 0x0001,
+    kRtcpSr             = 0x0002,
+    kRtcpRr             = 0x0004,
+    kRtcpBye            = 0x0008,
+    kRtcpPli            = 0x0010,
+    kRtcpNack           = 0x0020,
+    kRtcpFir            = 0x0040,
+    kRtcpTmmbr          = 0x0080,
+    kRtcpTmmbn          = 0x0100,
+    kRtcpSrReq         = 0x0200,
+    kRtcpXrVoipMetric = 0x0400,
+    kRtcpApp            = 0x0800,
+    kRtcpAppBwe        = 0x0801,
+    kRtcpSli            = 0x4000,
+    kRtcpRpsi           = 0x8000
+};
+
+enum KeyFrameRequestMethod
+{
+    kKeyFrameReqFirRtp    = 1,
+    kKeyFrameReqPliRtcp   = 2,
+    kKeyFrameReqFirRtcp   = 3
+};
+
+enum RtpRtcpPacketType
+{
+    kPacketRtp        = 0,
+    kPacketKeepAlive = 1
+};
+
+enum NACKMethod
+{
+    kNackOff      = 0,
+    kNackRtcp     = 2
+};
+
+struct RTCPSenderInfo
+{
+    WebRtc_UWord32 NTPseconds;
+    WebRtc_UWord32 NTPfraction;
+    WebRtc_UWord32 RTPtimeStamp;
+    WebRtc_UWord32 sendPacketCount;
+    WebRtc_UWord32 sendOctetCount;
+};
+
+struct RTCPReportBlock
+{
+    WebRtc_UWord8 fractionLost;
+    WebRtc_UWord32 cumulativeLost;  // 24 bits valid
+    WebRtc_UWord32 extendedHighSeqNum;
+    WebRtc_UWord32 jitter;
+    WebRtc_UWord32 lastSR;
+    WebRtc_UWord32 delaySinceLastSR;
+};
+
+class RtpData
+{
+public:
+    virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
+                                              const WebRtc_UWord16 payloadSize,
+                                              const WebRtcRTPHeader* rtpHeader) = 0;
+protected:
+    virtual ~RtpData() {}
+};
+
+class RtcpFeedback
+{
+public:
+    // if audioVideoOffset > 0 video is behind audio
+    virtual void OnLipSyncUpdate(const WebRtc_Word32 /*id*/,
+                                 const WebRtc_Word32 /*audioVideoOffset*/)  {};
+
+    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 OnXRVoIPMetricReceived(const WebRtc_Word32 /*id*/,
+                                        const RTCPVoIPMetric* /*metric*/,
+                                        const WebRtc_Word8 /*VoIPmetricBuffer*/[28])  {};
+
+    virtual void OnRTCPPacketTimeout(const WebRtc_Word32 /*id*/)  {};
+
+    virtual void OnTMMBRReceived(const WebRtc_Word32 /*id*/,
+                                 const WebRtc_UWord16 /*bwEstimateKbit*/)  {};
+
+    virtual void OnSLIReceived(const WebRtc_Word32 /*id*/,
+                               const WebRtc_UWord8 /*pictureId*/) {};
+
+    virtual void OnRPSIReceived(const WebRtc_Word32 /*id*/,
+                                const WebRtc_UWord64 /*pictureId*/) {};
+
+    virtual void OnSendReportReceived(const WebRtc_Word32 id,
+                                      const WebRtc_UWord32 senderSSRC)  {};
+
+    virtual void OnReceiveReportReceived(const WebRtc_Word32 id,
+                                         const WebRtc_UWord32 senderSSRC)  {};
+
+protected:
+    virtual ~RtcpFeedback() {}
+};
+
+class RtpFeedback
+{
+public:
+    // Receiving payload change or SSRC change. (return success!)
+    /*
+    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
+    */
+    virtual WebRtc_Word32 OnInitializeDecoder(const WebRtc_Word32 id,
+                                            const WebRtc_Word8 payloadType,
+                                            const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
+                                            const WebRtc_UWord32 frequency,
+                                            const WebRtc_UWord8 channels,
+                                            const WebRtc_UWord32 rate) = 0;
+
+    virtual void OnPacketTimeout(const WebRtc_Word32 id) = 0;
+
+    virtual void OnReceivedPacket(const WebRtc_Word32 id,
+                                  const RtpRtcpPacketType packetType) = 0;
+
+    virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+                                       const RTPAliveType alive) = 0;
+
+    virtual void OnIncomingSSRCChanged( const WebRtc_Word32 id,
+                                        const WebRtc_UWord32 SSRC) = 0;
+
+    virtual void OnIncomingCSRCChanged( const WebRtc_Word32 id,
+                                        const WebRtc_UWord32 CSRC,
+                                        const bool added) = 0;
+
+protected:
+    virtual ~RtpFeedback() {}
+};
+
+class RtpAudioFeedback
+{
+public:
+    virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
+                                          const WebRtc_UWord8 event,
+                                          const bool endOfEvent) = 0;
+
+    virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
+                                      const WebRtc_UWord8 event,
+                                      const WebRtc_UWord16 lengthMs,
+                                      const WebRtc_UWord8 volume) = 0;
+
+protected:
+    virtual ~RtpAudioFeedback() {}
+};
+
+
+class RtpVideoFeedback
+{
+public:
+    // this function should call codec module to inform it about the request
+    virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
+                                             const WebRtc_UWord8 message = 0) = 0;
+
+    virtual void OnNetworkChanged(const WebRtc_Word32 id,
+                                  const WebRtc_UWord32 minBitrateBps,
+                                  const WebRtc_UWord32 maxBitrateBps,
+                                  const WebRtc_UWord8 fractionLost,
+                                  const WebRtc_UWord16 roundTripTimeMs,
+                                  const WebRtc_UWord16 bwEstimateKbitMin,
+                                  const WebRtc_UWord16 bwEstimateKbitMax) = 0;
+
+protected:
+    virtual ~RtpVideoFeedback() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_