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

2011-05-30 11:22:19 +00:00
parent 01813fe945
commit 77ae29bc81
1153 changed files with 404089 additions and 0 deletions
--- a/modules/rtp_rtcp/source/rtp_rtcp_private.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_private.h
@@ -0,0 +1,115 @@
+/*
+ *  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_SOURCE_RTP_RTCP_PRIVATE_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_PRIVATE_H_
+
+#include "bwe_defines.h"
+#include "rtp_rtcp.h"
+#include "tmmbr_help.h"
+#include "rtp_utility.h"
+
+namespace webrtc {
+class ModuleRtpRtcpPrivate : public RtpRtcp
+{
+public:
+    virtual void RegisterChildModule(RtpRtcp* module) = 0;
+    virtual void DeRegisterChildModule(RtpRtcp* module) = 0;
+
+    virtual WebRtc_Word32 RegisterVideoModule(RtpRtcp* videoModule) = 0;
+    virtual void DeRegisterVideoModule() = 0;
+
+    virtual void SetRemoteSSRC(const WebRtc_UWord32 SSRC) = 0;
+
+    virtual WebRtc_Word8 SendPayloadType() const = 0;
+
+    virtual RtpVideoCodecTypes ReceivedVideoCodec() const = 0;
+
+    virtual RtpVideoCodecTypes SendVideoCodec() const = 0;
+
+    // lipsync
+    virtual void OnReceivedNTP() = 0;
+
+    // bw estimation
+    virtual void OnPacketLossStatisticsUpdate(const WebRtc_UWord8 fractionLost,
+                                              const WebRtc_UWord16 roundTripTime,
+                                              const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
+                                              const WebRtc_UWord32 jitter) = 0;
+
+    // bw estimation
+    virtual void OnReceivedTMMBR() = 0;
+
+    // bw estimation
+    virtual void OnReceivedBandwidthEstimateUpdate( const WebRtc_UWord16 bwEstimateMinKbit,
+                                                    const WebRtc_UWord16 bwEstimateMaxKbit ) = 0;
+
+    //
+    virtual RateControlRegion OnOverUseStateUpdate(const RateControlInput& rateControlInput) = 0;
+
+    // received a request for a new key frame
+    virtual void OnReceivedIntraFrameRequest(const WebRtc_UWord8 message) = 0;
+
+    // received a request for a new SLI
+    virtual void OnReceivedSliceLossIndication(const WebRtc_UWord8 pictureID) = 0;
+
+    // received a new refereence frame
+    virtual void OnReceivedReferencePictureSelectionIndication(const WebRtc_UWord64 pitureID) = 0;
+
+    // request for a RTCP send report
+    virtual void OnRequestSendReport() = 0;
+
+    // Get remote SequenceNumber
+    virtual WebRtc_UWord16 RemoteSequenceNumber() const = 0;
+
+    virtual WebRtc_UWord32 PacketCountSent() const = 0;
+
+    virtual int CurrentSendFrequencyHz() const = 0;
+
+    virtual WebRtc_UWord32 ByteCountSent() const = 0;
+
+    virtual WebRtc_UWord32 BitrateReceivedNow() const = 0;
+
+    virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport) = 0;
+
+    virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs, // when we received the last report
+                                        WebRtc_UWord32& NTPfrac,
+                                        WebRtc_UWord32& remoteSR) = 0; // NTP inside the last received (mid 16 bits from sec and frac)
+
+    virtual WebRtc_Word32 ReportBlockStatistics(WebRtc_UWord8 *fraction_lost,
+                                              WebRtc_UWord32 *cum_lost,
+                                              WebRtc_UWord32 *ext_max,
+                                              WebRtc_UWord32 *jitter) = 0;
+
+    // bad state of RTP receiver request a keyframe
+    virtual void OnRequestIntraFrame( const FrameType frameType) = 0;
+
+    /*
+    *   NACK
+    */
+    virtual void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
+                                const WebRtc_UWord16* nackSequenceNumbers) = 0;
+
+    /*
+    *   TMMBR
+    */
+    virtual WebRtc_Word32 UpdateTMMBR() = 0;
+
+    virtual WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
+                                 const WebRtc_UWord32 maxBitrateKbit) = 0;
+
+    virtual WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
+                                    TMMBRSet*& boundingSetRec)= 0;
+
+    virtual WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
+                                      const WebRtc_UWord32 accNumCandidates,
+                                      TMMBRSet* candidateSet) const = 0;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_PRIVATE_H_