1 /*
2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
12 #define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
13 
14 #include <list>
15 #include <vector>
16 
17 #include "webrtc/base/constructormagic.h"
18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/base/thread_annotations.h"
20 #include "webrtc/common_types.h"
21 #include "webrtc/system_wrappers/include/atomic32.h"
22 
23 namespace webrtc {
24 
25 class CriticalSectionWrapper;
26 class RTPFragmentationHeader;
27 class RtpRtcp;
28 struct RTPVideoHeader;
29 
30 // PayloadRouter routes outgoing data to the correct sending RTP module, based
31 // on the simulcast layer in RTPVideoHeader.
32 class PayloadRouter {
33  public:
34   PayloadRouter();
35   ~PayloadRouter();
36 
37   static size_t DefaultMaxPayloadLength();
38 
39   // Rtp modules are assumed to be sorted in simulcast index order.
40   void SetSendingRtpModules(const std::list<RtpRtcp*>& rtp_modules);
41 
42   // PayloadRouter will only route packets if being active, all packets will be
43   // dropped otherwise.
44   void set_active(bool active);
45   bool active();
46 
47   // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
48   // Returns true if the packet was routed / sent, false otherwise.
49   bool RoutePayload(FrameType frame_type,
50                     int8_t payload_type,
51                     uint32_t time_stamp,
52                     int64_t capture_time_ms,
53                     const uint8_t* payload_data,
54                     size_t payload_size,
55                     const RTPFragmentationHeader* fragmentation,
56                     const RTPVideoHeader* rtp_video_hdr);
57 
58   // Configures current target bitrate per module. 'stream_bitrates' is assumed
59   // to be in the same order as 'SetSendingRtpModules'.
60   void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
61 
62   // Returns the maximum allowed data payload length, given the configured MTU
63   // and RTP headers.
64   size_t MaxPayloadLength() const;
65 
AddRef()66   void AddRef() { ++ref_count_; }
Release()67   void Release() { if (--ref_count_ == 0) { delete this; } }
68 
69  private:
70   // TODO(mflodman): When the new video API has launched, remove crit_ and
71   // assume rtp_modules_ will never change during a call.
72   rtc::scoped_ptr<CriticalSectionWrapper> crit_;
73 
74   // Active sending RTP modules, in layer order.
75   std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
76   bool active_ GUARDED_BY(crit_.get());
77 
78   Atomic32 ref_count_;
79 
80   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
81 };
82 
83 }  // namespace webrtc
84 
85 #endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
86