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
23namespace webrtc {
24
25class CriticalSectionWrapper;
26class RTPFragmentationHeader;
27class RtpRtcp;
28struct RTPVideoHeader;
29
30// PayloadRouter routes outgoing data to the correct sending RTP module, based
31// on the simulcast layer in RTPVideoHeader.
32class 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
66  void AddRef() { ++ref_count_; }
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