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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_ENGINE_PAYLOAD_ROUTER_H_ | |
12 #define WEBRTC_VIDEO_ENGINE_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 | |
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_ENGINE_PAYLOAD_ROUTER_H_ | |
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