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1 /* | 1 /* |
2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2017 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_PC_RTPTRANSPORT_H_ | 11 #ifndef WEBRTC_PC_RTPTRANSPORT_H_ |
12 #define WEBRTC_PC_RTPTRANSPORT_H_ | 12 #define WEBRTC_PC_RTPTRANSPORT_H_ |
13 | 13 |
14 #include "webrtc/api/ortc/rtptransportinterface.h" | 14 #include "webrtc/api/ortc/rtptransportinterface.h" |
| 15 #include "webrtc/base/sigslot.h" |
15 | 16 |
16 namespace rtc { | 17 namespace rtc { |
17 | 18 |
| 19 class CopyOnWriteBuffer; |
| 20 struct PacketOptions; |
18 class PacketTransportInternal; | 21 class PacketTransportInternal; |
19 | 22 |
20 } // namespace rtc | 23 } // namespace rtc |
21 | 24 |
22 namespace webrtc { | 25 namespace webrtc { |
23 | 26 |
24 class RtpTransport : public RtpTransportInterface { | 27 class RtpTransport : public RtpTransportInterface, public sigslot::has_slots<> { |
25 public: | 28 public: |
26 RtpTransport(const RtpTransport&) = delete; | 29 RtpTransport(const RtpTransport&) = delete; |
27 RtpTransport& operator=(const RtpTransport&) = delete; | 30 RtpTransport& operator=(const RtpTransport&) = delete; |
28 | 31 |
29 explicit RtpTransport(bool rtcp_mux_required) | 32 explicit RtpTransport(bool rtcp_mux_enabled) |
30 : rtcp_mux_required_(rtcp_mux_required) {} | 33 : rtcp_mux_enabled_(rtcp_mux_enabled) {} |
31 | 34 |
32 bool rtcp_mux_required() const { return rtcp_mux_required_; } | 35 bool rtcp_mux_enabled() const { return rtcp_mux_enabled_; } |
| 36 void SetRtcpMuxEnabled(bool enable); |
33 | 37 |
34 rtc::PacketTransportInternal* rtp_packet_transport() const { | 38 rtc::PacketTransportInternal* rtp_packet_transport() const { |
35 return rtp_packet_transport_; | 39 return rtp_packet_transport_; |
36 } | 40 } |
37 void set_rtp_packet_transport(rtc::PacketTransportInternal* rtp); | 41 void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp); |
38 | 42 |
39 rtc::PacketTransportInternal* rtcp_packet_transport() const { | 43 rtc::PacketTransportInternal* rtcp_packet_transport() const { |
40 return rtcp_packet_transport_; | 44 return rtcp_packet_transport_; |
41 } | 45 } |
42 void set_rtcp_packet_transport(rtc::PacketTransportInternal* rtcp); | 46 void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp); |
43 | 47 |
44 PacketTransportInterface* GetRtpPacketTransport() const override; | 48 PacketTransportInterface* GetRtpPacketTransport() const override; |
45 PacketTransportInterface* GetRtcpPacketTransport() const override; | 49 PacketTransportInterface* GetRtcpPacketTransport() const override; |
46 | 50 |
47 // TODO(zstein): Use these RtcpParameters for configuration elsewhere. | 51 // TODO(zstein): Use these RtcpParameters for configuration elsewhere. |
48 RTCError SetRtcpParameters(const RtcpParameters& parameters) override; | 52 RTCError SetRtcpParameters(const RtcpParameters& parameters) override; |
49 RtcpParameters GetRtcpParameters() const override; | 53 RtcpParameters GetRtcpParameters() const override; |
50 | 54 |
| 55 // Called whenever a transport's ready-to-send state changes. The argument |
| 56 // is true if all used transports are ready to send. This is more specific |
| 57 // than just "writable"; it means the last send didn't return ENOTCONN. |
| 58 sigslot::signal1<bool> SignalReadyToSend; |
| 59 |
| 60 bool IsWritable(bool rtcp) const; |
| 61 |
| 62 bool SendPacket(bool rtcp, |
| 63 const rtc::CopyOnWriteBuffer* packet, |
| 64 const rtc::PacketOptions& options, |
| 65 int flags); |
| 66 |
51 protected: | 67 protected: |
52 // TODO(zstein): Remove this when we remove RtpTransportAdapter. | 68 // TODO(zstein): Remove this when we remove RtpTransportAdapter. |
53 RtpTransportAdapter* GetInternal() override; | 69 RtpTransportAdapter* GetInternal() override; |
54 | 70 |
55 private: | 71 private: |
56 // True if RTCP-multiplexing is required. rtcp_packet_transport_ should | 72 void OnReadyToSend(rtc::PacketTransportInternal* transport); |
57 // always be null in this case. | 73 |
58 const bool rtcp_mux_required_; | 74 // Updates "ready to send" for an individual channel and fires |
| 75 // SignalReadyToSend. |
| 76 void SetReadyToSend(bool rtcp, bool ready); |
| 77 |
| 78 void MaybeSignalReadyToSend(); |
| 79 |
| 80 bool rtcp_mux_enabled_; |
59 | 81 |
60 rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr; | 82 rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr; |
61 rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr; | 83 rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr; |
62 | 84 |
| 85 bool ready_to_send_ = false; |
| 86 bool rtp_ready_to_send_ = false; |
| 87 bool rtcp_ready_to_send_ = false; |
| 88 |
63 RtcpParameters rtcp_parameters_; | 89 RtcpParameters rtcp_parameters_; |
64 }; | 90 }; |
65 | 91 |
66 } // namespace webrtc | 92 } // namespace webrtc |
67 | 93 |
68 #endif // WEBRTC_PC_RTPTRANSPORT_H_ | 94 #endif // WEBRTC_PC_RTPTRANSPORT_H_ |
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