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1 /* | |
2 * Copyright (c) 2012 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_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_ | |
12 #define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_ | |
13 | |
14 #include <string> | |
15 #include <vector> | |
16 | |
17 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | |
18 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" | |
19 | |
20 namespace webrtc { | |
21 class Clock; | |
22 | |
23 namespace rtpplayer { | |
24 | |
25 class PayloadCodecTuple { | |
26 public: | |
27 PayloadCodecTuple(uint8_t payload_type, const std::string& codec_name, | |
28 VideoCodecType codec_type) | |
29 : name_(codec_name), | |
30 payload_type_(payload_type), | |
31 codec_type_(codec_type) { | |
32 } | |
33 | |
34 const std::string& name() const { return name_; } | |
35 uint8_t payload_type() const { return payload_type_; } | |
36 VideoCodecType codec_type() const { return codec_type_; } | |
37 | |
38 private: | |
39 std::string name_; | |
40 uint8_t payload_type_; | |
41 VideoCodecType codec_type_; | |
42 }; | |
43 | |
44 typedef std::vector<PayloadCodecTuple> PayloadTypes; | |
45 typedef std::vector<PayloadCodecTuple>::const_iterator PayloadTypesIterator; | |
46 | |
47 // Implemented by RtpPlayer and given to client as a means to retrieve | |
48 // information about a specific RTP stream. | |
49 class RtpStreamInterface { | |
50 public: | |
51 virtual ~RtpStreamInterface() {} | |
52 | |
53 // Ask for missing packets to be resent. | |
54 virtual void ResendPackets(const uint16_t* sequence_numbers, | |
55 uint16_t length) = 0; | |
56 | |
57 virtual uint32_t ssrc() const = 0; | |
58 virtual const PayloadTypes& payload_types() const = 0; | |
59 }; | |
60 | |
61 // Implemented by a sink. Wraps RtpData because its d-tor is protected. | |
62 class PayloadSinkInterface : public RtpData { | |
63 public: | |
64 virtual ~PayloadSinkInterface() {} | |
65 }; | |
66 | |
67 // Implemented to provide a sink for RTP data, such as hooking up a VCM to | |
68 // the incoming RTP stream. | |
69 class PayloadSinkFactoryInterface { | |
70 public: | |
71 virtual ~PayloadSinkFactoryInterface() {} | |
72 | |
73 // Return NULL if failed to create sink. 'stream' is guaranteed to be | |
74 // around for as long as the RtpData. The returned object is owned by | |
75 // the caller (RtpPlayer). | |
76 virtual PayloadSinkInterface* Create(RtpStreamInterface* stream) = 0; | |
77 }; | |
78 | |
79 // The client's view of an RtpPlayer. | |
80 class RtpPlayerInterface { | |
81 public: | |
82 virtual ~RtpPlayerInterface() {} | |
83 | |
84 virtual int NextPacket(int64_t timeNow) = 0; | |
85 virtual uint32_t TimeUntilNextPacket() const = 0; | |
86 virtual void Print() const = 0; | |
87 }; | |
88 | |
89 RtpPlayerInterface* Create(const std::string& inputFilename, | |
90 PayloadSinkFactoryInterface* payloadSinkFactory, Clock* clock, | |
91 const PayloadTypes& payload_types, float lossRate, int64_t rttMs, | |
92 bool reordering); | |
93 | |
94 } // namespace rtpplayer | |
95 } // namespace webrtc | |
96 | |
97 #endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_ | |
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