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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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