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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_VIDEO_ENGINE_VIE_RECEIVER_H_ | |
| 12 #define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_ | |
| 13 | |
| 14 #include <list> | |
| 15 #include <vector> | |
| 16 | |
| 17 #include "webrtc/base/scoped_ptr.h" | |
| 18 #include "webrtc/engine_configurations.h" | |
| 19 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | |
| 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | |
| 21 #include "webrtc/typedefs.h" | |
| 22 | |
| 23 namespace webrtc { | |
| 24 | |
| 25 class CriticalSectionWrapper; | |
| 26 class FecReceiver; | |
| 27 class RemoteNtpTimeEstimator; | |
| 28 class ReceiveStatistics; | |
| 29 class RemoteBitrateEstimator; | |
| 30 class RtpHeaderParser; | |
| 31 class RTPPayloadRegistry; | |
| 32 class RtpReceiver; | |
| 33 class RtpRtcp; | |
| 34 class VideoCodingModule; | |
| 35 struct ReceiveBandwidthEstimatorStats; | |
| 36 | |
| 37 class ViEReceiver : public RtpData { | |
| 38 public: | |
| 39 ViEReceiver(VideoCodingModule* module_vcm, | |
| 40 RemoteBitrateEstimator* remote_bitrate_estimator, | |
| 41 RtpFeedback* rtp_feedback); | |
| 42 ~ViEReceiver(); | |
| 43 | |
| 44 bool SetReceiveCodec(const VideoCodec& video_codec); | |
| 45 bool RegisterPayload(const VideoCodec& video_codec); | |
| 46 | |
| 47 void SetNackStatus(bool enable, int max_nack_reordering_threshold); | |
| 48 void SetRtxPayloadType(int payload_type, int associated_payload_type); | |
| 49 // If set to true, the RTX payload type mapping supplied in | |
| 50 // |SetRtxPayloadType| will be used when restoring RTX packets. Without it, | |
| 51 // RTX packets will always be restored to the last non-RTX packet payload type | |
| 52 // received. | |
| 53 void SetUseRtxPayloadMappingOnRestore(bool val); | |
| 54 void SetRtxSsrc(uint32_t ssrc); | |
| 55 bool GetRtxSsrc(uint32_t* ssrc) const; | |
| 56 | |
| 57 bool IsFecEnabled() const; | |
| 58 | |
| 59 uint32_t GetRemoteSsrc() const; | |
| 60 int GetCsrcs(uint32_t* csrcs) const; | |
| 61 | |
| 62 void SetRtpRtcpModule(RtpRtcp* module); | |
| 63 | |
| 64 RtpReceiver* GetRtpReceiver() const; | |
| 65 | |
| 66 void RegisterRtpRtcpModules(const std::vector<RtpRtcp*>& rtp_modules); | |
| 67 | |
| 68 bool SetReceiveTimestampOffsetStatus(bool enable, int id); | |
| 69 bool SetReceiveAbsoluteSendTimeStatus(bool enable, int id); | |
| 70 bool SetReceiveVideoRotationStatus(bool enable, int id); | |
| 71 bool SetReceiveTransportSequenceNumber(bool enable, int id); | |
| 72 | |
| 73 void StartReceive(); | |
| 74 void StopReceive(); | |
| 75 | |
| 76 // Receives packets from external transport. | |
| 77 int ReceivedRTPPacket(const void* rtp_packet, size_t rtp_packet_length, | |
| 78 const PacketTime& packet_time); | |
| 79 int ReceivedRTCPPacket(const void* rtcp_packet, size_t rtcp_packet_length); | |
| 80 | |
| 81 // Implements RtpData. | |
| 82 int32_t OnReceivedPayloadData(const uint8_t* payload_data, | |
| 83 const size_t payload_size, | |
| 84 const WebRtcRTPHeader* rtp_header) override; | |
| 85 bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override; | |
| 86 | |
| 87 ReceiveStatistics* GetReceiveStatistics() const; | |
| 88 | |
| 89 private: | |
| 90 int InsertRTPPacket(const uint8_t* rtp_packet, size_t rtp_packet_length, | |
| 91 const PacketTime& packet_time); | |
| 92 bool ReceivePacket(const uint8_t* packet, | |
| 93 size_t packet_length, | |
| 94 const RTPHeader& header, | |
| 95 bool in_order); | |
| 96 // Parses and handles for instance RTX and RED headers. | |
| 97 // This function assumes that it's being called from only one thread. | |
| 98 bool ParseAndHandleEncapsulatingHeader(const uint8_t* packet, | |
| 99 size_t packet_length, | |
| 100 const RTPHeader& header); | |
| 101 void NotifyReceiverOfFecPacket(const RTPHeader& header); | |
| 102 int InsertRTCPPacket(const uint8_t* rtcp_packet, size_t rtcp_packet_length); | |
| 103 bool IsPacketInOrder(const RTPHeader& header) const; | |
| 104 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; | |
| 105 void UpdateHistograms(); | |
| 106 | |
| 107 rtc::scoped_ptr<CriticalSectionWrapper> receive_cs_; | |
| 108 Clock* clock_; | |
| 109 rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; | |
| 110 rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_; | |
| 111 rtc::scoped_ptr<RtpReceiver> rtp_receiver_; | |
| 112 const rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_; | |
| 113 rtc::scoped_ptr<FecReceiver> fec_receiver_; | |
| 114 RtpRtcp* rtp_rtcp_; | |
| 115 std::vector<RtpRtcp*> rtp_rtcp_simulcast_; | |
| 116 VideoCodingModule* vcm_; | |
| 117 RemoteBitrateEstimator* remote_bitrate_estimator_; | |
| 118 | |
| 119 rtc::scoped_ptr<RemoteNtpTimeEstimator> ntp_estimator_; | |
| 120 | |
| 121 bool receiving_; | |
| 122 uint8_t restored_packet_[IP_PACKET_SIZE]; | |
| 123 bool restored_packet_in_use_; | |
| 124 bool receiving_ast_enabled_; | |
| 125 bool receiving_cvo_enabled_; | |
| 126 bool receiving_tsn_enabled_; | |
| 127 int64_t last_packet_log_ms_; | |
| 128 }; | |
| 129 | |
| 130 } // namespace webrtc | |
| 131 | |
| 132 #endif // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_ | |
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Chromium Code Reviews
Issue 1510183002:
Reland of Merge webrtc/video_engine/ into webrtc/video/ (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master