48df38114d
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state in the rtp receiver to never get valid. Also makes sure that only valid timestamps and receive times are used for audio/video sync. BUG=2608 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
124 lines
4.2 KiB
C++
124 lines
4.2 KiB
C++
/*
|
|
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
|
|
#define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
|
|
|
|
#include <list>
|
|
|
|
#include "webrtc/engine_configurations.h"
|
|
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
|
|
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
|
|
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
|
|
#include "webrtc/typedefs.h"
|
|
#include "webrtc/video_engine/vie_defines.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class CriticalSectionWrapper;
|
|
class Encryption;
|
|
class FecReceiver;
|
|
class ReceiveStatistics;
|
|
class RemoteBitrateEstimator;
|
|
class RtpDump;
|
|
class RtpHeaderParser;
|
|
class RTPPayloadRegistry;
|
|
class RtpReceiver;
|
|
class RtpRtcp;
|
|
class VideoCodingModule;
|
|
|
|
class ViEReceiver : public RtpData {
|
|
public:
|
|
ViEReceiver(const int32_t channel_id, VideoCodingModule* module_vcm,
|
|
RemoteBitrateEstimator* remote_bitrate_estimator,
|
|
RtpFeedback* rtp_feedback);
|
|
~ViEReceiver();
|
|
|
|
bool SetReceiveCodec(const VideoCodec& video_codec);
|
|
bool RegisterPayload(const VideoCodec& video_codec);
|
|
|
|
void SetNackStatus(bool enable, int max_nack_reordering_threshold);
|
|
void SetRtxStatus(bool enable, uint32_t ssrc);
|
|
void SetRtxPayloadType(uint32_t payload_type);
|
|
|
|
uint32_t GetRemoteSsrc() const;
|
|
int GetCsrcs(uint32_t* csrcs) const;
|
|
|
|
int RegisterExternalDecryption(Encryption* decryption);
|
|
int DeregisterExternalDecryption();
|
|
|
|
void SetRtpRtcpModule(RtpRtcp* module);
|
|
|
|
RtpReceiver* GetRtpReceiver() const;
|
|
|
|
void RegisterSimulcastRtpRtcpModules(const std::list<RtpRtcp*>& rtp_modules);
|
|
|
|
bool SetReceiveTimestampOffsetStatus(bool enable, int id);
|
|
bool SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
|
|
|
|
void StartReceive();
|
|
void StopReceive();
|
|
|
|
int StartRTPDump(const char file_nameUTF8[1024]);
|
|
int StopRTPDump();
|
|
|
|
// Receives packets from external transport.
|
|
int ReceivedRTPPacket(const void* rtp_packet, int rtp_packet_length);
|
|
int ReceivedRTCPPacket(const void* rtcp_packet, int rtcp_packet_length);
|
|
virtual bool OnRecoveredPacket(const uint8_t* packet,
|
|
int packet_length) OVERRIDE;
|
|
|
|
// Implements RtpData.
|
|
virtual int32_t OnReceivedPayloadData(
|
|
const uint8_t* payload_data,
|
|
const uint16_t payload_size,
|
|
const WebRtcRTPHeader* rtp_header);
|
|
|
|
void EstimatedReceiveBandwidth(unsigned int* available_bandwidth) const;
|
|
|
|
ReceiveStatistics* GetReceiveStatistics() const;
|
|
|
|
private:
|
|
int InsertRTPPacket(const int8_t* rtp_packet, int rtp_packet_length);
|
|
bool ReceivePacket(const uint8_t* packet, int packet_length,
|
|
const RTPHeader& header, bool in_order);
|
|
// Parses and handles for instance RTX and RED headers.
|
|
// This function assumes that it's being called from only one thread.
|
|
bool ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
|
|
int packet_length,
|
|
const RTPHeader& header);
|
|
int InsertRTCPPacket(const int8_t* rtcp_packet, int rtcp_packet_length);
|
|
bool IsPacketInOrder(const RTPHeader& header) const;
|
|
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
|
|
|
|
scoped_ptr<CriticalSectionWrapper> receive_cs_;
|
|
const int32_t channel_id_;
|
|
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
|
|
scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
|
|
scoped_ptr<RtpReceiver> rtp_receiver_;
|
|
scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
|
|
scoped_ptr<FecReceiver> fec_receiver_;
|
|
RtpRtcp* rtp_rtcp_;
|
|
std::list<RtpRtcp*> rtp_rtcp_simulcast_;
|
|
VideoCodingModule* vcm_;
|
|
RemoteBitrateEstimator* remote_bitrate_estimator_;
|
|
|
|
Encryption* external_decryption_;
|
|
uint8_t* decryption_buffer_;
|
|
RtpDump* rtp_dump_;
|
|
bool receiving_;
|
|
uint8_t restored_packet_[kViEMaxMtu];
|
|
bool restored_packet_in_use_;
|
|
};
|
|
|
|
} // namespace webrt
|
|
|
|
#endif // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
|