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
117 lines
3.7 KiB
C++
117 lines
3.7 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RtpReceiverImpl : public RtpReceiver {
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public:
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// Callbacks passed in here may not be NULL (use Null Object callbacks if you
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// want callbacks to do nothing). This class takes ownership of the media
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// receiver but nothing else.
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RtpReceiverImpl(int32_t id,
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Clock* clock,
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RtpAudioFeedback* incoming_audio_messages_callback,
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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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virtual ~RtpReceiverImpl();
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RTPReceiverStrategy* GetMediaReceiver() const;
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int32_t RegisterReceivePayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate);
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int32_t DeRegisterReceivePayload(const int8_t payload_type);
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bool IncomingRtpPacket(
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const RTPHeader& rtp_header,
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const uint8_t* payload,
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int payload_length,
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PayloadUnion payload_specific,
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bool in_order);
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NACKMethod NACK() const;
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// Turn negative acknowledgement requests on/off.
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void SetNACKStatus(const NACKMethod method);
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// Returns the last received timestamp.
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bool Timestamp(uint32_t* timestamp) const;
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bool LastReceivedTimeMs(int64_t* receive_time_ms) const;
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uint32_t SSRC() const;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const;
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int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
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// RTX.
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void SetRTXStatus(bool enable, uint32_t ssrc);
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void RTXStatus(bool* enable, uint32_t* ssrc, int* payload_type) const;
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void SetRtxPayloadType(int payload_type);
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TelephoneEventHandler* GetTelephoneEventHandler();
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private:
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bool HaveReceivedFrame() const;
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RtpVideoCodecTypes VideoCodecType() const;
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
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const int8_t first_payload_byte,
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bool& is_red,
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PayloadUnion* payload,
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bool* should_reset_statistics);
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Clock* clock_;
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RTPPayloadRegistry* rtp_payload_registry_;
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scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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int32_t id_;
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RtpFeedback* cb_rtp_feedback_;
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scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
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int64_t last_receive_time_;
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uint16_t last_received_payload_length_;
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// SSRCs.
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uint32_t ssrc_;
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uint8_t num_csrcs_;
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uint32_t current_remote_csrc_[kRtpCsrcSize];
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uint32_t last_received_timestamp_;
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int64_t last_received_frame_time_ms_;
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uint16_t last_received_sequence_number_;
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NACKMethod nack_method_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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