6071b0636d
This will make further changes to these classes safer by ensuring that the compile breaks if the base class changes and not all overrides are fixed. This also highlighted a number of unused functions which I've removed. -- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/, but -- a new cl was needed to resolve a small conflict before committing. BUG=none TEST=none TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22359004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
106 lines
3.6 KiB
C++
106 lines
3.6 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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virtual 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) OVERRIDE;
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virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) OVERRIDE;
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virtual 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) OVERRIDE;
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virtual NACKMethod NACK() const OVERRIDE;
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// Turn negative acknowledgement requests on/off.
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virtual void SetNACKStatus(const NACKMethod method) OVERRIDE;
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// Returns the last received timestamp.
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virtual bool Timestamp(uint32_t* timestamp) const OVERRIDE;
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virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const OVERRIDE;
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virtual uint32_t SSRC() const OVERRIDE;
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virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const OVERRIDE;
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virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const OVERRIDE;
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virtual TelephoneEventHandler* GetTelephoneEventHandler() OVERRIDE;
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private:
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bool HaveReceivedFrame() 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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