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fd77b78821
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and RtpPayloadRegistry. Only intended change in behavior is that packets with unknown payload types are not discarded at this level of the stack. They are discarded higher up, in Channel::ReceivePacket (audio) and RtpVideoStreamReceiver::ReceivePacket (video). Bug: webrtc:8995 Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf Reviewed-on: https://webrtc-review.googlesource.com/92087 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24196}
101 lines
3.6 KiB
C++
101 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 MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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#include <list>
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#include <memory>
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#include <unordered_map>
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#include <vector>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/include/rtp_receiver.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "rtc_base/criticalsection.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(Clock* clock,
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RTPPayloadRegistry* rtp_payload_registry,
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RTPReceiverStrategy* rtp_media_receiver);
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~RtpReceiverImpl() override;
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int32_t RegisterReceivePayload(int payload_type,
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const SdpAudioFormat& audio_format) override;
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int32_t RegisterReceivePayload(const VideoCodec& video_codec) override;
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int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
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bool IncomingRtpPacket(const RTPHeader& rtp_header,
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const uint8_t* payload,
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size_t payload_length,
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PayloadUnion payload_specific) override;
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bool GetLatestTimestamps(uint32_t* timestamp,
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int64_t* receive_time_ms) const override;
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uint32_t SSRC() const override;
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int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
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std::vector<RtpSource> GetSources() const override;
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const std::vector<RtpSource>& ssrc_sources_for_testing() const {
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return ssrc_sources_;
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}
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const std::list<RtpSource>& csrc_sources_for_testing() const {
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return csrc_sources_;
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}
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private:
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void CheckSSRCChanged(const RTPHeader& rtp_header);
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void CheckCSRC(const WebRtcRTPHeader& rtp_header);
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void UpdateSources(const absl::optional<uint8_t>& ssrc_audio_level);
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void RemoveOutdatedSources(int64_t now_ms);
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Clock* clock_;
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rtc::CriticalSection critical_section_rtp_receiver_;
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RTPPayloadRegistry* const rtp_payload_registry_
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RTC_PT_GUARDED_BY(critical_section_rtp_receiver_);
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const std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
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// SSRCs.
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uint32_t ssrc_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
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uint8_t num_csrcs_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
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uint32_t current_remote_csrc_[kRtpCsrcSize] RTC_GUARDED_BY(
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critical_section_rtp_receiver_);
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// Sequence number and timestamps for the latest in-order packet.
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absl::optional<uint16_t> last_received_sequence_number_
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RTC_GUARDED_BY(critical_section_rtp_receiver_);
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uint32_t last_received_timestamp_
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RTC_GUARDED_BY(critical_section_rtp_receiver_);
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int64_t last_received_frame_time_ms_
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RTC_GUARDED_BY(critical_section_rtp_receiver_);
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std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
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iterator_by_csrc_;
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// The RtpSource objects are sorted chronologically.
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std::list<RtpSource> csrc_sources_;
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std::vector<RtpSource> ssrc_sources_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
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