This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950. Reason for revert: Broken internal project. Original change's description: > Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." > > This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." > > > > This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e. > > > > Reason for revert: Broke chromium tests. > > Original change's description: > > > Replace BundleFilter with RtpDemuxer in RtpTransport. > > > > > > BundleFilter is replaced by RtpDemuxer in RtpTransport for payload > > > type-based demuxing. RtpTransport will support MID-based demuxing later. > > > > > > Each BaseChannel has its own RTP demuxing criteria and when connecting > > > to the RtpTransport, BaseChannel will register itself as a demuxer sink. > > > > > > The inheritance model is changed. New inheritance chain: > > > DtlsSrtpTransport->SrtpTransport->RtpTranpsort > > > > > > NOTE: > > > When RTCP packets are received, Call::DeliverRtcp will be called for > > > multiple times (webrtc:9035) which is an existing issue. With this CL, > > > it will become more of a problem and should be fixed. > > > > > > Bug: webrtc:8587 > > > Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0 > > > Reviewed-on: https://webrtc-review.googlesource.com/61360 > > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22613} > > > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > > > Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:8587 > > Reviewed-on: https://webrtc-review.googlesource.com/64860 > > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22614} > > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org > > Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8587 > Reviewed-on: https://webrtc-review.googlesource.com/64862 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22615} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8587 Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4 Reviewed-on: https://webrtc-review.googlesource.com/65381 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22665}
126 lines
3.9 KiB
C++
126 lines
3.9 KiB
C++
/*
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* Copyright 2017 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 PC_RTPTRANSPORT_H_
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#define PC_RTPTRANSPORT_H_
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#include <string>
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#include "pc/bundlefilter.h"
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#include "pc/rtptransportinternal.h"
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#include "rtc_base/sigslot.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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struct PacketOptions;
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struct PacketTime;
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class PacketTransportInternal;
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} // namespace rtc
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namespace webrtc {
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class RtpTransport : public RtpTransportInternal {
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public:
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RtpTransport(const RtpTransport&) = delete;
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RtpTransport& operator=(const RtpTransport&) = delete;
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explicit RtpTransport(bool rtcp_mux_enabled)
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: rtcp_mux_enabled_(rtcp_mux_enabled) {}
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bool rtcp_mux_enabled() const override { return rtcp_mux_enabled_; }
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void SetRtcpMuxEnabled(bool enable) override;
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rtc::PacketTransportInternal* rtp_packet_transport() const override {
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return rtp_packet_transport_;
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}
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void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) override;
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rtc::PacketTransportInternal* rtcp_packet_transport() const override {
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return rtcp_packet_transport_;
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}
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void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) override;
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PacketTransportInterface* GetRtpPacketTransport() const override;
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PacketTransportInterface* GetRtcpPacketTransport() const override;
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// TODO(zstein): Use these RtcpParameters for configuration elsewhere.
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RTCError SetParameters(const RtpTransportParameters& parameters) override;
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RtpTransportParameters GetParameters() const override;
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bool IsWritable(bool rtcp) const override;
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bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool HandlesPayloadType(int payload_type) const override;
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void AddHandledPayloadType(int payload_type) override;
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void SetMetricsObserver(
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rtc::scoped_refptr<MetricsObserverInterface> metrics_observer) override {}
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protected:
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// TODO(zstein): Remove this when we remove RtpTransportAdapter.
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RtpTransportAdapter* GetInternal() override;
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private:
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bool IsRtpTransportWritable();
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bool HandlesPacket(const uint8_t* data, size_t len);
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void OnReadyToSend(rtc::PacketTransportInternal* transport);
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void OnNetworkRouteChange(rtc::Optional<rtc::NetworkRoute> network_route);
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void OnWritableState(rtc::PacketTransportInternal* packet_transport);
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void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
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const rtc::SentPacket& sent_packet);
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// Updates "ready to send" for an individual channel and fires
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// SignalReadyToSend.
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void SetReadyToSend(bool rtcp, bool ready);
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void MaybeSignalReadyToSend();
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bool SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags);
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void OnReadPacket(rtc::PacketTransportInternal* transport,
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const char* data,
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size_t len,
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const rtc::PacketTime& packet_time,
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int flags);
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bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
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bool rtcp_mux_enabled_;
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rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
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rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
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bool ready_to_send_ = false;
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bool rtp_ready_to_send_ = false;
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bool rtcp_ready_to_send_ = false;
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RtpTransportParameters parameters_;
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cricket::BundleFilter bundle_filter_;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORT_H_
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