27f3bf5128
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}
106 lines
4.3 KiB
C++
106 lines
4.3 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_RTPTRANSPORTINTERNAL_H_
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#define PC_RTPTRANSPORTINTERNAL_H_
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#include <string>
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#include "api/ortc/srtptransportinterface.h"
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#include "api/umametrics.h"
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#include "call/rtp_demuxer.h"
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#include "p2p/base/icetransportinternal.h"
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#include "pc/sessiondescription.h"
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#include "rtc_base/networkroute.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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} // namespace rtc
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namespace webrtc {
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// This represents the internal interface beneath SrtpTransportInterface;
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// it is not accessible to API consumers but is accessible to internal classes
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// in order to send and receive RTP and RTCP packets belonging to a single RTP
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// session. Additional convenience and configuration methods are also provided.
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class RtpTransportInternal : public SrtpTransportInterface,
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public sigslot::has_slots<> {
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public:
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virtual void SetRtcpMuxEnabled(bool enable) = 0;
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// TODO(zstein): Remove PacketTransport setters. Clients should pass these
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// in to constructors instead and construct a new RtpTransportInternal instead
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// of updating them.
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virtual bool rtcp_mux_enabled() const = 0;
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virtual rtc::PacketTransportInternal* rtp_packet_transport() const = 0;
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virtual void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) = 0;
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virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0;
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virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 0;
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// Called whenever a transport's ready-to-send state changes. The argument
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// is true if all used transports are ready to send. This is more specific
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// than just "writable"; it means the last send didn't return ENOTCONN.
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sigslot::signal1<bool> SignalReadyToSend;
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// Called whenever an RTCP packet is received. There is no equivalent signal
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// for RTP packets because they would be forwarded to the BaseChannel through
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// the RtpDemuxer callback.
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sigslot::signal2<rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
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SignalRtcpPacketReceived;
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// Called whenever the network route of the P2P layer transport changes.
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// The argument is an optional network route.
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sigslot::signal1<rtc::Optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
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// Called whenever a transport's writable state might change. The argument is
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// true if the transport is writable, otherwise it is false.
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sigslot::signal1<bool> SignalWritableState;
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sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
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virtual bool IsWritable(bool rtcp) const = 0;
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// TODO(zhihuang): Pass the |packet| by copy so that the original data
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// wouldn't be modified.
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virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) = 0;
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virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) = 0;
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// This method updates the RTP header extension map so that the RTP transport
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// can parse the received packets and identify the MID. This is called by the
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// BaseChannel when setting the content description.
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//
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// Note: This doesn't take the BUNDLE case in account meaning the RTP header
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// extension maps are not merged when BUNDLE is enabled. This is fine because
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// the ID for MID should be consistent among all the RTP transports.
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virtual void UpdateRtpHeaderExtensionMap(
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const cricket::RtpHeaderExtensions& header_extensions) = 0;
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virtual void SetMetricsObserver(
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rtc::scoped_refptr<MetricsObserverInterface> metrics_observer) = 0;
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virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
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RtpPacketSinkInterface* sink) = 0;
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virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORTINTERNAL_H_
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