95e7dbb7c7
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}
94 lines
3.5 KiB
C++
94 lines
3.5 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/rtptransportinterface.h"
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#include "api/umametrics.h"
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#include "p2p/base/icetransportinternal.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 RtpTransportInterface;
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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 RtpTransportInterface,
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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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// TODO(zstein): Consider having two signals - RtpPacketReceived and
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// RtcpPacketReceived.
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// The first argument is true for RTCP packets and false for RTP packets.
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sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
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SignalPacketReceived;
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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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virtual bool HandlesPayloadType(int payload_type) const = 0;
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virtual void AddHandledPayloadType(int payload_type) = 0;
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virtual void SetMetricsObserver(
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rtc::scoped_refptr<MetricsObserverInterface> metrics_observer) = 0;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORTINTERNAL_H_
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