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}
80 lines
2.4 KiB
C++
80 lines
2.4 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_RTPTRANSPORTTESTUTIL_H_
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#define PC_RTPTRANSPORTTESTUTIL_H_
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#include "call/rtp_packet_sink_interface.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "pc/rtptransportinternal.h"
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#include "rtc_base/sigslot.h"
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namespace webrtc {
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// Used to handle the signals when the RtpTransport receives an RTP/RTCP packet.
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// Used in Rtp/Srtp/DtlsTransport unit tests.
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class TransportObserver : public RtpPacketSinkInterface,
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public sigslot::has_slots<> {
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public:
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TransportObserver() {}
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explicit TransportObserver(RtpTransportInternal* rtp_transport) {
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rtp_transport->SignalRtcpPacketReceived.connect(
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this, &TransportObserver::OnRtcpPacketReceived);
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rtp_transport->SignalReadyToSend.connect(this,
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&TransportObserver::OnReadyToSend);
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}
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// RtpPacketInterface override.
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void OnRtpPacket(const RtpPacketReceived& packet) override {
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rtp_count_++;
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last_recv_rtp_packet_ = packet.Buffer();
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}
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void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) {
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RTC_LOG(LS_INFO) << "Received an RTCP packet.";
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rtcp_count_++;
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last_recv_rtcp_packet_ = *packet;
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}
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int rtp_count() const { return rtp_count_; }
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int rtcp_count() const { return rtcp_count_; }
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rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
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return last_recv_rtp_packet_;
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}
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rtc::CopyOnWriteBuffer last_recv_rtcp_packet() {
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return last_recv_rtcp_packet_;
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}
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void OnReadyToSend(bool ready) {
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ready_to_send_signal_count_++;
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ready_to_send_ = ready;
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}
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bool ready_to_send() { return ready_to_send_; }
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int ready_to_send_signal_count() { return ready_to_send_signal_count_; }
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private:
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bool ready_to_send_ = false;
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int rtp_count_ = 0;
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int rtcp_count_ = 0;
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int ready_to_send_signal_count_ = 0;
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rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
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rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORTTESTUTIL_H_
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