365381fdf1
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 The JsepTransport2 is renamed to JsepTransport. 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: Ibd880e7b744bd912336a691309950bc18e42cf62 Reviewed-on: https://webrtc-review.googlesource.com/65786 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22867}
79 lines
2.4 KiB
C++
79 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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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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