e1e94ad4c8
With this cl, packets that are discarded in RtpTransport now notifies Call, so that they can be part of BWE even if they are dropped. These packets have been recevied on the transport, and has bin decrypted and parsed and thus can be accounted for. The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded. Bug: webrtc:14928 Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39727}
87 lines
2.7 KiB
C++
87 lines
2.7 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_TEST_RTP_TRANSPORT_TEST_UTIL_H_
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#define PC_TEST_RTP_TRANSPORT_TEST_UTIL_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/rtp_transport_internal.h"
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#include "rtc_base/third_party/sigslot/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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rtp_transport->SignalUnDemuxableRtpPacketReceived.connect(
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this, &TransportObserver::OnUndemuxableRtpPacket);
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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 OnUndemuxableRtpPacket(const RtpPacketReceived& packet) {
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un_demuxable_rtp_count_++;
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}
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void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
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int64_t packet_time_us) {
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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 un_demuxable_rtp_count() const { return un_demuxable_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 un_demuxable_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_TEST_RTP_TRANSPORT_TEST_UTIL_H_
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