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776c1a1a86
Bug: webrtc:15368 Change-Id: Ie2d982a9172759a65f7f7225eeddd64cfa82490d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41903}
109 lines
3.6 KiB
C++
109 lines
3.6 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 <utility>
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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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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:
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TransportObserver() {}
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explicit TransportObserver(RtpTransportInternal* rtp_transport) {
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rtp_transport->SubscribeRtcpPacketReceived(
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this, [this](rtc::CopyOnWriteBuffer* buffer, int64_t packet_time_ms) {
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OnRtcpPacketReceived(buffer, packet_time_ms);
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});
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rtp_transport->SubscribeReadyToSend(
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this, [this](bool arg) { OnReadyToSend(arg); });
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rtp_transport->SetUnDemuxableRtpPacketReceivedHandler(
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[this](RtpPacketReceived& packet) { OnUndemuxableRtpPacket(packet); });
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rtp_transport->SubscribeSentPacket(this,
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[this](const rtc::SentPacket& packet) {
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sent_packet_count_++;
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if (action_on_sent_packet_) {
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action_on_sent_packet_();
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}
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});
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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;
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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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int sent_packet_count() const { return sent_packet_count_; }
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const RtpPacketReceived& 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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if (action_on_ready_to_send_) {
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action_on_ready_to_send_(ready);
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}
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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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void SetActionOnReadyToSend(absl::AnyInvocable<void(bool)> action) {
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action_on_ready_to_send_ = std::move(action);
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}
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void SetActionOnSentPacket(absl::AnyInvocable<void()> action) {
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action_on_sent_packet_ = std::move(action);
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}
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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 sent_packet_count_ = 0;
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int ready_to_send_signal_count_ = 0;
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RtpPacketReceived last_recv_rtp_packet_;
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rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
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absl::AnyInvocable<void(bool)> action_on_ready_to_send_;
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absl::AnyInvocable<void()> action_on_sent_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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