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This reverts commit3e61f881cd
. Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104 Original change's description: > Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp" > > This reverts commit3b96f2c770
. > > Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio > > > Original change's description: > > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp > > > > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped. > > Therefore DirectTransport is provided with the extension mapping. > > > > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour. > > > > > > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1 > > Bug: webrtc:7135, webrtc:14795 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980 > > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39137} > > Bug: webrtc:7135, webrtc:14795, webrtc:14833 > Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220 > Owners-Override: Björn Terelius <terelius@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#39146} Bug: webrtc:7135, webrtc:14795, webrtc:14833 Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Owners-Override: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39157}
151 lines
4.5 KiB
C++
151 lines
4.5 KiB
C++
/*
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* Copyright (c) 2013 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 TEST_RTP_RTCP_OBSERVER_H_
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#define TEST_RTP_RTCP_OBSERVER_H_
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#include <map>
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#include <memory>
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#include <utility>
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#include <vector>
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#include "api/array_view.h"
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#include "api/test/simulated_network.h"
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#include "api/units/time_delta.h"
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#include "call/simulated_packet_receiver.h"
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#include "call/video_send_stream.h"
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#include "modules/rtp_rtcp/source/rtp_util.h"
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#include "rtc_base/event.h"
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#include "system_wrappers/include/field_trial.h"
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#include "test/direct_transport.h"
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#include "test/gtest.h"
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namespace {
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constexpr webrtc::TimeDelta kShortTimeout = webrtc::TimeDelta::Millis(500);
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}
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namespace webrtc {
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namespace test {
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class PacketTransport;
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class RtpRtcpObserver {
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public:
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enum Action {
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SEND_PACKET,
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DROP_PACKET,
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};
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virtual ~RtpRtcpObserver() {}
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virtual bool Wait() {
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if (field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
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observation_complete_.Wait(kShortTimeout);
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return true;
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}
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return observation_complete_.Wait(timeout_);
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}
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virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
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return SEND_PACKET;
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}
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virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
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return SEND_PACKET;
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}
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virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
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return SEND_PACKET;
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}
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virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
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return SEND_PACKET;
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}
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protected:
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RtpRtcpObserver() : RtpRtcpObserver(TimeDelta::Zero()) {}
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explicit RtpRtcpObserver(TimeDelta event_timeout) : timeout_(event_timeout) {}
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rtc::Event observation_complete_;
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private:
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const TimeDelta timeout_;
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};
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class PacketTransport : public test::DirectTransport {
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public:
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enum TransportType { kReceiver, kSender };
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PacketTransport(TaskQueueBase* task_queue,
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Call* send_call,
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RtpRtcpObserver* observer,
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TransportType transport_type,
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const std::map<uint8_t, MediaType>& payload_type_map,
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std::unique_ptr<SimulatedPacketReceiverInterface> nw_pipe,
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rtc::ArrayView<const RtpExtension> audio_extensions,
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rtc::ArrayView<const RtpExtension> video_extensions)
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: test::DirectTransport(task_queue,
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std::move(nw_pipe),
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send_call,
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payload_type_map,
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audio_extensions,
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video_extensions),
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observer_(observer),
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transport_type_(transport_type) {}
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private:
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bool SendRtp(const uint8_t* packet,
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size_t length,
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const PacketOptions& options) override {
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EXPECT_TRUE(IsRtpPacket(rtc::MakeArrayView(packet, length)));
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RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET;
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if (observer_) {
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if (transport_type_ == kSender) {
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action = observer_->OnSendRtp(packet, length);
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} else {
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action = observer_->OnReceiveRtp(packet, length);
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}
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}
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switch (action) {
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case RtpRtcpObserver::DROP_PACKET:
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// Drop packet silently.
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return true;
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case RtpRtcpObserver::SEND_PACKET:
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return test::DirectTransport::SendRtp(packet, length, options);
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}
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return true; // Will never happen, makes compiler happy.
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}
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bool SendRtcp(const uint8_t* packet, size_t length) override {
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EXPECT_TRUE(IsRtcpPacket(rtc::MakeArrayView(packet, length)));
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RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET;
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if (observer_) {
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if (transport_type_ == kSender) {
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action = observer_->OnSendRtcp(packet, length);
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} else {
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action = observer_->OnReceiveRtcp(packet, length);
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}
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}
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switch (action) {
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case RtpRtcpObserver::DROP_PACKET:
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// Drop packet silently.
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return true;
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case RtpRtcpObserver::SEND_PACKET:
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return test::DirectTransport::SendRtcp(packet, length);
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}
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return true; // Will never happen, makes compiler happy.
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}
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RtpRtcpObserver* const observer_;
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TransportType transport_type_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_RTP_RTCP_OBSERVER_H_
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