7e85d67031
Added functionality on the FakeNetworkPipe to introduce arbitrary clock offsets. This offset is added to the reported receive time of all packets. This prepares for a later CL using this to test correction of receive time stamps. Bug: webrtc:9054 Change-Id: I811b3aa8359bc917f59443088d8a418368242db9 Reviewed-on: https://webrtc-review.googlesource.com/64726 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22763}
131 lines
4.1 KiB
C++
131 lines
4.1 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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#include "test/direct_transport.h"
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#include "call/call.h"
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#include "rtc_base/ptr_util.h"
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#include "system_wrappers/include/clock.h"
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#include "test/single_threaded_task_queue.h"
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namespace webrtc {
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namespace test {
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DirectTransport::DirectTransport(
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SingleThreadedTaskQueueForTesting* task_queue,
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Call* send_call,
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const std::map<uint8_t, MediaType>& payload_type_map)
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: DirectTransport(task_queue,
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FakeNetworkPipe::Config(),
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send_call,
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payload_type_map) {
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}
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DirectTransport::DirectTransport(
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SingleThreadedTaskQueueForTesting* task_queue,
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const FakeNetworkPipe::Config& config,
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Call* send_call,
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const std::map<uint8_t, MediaType>& payload_type_map)
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: DirectTransport(
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task_queue,
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config,
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send_call,
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std::unique_ptr<Demuxer>(new DemuxerImpl(payload_type_map))) {
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}
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DirectTransport::DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
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const FakeNetworkPipe::Config& config,
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Call* send_call,
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std::unique_ptr<Demuxer> demuxer)
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: send_call_(send_call),
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clock_(Clock::GetRealTimeClock()),
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task_queue_(task_queue),
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fake_network_(rtc::MakeUnique<FakeNetworkPipe>(clock_, config,
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std::move(demuxer))) {
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Start();
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}
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DirectTransport::DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
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std::unique_ptr<FakeNetworkPipe> pipe,
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Call* send_call)
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: send_call_(send_call),
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clock_(Clock::GetRealTimeClock()),
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task_queue_(task_queue),
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fake_network_(std::move(pipe)) {
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Start();
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}
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DirectTransport::~DirectTransport() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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// Constructor updates |next_scheduled_task_|, so it's guaranteed to
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// be initialized.
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task_queue_->CancelTask(next_scheduled_task_);
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}
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void DirectTransport::SetClockOffset(int64_t offset_ms) {
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fake_network_->SetClockOffset(offset_ms);
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}
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void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
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fake_network_->SetConfig(config);
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}
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void DirectTransport::StopSending() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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task_queue_->CancelTask(next_scheduled_task_);
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}
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void DirectTransport::SetReceiver(PacketReceiver* receiver) {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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fake_network_->SetReceiver(receiver);
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}
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bool DirectTransport::SendRtp(const uint8_t* data,
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size_t length,
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const PacketOptions& options) {
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if (send_call_) {
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rtc::SentPacket sent_packet(options.packet_id,
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clock_->TimeInMilliseconds());
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send_call_->OnSentPacket(sent_packet);
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}
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fake_network_->SendPacket(data, length);
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return true;
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}
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bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
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fake_network_->SendPacket(data, length);
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return true;
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}
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int DirectTransport::GetAverageDelayMs() {
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return fake_network_->AverageDelay();
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}
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void DirectTransport::Start() {
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RTC_DCHECK(task_queue_);
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if (send_call_) {
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send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
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send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
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}
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SendPackets();
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}
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void DirectTransport::SendPackets() {
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RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
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fake_network_->Process();
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int64_t delay_ms = fake_network_->TimeUntilNextProcess();
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next_scheduled_task_ = task_queue_->PostDelayedTask([this]() {
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SendPackets();
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}, delay_ms);
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}
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} // namespace test
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} // namespace webrtc
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