DirectTransport has so far used its own thread, which led to a different threading-model for in the unit-tests than is used in actual WebRTC. Because of that, some critical-sections that weren't truly necessary in WebRTC could not be replaced with thread-checks, because those checks failed in unit-tests. This CL introduces SingleThreadedTaskQueue - a TaskQueue which guarantees to run all of its tasks on the same thread (rtc::TaskQueue doesn't guarantee that on Mac) - and uses that for DirectTransport. CLs based on top of this will uncomment thread-checks which had to be commented out before, and remove unnecessary critical-sections. Future work would probably replace the thread-checkers by more sophisticated serialized-access checks, allowing us to move from the SingleThreadedTaskQueue to a normal TaskQueue. Related implementation notes: * This CL has made DirectTransport::StopSending() superfluous, and so it was deleted. BUG=webrtc:8113, webrtc:7405, webrtc:8056, webrtc:8116 Review-Url: https://codereview.webrtc.org/2998923002 Cr-Commit-Position: refs/heads/master@{#19445}
154 lines
5.0 KiB
C++
154 lines
5.0 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 "webrtc/test/direct_transport.h"
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#include "webrtc/call/call.h"
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#include "webrtc/rtc_base/ptr_util.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/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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Call* send_call,
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const std::map<uint8_t, MediaType>& payload_type_map)
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: DirectTransport(FakeNetworkPipe::Config(), send_call, payload_type_map) {}
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DirectTransport::DirectTransport(
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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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config,
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send_call,
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std::unique_ptr<Demuxer>(new DemuxerImpl(payload_type_map))) {}
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DirectTransport::DirectTransport(const FakeNetworkPipe::Config& config,
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Call* send_call,
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std::unique_ptr<Demuxer> demuxer)
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: DirectTransport(nullptr, config, send_call, std::move(demuxer)) {}
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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_(clock_, config, std::move(demuxer)) {
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// TODO(eladalon): When the deprecated ctors are removed, this check
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// can be restored. https://bugs.chromium.org/p/webrtc/issues/detail?id=8125
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// RTC_DCHECK(task_queue);
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if (!task_queue) {
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deprecated_task_queue_ =
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rtc::MakeUnique<SingleThreadedTaskQueueForTesting>("deprecated_queue");
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task_queue_ = deprecated_task_queue_.get();
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}
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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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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::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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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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DirectTransport::ForceDemuxer::ForceDemuxer(MediaType media_type)
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: media_type_(media_type) {}
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void DirectTransport::ForceDemuxer::SetReceiver(PacketReceiver* receiver) {
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packet_receiver_ = receiver;
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}
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void DirectTransport::ForceDemuxer::DeliverPacket(
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const NetworkPacket* packet,
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const PacketTime& packet_time) {
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if (!packet_receiver_)
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return;
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packet_receiver_->DeliverPacket(media_type_, packet->data(),
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packet->data_length(), packet_time);
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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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