6516f76f9b
This class doesn't strictly follow rtc::TaskQueue semantics, which makes it surprising and hard to use correctly. Please use TaskQueueForTest instead. This CL follows usual deprecation process: 1/ Rename. % for i in `git ls-files` ; sed -i "s:SingleThreadedTaskQueueForTesting:DEPRECATED_SingleThreadedTaskQueueForTesting:" $i 2/ Annotate old name for downstream users and accidental new uses. Bug: webrtc:10933 Change-Id: I80b4ee5a48df1f63f63a43ed0efdb50eb7fb156a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150788 Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Yves Gerey <yvesg@google.com> Cr-Commit-Position: refs/heads/master@{#29045}
124 lines
3.9 KiB
C++
124 lines
3.9 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 "absl/memory/memory.h"
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#include "call/call.h"
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#include "call/fake_network_pipe.h"
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#include "rtc_base/time_utils.h"
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#include "test/rtp_header_parser.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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Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
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: payload_type_map_(payload_type_map) {}
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MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
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const size_t packet_length) const {
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if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
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RTC_CHECK_GE(packet_length, 2);
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const uint8_t payload_type = packet_data[1] & 0x7f;
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std::map<uint8_t, MediaType>::const_iterator it =
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payload_type_map_.find(payload_type);
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RTC_CHECK(it != payload_type_map_.end())
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<< "payload type " << static_cast<int>(payload_type) << " unknown.";
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return it->second;
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}
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return MediaType::ANY;
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}
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DirectTransport::DirectTransport(
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DEPRECATED_SingleThreadedTaskQueueForTesting* task_queue,
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std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
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Call* send_call,
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const std::map<uint8_t, MediaType>& payload_type_map)
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: send_call_(send_call),
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task_queue_(task_queue),
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demuxer_(payload_type_map),
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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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if (next_process_task_)
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task_queue_->CancelTask(*next_process_task_);
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}
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void DirectTransport::StopSending() {
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rtc::CritScope cs(&process_lock_);
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if (next_process_task_)
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task_queue_->CancelTask(*next_process_task_);
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}
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void DirectTransport::SetReceiver(PacketReceiver* receiver) {
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rtc::CritScope cs(&process_lock_);
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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, rtc::TimeMillis());
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sent_packet.info.included_in_feedback = options.included_in_feedback;
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sent_packet.info.included_in_allocation = options.included_in_allocation;
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sent_packet.info.packet_size_bytes = length;
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sent_packet.info.packet_type = rtc::PacketType::kData;
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send_call_->OnSentPacket(sent_packet);
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}
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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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SendPacket(data, length);
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return true;
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}
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void DirectTransport::SendPacket(const uint8_t* data, size_t length) {
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MediaType media_type = demuxer_.GetMediaType(data, length);
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int64_t send_time_us = rtc::TimeMicros();
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fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
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send_time_us);
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rtc::CritScope cs(&process_lock_);
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if (!next_process_task_)
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ProcessPackets();
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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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}
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void DirectTransport::ProcessPackets() {
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next_process_task_.reset();
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auto delay_ms = fake_network_->TimeUntilNextProcess();
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if (delay_ms) {
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next_process_task_ = task_queue_->PostDelayedTask(
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[this]() {
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fake_network_->Process();
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rtc::CritScope cs(&process_lock_);
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ProcessPackets();
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},
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*delay_ms);
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}
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}
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} // namespace test
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} // namespace webrtc
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