
This CL moves the responsibility for demuxing from FakeNetworkPipe to DirectTransport. This makes the interface for FakeNetworkPipe more consistent. It exposes fewer different interfaces for different usages. It also means that any time degradations applied to the packets due in FakeNetworkPipe in tests will now be propagated to Call in a more realistic manner. Previously the time was set to uninitialized which meant that Call filled in values based on the system clock. Bug: webrtc:9054 Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0 Reviewed-on: https://webrtc-review.googlesource.com/64260 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23017}
148 lines
4.6 KiB
C++
148 lines
4.6 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 "modules/rtp_rtcp/include/rtp_header_parser.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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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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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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: send_call_(send_call),
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clock_(Clock::GetRealTimeClock()),
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task_queue_(task_queue),
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demuxer_(payload_type_map),
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fake_network_(rtc::MakeUnique<FakeNetworkPipe>(clock_, config)) {
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Start();
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}
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DirectTransport::DirectTransport(
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SingleThreadedTaskQueueForTesting* task_queue,
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std::unique_ptr<FakeNetworkPipe> 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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clock_(Clock::GetRealTimeClock()),
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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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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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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 = clock_->TimeInMicroseconds();
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fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
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PacketTime(send_time, -1));
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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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