0378997db3
This CL adds flags to the PacketOptions and PacktInfo struct that are intended to be used to indicate if the packet belongs to a media stream that is part of bitrate allocation as well as if it is included in transport wide packet feedback. This is part of a series of CLs that allows GoogCC to track sent bitrate that is included in bitrate allocation but without transport feedback. Bug: webrtc:9796 Change-Id: Icdf3e1e13d3f119574ee1b2c574f2d3329a7e303 Reviewed-on: https://webrtc-review.googlesource.com/c/104920 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25069}
121 lines
4.0 KiB
C++
121 lines
4.0 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 "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 "modules/rtp_rtcp/include/rtp_header_parser.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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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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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::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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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 = clock_->TimeInMicroseconds();
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fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
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send_time);
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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_ =
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task_queue_->PostDelayedTask([this]() { SendPackets(); }, delay_ms);
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}
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} // namespace test
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} // namespace webrtc
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