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find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
123 lines
4.1 KiB
C++
123 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 "absl/memory/memory.h"
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#include "api/task_queue/task_queue_base.h"
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#include "api/units/time_delta.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/task_utils/repeating_task.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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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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TaskQueueBase* 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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next_process_task_.Stop();
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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_.Running())
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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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absl::optional<int64_t> initial_delay_ms =
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fake_network_->TimeUntilNextProcess();
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if (initial_delay_ms == absl::nullopt)
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return;
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next_process_task_ = RepeatingTaskHandle::DelayedStart(
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task_queue_, TimeDelta::Millis(*initial_delay_ms), [this] {
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fake_network_->Process();
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if (auto delay_ms = fake_network_->TimeUntilNextProcess())
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return TimeDelta::Millis(*delay_ms);
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// Otherwise stop the task.
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rtc::CritScope cs(&process_lock_);
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next_process_task_.Stop();
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// Since this task is stopped, return value doesn't matter.
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return TimeDelta::Zero();
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});
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}
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} // namespace test
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} // namespace webrtc
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