097085140e
This is especially useful in Chrome, allowing use to emulate network conditions in incoming or outgoing media without the need for platform specific tools or hacks. It also doesn't interfere with the rest of the network traffic. Also includes some refactorings. TBR=stefan@webrtc.org, philipel@webrtc.org Originally reviewed on: https://webrtc-review.googlesource.com/33013 Bug: webrtc:8910 Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0 Reviewed-on: https://webrtc-review.googlesource.com/61782 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22430}
90 lines
2.9 KiB
C++
90 lines
2.9 KiB
C++
/*
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* Copyright 2017 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 "call/callfactory.h"
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#include <memory>
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#include <string>
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#include <utility>
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#include "api/optional.h"
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#include "call/call.h"
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#include "call/degraded_call.h"
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#include "call/fake_network_pipe.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace {
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bool ParseConfigParam(std::string exp_name, int* field) {
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std::string group = field_trial::FindFullName(exp_name);
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if (group == "")
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return false;
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return (sscanf(group.c_str(), "%d", field) == 1);
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}
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rtc::Optional<webrtc::FakeNetworkPipe::Config> ParseDegradationConfig(
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bool send) {
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std::string exp_prefix = "WebRTCFakeNetwork";
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if (send) {
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exp_prefix += "Send";
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} else {
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exp_prefix += "Receive";
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}
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webrtc::FakeNetworkPipe::Config config;
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bool configured = false;
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configured |=
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ParseConfigParam(exp_prefix + "DelayMs", &config.queue_delay_ms);
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configured |= ParseConfigParam(exp_prefix + "DelayStdDevMs",
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&config.delay_standard_deviation_ms);
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int queue_length = 0;
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if (ParseConfigParam(exp_prefix + "QueueLength", &queue_length)) {
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RTC_CHECK_GE(queue_length, 0);
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config.queue_length_packets = queue_length;
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configured = true;
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}
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configured |=
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ParseConfigParam(exp_prefix + "CapacityKbps", &config.link_capacity_kbps);
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configured |=
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ParseConfigParam(exp_prefix + "LossPercent", &config.loss_percent);
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int allow_reordering = 0;
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if (ParseConfigParam(exp_prefix + "AllowReordering", &allow_reordering)) {
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config.allow_reordering = true;
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configured = true;
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}
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configured |= ParseConfigParam(exp_prefix + "AvgBurstLossLength",
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&config.avg_burst_loss_length);
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return configured ? rtc::Optional<webrtc::FakeNetworkPipe::Config>(config)
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: rtc::nullopt;
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}
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} // namespace
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Call* CallFactory::CreateCall(const Call::Config& config) {
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rtc::Optional<webrtc::FakeNetworkPipe::Config> send_degradation_config =
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ParseDegradationConfig(true);
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rtc::Optional<webrtc::FakeNetworkPipe::Config> receive_degradation_config =
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ParseDegradationConfig(false);
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if (send_degradation_config || receive_degradation_config) {
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return new DegradedCall(std::unique_ptr<Call>(Call::Create(config)),
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send_degradation_config,
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receive_degradation_config);
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}
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return Call::Create(config);
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}
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std::unique_ptr<CallFactoryInterface> CreateCallFactory() {
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return std::unique_ptr<CallFactoryInterface>(new CallFactory());
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}
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} // namespace webrtc
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