mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
Add support for time-varying constraints in DegradedCall.
The fake network configs are now specified using just two field trials: WebRTC-FakeNetworkSendConfig and WebRTC-FakeNetworkReceiveConfig. Both of them have the following parameters from BuiltInNetworkBehaviorConfig: * queue_length_packets // Queue length in number of packets. * queue_delay_ms // Delay in addition to capacity induced delay. * delay_standard_deviation_ms // Standard deviation of the extra delay. * link_capacity_kbps // Link capacity in kbps. * loss_percent // Random packet loss. * allow_reordering // If packets are allowed to be reordered. * avg_burst_loss_length // The average length of a burst of lost packets. * packet_overhead // Additional bytes to add to packet size. * codel_active_queue_management // Enable CoDel active queue management. Plus: * duration // For how long to use this config before progressing. Example: WebRTC-FakeNetworkSendConfig/queue_delay_ms:66|1,loss_percent:1|0,link_capacity_kbps:200|10000,queue_length_packets:10|100,duration:15s|45s/ This creates two configs: 1. For 15s, apply 66ms delay, 1% loss, 200kbps bandwidth, 10 packet queue size 2. For 45s, apply 1ms delay, 0% loss, 10Mbps bandwidth, 100 packets queue size (then repeat) Bug: webrtc:13655 Change-Id: I0524f572de480731df4d414724203772182c628b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251043 Reviewed-by: Stefan Holmer <holmer@google.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35952}
This commit is contained in:
Erik Språng
WebRTC LUCI CQ
parent
27c1452fae
commit
b5cba85c2f
3 changed files with 154 additions and 44 deletions
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@ -15,26 +15,34 @@
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/test/simulated_network.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/degraded_call.h"
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#include "call/rtp_transport_config.h"
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#include "rtc_base/checks.h"
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#include "system_wrappers/include/field_trial.h"
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#include "rtc_base/experiments/field_trial_list.h"
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#include "rtc_base/experiments/field_trial_parser.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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using TimeScopedNetworkConfig = DegradedCall::TimeScopedNetworkConfig;
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bool ParseConfigParam(const WebRtcKeyValueConfig& trials,
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absl::string_view exp_name,
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int* field) {
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std::string group = trials.Lookup(exp_name);
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if (group.empty())
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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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absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
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absl::optional<TimeScopedNetworkConfig> ParseDegradationConfig(
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const WebRtcKeyValueConfig& trials,
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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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@ -43,33 +51,92 @@ absl::optional<webrtc::BuiltInNetworkBehaviorConfig> ParseDegradationConfig(
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exp_prefix += "Receive";
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}
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webrtc::BuiltInNetworkBehaviorConfig config;
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TimeScopedNetworkConfig 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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ParseConfigParam(trials, exp_prefix + "DelayMs", &config.queue_delay_ms);
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configured |= ParseConfigParam(trials, 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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if (ParseConfigParam(trials, 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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configured |= ParseConfigParam(trials, exp_prefix + "CapacityKbps",
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&config.link_capacity_kbps);
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configured |= ParseConfigParam(trials, exp_prefix + "LossPercent",
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&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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if (ParseConfigParam(trials, exp_prefix + "AllowReordering",
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&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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configured |= ParseConfigParam(trials, exp_prefix + "AvgBurstLossLength",
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&config.avg_burst_loss_length);
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return configured
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? absl::optional<webrtc::BuiltInNetworkBehaviorConfig>(config)
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: absl::nullopt;
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return configured ? absl::optional<TimeScopedNetworkConfig>(config)
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: absl::nullopt;
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}
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std::vector<TimeScopedNetworkConfig> GetNetworkConfigs(
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const WebRtcKeyValueConfig& trials,
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bool send) {
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FieldTrialStructList<TimeScopedNetworkConfig> trials_list(
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{FieldTrialStructMember("queue_length_packets",
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[](TimeScopedNetworkConfig* p) {
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// FieldTrialParser does not natively support
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// size_t type, so use this ugly cast as
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// workaround.
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return reinterpret_cast<unsigned*>(
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&p->queue_length_packets);
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}),
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FieldTrialStructMember(
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"queue_delay_ms",
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[](TimeScopedNetworkConfig* p) { return &p->queue_delay_ms; }),
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FieldTrialStructMember("delay_standard_deviation_ms",
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[](TimeScopedNetworkConfig* p) {
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return &p->delay_standard_deviation_ms;
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}),
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FieldTrialStructMember(
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"link_capacity_kbps",
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[](TimeScopedNetworkConfig* p) { return &p->link_capacity_kbps; }),
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FieldTrialStructMember(
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"loss_percent",
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[](TimeScopedNetworkConfig* p) { return &p->loss_percent; }),
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FieldTrialStructMember(
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"allow_reordering",
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[](TimeScopedNetworkConfig* p) { return &p->allow_reordering; }),
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FieldTrialStructMember("avg_burst_loss_length",
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[](TimeScopedNetworkConfig* p) {
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return &p->avg_burst_loss_length;
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}),
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FieldTrialStructMember(
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"packet_overhead",
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[](TimeScopedNetworkConfig* p) { return &p->packet_overhead; }),
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FieldTrialStructMember("codel_active_queue_management",
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[](TimeScopedNetworkConfig* p) {
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return &p->codel_active_queue_management;
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}),
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FieldTrialStructMember(
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"duration",
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[](TimeScopedNetworkConfig* p) { return &p->duration; })},
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{});
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ParseFieldTrial({&trials_list},
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trials.Lookup(send ? "WebRTC-FakeNetworkSendConfig"
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: "WebRTC-FakeNetworkReceiveConfig"));
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std::vector<TimeScopedNetworkConfig> configs = trials_list.Get();
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if (configs.empty()) {
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// Try legacy fallback trials.
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absl::optional<DegradedCall::TimeScopedNetworkConfig> fallback_config =
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ParseDegradationConfig(trials, send);
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if (fallback_config.has_value()) {
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configs.push_back(*fallback_config);
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}
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}
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return configs;
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}
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} // namespace
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CallFactory::CallFactory() {
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@ -78,14 +145,18 @@ CallFactory::CallFactory() {
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Call* CallFactory::CreateCall(const Call::Config& config) {
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RTC_DCHECK_RUN_ON(&call_thread_);
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absl::optional<webrtc::BuiltInNetworkBehaviorConfig> send_degradation_config =
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ParseDegradationConfig(true);
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absl::optional<webrtc::BuiltInNetworkBehaviorConfig>
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receive_degradation_config = ParseDegradationConfig(false);
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RTC_DCHECK(config.trials);
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std::vector<DegradedCall::TimeScopedNetworkConfig> send_degradation_configs =
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GetNetworkConfigs(*config.trials, /*send=*/true);
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std::vector<DegradedCall::TimeScopedNetworkConfig>
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receive_degradation_configs =
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GetNetworkConfigs(*config.trials, /*send=*/false);
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RtpTransportConfig transportConfig = config.ExtractTransportConfig();
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if (send_degradation_config || receive_degradation_config) {
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if (!send_degradation_configs.empty() ||
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!receive_degradation_configs.empty()) {
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return new DegradedCall(
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std::unique_ptr<Call>(Call::Create(
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config, Clock::GetRealTimeClock(),
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@ -94,7 +165,7 @@ Call* CallFactory::CreateCall(const Call::Config& config) {
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config.rtp_transport_controller_send_factory->Create(
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transportConfig, Clock::GetRealTimeClock(),
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ProcessThread::Create("PacerThread")))),
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send_degradation_config, receive_degradation_config,
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send_degradation_configs, receive_degradation_configs,
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config.task_queue_factory);
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}
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@ -127,27 +127,39 @@ bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp(
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DegradedCall::DegradedCall(
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std::unique_ptr<Call> call,
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absl::optional<BuiltInNetworkBehaviorConfig> send_config,
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absl::optional<BuiltInNetworkBehaviorConfig> receive_config,
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const std::vector<TimeScopedNetworkConfig>& send_configs,
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const std::vector<TimeScopedNetworkConfig>& receive_configs,
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TaskQueueFactory* task_queue_factory)
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: clock_(Clock::GetRealTimeClock()),
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call_(std::move(call)),
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task_queue_factory_(task_queue_factory),
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send_config_(send_config),
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send_config_index_(0),
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send_configs_(send_configs),
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send_simulated_network_(nullptr),
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receive_config_(receive_config) {
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if (receive_config_) {
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auto network = std::make_unique<SimulatedNetwork>(*receive_config_);
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receive_config_index_(0),
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receive_configs_(receive_configs) {
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if (!receive_configs_.empty()) {
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auto network = std::make_unique<SimulatedNetwork>(receive_configs_[0]);
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receive_simulated_network_ = network.get();
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receive_pipe_ =
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std::make_unique<webrtc::FakeNetworkPipe>(clock_, std::move(network));
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receive_pipe_->SetReceiver(call_->Receiver());
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if (receive_configs_.size() > 1) {
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call_->network_thread()->PostDelayedTask(
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ToQueuedTask(task_safety_, [this] { UpdateReceiveNetworkConfig(); }),
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receive_configs_[0].duration.ms());
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}
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}
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if (send_config_) {
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auto network = std::make_unique<SimulatedNetwork>(*send_config_);
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if (!send_configs_.empty()) {
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auto network = std::make_unique<SimulatedNetwork>(send_configs_[0]);
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send_simulated_network_ = network.get();
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send_pipe_ = std::make_unique<FakeNetworkPipeOnTaskQueue>(
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task_queue_factory_, clock_, std::move(network));
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if (send_configs_.size() > 1) {
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call_->network_thread()->PostDelayedTask(
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ToQueuedTask(task_safety_, [this] { UpdateSendNetworkConfig(); }),
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send_configs_[0].duration.ms());
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}
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}
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}
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@ -155,7 +167,7 @@ DegradedCall::~DegradedCall() = default;
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AudioSendStream* DegradedCall::CreateAudioSendStream(
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const AudioSendStream::Config& config) {
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if (send_config_) {
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if (!send_configs_.empty()) {
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auto transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
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send_pipe_.get(), call_.get(), clock_, config.send_transport);
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AudioSendStream::Config degrade_config = config;
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@ -189,7 +201,7 @@ VideoSendStream* DegradedCall::CreateVideoSendStream(
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VideoSendStream::Config config,
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VideoEncoderConfig encoder_config) {
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std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
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if (send_config_) {
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if (!send_configs_.empty()) {
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transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
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send_pipe_.get(), call_.get(), clock_, config.send_transport);
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config.send_transport = transport_adapter.get();
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@ -207,7 +219,7 @@ VideoSendStream* DegradedCall::CreateVideoSendStream(
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VideoEncoderConfig encoder_config,
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std::unique_ptr<FecController> fec_controller) {
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std::unique_ptr<FakeNetworkPipeTransportAdapter> transport_adapter;
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if (send_config_) {
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if (!send_configs_.empty()) {
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transport_adapter = std::make_unique<FakeNetworkPipeTransportAdapter>(
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send_pipe_.get(), call_.get(), clock_, config.send_transport);
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config.send_transport = transport_adapter.get();
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@ -251,7 +263,7 @@ void DegradedCall::AddAdaptationResource(
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}
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PacketReceiver* DegradedCall::Receiver() {
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if (receive_config_) {
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if (!receive_configs_.empty()) {
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return this;
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}
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return call_->Receiver();
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@ -299,7 +311,7 @@ void DegradedCall::OnUpdateSyncGroup(AudioReceiveStream& stream,
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}
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void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
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if (send_config_) {
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if (!send_configs_.empty()) {
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// If we have a degraded send-transport, we have already notified call
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// about the supposed network send time. Discard the actual network send
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// time in order to properly fool the BWE.
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@ -325,4 +337,21 @@ PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
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receive_pipe_->Process();
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return status;
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}
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void DegradedCall::UpdateSendNetworkConfig() {
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send_config_index_ = (send_config_index_ + 1) % send_configs_.size();
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send_simulated_network_->SetConfig(send_configs_[send_config_index_]);
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call_->network_thread()->PostDelayedTask(
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ToQueuedTask(task_safety_, [this] { UpdateSendNetworkConfig(); }),
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send_configs_[send_config_index_].duration.ms());
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}
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void DegradedCall::UpdateReceiveNetworkConfig() {
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receive_config_index_ = (receive_config_index_ + 1) % receive_configs_.size();
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receive_simulated_network_->SetConfig(
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receive_configs_[receive_config_index_]);
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call_->network_thread()->PostDelayedTask(
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ToQueuedTask(task_safety_, [this] { UpdateReceiveNetworkConfig(); }),
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receive_configs_[receive_config_index_].duration.ms());
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}
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} // namespace webrtc
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@ -17,6 +17,7 @@
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/call/transport.h"
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@ -38,15 +39,20 @@
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/network/sent_packet.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/task_utils/pending_task_safety_flag.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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class DegradedCall : public Call, private PacketReceiver {
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public:
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struct TimeScopedNetworkConfig : public BuiltInNetworkBehaviorConfig {
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TimeDelta duration = TimeDelta::PlusInfinity();
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};
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explicit DegradedCall(
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std::unique_ptr<Call> call,
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absl::optional<BuiltInNetworkBehaviorConfig> send_config,
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absl::optional<BuiltInNetworkBehaviorConfig> receive_config,
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const std::vector<TimeScopedNetworkConfig>& send_configs,
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const std::vector<TimeScopedNetworkConfig>& receive_configs,
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TaskQueueFactory* task_queue_factory);
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~DegradedCall() override;
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@ -157,14 +163,17 @@ class DegradedCall : public Call, private PacketReceiver {
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Transport* const real_transport_;
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};
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Clock* const clock_;
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const std::unique_ptr<Call> call_;
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TaskQueueFactory* const task_queue_factory_;
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void SetClientBitratePreferences(
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const webrtc::BitrateSettings& preferences) override {}
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void UpdateSendNetworkConfig();
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void UpdateReceiveNetworkConfig();
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const absl::optional<BuiltInNetworkBehaviorConfig> send_config_;
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Clock* const clock_;
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const std::unique_ptr<Call> call_;
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ScopedTaskSafety task_safety_;
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TaskQueueFactory* const task_queue_factory_;
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size_t send_config_index_;
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const std::vector<TimeScopedNetworkConfig> send_configs_;
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SimulatedNetwork* send_simulated_network_;
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std::unique_ptr<FakeNetworkPipeOnTaskQueue> send_pipe_;
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std::map<AudioSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
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@ -172,7 +181,8 @@ class DegradedCall : public Call, private PacketReceiver {
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std::map<VideoSendStream*, std::unique_ptr<FakeNetworkPipeTransportAdapter>>
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video_send_transport_adapters_;
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const absl::optional<BuiltInNetworkBehaviorConfig> receive_config_;
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size_t receive_config_index_;
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const std::vector<TimeScopedNetworkConfig> receive_configs_;
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SimulatedNetwork* receive_simulated_network_;
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std::unique_ptr<FakeNetworkPipe> receive_pipe_;
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};
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