Simplifies transport overhead mechanism in Scenario test framework.
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.
Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
diff --git a/api/test/network_emulation/network_emulation_interfaces.cc b/api/test/network_emulation/network_emulation_interfaces.cc
index 50f8bed..e023334 100644
--- a/api/test/network_emulation/network_emulation_interfaces.cc
+++ b/api/test/network_emulation/network_emulation_interfaces.cc
@@ -12,19 +12,23 @@
namespace webrtc {
namespace {
-constexpr size_t kIPv4HeaderSize = 20;
-constexpr size_t kIPv6HeaderSize = 40;
+constexpr int kIPv4HeaderSize = 20;
+constexpr int kIPv6HeaderSize = 40;
+constexpr int kUdpHeaderSize = 8;
+int IpHeaderSize(const rtc::SocketAddress& address) {
+ return (address.family() == AF_INET) ? kIPv4HeaderSize : kIPv6HeaderSize;
+}
} // namespace
EmulatedIpPacket::EmulatedIpPacket(const rtc::SocketAddress& from,
const rtc::SocketAddress& to,
rtc::CopyOnWriteBuffer data,
- Timestamp arrival_time)
+ Timestamp arrival_time,
+ uint16_t application_overhead)
: from(from),
to(to),
data(data),
- ip_header_size((to.family() == AF_INET) ? kIPv4HeaderSize
- : kIPv6HeaderSize),
+ headers_size(IpHeaderSize(to) + application_overhead + kUdpHeaderSize),
arrival_time(arrival_time) {
RTC_DCHECK(to.family() == AF_INET || to.family() == AF_INET6);
}
diff --git a/api/test/network_emulation/network_emulation_interfaces.h b/api/test/network_emulation/network_emulation_interfaces.h
index 35ebabc..5d75bf3 100644
--- a/api/test/network_emulation/network_emulation_interfaces.h
+++ b/api/test/network_emulation/network_emulation_interfaces.h
@@ -22,12 +22,11 @@
struct EmulatedIpPacket {
public:
- static constexpr int kUdpHeaderSize = 8;
-
EmulatedIpPacket(const rtc::SocketAddress& from,
const rtc::SocketAddress& to,
rtc::CopyOnWriteBuffer data,
- Timestamp arrival_time);
+ Timestamp arrival_time,
+ uint16_t application_overhead = 0);
~EmulatedIpPacket() = default;
// This object is not copyable or assignable.
EmulatedIpPacket(const EmulatedIpPacket&) = delete;
@@ -39,14 +38,12 @@
size_t size() const { return data.size(); }
const uint8_t* cdata() const { return data.cdata(); }
- size_t ip_packet_size() const {
- return size() + kUdpHeaderSize + ip_header_size;
- }
+ size_t ip_packet_size() const { return size() + headers_size; }
rtc::SocketAddress from;
rtc::SocketAddress to;
// Holds the UDP payload.
rtc::CopyOnWriteBuffer data;
- int ip_header_size;
+ uint16_t headers_size;
Timestamp arrival_time;
};
diff --git a/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc b/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc
index 8cf4d17..3e5403a 100644
--- a/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc
+++ b/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc
@@ -132,9 +132,10 @@
auto ret_net = s.CreateMutableSimulationNode(
[](NetworkSimulationConfig* c) { c->delay = TimeDelta::ms(100); });
auto* client = s.CreateClient("send", config);
- auto routes = s.CreateRoutes(client, {send_net->node()},
+ const DataSize kOverhead = DataSize::bytes(38); // IPV4 + UDP + SRTP
+ auto routes = s.CreateRoutes(client, {send_net->node()}, kOverhead,
s.CreateClient("recv", CallClientConfig()),
- {ret_net->node()});
+ {ret_net->node()}, kOverhead);
s.CreateVideoStream(routes->forward(), VideoStreamConfig());
s.RunFor(TimeDelta::seconds(25));
@@ -156,7 +157,7 @@
[](NetworkSimulationConfig* c) { c->delay = TimeDelta::ms(200); });
s.RunFor(TimeDelta::seconds(35));
- EXPECT_NEAR(client->send_bandwidth().kbps(), 180, 50);
+ EXPECT_NEAR(client->send_bandwidth().kbps(), 170, 50);
}
} // namespace test
diff --git a/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
index 0227027..09aec43 100644
--- a/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
+++ b/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
@@ -410,7 +410,7 @@
// This will cause the RTT to be large for a while.
s.TriggerPacketBurst({send_net}, kBloatPacketCount, kBloatPacketSize.bytes());
// Wait to allow the high RTT to be detected and acted upon.
- s.RunFor(TimeDelta::seconds(4));
+ s.RunFor(TimeDelta::seconds(6));
// By now the target rate should have dropped to the minimum configured rate.
EXPECT_NEAR(client->target_rate().kbps(), kBandwidthFloor.kbps(), 5);
}
diff --git a/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc b/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
index 2383378..9910a03 100644
--- a/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
+++ b/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
@@ -109,7 +109,7 @@
ret_net->UpdateConfig(
[](NetworkSimulationConfig* c) { c->delay = TimeDelta::ms(200); });
s.RunFor(TimeDelta::seconds(35));
- EXPECT_NEAR(client->target_rate().kbps(), 180, 40);
+ EXPECT_NEAR(client->target_rate().kbps(), 170, 50);
}
} // namespace test
diff --git a/test/scenario/call_client.cc b/test/scenario/call_client.cc
index 5823656..61612c5 100644
--- a/test/scenario/call_client.cc
+++ b/test/scenario/call_client.cc
@@ -263,12 +263,6 @@
}
void CallClient::OnPacketReceived(EmulatedIpPacket packet) {
- // Removes added overhead before delivering packet to sender.
- size_t size =
- packet.data.size() - route_overhead_.at(packet.to.ipaddr()).bytes();
- RTC_DCHECK_GE(size, 0);
- packet.data.SetSize(size);
-
MediaType media_type = MediaType::ANY;
if (!RtpHeaderParser::IsRtcp(packet.cdata(), packet.data.size())) {
auto ssrc = RtpHeaderParser::GetSsrc(packet.cdata(), packet.data.size());
diff --git a/test/scenario/call_client.h b/test/scenario/call_client.h
index 34a15c1..a4c04af 100644
--- a/test/scenario/call_client.h
+++ b/test/scenario/call_client.h
@@ -141,9 +141,6 @@
std::unique_ptr<NetworkNodeTransport> transport_;
std::unique_ptr<RtpHeaderParser> const header_parser_;
- // Stores the configured overhead per known destination endpoint. This is used
- // to subtract the overhead before processing.
- std::map<rtc::IPAddress, DataSize> route_overhead_;
int next_video_ssrc_index_ = 0;
int next_video_local_ssrc_index_ = 0;
int next_rtx_ssrc_index_ = 0;
diff --git a/test/scenario/network_node.cc b/test/scenario/network_node.cc
index a33df83..c430b87 100644
--- a/test/scenario/network_node.cc
+++ b/test/scenario/network_node.cc
@@ -89,11 +89,10 @@
rtc::CritScope crit(&crit_sect_);
if (!send_net_)
return false;
- rtc::CopyOnWriteBuffer buffer(packet, length,
- length + packet_overhead_.bytes());
- buffer.SetSize(length + packet_overhead_.bytes());
+ rtc::CopyOnWriteBuffer buffer(packet, length);
send_net_->OnPacketReceived(
- EmulatedIpPacket(local_address_, receiver_address_, buffer, send_time));
+ EmulatedIpPacket(local_address_, receiver_address_, buffer, send_time,
+ packet_overhead_.bytes()));
return true;
}
@@ -101,11 +100,11 @@
rtc::CopyOnWriteBuffer buffer(packet, length);
Timestamp send_time = sender_clock_->CurrentTime();
rtc::CritScope crit(&crit_sect_);
- buffer.SetSize(length + packet_overhead_.bytes());
if (!send_net_)
return false;
send_net_->OnPacketReceived(
- EmulatedIpPacket(local_address_, receiver_address_, buffer, send_time));
+ EmulatedIpPacket(local_address_, receiver_address_, buffer, send_time,
+ packet_overhead_.bytes()));
return true;
}
diff --git a/test/scenario/scenario.cc b/test/scenario/scenario.cc
index ad382bd..9d27e6a 100644
--- a/test/scenario/scenario.cc
+++ b/test/scenario/scenario.cc
@@ -166,7 +166,6 @@
std::vector<EmulatedNetworkNode*> over_nodes,
DataSize overhead) {
rtc::IPAddress route_ip(next_route_id_++);
- clients.second->route_overhead_.insert({route_ip, overhead});
EmulatedNetworkNode::CreateRoute(route_ip, over_nodes, clients.second);
clients.first->transport_->Connect(over_nodes.front(), route_ip, overhead);
}
diff --git a/test/scenario/scenario_config.h b/test/scenario/scenario_config.h
index 7b9c633..282d471 100644
--- a/test/scenario/scenario_config.h
+++ b/test/scenario/scenario_config.h
@@ -28,9 +28,6 @@
namespace webrtc {
namespace test {
struct PacketOverhead {
- static constexpr size_t kIpv4 = 20;
- static constexpr size_t kIpv6 = 40;
- static constexpr size_t kUdp = 8;
static constexpr size_t kSrtp = 10;
static constexpr size_t kStun = 4;
// TURN messages can be sent either with or without an establieshed channel.
@@ -38,7 +35,7 @@
// significantly more overhead.
static constexpr size_t kTurnChannelMessage = 4;
static constexpr size_t kTurnIndicationMessage = 36;
- static constexpr size_t kDefault = kIpv4 + kUdp + kSrtp;
+ static constexpr size_t kDefault = kSrtp;
};
struct TransportControllerConfig {
struct Rates {
diff --git a/test/scenario/stats_collection_unittest.cc b/test/scenario/stats_collection_unittest.cc
index 6ea03c6a..4159eea 100644
--- a/test/scenario/stats_collection_unittest.cc
+++ b/test/scenario/stats_collection_unittest.cc
@@ -78,7 +78,7 @@
EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 75, 50);
EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 100, 50);
EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
- EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 150, 130);
+ EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 200, 150);
}
TEST(ScenarioAnalyzerTest, CountsCapturedButNotRendered) {
