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In ReceiveSideCongestionController unittest use modern version of OnReceivedPacket

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Bug: webrtc:14859
Change-Id: I8fd93272b0292947c93b5faaec2579a9c5ed3c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296761
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39529}
This commit is contained in:
Danil Chapovalov 2023-03-09 11:02:51 +01:00 committed by WebRTC LUCI CQ
parent 0791e8f8df
commit 1362c8ddcf
2 changed files with 29 additions and 26 deletions
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2
modules/congestion_controller/BUILD.gn
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@ -54,12 +54,14 @@ if (rtc_include_tests && !build_with_chromium) {
"../../api/test/network_emulation",
"../../api/test/network_emulation:create_cross_traffic",
"../../api/units:data_rate",
"../../api/units:data_size",
"../../api/units:time_delta",
"../../api/units:timestamp",
"../../system_wrappers",
"../../test:test_support",
"../../test/scenario",
"../pacing",
"../rtp_rtcp:rtp_rtcp_format",
"goog_cc:estimators",
"goog_cc:goog_cc_unittests",
"pcc:pcc_unittests",

53
modules/congestion_controller/receive_side_congestion_controller_unittest.cc
View File

@ -12,36 +12,32 @@
#include "api/test/network_emulation/create_cross_traffic.h"
#include "api/test/network_emulation/cross_traffic.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/scenario/scenario.h"
namespace webrtc {
namespace test {
namespace {
using ::testing::_;
using ::testing::AtLeast;
using ::testing::ElementsAre;
using ::testing::MockFunction;
namespace webrtc {
namespace {
// Helper to convert some time format to resolution used in absolute send time
// header extension, rounded upwards. `t` is the time to convert, in some
// resolution. `denom` is the value to divide `t` by to get whole seconds,
// e.g. `denom` = 1000 if `t` is in milliseconds.
uint32_t AbsSendTime(int64_t t, int64_t denom) {
return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
}
const uint32_t kInitialBitrateBps = 60000;
} // namespace
namespace test {
constexpr DataRate kInitialBitrate = DataRate::BitsPerSec(60'000);
TEST(ReceiveSideCongestionControllerTest, SendsRembWithAbsSendTime) {
static constexpr DataSize kPayloadSize = DataSize::Bytes(1000);
MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
feedback_sender;
MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
@ -51,18 +47,22 @@ TEST(ReceiveSideCongestionControllerTest, SendsRembWithAbsSendTime) {
&clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
nullptr);
size_t payload_size = 1000;
RTPHeader header;
header.ssrc = 0x11eb21c;
header.extension.hasAbsoluteSendTime = true;
RtpHeaderExtensionMap extensions;
extensions.Register<AbsoluteSendTime>(1);
RtpPacketReceived packet(&extensions);
packet.SetSsrc(0x11eb21c);
packet.ReserveExtension<AbsoluteSendTime>();
packet.SetPayloadSize(kPayloadSize.bytes());
EXPECT_CALL(remb_sender, Call(_, ElementsAre(header.ssrc))).Times(AtLeast(1));
EXPECT_CALL(remb_sender, Call(_, ElementsAre(packet.Ssrc())))
.Times(AtLeast(1));
for (int i = 0; i < 10; ++i) {
clock_.AdvanceTimeMilliseconds((1000 * payload_size) / kInitialBitrateBps);
int64_t now_ms = clock_.TimeInMilliseconds();
header.extension.absoluteSendTime = AbsSendTime(now_ms, 1000);
controller.OnReceivedPacket(now_ms, payload_size, header);
clock_.AdvanceTime(kPayloadSize / kInitialBitrate);
Timestamp now = clock_.CurrentTime();
packet.SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
packet.set_arrival_time(now);
controller.OnReceivedPacket(packet, MediaType::VIDEO);
}
}
@ -122,5 +122,6 @@ TEST(ReceiveSideCongestionControllerTest, IsFairToTCP) {
// fixed and a lower bound should be added to the test.
EXPECT_LT(client->send_bandwidth().kbps(), 750);
}
} // namespace
} // namespace test
} // namespace webrtc