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in call RtpVideoSenderTests rely on simulated time

Browse Source 
instead of waiting on an rtc::Event
to make tests faster and potentially less flaky

Bug: None
Change-Id: I04e8fa79761e782f60838b924d40e6d6a104b14b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168644
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30540}
This commit is contained in:
Danil Chapovalov
2020-02-17 16:13:14 +01:00
committed by Commit Bot
parent 93d9ae8a17
commit a68eb8c1cb
1 changed files with 47 additions and 72 deletions
Download Patch File Download Diff File Expand all files Collapse all files

119
call/rtp_video_sender_unittest.cc
View File

@ -21,7 +21,6 @@
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/video_coding/fec_controller_default.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/event.h"
#include "rtc_base/rate_limiter.h"
#include "test/field_trial.h"
#include "test/gmock.h"
@ -123,15 +122,16 @@ class RtpVideoSenderTestFixture {
payload_type)),
send_delay_stats_(time_controller_.GetClock()),
bitrate_config_(GetBitrateConfig()),
transport_controller_(time_controller_.GetClock(),
&event_log_,
nullptr,
nullptr,
bitrate_config_,
ProcessThread::Create("PacerThread"),
time_controller_.GetTaskQueueFactory(),
&field_trials_),
process_thread_(ProcessThread::Create("test_thread")),
transport_controller_(
time_controller_.GetClock(),
&event_log_,
nullptr,
nullptr,
bitrate_config_,
time_controller_.CreateProcessThread("PacerThread"),
time_controller_.GetTaskQueueFactory(),
&field_trials_),
process_thread_(time_controller_.CreateProcessThread("test_thread")),
call_stats_(time_controller_.GetClock(), process_thread_.get()),
stats_proxy_(time_controller_.GetClock(),
config_,
@ -392,8 +392,6 @@ TEST(RtpVideoSenderTest, FrameCountCallbacks) {
// that the packet is removed from RtpPacketHistory and won't be retransmitted
// again.
TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
const int64_t kTimeoutMs = 500;
RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2},
kPayloadType, {});
test.router()->SetActive(true);
@ -406,24 +404,19 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
encoded_image.SetEncodedData(EncodedImageBuffer::Create(&kPayload, 1));
// Send two tiny images, mapping to two RTP packets. Capture sequence numbers.
rtc::Event event;
std::vector<uint16_t> rtp_sequence_numbers;
std::vector<uint16_t> transport_sequence_numbers;
EXPECT_CALL(test.transport(), SendRtp)
.Times(2)
.WillRepeatedly(
[&event, &rtp_sequence_numbers, &transport_sequence_numbers](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
rtp_sequence_numbers.push_back(rtp_packet.SequenceNumber());
transport_sequence_numbers.push_back(options.packet_id);
if (transport_sequence_numbers.size() == 2) {
event.Set();
}
return true;
});
.WillRepeatedly([&rtp_sequence_numbers, &transport_sequence_numbers](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
rtp_sequence_numbers.push_back(rtp_packet.SequenceNumber());
transport_sequence_numbers.push_back(options.packet_id);
return true;
});
EXPECT_EQ(
EncodedImageCallback::Result::OK,
test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error);
@ -435,8 +428,6 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
test.AdvanceTime(TimeDelta::Millis(33));
ASSERT_TRUE(event.Wait(kTimeoutMs));
// Construct a NACK message for requesting retransmission of both packet.
rtcp::Nack nack;
nack.SetMediaSsrc(kSsrc1);
@ -446,7 +437,7 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
std::vector<uint16_t> retransmitted_rtp_sequence_numbers;
EXPECT_CALL(test.transport(), SendRtp)
.Times(2)
.WillRepeatedly([&event, &retransmitted_rtp_sequence_numbers](
.WillRepeatedly([&retransmitted_rtp_sequence_numbers](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
@ -456,14 +447,10 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
rtc::ArrayView<const uint8_t> payload = rtp_packet.payload();
retransmitted_rtp_sequence_numbers.push_back(
ByteReader<uint16_t>::ReadBigEndian(payload.data()));
if (retransmitted_rtp_sequence_numbers.size() == 2) {
event.Set();
}
return true;
});
test.router()->DeliverRtcp(nack_buffer.data(), nack_buffer.size());
test.AdvanceTime(TimeDelta::Millis(33));
ASSERT_TRUE(event.Wait(kTimeoutMs));
// Verify that both packets were retransmitted.
EXPECT_EQ(retransmitted_rtp_sequence_numbers, rtp_sequence_numbers);
@ -490,9 +477,8 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
// still be retransmitted.
test.AdvanceTime(TimeDelta::Millis(33));
EXPECT_CALL(test.transport(), SendRtp)
.WillOnce([&event, &lost_packet_feedback](const uint8_t* packet,
size_t length,
const PacketOptions& options) {
.WillOnce([&lost_packet_feedback](const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1);
@ -500,12 +486,10 @@ TEST(RtpVideoSenderTest, DoesNotRetrasmitAckedPackets) {
rtc::ArrayView<const uint8_t> payload = rtp_packet.payload();
EXPECT_EQ(lost_packet_feedback.rtp_sequence_number,
ByteReader<uint16_t>::ReadBigEndian(payload.data()));
event.Set();
return true;
});
test.router()->DeliverRtcp(nack_buffer.data(), nack_buffer.size());
test.AdvanceTime(TimeDelta::Millis(33));
ASSERT_TRUE(event.Wait(kTimeoutMs));
}
// This tests that we utilize transport wide feedback to retransmit lost
@ -567,8 +551,6 @@ TEST(RtpVideoSenderTest, RetransmitsOnTransportWideLossInfo) {
// Integration test verifying that retransmissions are sent for packets which
// can be detected as lost early, using transport wide feedback.
TEST(RtpVideoSenderTest, EarlyRetransmits) {
const int64_t kTimeoutMs = 500;
RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, {kRtxSsrc1, kRtxSsrc2},
kPayloadType, {});
test.router()->SetActive(true);
@ -587,61 +569,56 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
// Send two tiny images, mapping to single RTP packets. Capture sequence
// numbers.
rtc::Event event;
uint16_t frame1_rtp_sequence_number = 0;
uint16_t frame1_transport_sequence_number = 0;
EXPECT_CALL(test.transport(), SendRtp)
.WillOnce([&event, &frame1_rtp_sequence_number,
&frame1_transport_sequence_number](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
frame1_rtp_sequence_number = rtp_packet.SequenceNumber();
frame1_transport_sequence_number = options.packet_id;
EXPECT_EQ(rtp_packet.Ssrc(), kSsrc1);
event.Set();
return true;
});
.WillOnce(
[&frame1_rtp_sequence_number, &frame1_transport_sequence_number](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
frame1_rtp_sequence_number = rtp_packet.SequenceNumber();
frame1_transport_sequence_number = options.packet_id;
EXPECT_EQ(rtp_packet.Ssrc(), kSsrc1);
return true;
});
EXPECT_EQ(test.router()
->OnEncodedImage(encoded_image, &codec_specific, nullptr)
.error,
EncodedImageCallback::Result::OK);
test.AdvanceTime(TimeDelta::Millis(33));
ASSERT_TRUE(event.Wait(kTimeoutMs));
uint16_t frame2_rtp_sequence_number = 0;
uint16_t frame2_transport_sequence_number = 0;
encoded_image.SetSpatialIndex(1);
EXPECT_CALL(test.transport(), SendRtp)
.WillOnce([&event, &frame2_rtp_sequence_number,
&frame2_transport_sequence_number](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
frame2_rtp_sequence_number = rtp_packet.SequenceNumber();
frame2_transport_sequence_number = options.packet_id;
EXPECT_EQ(rtp_packet.Ssrc(), kSsrc2);
event.Set();
return true;
});
.WillOnce(
[&frame2_rtp_sequence_number, &frame2_transport_sequence_number](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
frame2_rtp_sequence_number = rtp_packet.SequenceNumber();
frame2_transport_sequence_number = options.packet_id;
EXPECT_EQ(rtp_packet.Ssrc(), kSsrc2);
return true;
});
EXPECT_EQ(test.router()
->OnEncodedImage(encoded_image, &codec_specific, nullptr)
.error,
EncodedImageCallback::Result::OK);
test.AdvanceTime(TimeDelta::Millis(33));
ASSERT_TRUE(event.Wait(kTimeoutMs));
EXPECT_NE(frame1_transport_sequence_number, frame2_transport_sequence_number);
// Inject a transport feedback where the packet for the first frame is lost,
// expect a retransmission for it.
EXPECT_CALL(test.transport(), SendRtp)
.WillOnce([&event, &frame1_rtp_sequence_number](
const uint8_t* packet, size_t length,
const PacketOptions& options) {
.WillOnce([&frame1_rtp_sequence_number](const uint8_t* packet,
size_t length,
const PacketOptions& options) {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
EXPECT_EQ(rtp_packet.Ssrc(), kRtxSsrc1);
@ -651,7 +628,6 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
rtc::ArrayView<const uint8_t> payload = rtp_packet.payload();
EXPECT_EQ(ByteReader<uint16_t>::ReadBigEndian(payload.data()),
frame1_rtp_sequence_number);
event.Set();
return true;
});
@ -670,7 +646,6 @@ TEST(RtpVideoSenderTest, EarlyRetransmits) {
// Wait for pacer to run and send the RTX packet.
test.AdvanceTime(TimeDelta::Millis(33));
ASSERT_TRUE(event.Wait(kTimeoutMs));
}
TEST(RtpVideoSenderTest, CanSetZeroBitrateWithOverhead) {