zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Use RTX SSRCs in scenario test framework.

Browse Source 
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.

Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
This commit is contained in:
Sebastian Jansson
2020-01-14 12:30:13 +01:00
committed by Commit Bot
parent 2a92d2b461
commit 3e66a498c3
4 changed files with 26 additions and 10 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
View File

@ -412,7 +412,7 @@ TEST_F(GoogCcNetworkControllerTest, LimitsToFloorIfRttIsHighInTrial) {
// Wait to allow the high RTT to be detected and acted upon.
s.RunFor(TimeDelta::seconds(4));
// By now the target rate should have dropped to the minimum configured rate.
EXPECT_NEAR(client->target_rate().kbps(), kBandwidthFloor.kbps(), 1);
EXPECT_NEAR(client->target_rate().kbps(), kBandwidthFloor.kbps(), 5);
}
TEST_F(GoogCcNetworkControllerTest, UpdatesTargetRateBasedOnLinkCapacity) {
@ -502,7 +502,10 @@ DataRate AverageBitrateAfterCrossInducedLoss(std::string name) {
auto* client = s.CreateClient("send", CallClientConfig());
auto* route = s.CreateRoutes(
client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
auto* video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
// TODO(srte): Make this work with RTX enabled or remove it.
auto* video = s.CreateVideoStream(route->forward(), [](VideoStreamConfig* c) {
c->stream.use_rtx = false;
});
s.RunFor(TimeDelta::seconds(10));
for (int i = 0; i < 4; ++i) {
// Sends TCP cross traffic inducing loss.
@ -521,7 +524,7 @@ DataRate AverageBitrateAfterCrossInducedLoss(std::string name) {
TEST_F(GoogCcNetworkControllerTest,
NoLossBasedRecoversSlowerAfterCrossInducedLoss) {
// This test acts as a reference for the test below, showing that wihtout the
// This test acts as a reference for the test below, showing that without the
// trial, we have worse behavior.
DataRate average_bitrate =
AverageBitrateAfterCrossInducedLoss("googcc_unit/no_cross_loss_based");

4
modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
View File

@ -91,7 +91,9 @@ TEST(PccNetworkControllerTest, UpdatesTargetSendRate) {
auto* route = s.CreateRoutes(client, {send_net->node()},
s.CreateClient("return", CallClientConfig()),
{ret_net->node()});
s.CreateVideoStream(route->forward(), VideoStreamConfig());
VideoStreamConfig video;
video.stream.use_rtx = false;
s.CreateVideoStream(route->forward(), video);
s.RunFor(TimeDelta::seconds(30));
EXPECT_NEAR(client->target_rate().kbps(), 450, 100);
send_net->UpdateConfig([](NetworkSimulationConfig* c) {

14
test/scenario/video_stream.cc
View File

@ -120,9 +120,11 @@ std::string TransformFilePath(std::string path) {
return path;
}
VideoSendStream::Config CreateVideoSendStreamConfig(VideoStreamConfig config,
std::vector<uint32_t> ssrcs,
Transport* send_transport) {
VideoSendStream::Config CreateVideoSendStreamConfig(
VideoStreamConfig config,
std::vector<uint32_t> ssrcs,
std::vector<uint32_t> rtx_ssrcs,
Transport* send_transport) {
VideoSendStream::Config send_config(send_transport);
send_config.rtp.payload_name = CodecTypeToPayloadString(config.encoder.codec);
send_config.rtp.payload_type = CodecTypeToPayloadType(config.encoder.codec);
@ -132,6 +134,10 @@ VideoSendStream::Config CreateVideoSendStreamConfig(VideoStreamConfig config,
send_config.rtp.ssrcs = ssrcs;
send_config.rtp.extensions = GetVideoRtpExtensions(config);
if (config.stream.use_rtx) {
send_config.rtp.rtx.payload_type = CallTest::kSendRtxPayloadType;
send_config.rtp.rtx.ssrcs = rtx_ssrcs;
}
if (config.stream.use_flexfec) {
send_config.rtp.flexfec.payload_type = CallTest::kFlexfecPayloadType;
send_config.rtp.flexfec.ssrc = CallTest::kFlexfecSendSsrc;
@ -398,7 +404,7 @@ SendVideoStream::SendVideoStream(CallClient* sender,
rtx_ssrcs_.push_back(sender->GetNextRtxSsrc());
}
VideoSendStream::Config send_config =
CreateVideoSendStreamConfig(config, ssrcs_, send_transport);
CreateVideoSendStreamConfig(config, ssrcs_, rtx_ssrcs_, send_transport);
send_config.encoder_settings.encoder_factory = encoder_factory_.get();
send_config.encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();

9
test/scenario/video_stream_unittest.cc
View File

@ -126,8 +126,11 @@ TEST(VideoStreamTest, SendsNacksOnLoss) {
// NACK retransmissions are enabled by default.
auto video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
s.RunFor(TimeDelta::seconds(1));
auto stream_stats = video->send()->GetStats().substreams.begin()->second;
EXPECT_GT(stream_stats.rtp_stats.retransmitted.packets, 0u);
int retransmit_packets = 0;
for (const auto& substream : video->send()->GetStats().substreams) {
retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
}
EXPECT_GT(retransmit_packets, 0);
}
TEST(VideoStreamTest, SendsFecWithUlpFec) {
@ -136,6 +139,7 @@ TEST(VideoStreamTest, SendsFecWithUlpFec) {
s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
c->loss_rate = 0.1;
c->delay = TimeDelta::ms(100);
})},
s.CreateClient("callee", CallClientConfig()),
{s.CreateSimulationNode(NetworkSimulationConfig())});
@ -154,6 +158,7 @@ TEST(VideoStreamTest, SendsFecWithFlexFec) {
s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
c->loss_rate = 0.1;
c->delay = TimeDelta::ms(100);
})},
s.CreateClient("callee", CallClientConfig()),
{s.CreateSimulationNode(NetworkSimulationConfig())});