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/*
* Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/stats/rtc_stats_collector_callback.h"
#include "api/stats/rtcstats_objects.h"
#include "api/units/data_rate.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/peer_scenario/peer_scenario.h"
#include "test/peer_scenario/peer_scenario_client.h"
namespace webrtc {
namespace test {
// TODO(terelius): Use fake encoder and enable on Android once
#if defined(WEBRTC_ANDROID)
#define MAYBE_NoBweChangeFromVideoUnmute DISABLED_NoBweChangeFromVideoUnmute
#else
#define MAYBE_NoBweChangeFromVideoUnmute NoBweChangeFromVideoUnmute
#endif
TEST(GoogCcPeerScenarioTest, MAYBE_NoBweChangeFromVideoUnmute) {
// If transport wide sequence numbers are used for audio, and the call
// switches from audio only to video only, there will be a sharp change in
// packets sizes. This will create a change in propagation time which might be
// detected as an overuse. Using separate overuse detectors for audio and
// video avoids the issue.
std::string audio_twcc_trials("WebRTC-Audio-AlrProbing/Disabled/");
std::string separate_audio_video(
"WebRTC-Bwe-SeparateAudioPackets/"
"enabled:true,packet_threshold:15,time_threshold:1000ms/");
ScopedFieldTrials field_trial(audio_twcc_trials + separate_audio_video);
PeerScenario s(*test_info_);
auto* caller = s.CreateClient(PeerScenarioClient::Config());
auto* callee = s.CreateClient(PeerScenarioClient::Config());
BuiltInNetworkBehaviorConfig net_conf;
net_conf.link_capacity = DataRate::KilobitsPerSec(350);
net_conf.queue_delay_ms = 50;
auto send_node = s.net()->CreateEmulatedNode(net_conf);
auto ret_node = s.net()->CreateEmulatedNode(net_conf);
PeerScenarioClient::VideoSendTrackConfig video_conf;
video_conf.generator.squares_video->framerate = 15;
auto video = caller->CreateVideo("VIDEO", video_conf);
auto audio = caller->CreateAudio("AUDIO", cricket::AudioOptions());
// Start ICE and exchange SDP.
s.SimpleConnection(caller, callee, {send_node}, {ret_node});
// Limit the encoder bitrate to ensure that there are no actual BWE overuses.
ASSERT_EQ(caller->pc()->GetSenders().size(), 2u); // 2 senders.
int num_video_streams = 0;
for (auto& rtp_sender : caller->pc()->GetSenders()) {
auto parameters = rtp_sender->GetParameters();
ASSERT_EQ(parameters.encodings.size(), 1u); // 1 stream per sender.
for (auto& encoding_parameters : parameters.encodings) {
if (encoding_parameters.ssrc == video.sender->ssrc()) {
num_video_streams++;
encoding_parameters.max_bitrate_bps = 220000;
encoding_parameters.max_framerate = 15;
}
}
rtp_sender->SetParameters(parameters);
}
ASSERT_EQ(num_video_streams, 1); // Exactly 1 video stream.
auto get_bwe = [&] {
auto callback =
rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
caller->pc()->GetStats(callback.get());
s.net()->time_controller()->Wait([&] { return callback->called(); });
auto stats =
callback->report()->GetStatsOfType<RTCIceCandidatePairStats>()[0];
return DataRate::BitsPerSec(*stats->available_outgoing_bitrate);
};
s.ProcessMessages(TimeDelta::Seconds(15));
const DataRate initial_bwe = get_bwe();
EXPECT_GE(initial_bwe, DataRate::KilobitsPerSec(300));
// 10 seconds audio only. Bandwidth should not drop.
video.capturer->Stop();
s.ProcessMessages(TimeDelta::Seconds(10));
EXPECT_GE(get_bwe(), initial_bwe);
// Resume video but stop audio. Bandwidth should not drop.
video.capturer->Start();
RTCError status = caller->pc()->RemoveTrackOrError(audio.sender);
ASSERT_TRUE(status.ok());
audio.track->set_enabled(false);
for (int i = 0; i < 10; i++) {
s.ProcessMessages(TimeDelta::Seconds(1));
EXPECT_GE(get_bwe(), initial_bwe);
}
caller->pc()->Close();
callee->pc()->Close();
}
} // namespace test
} // namespace webrtc