Adding upstream version 115.7.0esr.upstream/115.7.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2 files changed, 133 insertions, 0 deletions

diff --git a/third_party/libwebrtc/pc/scenario_tests/BUILD.gn b/third_party/libwebrtc/pc/scenario_tests/BUILD.gn
new file mode 100644
index 0000000000..fa3a67c9a2
--- /dev/null
+++ b/third_party/libwebrtc/pc/scenario_tests/BUILD.gn
@@ -0,0 +1,24 @@
+# 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.
+
+import("../../webrtc.gni")
+
+if (rtc_include_tests) {
+  rtc_library("pc_scenario_tests") {
+    testonly = true
+    sources = [ "goog_cc_test.cc" ]
+    deps = [
+      "../../api:rtc_stats_api",
+      "../../modules/rtp_rtcp:rtp_rtcp",
+      "../../pc:pc_test_utils",
+      "../../test:field_trial",
+      "../../test:test_support",
+      "../../test/peer_scenario:peer_scenario",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/pc/scenario_tests/goog_cc_test.cc b/third_party/libwebrtc/pc/scenario_tests/goog_cc_test.cc
new file mode 100644
index 0000000000..ea96408ac7
--- /dev/null
+++ b/third_party/libwebrtc/pc/scenario_tests/goog_cc_test.cc
@@ -0,0 +1,109 @@
+/*
+ *  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 "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
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=11408 is fixed.
+#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_kbps = 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