1 files changed, 112 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc
new file mode 100644
index 0000000000..2dfbfdd3c9
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+++ b/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2019 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 "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.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "pc/media_session.h"
+#include "pc/session_description.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/peer_scenario/peer_scenario.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+RtpHeaderExtensionMap AudioExtensions(
+    const SessionDescriptionInterface& session) {
+  auto* audio_desc =
+      cricket::GetFirstAudioContentDescription(session.description());
+  return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
+}
+
+}  // namespace
+
+TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
+  PeerScenario s(*test_info_);
+
+  auto* caller = s.CreateClient(PeerScenarioClient::Config());
+  auto* callee = s.CreateClient(PeerScenarioClient::Config());
+
+  auto send_link = {s.net()->NodeBuilder().Build().node};
+  auto ret_link = {s.net()->NodeBuilder().Build().node};
+
+  s.net()->CreateRoute(caller->endpoint(), send_link, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), ret_link, caller->endpoint());
+
+  auto signaling = s.ConnectSignaling(caller, callee, send_link, ret_link);
+  caller->CreateVideo("VIDEO", PeerScenarioClient::VideoSendTrackConfig());
+  std::atomic<bool> offer_exchange_done(false);
+  signaling.NegotiateSdp(
+      [](SessionDescriptionInterface* offer) {
+        for (auto& cont : offer->description()->contents()) {
+          cont.media_description()->set_remote_estimate(true);
+        }
+      },
+      [&](const SessionDescriptionInterface& answer) {
+        for (auto& cont : answer.description()->contents()) {
+          EXPECT_TRUE(cont.media_description()->remote_estimate());
+        }
+        offer_exchange_done = true;
+      });
+  RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
+}
+
+TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
+  // Defined before PeerScenario so it gets destructed after, to avoid use after free.
+  std::atomic<bool> received_abs_send_time(false);
+  PeerScenario s(*test_info_);
+
+  auto* caller = s.CreateClient(PeerScenarioClient::Config());
+  auto* callee = s.CreateClient(PeerScenarioClient::Config());
+
+  auto send_node = s.net()->NodeBuilder().Build().node;
+  auto ret_node = s.net()->NodeBuilder().Build().node;
+
+  s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
+
+  auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
+  caller->CreateAudio("AUDIO", cricket::AudioOptions());
+  signaling.StartIceSignaling();
+  RtpHeaderExtensionMap extension_map;
+  std::atomic<bool> offer_exchange_done(false);
+  signaling.NegotiateSdp(
+      [&extension_map](SessionDescriptionInterface* offer) {
+        extension_map = AudioExtensions(*offer);
+        EXPECT_TRUE(extension_map.IsRegistered(kRtpExtensionAbsoluteSendTime));
+      },
+      [&](const SessionDescriptionInterface& answer) {
+        EXPECT_TRUE(AudioExtensions(answer).IsRegistered(
+            kRtpExtensionAbsoluteSendTime));
+        offer_exchange_done = true;
+      });
+  RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
+  send_node->router()->SetWatcher(
+      [extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
+        // The dummy packets used by the fake signaling are filled with 0. We
+        // want to ignore those and we can do that on the basis that the first
+        // byte of RTP packets are guaranteed to not be 0.
+        RtpPacket rtp_packet(&extension_map);
+        // TODO(bugs.webrtc.org/14525): Look why there are RTP packets with
+        // payload 72 or 73 (these don't have the RTP AbsoluteSendTime
+        // Extension).
+        if (rtp_packet.Parse(packet.data) && rtp_packet.PayloadType() == 111) {
+          EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
+          received_abs_send_time = true;
+        }
+      });
+  RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
+  caller->pc()->Close();
+  callee->pc()->Close();
+}
+}  // namespace test
+}  // namespace webrtc