Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 66 insertions, 0 deletions

diff --git a/media/webrtc/signaling/gtest/MockConduit.h b/media/webrtc/signaling/gtest/MockConduit.h
new file mode 100644
index 0000000000..c2c5becd1b
--- /dev/null
+++ b/media/webrtc/signaling/gtest/MockConduit.h
@@ -0,0 +1,66 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MEDIA_WEBRTC_SIGNALING_GTEST_MOCKCONDUIT_H_
+#define MEDIA_WEBRTC_SIGNALING_GTEST_MOCKCONDUIT_H_
+
+#include "gmock/gmock.h"
+#include "MediaConduitInterface.h"
+#include "libwebrtcglue/FrameTransformer.h"
+
+namespace webrtc {
+std::ostream& operator<<(std::ostream& aStream,
+                         const webrtc::Call::Stats& aObj) {
+  aStream << aObj.ToString(0);
+  return aStream;
+}
+}  // namespace webrtc
+
+namespace mozilla {
+class MockConduit : public MediaSessionConduit {
+ public:
+  MockConduit() = default;
+
+  MOCK_CONST_METHOD0(type, Type());
+  MOCK_CONST_METHOD0(ActiveSendPayloadType, Maybe<int>());
+  MOCK_CONST_METHOD0(ActiveRecvPayloadType, Maybe<int>());
+  MOCK_METHOD1(SetTransportActive, void(bool));
+  MOCK_METHOD0(SenderRtpSendEvent, MediaEventSourceExc<MediaPacket>&());
+  MOCK_METHOD0(SenderRtcpSendEvent, MediaEventSourceExc<MediaPacket>&());
+  MOCK_METHOD0(ReceiverRtcpSendEvent, MediaEventSourceExc<MediaPacket>&());
+  MOCK_METHOD1(
+      ConnectReceiverRtpEvent,
+      void(MediaEventSourceExc<webrtc::RtpPacketReceived, webrtc::RTPHeader>&));
+  MOCK_METHOD1(ConnectReceiverRtcpEvent,
+               void(MediaEventSourceExc<MediaPacket>&));
+  MOCK_METHOD1(ConnectSenderRtcpEvent, void(MediaEventSourceExc<MediaPacket>&));
+  MOCK_CONST_METHOD0(LastRtcpReceived, Maybe<DOMHighResTimeStamp>());
+  MOCK_CONST_METHOD1(RtpSendBaseSeqFor, Maybe<uint16_t>(uint32_t));
+  MOCK_CONST_METHOD0(GetNow, DOMHighResTimeStamp());
+  MOCK_CONST_METHOD0(GetTimestampMaker, dom::RTCStatsTimestampMaker&());
+  MOCK_CONST_METHOD0(GetLocalSSRCs, Ssrcs());
+  MOCK_CONST_METHOD0(GetRemoteSSRC, Maybe<Ssrc>());
+  MOCK_METHOD1(UnsetRemoteSSRC, void(Ssrc));
+  MOCK_METHOD0(DisableSsrcChanges, void());
+  MOCK_CONST_METHOD1(HasCodecPluginID, bool(uint64_t));
+  MOCK_METHOD0(RtcpByeEvent, MediaEventSource<void>&());
+  MOCK_METHOD0(RtcpTimeoutEvent, MediaEventSource<void>&());
+  MOCK_METHOD0(RtpPacketEvent, MediaEventSource<void>&());
+  MOCK_METHOD3(SendRtp,
+               bool(const uint8_t*, size_t, const webrtc::PacketOptions&));
+  MOCK_METHOD2(SendSenderRtcp, bool(const uint8_t*, size_t));
+  MOCK_METHOD2(SendReceiverRtcp, bool(const uint8_t*, size_t));
+  MOCK_METHOD2(DeliverPacket, void(rtc::CopyOnWriteBuffer, PacketType));
+  MOCK_METHOD0(Shutdown, RefPtr<GenericPromise>());
+  MOCK_METHOD0(AsAudioSessionConduit, Maybe<RefPtr<AudioSessionConduit>>());
+  MOCK_METHOD0(AsVideoSessionConduit, Maybe<RefPtr<VideoSessionConduit>>());
+  MOCK_CONST_METHOD0(GetCallStats, Maybe<webrtc::Call::Stats>());
+  MOCK_METHOD1(SetJitterBufferTarget, void(DOMHighResTimeStamp));
+  MOCK_CONST_METHOD0(GetUpstreamRtpSources, std::vector<webrtc::RtpSource>());
+};
+}  // namespace mozilla
+
+#endif