Adding upstream version 110.0.1.upstream/110.0.1upstream

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

1 files changed, 53 insertions, 0 deletions

diff --git a/media/webrtc/signaling/gtest/MockCall.cpp b/media/webrtc/signaling/gtest/MockCall.cpp
new file mode 100644
index 0000000000..88439a0d14
--- /dev/null
+++ b/media/webrtc/signaling/gtest/MockCall.cpp
@@ -0,0 +1,53 @@
+/* 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/. */
+
+#include "MockCall.h"
+
+namespace test {
+
+const webrtc::AudioSendStream::Config& MockAudioSendStream::GetConfig() const {
+  return *mCallWrapper->GetMockCall()->mAudioSendConfig;
+}
+
+void MockAudioSendStream::Reconfigure(const Config& config) {
+  mCallWrapper->GetMockCall()->mAudioSendConfig = mozilla::Some(config);
+}
+
+void MockAudioReceiveStream::SetDecoderMap(
+    std::map<int, webrtc::SdpAudioFormat> decoder_map) {
+  MOZ_ASSERT(mCallWrapper->GetMockCall()->mAudioReceiveConfig.isSome());
+  mCallWrapper->GetMockCall()->mAudioReceiveConfig->decoder_map =
+      std::move(decoder_map);
+}
+
+void MockAudioReceiveStream::SetRtpExtensions(
+    std::vector<webrtc::RtpExtension> extensions) {
+  MOZ_ASSERT(mCallWrapper->GetMockCall()->mAudioReceiveConfig.isSome());
+  mCallWrapper->GetMockCall()->mAudioReceiveConfig->rtp.extensions =
+      std::move(extensions);
+}
+
+const std::vector<webrtc::RtpExtension>&
+MockAudioReceiveStream::GetRtpExtensions() const {
+  static std::vector<webrtc::RtpExtension> rtpExtensions;
+  return rtpExtensions;
+}
+
+webrtc::RtpHeaderExtensionMap MockAudioReceiveStream::GetRtpExtensionMap()
+    const {
+  return webrtc::RtpHeaderExtensionMap();
+}
+
+void MockVideoSendStream::ReconfigureVideoEncoder(
+    webrtc::VideoEncoderConfig config) {
+  mCallWrapper->GetMockCall()->mVideoSendEncoderConfig =
+      mozilla::Some(config.Copy());
+}
+
+webrtc::RtpHeaderExtensionMap MockVideoReceiveStream::GetRtpExtensionMap()
+    const {
+  return webrtc::RtpHeaderExtensionMap();
+}
+
+}  // namespace test