--- a/mozilla-silence-no-return-type.patch Fri Mar 22 10:02:25 2024 +0100
+++ b/mozilla-silence-no-return-type.patch Sun Apr 21 06:46:25 2024 +0200
@@ -1,5 +1,5 @@
# HG changeset patch
-# Parent d1908d68e16e148fcc012caac881a03417eccc7e
+# Parent 831d03cde86aa6b8803d5ac431e2d28bf85c9289
diff --git a/gfx/skia/skia/include/codec/SkEncodedOrigin.h b/gfx/skia/skia/include/codec/SkEncodedOrigin.h
--- a/gfx/skia/skia/include/codec/SkEncodedOrigin.h
@@ -875,6 +875,28 @@
int sample_rate_hz,
int detector_rate_hz,
int num_channels)
+diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
+--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
++++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
+@@ -90,16 +90,18 @@ BandwidthLimitedCause GetBandwidthLimite
+ // Probes may not be sent in this state.
+ return BandwidthLimitedCause::kLossLimitedBwe;
+ case LossBasedState::kIncreasing:
+ // Probes may be sent in this state.
+ return BandwidthLimitedCause::kLossLimitedBweIncreasing;
+ case LossBasedState::kDelayBasedEstimate:
+ return BandwidthLimitedCause::kDelayBasedLimited;
+ }
++ // just return something by default
++ return BandwidthLimitedCause::kLossLimitedBwe;
+ }
+
+ } // namespace
+
+ GoogCcNetworkController::GoogCcNetworkController(NetworkControllerConfig config,
+ GoogCcConfig goog_cc_config)
+ : key_value_config_(config.key_value_config ? config.key_value_config
+ : &trial_based_config_),
diff --git a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc
--- a/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc
+++ b/third_party/libwebrtc/modules/desktop_capture/linux/wayland/screencast_portal.cc
@@ -957,7 +979,7 @@
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
-@@ -41,16 +41,17 @@ namespace {
+@@ -42,16 +42,17 @@ namespace {
case AudioFrameType::kEmptyFrame:
return "empty";
case AudioFrameType::kAudioFrameSpeech:
@@ -1020,7 +1042,7 @@
diff --git a/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc b/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc
--- a/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc
+++ b/third_party/libwebrtc/video/adaptation/video_stream_encoder_resource_manager.cc
-@@ -58,16 +58,17 @@ bool IsFramerateScalingEnabled(Degradati
+@@ -59,16 +59,17 @@ bool IsFramerateScalingEnabled(Degradati
std::string ToString(VideoAdaptationReason reason) {
switch (reason) {
case VideoAdaptationReason::kQuality: