--- a/mozilla-silence-no-return-type.patch	Tue Jan 23 17:32:46 2024 +0100
+++ b/mozilla-silence-no-return-type.patch	Thu Feb 22 20:31:18 2024 +0100
@@ -1,5 +1,5 @@
 # HG changeset patch
-# Parent  e7eb7e9e99204275532b04de030879c9548b88a3
+# Parent  f5fd2bbd77ef4b6554a7180c9c4768e64aca3b2a
 
 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
@@ -526,7 +526,7 @@
 diff --git a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
 --- a/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
 +++ b/third_party/libwebrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
-@@ -158,16 +158,17 @@ class VideoEncoderSoftwareFallbackWrappe
+@@ -183,16 +183,17 @@ class VideoEncoderSoftwareFallbackWrappe
          [[fallthrough]];
        case EncoderState::kMainEncoderUsed:
          return encoder_.get();
@@ -544,7 +544,7 @@
  
    // Settings used in the last InitEncode call and used if a dynamic fallback to
    // software is required.
-@@ -338,16 +339,17 @@ int32_t VideoEncoderSoftwareFallbackWrap
+@@ -363,16 +364,17 @@ int32_t VideoEncoderSoftwareFallbackWrap
      case EncoderState::kMainEncoderUsed: {
        return EncodeWithMainEncoder(frame, frame_types);
      }
@@ -684,7 +684,7 @@
 diff --git a/third_party/libwebrtc/media/base/codec.cc b/third_party/libwebrtc/media/base/codec.cc
 --- a/third_party/libwebrtc/media/base/codec.cc
 +++ b/third_party/libwebrtc/media/base/codec.cc
-@@ -195,16 +195,17 @@ bool Codec::Matches(const Codec& codec,
+@@ -201,16 +201,17 @@ bool Codec::Matches(const Codec& codec,
                 (codec.bitrate == 0 || bitrate <= 0 ||
                  bitrate == codec.bitrate) &&
                 ((codec.channels < 2 && channels < 2) ||
@@ -699,9 +699,9 @@
    return matches_id && matches_type_specific();
  }
  
- bool Codec::MatchesCapability(
-     const webrtc::RtpCodecCapability& codec_capability) const {
+ bool Codec::MatchesRtpCodec(const webrtc::RtpCodec& codec_capability) const {
    webrtc::RtpCodecParameters codec_parameters = ToCodecParameters();
+ 
 diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc
 --- a/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc
 +++ b/third_party/libwebrtc/modules/audio_processing/agc2/clipping_predictor.cc
@@ -957,7 +957,7 @@
 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
-@@ -40,16 +40,17 @@ namespace {
+@@ -41,16 +41,17 @@ namespace {
      case AudioFrameType::kEmptyFrame:
        return "empty";
      case AudioFrameType::kAudioFrameSpeech: