2025-09-07 02:00:19 +00:00
9 changed files with 591 additions and 27 deletions
--- a/.forgejo/workflows/build-aarch64.yaml
+++ b/.forgejo/workflows/build-aarch64.yaml
@ -28,7 +28,7 @@ jobs:
          ${{ github.workspace }}/.forgejo/bin/build.sh
          touch packages/dummy
      - name: Package upload
-        uses: forgejo/upload-artifact@v3
+        uses: actions/upload-artifact@v3
        with:
          name: package
          path: packages
@ -49,6 +49,6 @@ jobs:
      - name: Repo pull
        uses: actions/checkout@v4
      - name: Package download
-        uses: forgejo/download-artifact@v3
+        uses: actions/download-artifact@v3
      - name: Package deployment
        run: ${{ github.workspace }}/.forgejo/bin/deploy.sh
--- a/.forgejo/workflows/build-cross.yaml
+++ b/.forgejo/workflows/build-cross.yaml
@ -33,7 +33,7 @@ jobs:
          ${{ github.workspace }}/.forgejo/bin/generate-cross.sh ${{ github.ref_name }} ${{ inputs.target_arch }}
          mv -v /home/buildozer/packages/main ${{ github.workspace }}/packages/cross
      - name: Package upload
-        uses: forgejo/upload-artifact@v3
+        uses: actions/upload-artifact@v3
        with:
          name: package
          path: packages
@ -54,6 +54,6 @@ jobs:
      - name: Repo pull
        uses: actions/checkout@v4
      - name: Package download
-        uses: forgejo/download-artifact@v3
+        uses: actions/download-artifact@v3
      - name: Package deployment
        run: ${{ github.workspace }}/.forgejo/bin/deploy.sh
--- a/.forgejo/workflows/build-x86_64.yaml
+++ b/.forgejo/workflows/build-x86_64.yaml
@ -28,7 +28,7 @@ jobs:
          ${{ github.workspace }}/.forgejo/bin/build.sh
          touch packages/dummy
      - name: Package upload
-        uses: forgejo/upload-artifact@v3
+        uses: actions/upload-artifact@v3
        with:
          name: package
          path: packages
@ -49,6 +49,6 @@ jobs:
      - name: Repo pull
        uses: actions/checkout@v4
      - name: Package download
-        uses: forgejo/download-artifact@v3
+        uses: actions/download-artifact@v3
      - name: Package deployment
        run: ${{ github.workspace }}/.forgejo/bin/deploy.sh
--- a/backports/signal-desktop/APKBUILD
+++ b/backports/signal-desktop/APKBUILD
@ -2,8 +2,8 @@
 # Contributor: Antoine Martin (ayakael) <dev@ayakael.net>
 # Maintainer: Antoine Martin (ayakael) <dev@ayakael.net>
 pkgname=signal-desktop
-pkgver=7.56.1
-pkgrel=1
+pkgver=7.69.0
+pkgrel=0
 pkgdesc="A messaging app for simple private communication with friends"
 url="https://github.com/signalapp/Signal-Desktop/"
 # same as electron
@ -37,7 +37,6 @@ makedepends="
 	glib-dev
 	gn
 	jsoncpp-dev
-	libavif-dev
 	libjpeg-turbo-dev
 	libepoxy-dev
 	libsecret-dev
@ -67,10 +66,10 @@ makedepends="
 options="net !check"

 # use _check_depends to validate this
-_libsignalver=0.72.1
-_ringrtcver=2.52.0
-_webrtcver=6998c
-_sqlcipherver=2.0.3
+_libsignalver=0.78.3
+_ringrtcver=2.57.0
+_webrtcver=7204c
+_sqlcipherver=2.4.4

 source="
 	https://github.com/signalapp/Signal-Desktop/archive/refs/tags/v$pkgver/Signal-Desktop-$pkgver.tar.gz
@ -87,12 +86,14 @@ source="
 	signal-rollback-locale-changes.patch
 	signal-do-not-package-sqlcipher-deps.patch
 	ringrtc-webrtc-renamed.patch
+	ringrtc-use-sh.patch
 	webrtc-shared-libs.patch
 	webrtc-compiler.patch
 	webrtc-gcc13.patch
-	webrtc-rollback-red.patch
 	webrtc-rollback-3rdparty-build-gn.patch
-	webrtc-pipewire-1.4.patch
+	webrtc-add-cstdint-include-patch-event.patch
+	webrtc-use-only-payloadtypesuggester-for-pt-assignement.patch
+	webrtc-fix-declaration-that-change-meaning-inside-class.patch

 	signal-desktop.sh
 	"
@ -112,6 +113,10 @@ export CFLAGS="${CFLAGS/-g/} -O2 -Wno-error -Wno-deprecated-builtins -Wno-unknow
 export CXXFLAGS="${CXXFLAGS/-g/} -O2 -Wno-error -Wno-deprecated-builtins -Wno-unknown-warning-option -Wno-builtin-macro-redefined -Wno-deprecated-declarations"
 export CPPFLAGS="$CPPFLAGS -D__DATE__=  -D__TIME__=  -D__TIMESTAMP__="

+# stdatomic from compiler-rt uses implicit function declaration
+export CFLAGS="$CFLAGS -Wno-implicit-function-declaration"
+export CXXFLAGS="$CXXFLAGS -Wno-implicit-function-declaration"
+
 export CARGO_PROFILE_RELEASE_OPT_LEVEL=2
 export CARGO_PROFILE_RELEASE_STRIP="symbols"
 #export RUSTFLAGS="$RUSTFLAGS -C linker=clang"
@ -136,7 +141,7 @@ _update_depends() {
 		-e "s|^_ringrtcver=.*|_ringrtcver=$_ringrtcver|" \
 		-e "s|^_webrtcver=.*|_webrtcver=$_webrtcver|" \
 		-e "s|^_sqlcipherver=.*|_sqlcipherver=$_sqlcipherver|" \
-		APKBUILD
+		$APKBUILD
 }

 # webrtc only, the other dependencies are fine with tarballs
@ -223,7 +228,6 @@ prepare() {
 			icu
 			jsoncpp
 			libaom
-			libavif
 			libdrm
 			libjpeg
 			libpng
@ -329,6 +333,9 @@ build() {
 			rtc_include_ilbc=false
 			rtc_libvpx_build_vp9=true
 			rtc_use_x11=false
+			rustc_version=\"yes\"
+			rust_bindgen_root=\"/usr\"
+			rust_sysroot_absolute=\"/usr\"

 			build_with_mozilla=false
 			chrome_pgo_phase=0
@ -478,24 +485,26 @@ package() {
 }

 sha512sums="
-cb5a3471ba6269bb36b95183152ec8e74e77cf605970fd1310b2435ee87ef024eb7a51750a5993abd10c036198e30ccdfdc5b6195c0381035bf7b1e3735e0e34  Signal-Desktop-7.56.1.tar.gz
-e7e08e2909080f34a6c9e65065cfd107e922415727dcdaf9b61aa0e9fb91c7906326c233b3bb7e26eb665e6b4a1b771e2af13e58d8f6ea0f79c39306c9a071cf  libsignal-0.72.1.tar.gz
-37bf7c4d847117c4fcdf371ac84193b005e5f1ff37eec89e2a54ca8968e59945104587b3742b3de116aa0f6f825f3695baeb49b4b96f0268a4c9da0e324b5678  ringrtc-2.52.0.tar.gz
-ec503ba505ea84673773ce761f2108272a890c14fb33d194828045d8170023e8b450f15c3888af89e236cb59a812138216cd0dbd68a7ec62773318b71705c046  node-sqlcipher-2.0.3.tar.gz
-b43d6ba8d8a64de0715efdcf3c17f174eec359d3f4af123908d18d94c15d5d87339184aa4acd640b6a8c28b1a8ed4ca4e1a36e16bd66aa7914dc47d88be89002  webrtc-6998c.tar.zst
+d322c7d0e0eb947bf579d3b4a0c2de4850ea729ebd3871d3aee7d99a10cf1c00abe3c72b99a12b3afb7265dc9f5c0cda9de6c242eb490d6e4e0e9a7b6361ad81  Signal-Desktop-7.69.0.tar.gz
+fc2170bb029628efed5dbae795471026387e501bc2ca7075a5e210df6d81103608361c8163759c7d73f9e82b61e2a5e46d8bf32f40a0c5a19e5fc556a2ff1413  libsignal-0.78.3.tar.gz
+205da87c64a000cc439b2c7bfddb923eac5ab70a0611dab9858eb2e0c0a71227722e6d13eaba2a458807cf27ff93641b8f549580f5e1fc3781ab3c7fb29d9edb  ringrtc-2.57.0.tar.gz
+79dd21548329b4ee409fa73b013c18ed4e85038aeef7a4cfe196fd6b5113279da8ad287a259fa6a6b546b52a5a6dd6ae2cb1050007043d25f76d7917dbbbc02d  node-sqlcipher-2.4.4.tar.gz
+5dd61ebe525c592020a7246337dd837a8cd5de45b201afdc5b75cf3b25c5afcf6bbbae2bb677677807157697fdfb0be51681c0a8e68839e4f1eee30d3dbcad48  webrtc-7204c.tar.zst
 8d2d2d82c8546c2dd1fef161b61df79918f8c22235a56a46adb375a0beb4acef12c5fe53e67242a4be97f77adc522ff79b47949c352956c742a70d50f4179f7f  libsignal-auditable.patch
 7ee747c8aaa9d6e9149fe0a031e3d4cc9e9c08f22c42076bd05e2acb86952f8170032613d13a53716010edfa3f26ad97b530e82460318eb46a42e28cf5faeb9d  signal-use-system-sqlcipher.patch
-fc2af28645364cd85c106304a26811b036f6be048c52137fbbcc112870a84132a181f532563e0569b560fde3cf8c02480666a9b01d804b0008f97c728005b1c7  signal-disable-updates.patch
+ef3622da416a5bd2d4bea4f2a4fbb21a985f660d4acc17bbe66ce51ac1180ab92c0c843a5414ff56ea1deda87c2b0f611a299ca8ebe4d6a24df53626b36ceea8  signal-disable-updates.patch
 853de84d636f730694f17bcec63463fa7bfbdd0a7f7a64618a8fc6bc523ce1a9854b4c651753735af735c18101295d3efbe54f83ae0bccbed83c8c1fee3e7049  signal-update-links.patch
 882d6889b23a3ebc6449c8b6acec8c3853674a7e94f066d65b57bab674ba8c11d582ba2c760825cb67b9202716e6d8b7123001d1e9f9229e49a0b77e9d978265  signal-show-window-please.patch
 aca92e4de6cd005d660d7e8b99607de96ca2d7dc220e3465247d517e915e4d41a4b0eb519e85c5eca1c81cd8bac9821acd03ff57f1603918987829c6ea7757b4  signal-rollback-locale-changes.patch
 961568777b86f8fbcc73360252123686c9d1e16b2650f23d8afbc6d7580d53024f81b62e9e9cbdcd0031b5cf99854bd47c6dd4580197f2b27b8b4cbb51c6c9c9  signal-do-not-package-sqlcipher-deps.patch
 02a648bb8541a39c75fec2db39f024a27976afca41908eefa2fe1f2e30b05b5d59f980ccc930021c0b0c09ba8b0cbcae071e9f0dd530543c667adbc4272af552  ringrtc-webrtc-renamed.patch
+19d2e07bdc0b160ec542fcb0a3d94ae1e37dcb1b3455e57b278cf074f8aac625341b47f4f06a1f7eb5a197cb0f11754de8785ffd10876852972cbfafdc2615db  ringrtc-use-sh.patch
 98a7e4df27c3ed56c5968ebc3ae8e08bfb54362d0c910f9af1d414079a6d5fe0f18aa570ae3c38042f4691af6a130cb3ff689625e26a3987720dd319a5f587b1  webrtc-shared-libs.patch
 897174fa3b1eaa74a3b2d7d861863a9f60550ddb0d7be700cd590600efdc3bd962f52c21e371276eaf8d5657c98fe8ade9853b0197b61a68d5dce6fa03b5899b  webrtc-compiler.patch
 3f1095861a79862ab5a55c86d7e353d272974617afba39e4d4441024bede8d7b9b8e122da90d56a9c3c0965171803d6b9e29e91d78167a08c6bcc86b3eedab18  webrtc-gcc13.patch
-8f53dcf3264c5cec338f037e57a7770ec939ee06dfb0c5dfeab23967a3a64d3f861a6b5419766ac892676036b4522bce2f47279ca3658b5c0b6a31c82ac05053  webrtc-rollback-red.patch
 f8bd574a0de077a643ced26e8a0e3f162e014bbf91c957bbefd113883a70e2b63e483bf400b7e2da8d09edfe76d4f6a257f194a14124f344f1625c5632d12acb  webrtc-rollback-3rdparty-build-gn.patch
-7fa0344a144d5b8f05d0962799b8ce1dbe8f2d403a311d4fbf802a2d032e6586a1268387052b529e809c4740a79d6737a63d7e37ea8902ce926b74309dd44fc3  webrtc-pipewire-1.4.patch
+2a83171eae579a756edbf95c3ded80cbe0cd0fdb4f19b4c73612837f7ea6f573125dde013a17c8e7ce08f101568bb73bfb86a945dc2d10dd91ad45632c69a132  webrtc-add-cstdint-include-patch-event.patch
+1520ca0993cd9df44401de001d1f49da924b8c9bd818c8f7fb47863af710cc2796306e786ad8522c9729dec6a1191d1f2d9e0e2ba2b678b9b8ea534b7bddc762  webrtc-use-only-payloadtypesuggester-for-pt-assignement.patch
+ace87a2be03c6965012a87a10918c5923b10d5e74a7169c1e1384c15b34a367d0c47e806764c368cc30dad01a3dce54b50587abea9831130178db3806619abc8  webrtc-fix-declaration-that-change-meaning-inside-class.patch
 87534e7b5ad7365509eab75629e6bd1a9ed61ee92f7e358405a0abaf0df57de14623fb3894eb082f8785422e5c087e1c50f9e2e5cafbb2529591fd7bf447f7f5  signal-desktop.sh
 "
--- a/backports/signal-desktop/ringrtc-use-sh.patch
+++ b/backports/signal-desktop/ringrtc-use-sh.patch
@ -0,0 +1,14 @@
+diff --git a/ringrtc/src/node/package.json.orig b/ringrtc/src/node/package.json
+index 366caf2927f..eedeca3895e 100644
+--- a/ringrtc/src/node/package.json.orig
+++ b/ringrtc/src/node/package.json
+@@ -16,7 +16,7 @@
+     "virtual_audio": "dist/bin/virtual_audio.sh"
+   },
+   "scripts": {
+-    "build": "tsc && bash scripts/build-help.sh",
+    "build": "tsc && sh scripts/build-help.sh",
+     "clean": "rimraf dist",
+     "test": "electron-mocha --renderer --recursive dist/test --timeout 10000 --require source-map-support/register",
+     "eslint": "eslint .",
+
--- a/backports/signal-desktop/signal-disable-updates.patch
+++ b/backports/signal-desktop/signal-disable-updates.patch
@ -1,9 +1,11 @@
--- a/Signal-Desktop/config/production.json
+diff --git a/Signal-Desktop/config/production.json.orig b/Signal-Desktop/config/production.json
+index ee1fc6523d8..6fcb97f3a36 100644
+--- a/Signal-Desktop/config/production.json.orig
 +++ b/Signal-Desktop/config/production.json
@@ -16,5 +16,5 @@
-   "serverTrustRoot": "BXu6QIKVz5MA8gstzfOgRQGqyLqOwNKHL6INkv3IHWMF",
   "genericServerPublicParams": "AByD873dTilmOSG0TjKrvpeaKEsUmIO8Vx9BeMmftwUs9v7ikPwM8P3OHyT0+X3EUMZrSe9VUp26Wai51Q9I8mdk0hX/yo7CeFGJyzoOqn8e/i4Ygbn5HoAyXJx5eXfIbqpc0bIxzju4H/HOQeOpt6h742qii5u/cbwOhFZCsMIbElZTaeU+BWMBQiZHIGHT5IE0qCordQKZ5iPZom0HeFa8Yq0ShuEyAl0WINBiY6xE3H/9WnvzXBbMuuk//eRxXgzO8ieCeK8FwQNxbfXqZm6Ro1cMhCOF3u7xoX83QhpN",
   "backupServerPublicParams": "AJwNSU55fsFCbgaxGRD11wO1juAs8Yr5GF8FPlGzzvdJJIKH5/4CC7ZJSOe3yL2vturVaRU2Cx0n751Vt8wkj1bozK3CBV1UokxV09GWf+hdVImLGjXGYLLhnI1J2TWEe7iWHyb553EEnRb5oxr9n3lUbNAJuRmFM7hrr0Al0F0wrDD4S8lo2mGaXe0MJCOM166F8oYRQqpFeEHfiLnxA1O8ZLh7vMdv4g9jI5phpRBTsJ5IjiJrWeP0zdIGHEssUeprDZ9OUJ14m0v61eYJMKsf59Bn+mAT2a7YfB+Don9O",
+   "stripePublishableKey": "pk_live_6cmGZopuTsV8novGgJJW9JpC00vLIgtQ1D",
 -  "updatesEnabled": true
 +  "updatesEnabled": false
 }
--- a/backports/signal-desktop/webrtc-add-cstdint-include-patch-event.patch
+++ b/backports/signal-desktop/webrtc-add-cstdint-include-patch-event.patch
@ -0,0 +1,12 @@
+diff --git a/webrtc/rtc_base/trace_event.h b/webrtc/rtc_base/trace_event.h
+index 3329abd..9ec7088 100644
+--- a/webrtc/rtc_base/trace_event.h
+++ b/webrtc/rtc_base/trace_event.h
+@@ -11,6 +11,7 @@
+ #ifndef RTC_BASE_TRACE_EVENT_H_
+ #define RTC_BASE_TRACE_EVENT_H_
+ 
+#include <cstdint>
+ #if defined(RTC_DISABLE_TRACE_EVENTS)
+ #define RTC_TRACE_EVENTS_ENABLED 0
+ #else
--- a/backports/signal-desktop/webrtc-fix-declaration-that-change-meaning-inside-class.patch
+++ b/backports/signal-desktop/webrtc-fix-declaration-that-change-meaning-inside-class.patch
@ -0,0 +1,58 @@
+diff --git a/webrtc/p2p/base/port_interface.h b/webrtc/p2p/base/port_interface.h
+index 2335fc2..1484a14 100644
+--- a/webrtc/p2p/base/port_interface.h
+++ b/webrtc/p2p/base/port_interface.h
+@@ -55,7 +55,7 @@
+   virtual ~PortInterface();
+ 
+   virtual IceCandidateType Type() const = 0;
+-  virtual const Network* Network() const = 0;
+  virtual const ::webrtc::Network* Network() const = 0;
+ 
+   // Methods to set/get ICE role and tiebreaker values.
+   virtual void SetIceRole(IceRole role) = 0;
+diff --git a/webrtc/pc/codec_vendor.h b/webrtc/pc/codec_vendor.h
+index aaddc4f..719a082 100644
+--- a/webrtc/pc/codec_vendor.h
+++ b/webrtc/pc/codec_vendor.h
+@@ -119,7 +119,7 @@
+ class CodecLookupHelper {
+  public:
+   virtual ~CodecLookupHelper() = default;
+-  virtual PayloadTypeSuggester* PayloadTypeSuggester() = 0;
+  virtual ::webrtc::PayloadTypeSuggester* PayloadTypeSuggester() = 0;
+   // Look up the codec vendor to use, depending on context.
+   // This call may get additional arguments in the future, to aid
+   // in selection of the correct context.
+diff --git a/webrtc/pc/jsep_transport.h b/webrtc/pc/jsep_transport.h
+index 017aad7..50dc991 100644
+--- a/webrtc/pc/jsep_transport.h
+++ b/webrtc/pc/jsep_transport.h
+@@ -115,14 +115,12 @@
+   }
+ 
+   RTCError SetLocalJsepTransportDescription(
+-      const JsepTransportDescription& jsep_description,
+-      SdpType type);
+      const JsepTransportDescription& jsep_description, SdpType type);
+ 
+   // Set the remote TransportDescription to be used by DTLS and ICE channels
+   // that are part of this Transport.
+   RTCError SetRemoteJsepTransportDescription(
+-      const JsepTransportDescription& jsep_description,
+-      SdpType type);
+      const JsepTransportDescription& jsep_description, SdpType type);
+   RTCError AddRemoteCandidates(const Candidates& candidates);
+ 
+   // Set the "needs-ice-restart" flag as described in JSEP. After the flag is
+@@ -204,7 +202,9 @@
+     return rtp_dtls_transport_;
+   }
+ 
+-  scoped_refptr<SctpTransport> SctpTransport() const { return sctp_transport_; }
+  scoped_refptr<::webrtc::SctpTransport> SctpTransport() const {
+    return sctp_transport_;
+  }
+ 
+   // TODO(bugs.webrtc.org/9719): Delete method, update callers to use
+   // SctpTransport() instead.
--- a/backports/signal-desktop/webrtc-use-only-payloadtypesuggester-for-pt-assignement.patch
+++ b/backports/signal-desktop/webrtc-use-only-payloadtypesuggester-for-pt-assignement.patch
@ -0,0 +1,469 @@
+diff --git a/webrtc/pc/codec_vendor.cc b/webrtc/pc/codec_vendor.cc
+index 42d52f6..e45516c 100644
+--- a/webrtc/pc/codec_vendor.cc
+++ b/webrtc/pc/codec_vendor.cc
+@@ -39,7 +39,6 @@
+ #include "pc/rtp_media_utils.h"
+ #include "pc/session_description.h"
+ #include "pc/typed_codec_vendor.h"
+-#include "pc/used_ids.h"
+ #include "rtc_base/checks.h"
+ #include "rtc_base/logging.h"
+ #include "rtc_base/string_encode.h"
+@@ -54,11 +53,6 @@
+ 
+ namespace {
+ 
+-using webrtc::PayloadTypeSuggester;
+-using webrtc::RTCError;
+-using webrtc::RTCErrorOr;
+-using webrtc::RtpTransceiverDirection;
+-
+ bool IsRtxCodec(const RtpCodecCapability& capability) {
+   return absl::EqualsIgnoreCase(capability.name, kRtxCodecName);
+ }
+@@ -75,8 +69,7 @@
+ std::optional<Codec> FindMatchingCodec(const CodecList& codecs1,
+                                        const CodecList& codecs2,
+                                        const Codec& codec_to_match) {
+-  return webrtc::FindMatchingCodec(codecs1.codecs(), codecs2.codecs(),
+-                                   codec_to_match);
+  return FindMatchingCodec(codecs1.codecs(), codecs2.codecs(), codec_to_match);
+ }
+ 
+ void StripCNCodecs(CodecList& audio_codecs) {
+@@ -105,7 +98,7 @@
+   }
+ 
+   int associated_pt;
+-  if (!webrtc::FromString(associated_pt_str, &associated_pt)) {
+  if (!FromString(associated_pt_str, &associated_pt)) {
+     RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
+                         << " of RTX codec " << rtx_codec.id
+                         << " to an integer.";
+@@ -136,14 +129,14 @@
+     return nullptr;
+   }
+ 
+-  std::vector<absl::string_view> redundant_payloads = webrtc::split(fmtp, '/');
+  std::vector<absl::string_view> redundant_payloads = split(fmtp, '/');
+   if (redundant_payloads.size() < 2) {
+     return nullptr;
+   }
+ 
+   absl::string_view associated_pt_str = redundant_payloads[0];
+   int associated_pt;
+-  if (!webrtc::FromString(associated_pt_str, &associated_pt)) {
+  if (!FromString(associated_pt_str, &associated_pt)) {
+     RTC_LOG(LS_WARNING) << "Couldn't convert first payload type "
+                         << associated_pt_str << " of RED codec " << red_codec.id
+                         << " to an integer.";
+@@ -182,7 +175,21 @@
+         return suggestion.MoveError();
+       }
+       codec.id = suggestion.value();
+-      offered_codecs.push_back(codec);
+      // The rewrite of the parameter may have turned the codec into
+      // one that is already present.
+      bool skip = false;
+      for (const Codec& present_codec : offered_codecs) {
+        if (present_codec.id == codec.id) {
+          RTC_DCHECK(MatchesWithReferenceAttributes(present_codec, codec))
+              << "Codec mismatch: present " << present_codec << ", new "
+              << codec;
+          skip = true;
+          break;
+        }
+      }
+      if (!skip) {
+        offered_codecs.push_back(codec);
+      }
+     }
+   }
+ 
+@@ -232,7 +239,7 @@
+         }
+         std::string red_param = absl::StrCat(matching_codec->id);
+         red_codec.params[kCodecParamNotInNameValueFormat] =
+-            webrtc::StrJoin(std::vector{red_param, red_param}, "/");
+            StrJoin(std::vector{red_param, red_param}, "/");
+       }
+       RTCErrorOr<PayloadType> suggestion =
+           pt_suggester.SuggestPayloadType(mid, red_codec);
+@@ -240,84 +247,25 @@
+         return suggestion.MoveError();
+       }
+       red_codec.id = suggestion.value();
+-      offered_codecs.push_back(red_codec);
+      // The rewrite of the parameter may have turned the RED codec into
+      // one that is already present.
+      bool skip = false;
+      for (const Codec& present_codec : offered_codecs) {
+        if (present_codec.id == red_codec.id) {
+          RTC_DCHECK(MatchesWithReferenceAttributes(present_codec, red_codec));
+          skip = true;
+          break;
+        }
+      }
+      if (!skip) {
+        offered_codecs.push_back(red_codec);
+      }
+     }
+   }
+   offered_codecs.CheckConsistency();
+   return RTCError::OK();
+ }
+ 
+-// Adds all codecs from `reference_codecs` to `offered_codecs` that don't
+-// already exist in `offered_codecs` and ensure the payload types don't
+-// collide.
+-// OLD VERSION - uses UsedPayloadTypes
+-void MergeCodecs(const CodecList& reference_codecs,
+-                 CodecList& offered_codecs,
+-                 UsedPayloadTypes* used_pltypes) {
+-  // Add all new codecs that are not RTX/RED codecs.
+-  // The two-pass splitting of the loops means preferring payload types
+-  // of actual codecs with respect to collisions.
+-  for (const Codec& reference_codec : reference_codecs) {
+-    if (reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRtx &&
+-        reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRed &&
+-        !FindMatchingCodec(reference_codecs, offered_codecs, reference_codec)) {
+-      Codec codec = reference_codec;
+-      used_pltypes->FindAndSetIdUsed(&codec);
+-      offered_codecs.push_back(codec);
+-    }
+-  }
+-
+-  // Add all new RTX or RED codecs.
+-  for (const Codec& reference_codec : reference_codecs) {
+-    if (reference_codec.GetResiliencyType() == Codec::ResiliencyType::kRtx &&
+-        !FindMatchingCodec(reference_codecs, offered_codecs, reference_codec)) {
+-      Codec rtx_codec = reference_codec;
+-      const Codec* associated_codec =
+-          GetAssociatedCodecForRtx(reference_codecs, rtx_codec);
+-      if (!associated_codec) {
+-        continue;
+-      }
+-      // Find a codec in the offered list that matches the reference codec.
+-      // Its payload type may be different than the reference codec.
+-      std::optional<Codec> matching_codec = FindMatchingCodec(
+-          reference_codecs, offered_codecs, *associated_codec);
+-      if (!matching_codec) {
+-        RTC_LOG(LS_WARNING)
+-            << "Couldn't find matching " << associated_codec->name << " codec.";
+-        continue;
+-      }
+-
+-      rtx_codec.params[kCodecParamAssociatedPayloadType] =
+-          absl::StrCat(matching_codec->id);
+-      used_pltypes->FindAndSetIdUsed(&rtx_codec);
+-      offered_codecs.push_back(rtx_codec);
+-    } else if (reference_codec.GetResiliencyType() ==
+-                   Codec::ResiliencyType::kRed &&
+-               !FindMatchingCodec(reference_codecs, offered_codecs,
+-                                  reference_codec)) {
+-      Codec red_codec = reference_codec;
+-      const Codec* associated_codec =
+-          GetAssociatedCodecForRed(reference_codecs, red_codec);
+-      if (associated_codec) {
+-        std::optional<Codec> matching_codec = FindMatchingCodec(
+-            reference_codecs, offered_codecs, *associated_codec);
+-        if (!matching_codec) {
+-          RTC_LOG(LS_WARNING) << "Couldn't find matching "
+-                              << associated_codec->name << " codec.";
+-          continue;
+-        }
+-
+-        red_codec.params[kCodecParamNotInNameValueFormat] =
+-            absl::StrCat(matching_codec->id) + "/" +
+-            absl::StrCat(matching_codec->id);
+-      }
+-      used_pltypes->FindAndSetIdUsed(&red_codec);
+-      offered_codecs.push_back(red_codec);
+-    }
+-  }
+-  offered_codecs.CheckConsistency();
+-}
+-
+ // `codecs` is a full list of codecs with correct payload type mappings, which
+ // don't conflict with mappings of the other media type; `supported_codecs` is
+ // a list filtered for the media section`s direction but with default payload
+@@ -382,7 +330,7 @@
+                   codec.params.find(kCodecParamNotInNameValueFormat);
+               if (fmtp != codec.params.end()) {
+                 std::vector<absl::string_view> redundant_payloads =
+-                    webrtc::split(fmtp->second, '/');
+                    split(fmtp->second, '/');
+                 if (!redundant_payloads.empty() &&
+                     redundant_payloads[0] == id) {
+                   if (!red_was_added) {
+@@ -443,7 +391,7 @@
+     for (const Codec& supported_codec : supported_codecs) {
+       if (absl::EqualsIgnoreCase(supported_codec.name, kH265CodecName)) {
+         std::optional<H265ProfileTierLevel> supported_ptl =
+-            webrtc::ParseSdpForH265ProfileTierLevel(supported_codec.params);
+            ParseSdpForH265ProfileTierLevel(supported_codec.params);
+         if (supported_ptl.has_value()) {
+           supported_h265_profiles[supported_ptl->profile] =
+               supported_ptl->level;
+@@ -458,14 +406,14 @@
+     for (auto& filtered_codec : filtered_codecs) {
+       if (absl::EqualsIgnoreCase(filtered_codec.name, kH265CodecName)) {
+         std::optional<H265ProfileTierLevel> filtered_ptl =
+-            webrtc::ParseSdpForH265ProfileTierLevel(filtered_codec.params);
+            ParseSdpForH265ProfileTierLevel(filtered_codec.params);
+         if (filtered_ptl.has_value()) {
+           auto it = supported_h265_profiles.find(filtered_ptl->profile);
+ 
+           if (it != supported_h265_profiles.end() &&
+               filtered_ptl->level != it->second) {
+             filtered_codec.params[kH265FmtpLevelId] =
+-                webrtc::H265LevelToString(it->second);
+                H265LevelToString(it->second);
+           }
+         }
+       }
+@@ -506,13 +454,13 @@
+         }
+       }
+       if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
+-        webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params,
+-                                                    &negotiated.params);
+        H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params,
+                                            &negotiated.params);
+       }
+ #ifdef RTC_ENABLE_H265
+       if (absl::EqualsIgnoreCase(ours.name, kH265CodecName)) {
+-        webrtc::H265GenerateProfileTierLevelForAnswer(
+-            ours.params, theirs->params, &negotiated.params);
+        H265GenerateProfileTierLevelForAnswer(ours.params, theirs->params,
+                                              &negotiated.params);
+         NegotiateTxMode(ours, *theirs, &negotiated);
+       }
+ #endif
+@@ -534,7 +482,7 @@
+         continue;
+       }
+       int apt_value;
+-      if (!webrtc::FromString(apt_str, &apt_value)) {
+      if (!FromString(apt_str, &apt_value)) {
+         RTC_LOG(LS_WARNING) << "Unconvertable apt value";
+         continue;
+       }
+@@ -634,9 +582,11 @@
+   }
+   // Add our codecs that are not in the current description.
+   if (media_description_options.type == MediaType::AUDIO) {
+-    MergeCodecs(all_audio_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(audio_recv_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(audio_send_codecs(), mid, codecs, pt_suggester);
+   } else {
+-    MergeCodecs(all_video_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(video_recv_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(video_send_codecs(), mid, codecs, pt_suggester);
+   }
+   CodecList filtered_codecs;
+   CodecList supported_codecs =
+@@ -667,20 +617,11 @@
+         const MediaContentDescription* mcd =
+             current_content->media_description();
+         for (const Codec& codec : mcd->codecs()) {
+-          if (webrtc::FindMatchingCodec(mcd->codecs(), codecs.codecs(),
+-                                        codec)) {
+          if (FindMatchingCodec(mcd->codecs(), codecs.codecs(), codec)) {
+             filtered_codecs.push_back(codec);
+           }
+         }
+       }
+-      // Note what PTs are already in use.
+-      UsedPayloadTypes
+-          used_pltypes;  // Used to avoid pt collisions in filtered_codecs
+-      for (auto& codec : filtered_codecs) {
+-        // Note: This may change PTs. Doing so woud indicate an error, but
+-        // UsedPayloadTypes doesn't offer a means to make the distinction.
+-        used_pltypes.FindAndSetIdUsed(&codec);
+-      }
+       // Add other supported codecs.
+       for (const Codec& codec : supported_codecs) {
+         std::optional<Codec> found_codec =
+@@ -707,8 +648,11 @@
+                                     changed_referenced_codec->id);
+             }
+           }
+-          // Quick fix for b/395077842: Remap the codec if it collides.
+-          used_pltypes.FindAndSetIdUsed(&(*found_codec));
+          auto pt_or_error = pt_suggester.SuggestPayloadType(mid, *found_codec);
+          if (!pt_or_error.ok()) {
+            return pt_or_error.MoveError();
+          }
+          found_codec->id = pt_or_error.value();
+           filtered_codecs.push_back(*found_codec);
+         }
+       }
+@@ -763,9 +707,11 @@
+   }
+   // Add all our supported codecs
+   if (media_description_options.type == MediaType::AUDIO) {
+-    MergeCodecs(all_audio_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(audio_recv_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(audio_send_codecs(), mid, codecs, pt_suggester);
+   } else {
+-    MergeCodecs(all_video_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(video_recv_codecs(), mid, codecs, pt_suggester);
+    MergeCodecs(video_send_codecs(), mid, codecs, pt_suggester);
+   }
+   CodecList filtered_codecs;
+   CodecList negotiated_codecs;
+@@ -794,8 +740,8 @@
+         const MediaContentDescription* mcd =
+             current_content->media_description();
+         for (const Codec& codec : mcd->codecs()) {
+-          if (std::optional<Codec> found_codec = webrtc::FindMatchingCodec(
+-                  mcd->codecs(), codecs.codecs(), codec)) {
+          if (std::optional<Codec> found_codec =
+                  FindMatchingCodec(mcd->codecs(), codecs.codecs(), codec)) {
+             filtered_codecs.push_back(*found_codec);
+           }
+         }
+@@ -913,8 +859,7 @@
+     case RtpTransceiverDirection::kSendRecv:
+     case RtpTransceiverDirection::kStopped:
+     case RtpTransceiverDirection::kInactive:
+-      return GetVideoCodecsForOffer(
+-          webrtc::RtpTransceiverDirectionReversed(offer));
+      return GetVideoCodecsForOffer(RtpTransceiverDirectionReversed(offer));
+     case RtpTransceiverDirection::kSendOnly:
+       return video_send_codecs_.codecs();
+     case RtpTransceiverDirection::kRecvOnly:
+@@ -948,8 +893,7 @@
+     case RtpTransceiverDirection::kSendRecv:
+     case RtpTransceiverDirection::kStopped:
+     case RtpTransceiverDirection::kInactive:
+-      return GetAudioCodecsForOffer(
+-          webrtc::RtpTransceiverDirectionReversed(offer));
+      return GetAudioCodecsForOffer(RtpTransceiverDirectionReversed(offer));
+     case RtpTransceiverDirection::kSendOnly:
+       return audio_send_codecs_.codecs();
+     case RtpTransceiverDirection::kRecvOnly:
+@@ -958,43 +902,6 @@
+   RTC_CHECK_NOTREACHED();
+ }
+ 
+-CodecList CodecVendor::all_video_codecs() const {
+-  CodecList all_codecs;
+-  UsedPayloadTypes used_payload_types;
+-  for (const Codec& codec : video_recv_codecs_.codecs()) {
+-    Codec codec_mutable = codec;
+-    used_payload_types.FindAndSetIdUsed(&codec_mutable);
+-    all_codecs.push_back(codec_mutable);
+-  }
+-
+-  // Use MergeCodecs to merge the second half of our list as it already checks
+-  // and fixes problems with duplicate payload types.
+-  MergeCodecs(video_send_codecs_.codecs(), all_codecs, &used_payload_types);
+-
+-  return all_codecs;
+-}
+-
+-CodecList CodecVendor::all_audio_codecs() const {
+-  // Compute the audio codecs union.
+-  CodecList codecs;
+-  for (const Codec& send : audio_send_codecs_.codecs()) {
+-    codecs.push_back(send);
+-    if (!FindMatchingCodec(audio_send_codecs_.codecs(),
+-                           audio_recv_codecs_.codecs(), send)) {
+-      // It doesn't make sense to have an RTX codec we support sending but not
+-      // receiving.
+-      RTC_DCHECK(send.GetResiliencyType() != Codec::ResiliencyType::kRtx);
+-    }
+-  }
+-  for (const Codec& recv : audio_recv_codecs_.codecs()) {
+-    if (!FindMatchingCodec(audio_recv_codecs_.codecs(),
+-                           audio_send_codecs_.codecs(), recv)) {
+-      codecs.push_back(recv);
+-    }
+-  }
+-  return codecs;
+-}
+-
+ CodecList CodecVendor::audio_sendrecv_codecs() const {
+   // Use NegotiateCodecs to merge our codec lists, since the operation is
+   // essentially the same. Put send_codecs as the offered_codecs, which is the
+diff --git a/webrtc/pc/wcodec_vendor.h b/webrtc/pc/codec_vendor.h
+index 1472225..4a3a8d9 100644
+--- a/webrtc/pc/codec_vendor.h
+++ b/webrtc/pc/codec_vendor.h
+@@ -102,9 +102,6 @@
+       const RtpTransceiverDirection& offer,
+       const RtpTransceiverDirection& answer) const;
+ 
+-  CodecList all_video_codecs() const;
+-  CodecList all_audio_codecs() const;
+-
+   TypedCodecVendor audio_send_codecs_;
+   TypedCodecVendor audio_recv_codecs_;
+ 
+diff --git a/webrtc/pc/used_ids.h b/webrtc/pc/used_ids.h
+index fe80531..f9d825c 100644
+--- a/webrtc/pc/used_ids.h
+++ b/webrtc/pc/used_ids.h
+@@ -14,9 +14,7 @@
+ #include <vector>
+ 
+ #include "api/rtp_parameters.h"
+-#include "media/base/codec.h"
+ #include "rtc_base/checks.h"
+-#include "rtc_base/logging.h"
+ 
+ namespace webrtc {
+ template <typename IdStruct>
+@@ -88,41 +86,6 @@
+   std::set<int> id_set_;
+ };
+ 
+-// Helper class used for finding duplicate RTP payload types among audio, video
+-// and data codecs. When bundle is used the payload types may not collide.
+-class UsedPayloadTypes : public UsedIds<Codec> {
+- public:
+-  UsedPayloadTypes()
+-      : UsedIds<Codec>(kFirstDynamicPayloadTypeLowerRange,
+-                       kLastDynamicPayloadTypeUpperRange) {}
+-
+-  // Check if a payload type is valid. The range [64-95] is forbidden
+-  // when rtcp-mux is used.
+-  static bool IsIdValid(Codec codec, bool rtcp_mux) {
+-    if (rtcp_mux && (codec.id > kLastDynamicPayloadTypeLowerRange &&
+-                     codec.id < kFirstDynamicPayloadTypeUpperRange)) {
+-      return false;
+-    }
+-    return codec.id >= 0 && codec.id <= kLastDynamicPayloadTypeUpperRange;
+-  }
+-
+- protected:
+-  bool IsIdUsed(int new_id) override {
+-    // Range marked for RTCP avoidance is "used".
+-    if (new_id > kLastDynamicPayloadTypeLowerRange &&
+-        new_id < kFirstDynamicPayloadTypeUpperRange)
+-      return true;
+-    return UsedIds<Codec>::IsIdUsed(new_id);
+-  }
+-
+- private:
+-  static const int kFirstDynamicPayloadTypeLowerRange = 35;
+-  static const int kLastDynamicPayloadTypeLowerRange = 63;
+-
+-  static const int kFirstDynamicPayloadTypeUpperRange = 96;
+-  static const int kLastDynamicPayloadTypeUpperRange = 127;
+-};
+-
+ // Helper class used for finding duplicate RTP Header extension ids among
+ // audio and video extensions.
+ class UsedRtpHeaderExtensionIds : public UsedIds<RtpExtension> {
+@@ -190,7 +153,6 @@
+ #ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
+ namespace cricket {
+ using ::webrtc::UsedIds;
+-using ::webrtc::UsedPayloadTypes;
+ using ::webrtc::UsedRtpHeaderExtensionIds;
+ }  // namespace cricket
+ #endif  // WEBRTC_ALLOW_DEPRECATED_NAMESPACES