gnu: webrtc-audio-processing: Update to 2.1.

* gnu/packages/audio.scm (webrtc-audio-processing): Update to 2.1.
[source]: Apply new patches.
[#:phases]: Delete.
* gnu/packages/patches/webrtc-audio-processing-x86-no-sse.patch: Delete file.
* gnu/packages/patches/webrtc-audio-processing-byte-order-pointer-size.patch: Likewise.
* gnu/packages/patches/webrtc-audio-processing-abseil-cpp.patch: New file.
* gnu/packages/patches/webrtc-audio-processing-gcc-14.patch: Likewise.
* gnu/local.mk (dist_patch_DATA): Adjust accordingly.

Change-Id: I7b9c408a7e256d5fb83e14b4d21f6c44a9cd4616
Signed-off-by: Noé Lopez <noelopez@free.fr>

6 files changed, 355 insertions, 69 deletions

diff --git a/gnu/local.mk b/gnu/local.mk
index 24aaaf3e007..c2552790fbd 100644
--- a/gnu/local.mk
+++ b/gnu/local.mk
@@ -2681,9 +2681,9 @@ dist_patch_DATA =						\
   %D%/packages/patches/wcstools-extend-makefiles.patch  \
   %D%/packages/patches/wdl-link-libs-and-fix-jnetlib.patch      \
   %D%/packages/patches/webkitgtk-adjust-bubblewrap-paths.patch  \
+  %D%/packages/patches/webrtc-audio-processing-abseil-cpp.patch \
   %D%/packages/patches/webrtc-audio-processing-big-endian.patch \
-  %D%/packages/patches/webrtc-audio-processing-byte-order-pointer-size.patch \
-  %D%/packages/patches/webrtc-audio-processing-x86-no-sse.patch \
+  %D%/packages/patches/webrtc-audio-processing-gcc-14.patch     \
   %D%/packages/patches/websocketpp-boost-compat.patch           \
   %D%/packages/patches/websocketpp-fix-for-cmake-3.15.patch     \
   %D%/packages/patches/wmctrl-64-fix.patch                      \
diff --git a/gnu/packages/audio.scm b/gnu/packages/audio.scm
index 39b27ce60c6..14b5ccdacb8 100644
--- a/gnu/packages/audio.scm
+++ b/gnu/packages/audio.scm
@@ -11,7 +11,7 @@
 ;;; Copyright © 2016–2023 Tobias Geerinckx-Rice <me@tobias.gr>
 ;;; Copyright © 2018, 2020, 2024 Oleg Pykhalov <go.wigust@gmail.com>
 ;;; Copyright © 2018 okapi <okapi@firemail.cc>
-;;; Copyright © 2018, 2020, 2022-2025 Maxim Cournoyer <maxim@guixotic.coop>
+;;; Copyright © 2018, 2020, 2022-2026 Maxim Cournoyer <maxim@guixotic.coop>
 ;;; Copyright © 2018 Clément Lassieur <clement@lassieur.org>
 ;;; Copyright © 2018 Brett Gilio <brettg@gnu.org>
 ;;; Copyright © 2018, 2019, 2022 Marius Bakke <marius@gnu.org>
@@ -404,7 +404,7 @@ displays a histogram of the roundtrip time jitter.")
 (define-public webrtc-audio-processing
   (package
     (name "webrtc-audio-processing")
-    (version "1.3")
+    (version "2.1")
     (source
      (origin
        (method url-fetch)
@@ -412,30 +412,14 @@ displays a histogram of the roundtrip time jitter.")
         (string-append "http://freedesktop.org/software/pulseaudio/"
                        name "/" name "-" version ".tar.gz"))
        (sha256
-        (base32 "0xfvq5lxg612vfzk3zk6896zcb4cgrrb7fq76w9h40magz0jymcm"))))
+        (base32 "0lrjzhshwj1rkh4fxh7sq2dkkfimlyjs2ha7s3ribsh2dnc6ps1m"))
+       (patches (search-patches "webrtc-audio-processing-abseil-cpp.patch"
+                                "webrtc-audio-processing-gcc-14.patch"))))
     (build-system meson-build-system)
     (arguments
      (list #:configure-flags (if (target-x86-32?)
                                  #~(list "-Dc_args=-DPFFFT_SIMD_DISABLE")
-                                 #~'())
-           #:phases
-           (if (or (target-x86-32?) (target-powerpc?))
-               #~(modify-phases %standard-phases
-                   (add-after 'unpack 'apply-patches
-                     (lambda _
-                       (define (patch file)
-                         (invoke "patch" "-p1" "--force" "-i" file))
-
-                       ;; https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/5
-                       ;; TODO: Move to the 'patches' field of the origin on
-                       ;; the next rebuild.
-                       (patch #$(local-file
-                                 (search-patch
-                                  "webrtc-audio-processing-byte-order-pointer-size.patch")))
-                       (patch #$(local-file
-                                 (search-patch
-                                  "webrtc-audio-processing-x86-no-sse.patch"))))))
-               #~%standard-phases)))
+                                 #~'())))
     (native-inputs (list pkg-config))
     (inputs (list abseil-cpp))
     (synopsis "WebRTC's Audio Processing Library")
diff --git a/gnu/packages/patches/webrtc-audio-processing-abseil-cpp.patch b/gnu/packages/patches/webrtc-audio-processing-abseil-cpp.patch
new file mode 100644
index 00000000000..82d57b2cc6f
--- /dev/null
+++ b/gnu/packages/patches/webrtc-audio-processing-abseil-cpp.patch
@@ -0,0 +1,299 @@
+From c8896801dfbfe03b56f85c1533abc077ff74a533 Mon Sep 17 00:00:00 2001
+From: Reilly Brogan <reilly@reillybrogan.com>
+Date: Tue, 19 Aug 2025 14:47:03 -0500
+Subject: [PATCH 1/2] Fix build with abseil-cpp 202508
+
+---
+ webrtc/api/audio/audio_processing.h           | 12 +++++++++
+ webrtc/api/make_ref_counted.h                 | 13 ++++++++++
+ webrtc/api/scoped_refptr.h                    | 15 +++++++++++
+ .../aec_dump/aec_dump_factory.h               | 15 +++++++++++
+ .../aec_dump/null_aec_dump_factory.cc         | 25 +++++++++++++++++++
+ .../audio_processing/audio_processing_impl.cc |  9 +++++++
+ .../audio_processing/audio_processing_impl.h  |  9 +++++++
+ 7 files changed, 98 insertions(+)
+
+diff --git a/webrtc/api/audio/audio_processing.h b/webrtc/api/audio/audio_processing.h
+index dca75f2..4580ba9 100644
+--- a/webrtc/api/audio/audio_processing.h
++++ b/webrtc/api/audio/audio_processing.h
+@@ -28,6 +28,7 @@
+ #include <string>
+ #include <utility>
+ 
++#include "absl/base/config.h"
+ #include "absl/base/nullability.h"
+ #include "absl/strings/string_view.h"
+ #include "api/array_view.h"
+@@ -632,6 +633,7 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
+   // return value of true indicates that the file has been
+   // sucessfully opened, while a value of false indicates that
+   // opening the file failed.
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+   virtual bool CreateAndAttachAecDump(
+       absl::string_view file_name,
+       int64_t max_log_size_bytes,
+@@ -640,6 +642,16 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
+       absl::Nonnull<FILE*> handle,
+       int64_t max_log_size_bytes,
+       absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
++#else
++  virtual bool CreateAndAttachAecDump(absl::string_view file_name,
++                                      int64_t max_log_size_bytes,
++                                      TaskQueueBase* absl_nonnull
++                                          worker_queue) = 0;
++  virtual bool CreateAndAttachAecDump(FILE* absl_nonnull handle,
++                                      int64_t max_log_size_bytes,
++                                      TaskQueueBase* absl_nonnull
++                                          worker_queue) = 0;
++#endif
+ 
+   // TODO(webrtc:5298) Deprecated variant.
+   // Attaches provided webrtc::AecDump for recording debugging
+diff --git a/webrtc/api/make_ref_counted.h b/webrtc/api/make_ref_counted.h
+index b5f4e99..080023a 100644
+--- a/webrtc/api/make_ref_counted.h
++++ b/webrtc/api/make_ref_counted.h
+@@ -13,6 +13,7 @@
+ #include <type_traits>
+ #include <utility>
+ 
++#include "absl/base/config.h"
+ #include "absl/base/nullability.h"
+ #include "api/ref_count.h"
+ #include "api/scoped_refptr.h"
+@@ -86,7 +87,11 @@ template <
+     typename std::enable_if<std::is_convertible_v<T*, RefCountInterface*> &&
+                                 std::is_abstract_v<T>,
+                             T>::type* = nullptr>
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+ absl::Nonnull<scoped_refptr<T>> make_ref_counted(Args&&... args) {
++#else
++absl_nonnull scoped_refptr<T> make_ref_counted(Args&&... args) {
++#endif
+   return scoped_refptr<T>(new RefCountedObject<T>(std::forward<Args>(args)...));
+ }
+ 
+@@ -99,7 +104,11 @@ template <
+         !std::is_convertible_v<T*, RefCountInterface*> &&
+             webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
+         T>::type* = nullptr>
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+ absl::Nonnull<scoped_refptr<T>> make_ref_counted(Args&&... args) {
++#else
++absl_nonnull scoped_refptr<T> make_ref_counted(Args&&... args) {
++#endif
+   return scoped_refptr<T>(new T(std::forward<Args>(args)...));
+ }
+ 
+@@ -113,7 +122,11 @@ template <
+             !webrtc_make_ref_counted_internal::HasAddRefAndRelease<T>::value,
+ 
+         T>::type* = nullptr>
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+ absl::Nonnull<scoped_refptr<FinalRefCountedObject<T>>> make_ref_counted(
++#else
++absl_nonnull scoped_refptr<FinalRefCountedObject<T>> make_ref_counted(
++#endif
+     Args&&... args) {
+   return scoped_refptr<FinalRefCountedObject<T>>(
+       new FinalRefCountedObject<T>(std::forward<Args>(args)...));
+diff --git a/webrtc/api/scoped_refptr.h b/webrtc/api/scoped_refptr.h
+index c6fb560..8c441ff 100644
+--- a/webrtc/api/scoped_refptr.h
++++ b/webrtc/api/scoped_refptr.h
+@@ -66,6 +66,7 @@
+ #include <cstddef>
+ #include <utility>
+ 
++#include "absl/base/config.h"
+ #include "absl/base/nullability.h"
+ 
+ namespace webrtc {
+@@ -73,13 +74,19 @@ namespace webrtc {
+ template <class T>
+ class ABSL_NULLABILITY_COMPATIBLE scoped_refptr {
+  public:
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+   using absl_nullability_compatible = void;
++#endif
+   using element_type = T;
+ 
+   scoped_refptr() : ptr_(nullptr) {}
+   scoped_refptr(std::nullptr_t) : ptr_(nullptr) {}  // NOLINT(runtime/explicit)
+ 
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+   explicit scoped_refptr(absl::Nullable<T*> p) : ptr_(p) {
++#else
++  explicit scoped_refptr(T* absl_nullable p) : ptr_(p) {
++#endif
+     if (ptr_)
+       ptr_->AddRef();
+   }
+@@ -122,7 +129,11 @@ class ABSL_NULLABILITY_COMPATIBLE scoped_refptr {
+     return retVal;
+   }
+ 
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+   scoped_refptr<T>& operator=(absl::Nullable<T*> p) {
++#else
++  scoped_refptr<T>& operator=(T* absl_nullable p) {
++#endif
+     // AddRef first so that self assignment should work
+     if (p)
+       p->AddRef();
+@@ -152,7 +163,11 @@ class ABSL_NULLABILITY_COMPATIBLE scoped_refptr {
+     return *this;
+   }
+ 
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+   void swap(absl::Nonnull<T**> pp) noexcept {
++#else
++  void swap(T** absl_nonnull pp) noexcept {
++#endif
+     T* p = ptr_;
+     ptr_ = *pp;
+     *pp = p;
+diff --git a/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
+index 0d258a9..14d8b39 100644
+--- a/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
++++ b/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
+@@ -29,6 +29,7 @@ class RTC_EXPORT AecDumpFactory {
+   // The AecDump takes responsibility for `handle` and closes it in the
+   // destructor. A non-null return value indicates that the file has been
+   // sucessfully opened.
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+   static absl::Nullable<std::unique_ptr<AecDump>> Create(
+       FileWrapper file,
+       int64_t max_log_size_bytes,
+@@ -41,6 +42,20 @@ class RTC_EXPORT AecDumpFactory {
+       absl::Nonnull<FILE*> handle,
+       int64_t max_log_size_bytes,
+       absl::Nonnull<TaskQueueBase*> worker_queue);
++#else
++  static absl_nullable std::unique_ptr<AecDump> Create(
++      FileWrapper file,
++      int64_t max_log_size_bytes,
++      TaskQueueBase* absl_nonnull worker_queue);
++  static absl_nullable std::unique_ptr<AecDump> Create(
++      absl::string_view file_name,
++      int64_t max_log_size_bytes,
++      TaskQueueBase* absl_nonnull worker_queue);
++  static absl_nullable std::unique_ptr<AecDump> Create(
++      FILE* absl_nonnull handle,
++      int64_t max_log_size_bytes,
++      TaskQueueBase* absl_nonnull worker_queue);
++#endif
+ };
+ 
+ }  // namespace webrtc
+diff --git a/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc b/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
+index 63929af..658bcee 100644
+--- a/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
++++ b/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
+@@ -16,6 +16,7 @@
+ 
+ namespace webrtc {
+ 
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+ absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+     FileWrapper file,
+     int64_t max_log_size_bytes,
+@@ -37,3 +38,27 @@ absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+   return nullptr;
+ }
+ }  // namespace webrtc
++#else
++absl_nullable std::unique_ptr<AecDump> AecDumpFactory::Create(
++    FileWrapper file,
++    int64_t max_log_size_bytes,
++    TaskQueueBase* absl_nonnull worker_queue) {
++  return nullptr;
++}
++
++absl_nullable std::unique_ptr<AecDump> AecDumpFactory::Create(
++    absl::string_view file_name,
++    int64_t max_log_size_bytes,
++    TaskQueueBase* absl_nonnull worker_queue) {
++  return nullptr;
++}
++
++absl_nullable std::unique_ptr<AecDump> AecDumpFactory::Create(
++    FILE* absl_nonnull handle,
++    int64_t max_log_size_bytes,
++    TaskQueueBase* absl_nonnull worker_queue) {
++  return nullptr;
++}
++}  // namespace webrtc
++
++#endif
+diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
+index a1cba51..1dfe26d 100644
+--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
++++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
+@@ -19,6 +19,7 @@
+ #include <type_traits>
+ #include <utility>
+ 
++#include "absl/base/config.h"
+ #include "absl/base/nullability.h"
+ #include "absl/strings/match.h"
+ #include "absl/strings/string_view.h"
+@@ -1787,7 +1788,11 @@ void AudioProcessingImpl::UpdateRecommendedInputVolumeLocked() {
+ bool AudioProcessingImpl::CreateAndAttachAecDump(
+     absl::string_view file_name,
+     int64_t max_log_size_bytes,
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+     absl::Nonnull<TaskQueueBase*> worker_queue) {
++#else
++    TaskQueueBase* absl_nonnull worker_queue) {
++#endif
+   std::unique_ptr<AecDump> aec_dump =
+       AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
+   if (!aec_dump) {
+@@ -1801,7 +1806,11 @@ bool AudioProcessingImpl::CreateAndAttachAecDump(
+ bool AudioProcessingImpl::CreateAndAttachAecDump(
+     FILE* handle,
+     int64_t max_log_size_bytes,
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+     absl::Nonnull<TaskQueueBase*> worker_queue) {
++#else
++    TaskQueueBase* absl_nonnull worker_queue) {
++#endif
+   std::unique_ptr<AecDump> aec_dump =
+       AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
+   if (!aec_dump) {
+diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
+index ecdc055..51a2bfb 100644
+--- a/webrtc/modules/audio_processing/audio_processing_impl.h
++++ b/webrtc/modules/audio_processing/audio_processing_impl.h
+@@ -20,6 +20,7 @@
+ #include <string>
+ #include <vector>
+ 
++#include "absl/base/config.h"
+ #include "absl/base/nullability.h"
+ #include "absl/strings/string_view.h"
+ #include "api/array_view.h"
+@@ -74,11 +75,19 @@ class AudioProcessingImpl : public AudioProcessing {
+   bool CreateAndAttachAecDump(
+       absl::string_view file_name,
+       int64_t max_log_size_bytes,
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+       absl::Nonnull<TaskQueueBase*> worker_queue) override;
++#else
++      TaskQueueBase* absl_nonnull worker_queue) override;
++#endif
+   bool CreateAndAttachAecDump(
+       FILE* handle,
+       int64_t max_log_size_bytes,
++#if defined(ABSL_LTS_RELEASE_VERSION) && ABSL_LTS_RELEASE_VERSION < 20250512
+       absl::Nonnull<TaskQueueBase*> worker_queue) override;
++#else
++      TaskQueueBase* absl_nonnull worker_queue) override;
++#endif
+   // TODO(webrtc:5298) Deprecated variant.
+   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
+   void DetachAecDump() override;
+-- 
+GitLab
diff --git a/gnu/packages/patches/webrtc-audio-processing-byte-order-pointer-size.patch b/gnu/packages/patches/webrtc-audio-processing-byte-order-pointer-size.patch
deleted file mode 100644
index e2d974afd97..00000000000
--- a/gnu/packages/patches/webrtc-audio-processing-byte-order-pointer-size.patch
+++ /dev/null
@@ -1,32 +0,0 @@
-https://bugs.gentoo.org/917493
-https://sources.debian.org/src/webrtc-audio-processing/1.0-0.2/debian/patches/Add-generic-byte-order-and-pointer-size-detection.patch/
-
-Description: Add generic byte order and pointer size detection
-Author: Than <than@redhat.com>
-Origin: https://bugs.freedesktop.org/show_bug.cgi?id=95738#c4
-Last-Update: 2022-02-01
----
-This patch header follows DEP-3: http://dep.debian.net/deps/dep3/
---- a/webrtc/rtc_base/system/arch.h
-+++ b/webrtc/rtc_base/system/arch.h
-@@ -58,7 +58,19 @@
- #define WEBRTC_ARCH_32_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
- #else
--#error Please add support for your architecture in rtc_base/system/arch.h
-+/* instead of failing, use typical unix defines... */
-+#if __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__
-+#define WEBRTC_ARCH_LITTLE_ENDIAN
-+#elif __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
-+#define WEBRTC_ARCH_BIG_ENDIAN
-+#else
-+#error __BYTE_ORDER__ is not defined
-+#endif
-+#if defined(__LP64__)
-+#define WEBRTC_ARCH_64_BITS
-+#else
-+#define WEBRTC_ARCH_32_BITS
-+#endif
- #endif
- 
- #if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
diff --git a/gnu/packages/patches/webrtc-audio-processing-gcc-14.patch b/gnu/packages/patches/webrtc-audio-processing-gcc-14.patch
new file mode 100644
index 00000000000..430e5020fb4
--- /dev/null
+++ b/gnu/packages/patches/webrtc-audio-processing-gcc-14.patch
@@ -0,0 +1,48 @@
+From e9c78dc4712fa6362b0c839ad57b6b46dce1ba83 Mon Sep 17 00:00:00 2001
+From: =?UTF-8?q?Fran=C3=A7ois=20Laignel?= <francois@centricular.com>
+Date: Mon, 14 Apr 2025 16:48:19 +0200
+Subject: [PATCH] Fix compilation with gcc-15
+
+This commit fixes compilation errors with gcc-15 (default for Fedora 42).
+
+From [1]:
+
+> Common C++ headers (like <memory>) in GCC 15.0.0 (combined with libstdc++)
+> don't transitively include uint64_t anymore.
+
+See also [2].
+
+[1]: https://github.com/ROCm/rocm_smi_lib/pull/198
+[2]: https://github.com/root-project/root/issues/17444
+---
+ .../audio_processing/aec3/multi_channel_content_detector.h       | 1 +
+ webrtc/rtc_base/trace_event.h                                    | 1 +
+ 2 files changed, 2 insertions(+)
+
+diff --git a/webrtc/modules/audio_processing/aec3/multi_channel_content_detector.h b/webrtc/modules/audio_processing/aec3/multi_channel_content_detector.h
+index 2b2f3b8..feb29fd 100644
+--- a/webrtc/modules/audio_processing/aec3/multi_channel_content_detector.h
++++ b/webrtc/modules/audio_processing/aec3/multi_channel_content_detector.h
+@@ -12,6 +12,7 @@
+ #define MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
+ 
+ #include <stddef.h>
++#include <cstdint>
+ 
+ #include <memory>
+ #include <optional>
+diff --git a/webrtc/rtc_base/trace_event.h b/webrtc/rtc_base/trace_event.h
+index 2aee713..f88a68e 100644
+--- a/webrtc/rtc_base/trace_event.h
++++ b/webrtc/rtc_base/trace_event.h
+@@ -28,6 +28,7 @@
+ 
+ #if !defined(RTC_USE_PERFETTO)
+ #include <string>
++#include <cstdint>
+ 
+ #include "rtc_base/event_tracer.h"
+ 
+-- 
+GitLab
+
diff --git a/gnu/packages/patches/webrtc-audio-processing-x86-no-sse.patch b/gnu/packages/patches/webrtc-audio-processing-x86-no-sse.patch
deleted file mode 100644
index c194dd9244e..00000000000
--- a/gnu/packages/patches/webrtc-audio-processing-x86-no-sse.patch
+++ /dev/null
@@ -1,13 +0,0 @@
-https://bugs.gentoo.org/921140
-https://gitlab.freedesktop.org/pulseaudio/webrtc-audio-processing/-/issues/5
---- a/webrtc/rtc_base/system/arch.h
-+++ b/webrtc/rtc_base/system/arch.h
-@@ -34,7 +34,7 @@
- #else
- #define WEBRTC_ARCH_32_BITS
- #endif
--#elif defined(_M_IX86) || defined(__i386__)
-+#elif defined(__SSE__) && (defined(_M_IX86) || defined(__i386__))
- #define WEBRTC_ARCH_X86_FAMILY
- #define WEBRTC_ARCH_X86
- #define WEBRTC_ARCH_32_BITS