guix_mirror_bot pushed a commit to branch gnome-team
in repository guix.
commit 48ae31bef3aba8d35e5aa2969fc61c49445938e8
Author: Maxim Cournoyer <[email protected]>
AuthorDate: Sun Mar 29 18:10:30 2026 +0900
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 <[email protected]>
---
gnu/local.mk | 4 +-
gnu/packages/audio.scm | 28 +-
.../webrtc-audio-processing-abseil-cpp.patch | 299 +++++++++++++++++++++
...-audio-processing-byte-order-pointer-size.patch | 32 ---
.../patches/webrtc-audio-processing-gcc-14.patch | 48 ++++
.../webrtc-audio-processing-x86-no-sse.patch | 13 -
6 files changed, 355 insertions(+), 69 deletions(-)
diff --git a/gnu/local.mk b/gnu/local.mk
index 24aaaf3e00..c2552790fb 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 39b27ce60c..14b5ccdacb 100644
--- a/gnu/packages/audio.scm
+++ b/gnu/packages/audio.scm
@@ -11,7 +11,7 @@
;;; Copyright © 2016–2023 Tobias Geerinckx-Rice <[email protected]>
;;; Copyright © 2018, 2020, 2024 Oleg Pykhalov <[email protected]>
;;; Copyright © 2018 okapi <[email protected]>
-;;; Copyright © 2018, 2020, 2022-2025 Maxim Cournoyer <[email protected]>
+;;; Copyright © 2018, 2020, 2022-2026 Maxim Cournoyer <[email protected]>
;;; Copyright © 2018 Clément Lassieur <[email protected]>
;;; Copyright © 2018 Brett Gilio <[email protected]>
;;; Copyright © 2018, 2019, 2022 Marius Bakke <[email protected]>
@@ -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 0000000000..82d57b2cc6
--- /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 <[email protected]>
+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 e2d974afd9..0000000000
--- 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 <[email protected]>
-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 0000000000..430e5020fb
--- /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?= <[email protected]>
+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 c194dd9244..0000000000
--- 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
