diff --git a/base/comps/components-publish-channels.toml b/base/comps/components-publish-channels.toml
index 38c39f49c09..44663d773e3 100644
--- a/base/comps/components-publish-channels.toml
+++ b/base/comps/components-publish-channels.toml
@@ -2519,7 +2519,6 @@ components = [
     "watchdog",
     "wayland",
     "web-assets",
-    "webrtc-audio-processing",
     "weldr-client",
     "wget2",
     "which",
diff --git a/base/comps/components.toml b/base/comps/components.toml
index a8b902fd201..c9001147970 100644
--- a/base/comps/components.toml
+++ b/base/comps/components.toml
@@ -6925,7 +6925,6 @@ includes = ["**/*.comp.toml", "component-check-disablement.toml", "component-min
 [components.wayland-protocols]
 [components.web-assets]
 [components.webkitgtk]
-[components.webrtc-audio-processing]
 [components.websocketpp]
 [components.weldr-client]
 [components.wget1]
diff --git a/base/comps/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.comp.toml b/base/comps/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.comp.toml
index 31fa0aab87f..5f029136eca 100644
--- a/base/comps/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.comp.toml
+++ b/base/comps/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.comp.toml
@@ -10,9 +10,25 @@
 # extras, so we must disable them too to avoid "unpackaged file" failures.
 # AZL does not ship openh264. Set openh264 bcond off to prevent building
 # gstreamer1-plugin-openh264 subpackage.
+#
+# AZL does not ship webrtc-audio-processing (Google-derived AEC library,
+# significantly behind upstream, downstream-maintenance burden — see PR
+# #17329). Disabling the `webrtc` bcond drops BuildRequires on
+# pkgconfig(webrtc-audio-coding-1) and pkgconfig(webrtc-audio-processing-1).
+# `webrtc1` cascades off automatically because its bcond expression chains
+# on `%{with webrtc}`. This also removes two plugins from the main
+# gstreamer1-plugins-bad-free RPM that have no in-tree replacement:
+#   - libgstwebrtcdsp.so  (WebRTC echo-cancel / AGC / noise-suppression
+#                          GStreamer element; %files gated on `with webrtc`)
+#   - libgstisac.so       (ISAC voice codec; %files gated on `with webrtc1`)
+# Pipelines that reference `webrtcdsp` or `isac` will fail with
+# "no such element" — there is no fallback. The unrelated GStreamer
+# WebRTC API plugins (libgstwebrtc-1.0.so, libgstwebrtcnice-1.0.so,
+# libgstwebrtc.so) are NOT affected; they come from a different source
+# tree and do not depend on webrtc-audio-processing.
 
 [components.gstreamer1-plugins-bad-free]
-build.without = ["extras", "dc1394", "onnx", "openh264"]
+build.without = ["extras", "dc1394", "onnx", "openh264", "webrtc"]
 
 # AZL does not ship opus — disable the opus parse plugin and remove its BR.
 
diff --git a/base/comps/pipewire/pipewire.comp.toml b/base/comps/pipewire/pipewire.comp.toml
index e15b3b30aaa..94d13f61f5a 100644
--- a/base/comps/pipewire/pipewire.comp.toml
+++ b/base/comps/pipewire/pipewire.comp.toml
@@ -10,7 +10,8 @@ release = { calculation = "manual" }
 #   1 - 5d461a2449 feat: introduce deterministic commit resolution via Azure Linux lock file
 #   2 - introduce azl_release for pipewire
 #   3 - disable opus codec support
-azl_release = "3"
+#   4 - drop webrtc-audio-processing BR
+azl_release = "4"
 
 [[components.pipewire.overlays]]
 description = "Add azl_release to upstream %{baserelease} so we can bump release independently of upstream while keeping a single-number Fedora-style Release schema"
@@ -19,9 +20,50 @@ regex = '^Release:\s+%\{baserelease\}%\{\?snapdate:\.%\{snapdate\}git%\{shortcom
 replacement = 'Release:        %[%{baserelease} + %{azl_release}]%{?snapdate:.%{snapdate}git%{shortcommit}}%{?dist}'
 
 [[components.pipewire.overlays]]
-description = "Disable opus codec support — opus is not shipped in AZL"
+description = """
+Force-disable meson features whose dependencies are not shipped in AZL.
+Required because Fedora's %meson macro always passes --auto-features=enabled,
+which flips every auto-default feature to mandatory and would otherwise turn
+these missing deps into hard build failures:
+  - opus / bluez5-codec-opus: opus codec library not shipped in AZL
+  - echo-cancel-webrtc: webrtc-audio-processing not shipped in AZL
+    (see the component removal and BR-drop overlays below)
+The modules themselves still build; only the codec/backend implementations
+that need the unavailable libraries are skipped.
+"""
 type = "spec-search-replace"
 section = "%build"
 regex = '%meson \\'
 replacement = '''%meson \
-    -D opus=disabled -D bluez5-codec-opus=disabled \'''
+    -D opus=disabled -D bluez5-codec-opus=disabled \
+    -D echo-cancel-webrtc=disabled \'''
+
+# Drop the webrtc-audio-processing BuildRequires lines. The library is not
+# shipped in AZL (Google-derived AEC code that is significantly behind upstream
+# and carries downstream-only patches just to keep building); see the matching
+# component removal in this PR.
+#
+# The BR is gated only by an unconditional `%if 0%{?fedora} >= 40 || ...`
+# block — there is no upstream %bcond, so `build.without` cannot reach it and
+# spec-remove-tag is the only path. The matching -D echo-cancel-webrtc=disabled
+# meson flag is injected by the combined opus/echo-cancel-webrtc overlay above
+# (required because Fedora's %meson uses --auto-features=enabled).
+#
+# libpipewire-module-echo-cancel.so still ships (with the speex / null
+# backends); only the WebRTC backend (libspa-aec-webrtc.so) is dropped.
+#
+# If a spec-remove-tag eventually stops matching after a Fedora rebase,
+# update the value to the new BR spelling — do NOT remove the overlay; the
+# constraint is permanent (webrtc-audio-processing is not shipped in AZL).
+
+[[components.pipewire.overlays]]
+description = "Drop webrtc-audio-processing-1 BR (fedora>=40 / rhel>=10 branch); library not shipped in AZL"
+type = "spec-remove-tag"
+tag = "BuildRequires"
+value = "pkgconfig(webrtc-audio-processing-1)"
+
+[[components.pipewire.overlays]]
+description = "Drop legacy webrtc-audio-processing BR (else branch, EL<10 / F<40); library not shipped in AZL"
+type = "spec-remove-tag"
+tag = "BuildRequires"
+value = "pkgconfig(webrtc-audio-processing) >= 0.2"
diff --git a/base/comps/pulseaudio/pulseaudio.comp.toml b/base/comps/pulseaudio/pulseaudio.comp.toml
index 24ba1e4100f..297aeb90c52 100644
--- a/base/comps/pulseaudio/pulseaudio.comp.toml
+++ b/base/comps/pulseaudio/pulseaudio.comp.toml
@@ -1,7 +1,32 @@
 [components.pulseaudio]
-# Release: 9%{?snap:.%{snap}git%{shortcommit}}%{?dist}
+# Release: %[9 + %{azl_release}]%{?snap:.%{snap}git%{shortcommit}}%{?dist}
 release = { calculation = "manual" }
 
+[components.pulseaudio.build.defines]
+# AZL-only release counter. Added to upstream's hardcoded Release number (9)
+# so the final Release tag is a single number (Fedora-style schema) that
+# bumps independently of upstream. Increment when shipping new AZL-only
+# changes on top of the pinned upstream commit.
+#   1 - 5d461a2449 feat: introduce deterministic commit resolution via Azure Linux lock file
+#   2 - 7c5a4c7e86 fix(pulseaudio): set manual release calculation
+#   3 - f9e776358e fix(pulseaudio): disable gdm-hooks subpackage
+#   4 - introduce azl_release for pulseaudio
+#   5 - disable WebRTC AEC backend
+azl_release = "5"
+
+# Add azl_release to upstream's hardcoded Release (9) so we can bump release
+# independently of upstream while keeping a single-number Fedora-style Release
+# schema. The regex matches the current upstream Release verbatim and acts as
+# a tripwire: if Fedora bumps the upstream number (e.g. 9 → 10), the regex
+# will stop matching and force a conscious re-evaluation of the baseline.
+# When that happens, update the regex *and* the baseline number in the
+# replacement to the new upstream value, and reset azl_release to "1".
+[[components.pulseaudio.overlays]]
+description = "Add azl_release to upstream Release so we can bump release independently of upstream while keeping a single-number Fedora-style Release schema"
+type = "spec-search-replace"
+regex = '^Release:\s+9%\{\?snap:\.%\{snap\}git%\{shortcommit\}\}%\{\?dist\}$'
+replacement = 'Release:        %[9 + %{azl_release}]%{?snap:.%{snap}git%{shortcommit}}%{?dist}'
+
 # The gdm-hooks subpackage was moved to gdm packaging in Fedora 28+,
 # so this subpackage should not be built. The upstream conditional mis-fires
 # when %fedora is undefined, but regardless we force gdm_hooks to 0 so the
@@ -11,3 +36,29 @@ description = "Disable gdm-hooks subpackage (moved to gdm packaging in Fedora 28
 type = "spec-search-replace"
 regex = "%global gdm_hooks 1"
 replacement = "%global gdm_hooks 0"
+
+# Disable WebRTC-based acoustic echo cancellation. AZL does not ship the
+# webrtc-audio-processing component (Google-derived code that is significantly
+# behind upstream and carries downstream patches just to keep building), so
+# PulseAudio's module-echo-cancel falls back to its other AEC implementations
+# (speex, null). Drops the BuildRequires on pkgconfig(webrtc-audio-processing-1)
+# and the module-echo-cancel webrtc backend (libwebrtc-util.so). See PR #17329.
+#
+# Note: we %undefine (rather than set to 0) because the spec also uses
+# `%{?with_webrtc:enabled}%{!?with_webrtc:disabled}` to compute the meson
+# `-D webrtc-aec=` flag — that conditional tests for *definition*, not
+# truthiness, so `%global with_webrtc 0` would still pass `enabled` to meson.
+# Leaving the macro undefined makes every `%if 0%{?with_webrtc}` and
+# `%{?with_webrtc:...}` evaluate to the disabled branch consistently.
+#
+# Regex is anchored to a full line (with optional trailing whitespace) to
+# guard against benign Fedora reformatting and to avoid accidentally
+# matching e.g. `%global with_webrtc 10` in some hypothetical future variant.
+# If this regex stops matching after a Fedora rebase, *update* it to the
+# new macro shape — do NOT remove the overlay; the constraint is permanent
+# (webrtc-audio-processing is not shipped in AZL).
+[[components.pulseaudio.overlays]]
+description = "Disable WebRTC AEC backend (webrtc-audio-processing not shipped in AZL)"
+type = "spec-search-replace"
+regex = '^%global with_webrtc 1\s*$'
+replacement = "%undefine with_webrtc"
diff --git a/locks/gstreamer1-plugins-bad-free.lock b/locks/gstreamer1-plugins-bad-free.lock
index 868ed4ade53..0361b2f01f3 100644
--- a/locks/gstreamer1-plugins-bad-free.lock
+++ b/locks/gstreamer1-plugins-bad-free.lock
@@ -2,5 +2,5 @@
 version = 1
 import-commit = 'b5597ad939bb024c2ab32ef8cf17e2ba2b86ebe5'
 upstream-commit = 'b5597ad939bb024c2ab32ef8cf17e2ba2b86ebe5'
-input-fingerprint = 'sha256:740487c359da4297d4f7dd7ae104f79fb9f6f41697e83588fffac023cdb245c9'
+input-fingerprint = 'sha256:393070c6177a80d1c864ad5f146e5716905c7c43697d06baba46a714252f146e'
 resolution-input-hash = 'sha256:466421704711c4fd3c71f0b2ed715a0e61d49e3e26f3a2637fee755795849c8e'
diff --git a/locks/pipewire.lock b/locks/pipewire.lock
index e7e4b66b30a..e3a2cc69ac7 100644
--- a/locks/pipewire.lock
+++ b/locks/pipewire.lock
@@ -2,5 +2,5 @@
 version = 1
 import-commit = 'f930436e78f3c6c32901254baf4979bab7c2710a'
 upstream-commit = 'f930436e78f3c6c32901254baf4979bab7c2710a'
-input-fingerprint = 'sha256:fc5fa06db1353654e6f6926fcd667ee31143ee244599e9cba56e96834bf7f88f'
+input-fingerprint = 'sha256:99d3b9e77574e8b7d13e5fd678e3e81000b2c0270105c45ce676725c6d93a7d0'
 resolution-input-hash = 'sha256:466421704711c4fd3c71f0b2ed715a0e61d49e3e26f3a2637fee755795849c8e'
diff --git a/locks/pulseaudio.lock b/locks/pulseaudio.lock
index 4540be31824..9385dbaec95 100644
--- a/locks/pulseaudio.lock
+++ b/locks/pulseaudio.lock
@@ -2,5 +2,5 @@
 version = 1
 import-commit = '9afb54a4dc0503fb2beb1853f6abe627713c0cf2'
 upstream-commit = '9afb54a4dc0503fb2beb1853f6abe627713c0cf2'
-input-fingerprint = 'sha256:d1579fccfc58e11114c6ad59360f542df440fcecc01b25a02cdc3b1ff95d41d2'
+input-fingerprint = 'sha256:b60c88cc394e0ef45d05f9d26f19f364d5c0186d583a96a71f8c3b4b37ddebb1'
 resolution-input-hash = 'sha256:466421704711c4fd3c71f0b2ed715a0e61d49e3e26f3a2637fee755795849c8e'
diff --git a/locks/webrtc-audio-processing.lock b/locks/webrtc-audio-processing.lock
deleted file mode 100644
index 71ca097c6cc..00000000000
--- a/locks/webrtc-audio-processing.lock
+++ /dev/null
@@ -1,6 +0,0 @@
-# Managed by azldev component update. Do not edit manually.
-version = 1
-import-commit = '3b7dc5b179158e24c207f280168809841e8ffc2a'
-upstream-commit = '3b7dc5b179158e24c207f280168809841e8ffc2a'
-input-fingerprint = 'sha256:f079fdc9071e3245760ff0bb1dad1f8e90901c5e7d138961dc395525475d92ee'
-resolution-input-hash = 'sha256:466421704711c4fd3c71f0b2ed715a0e61d49e3e26f3a2637fee755795849c8e'
diff --git a/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.azl.macros b/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.azl.macros
index 568e9e80ca7..327826ce9b0 100644
--- a/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.azl.macros
+++ b/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.azl.macros
@@ -4,3 +4,4 @@
 %_without_extras 1
 %_without_onnx 1
 %_without_openh264 1
+%_without_webrtc 1
diff --git a/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.spec b/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.spec
index 1071b2d4277..18dfb2b80ef 100644
--- a/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.spec
+++ b/specs/g/gstreamer1-plugins-bad-free/gstreamer1-plugins-bad-free.spec
@@ -36,7 +36,7 @@
 
 Name:           gstreamer1-plugins-bad-free
 Version:        1.26.10
-Release: 6%{?dist}
+Release: 7%{?dist}
 Summary:        GStreamer streaming media framework "bad" plugins
 
 # Automatically converted from old format: LGPLv2+ and LGPLv2 - review is highly recommended.
diff --git a/specs/p/pipewire/pipewire.azl.macros b/specs/p/pipewire/pipewire.azl.macros
index bc741136bc9..d235e1c54e4 100644
--- a/specs/p/pipewire/pipewire.azl.macros
+++ b/specs/p/pipewire/pipewire.azl.macros
@@ -1,3 +1,3 @@
 # Macros file automatically generated by azldev.
 # Do not edit manually; changes will be overwritten.
-%azl_release 3
+%azl_release 4
diff --git a/specs/p/pipewire/pipewire.spec b/specs/p/pipewire/pipewire.spec
index 7dd280c9a32..55a082aaf41 100644
--- a/specs/p/pipewire/pipewire.spec
+++ b/specs/p/pipewire/pipewire.spec
@@ -123,9 +123,7 @@ BuildRequires:  ncurses-devel
 BuildRequires:  pulseaudio-libs-devel
 BuildRequires:  avahi-devel
 %if 0%{?fedora} >= 40 || 0%{?rhel} >= 10
-BuildRequires:  pkgconfig(webrtc-audio-processing-1)
 %else
-BuildRequires:  pkgconfig(webrtc-audio-processing) >= 0.2
 %endif
 BuildRequires:  libusb1-devel
 BuildRequires:  readline-devel
@@ -453,6 +451,7 @@ cp %{SOURCE1} subprojects/packagefiles/
 %build
 %meson \
     -D opus=disabled -D bluez5-codec-opus=disabled \
+    -D echo-cancel-webrtc=disabled \
     -D docs=enabled -D man=enabled -D gstreamer=enabled -D systemd=enabled	\
     -D sdl2=disabled 								\
     -D audiotestsrc=disabled -D videotestsrc=disabled				\
diff --git a/specs/p/pulseaudio/pulseaudio.azl.macros b/specs/p/pulseaudio/pulseaudio.azl.macros
new file mode 100644
index 00000000000..0aa211138a9
--- /dev/null
+++ b/specs/p/pulseaudio/pulseaudio.azl.macros
@@ -0,0 +1,3 @@
+# Macros file automatically generated by azldev.
+# Do not edit manually; changes will be overwritten.
+%azl_release 5
diff --git a/specs/p/pulseaudio/pulseaudio.spec b/specs/p/pulseaudio/pulseaudio.spec
index 694fa3ee931..b517c2a6778 100644
--- a/specs/p/pulseaudio/pulseaudio.spec
+++ b/specs/p/pulseaudio/pulseaudio.spec
@@ -1,6 +1,9 @@
 # This spec file has been modified by azldev to include build configuration overlays.
 # Do not edit manually; changes may be overwritten.
 
+# All Azure Linux specs with overlays include this macro file, irrespective of whether new macros have been added.
+%{load:%{_sourcedir}/pulseaudio.azl.macros}
+
 %global pa_major   17.0
 #global pa_minor   0
 
@@ -9,7 +12,7 @@
 #global gitcommit  f5d3606fe76302c7dbdb0f6a80400df829a5f846
 #global shortcommit %%(c=%%{gitcommit}; echo ${c:0:5})
 
-%global with_webrtc 1
+%undefine with_webrtc
 
 %if 0%{?fedora}
 %global enable_daemon 1
@@ -37,7 +40,7 @@
 Name:           pulseaudio
 Summary:        Improved Linux Sound Server
 Version:        %{pa_major}%{?pa_minor:.%{pa_minor}}
-Release:        9%{?snap:.%{snap}git%{shortcommit}}%{?dist}
+Release:        %[9 + %{azl_release}]%{?snap:.%{snap}git%{shortcommit}}%{?dist}
 License:        LGPL-2.1-or-later
 URL:            http://www.freedesktop.org/wiki/Software/PulseAudio
 %if 0%{?gitrel}
@@ -50,6 +53,7 @@ Source1:        http://freedesktop.org/software/pulseaudio/releases/pulseaudio-%
 %endif
 
 Source5:        default.pa-for-gdm
+Source9999: pulseaudio.azl.macros
 
 # revert upstream commit to rely solely on autospawn for autostart, instead
 # include a fallback to manual launch when autospawn fails, like when
diff --git a/specs/w/webrtc-audio-processing/65f002e.patch b/specs/w/webrtc-audio-processing/65f002e.patch
deleted file mode 100644
index f9098ab3569..00000000000
--- a/specs/w/webrtc-audio-processing/65f002e.patch
+++ /dev/null
@@ -1,313 +0,0 @@
---- a/webrtc/common_audio/wav_file.cc
-+++ b/webrtc/common_audio/wav_file.cc
-@@ -10,6 +10,7 @@
- 
- #include "common_audio/wav_file.h"
- 
-+#include <byteswap.h>
- #include <errno.h>
- 
- #include <algorithm>
-@@ -34,6 +35,38 @@ bool FormatSupported(WavFormat format) {
-          format == WavFormat::kWavFormatIeeeFloat;
- }
- 
-+template <typename T>
-+void TranslateEndianness(T* destination, const T* source, size_t length) {
-+  static_assert(sizeof(T) == 2 || sizeof(T) == 4 || sizeof(T) == 8,
-+                "no converter, use integral types");
-+  if (sizeof(T) == 2) {
-+    const uint16_t* src = reinterpret_cast<const uint16_t*>(source);
-+    uint16_t* dst = reinterpret_cast<uint16_t*>(destination);
-+    for (size_t index = 0; index < length; index++) {
-+      dst[index] = bswap_16(src[index]);
-+    }
-+  }
-+  if (sizeof(T) == 4) {
-+    const uint32_t* src = reinterpret_cast<const uint32_t*>(source);
-+    uint32_t* dst = reinterpret_cast<uint32_t*>(destination);
-+    for (size_t index = 0; index < length; index++) {
-+      dst[index] = bswap_32(src[index]);
-+    }
-+  }
-+  if (sizeof(T) == 8) {
-+    const uint64_t* src = reinterpret_cast<const uint64_t*>(source);
-+    uint64_t* dst = reinterpret_cast<uint64_t*>(destination);
-+    for (size_t index = 0; index < length; index++) {
-+      dst[index] = bswap_64(src[index]);
-+    }
-+  }
-+}
-+
-+template <typename T>
-+void TranslateEndianness(T* buffer, size_t length) {
-+  TranslateEndianness(buffer, buffer, length);
-+}
-+
- // Doesn't take ownership of the file handle and won't close it.
- class WavHeaderFileReader : public WavHeaderReader {
-  public:
-@@ -89,10 +122,6 @@ void WavReader::Reset() {
- 
- size_t WavReader::ReadSamples(const size_t num_samples,
-                               int16_t* const samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to big-endian when reading from WAV file"
--#endif
--
-   size_t num_samples_left_to_read = num_samples;
-   size_t next_chunk_start = 0;
-   while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
-@@ -105,6 +134,9 @@ size_t WavReader::ReadSamples(const size_t num_samples,
-       num_bytes_read = file_.Read(samples_to_convert.data(),
-                                   chunk_size * sizeof(samples_to_convert[0]));
-       num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(samples_to_convert.data(), num_samples_read);
-+#endif
- 
-       for (size_t j = 0; j < num_samples_read; ++j) {
-         samples[next_chunk_start + j] = FloatToS16(samples_to_convert[j]);
-@@ -114,6 +146,10 @@ size_t WavReader::ReadSamples(const size_t num_samples,
-       num_bytes_read = file_.Read(&samples[next_chunk_start],
-                                   chunk_size * sizeof(samples[0]));
-       num_samples_read = num_bytes_read / sizeof(samples[0]);
-+
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(&samples[next_chunk_start], num_samples_read);
-+#endif
-     }
-     RTC_CHECK(num_samples_read == 0 || (num_bytes_read % num_samples_read) == 0)
-         << "Corrupt file: file ended in the middle of a sample.";
-@@ -129,10 +165,6 @@ size_t WavReader::ReadSamples(const size_t num_samples,
- }
- 
- size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to big-endian when reading from WAV file"
--#endif
--
-   size_t num_samples_left_to_read = num_samples;
-   size_t next_chunk_start = 0;
-   while (num_samples_left_to_read > 0 && num_unread_samples_ > 0) {
-@@ -145,6 +177,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
-       num_bytes_read = file_.Read(samples_to_convert.data(),
-                                   chunk_size * sizeof(samples_to_convert[0]));
-       num_samples_read = num_bytes_read / sizeof(samples_to_convert[0]);
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(samples_to_convert.data(), num_samples_read);
-+#endif
- 
-       for (size_t j = 0; j < num_samples_read; ++j) {
-         samples[next_chunk_start + j] =
-@@ -155,6 +190,9 @@ size_t WavReader::ReadSamples(const size_t num_samples, float* const samples) {
-       num_bytes_read = file_.Read(&samples[next_chunk_start],
-                                   chunk_size * sizeof(samples[0]));
-       num_samples_read = num_bytes_read / sizeof(samples[0]);
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(&samples[next_chunk_start], num_samples_read);
-+#endif
- 
-       for (size_t j = 0; j < num_samples_read; ++j) {
-         samples[next_chunk_start + j] =
-@@ -213,24 +251,32 @@ WavWriter::WavWriter(FileWrapper file,
- }
- 
- void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to little-endian when writing to WAV file"
--#endif
--
-   for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
-     const size_t num_remaining_samples = num_samples - i;
-     const size_t num_samples_to_write =
-         std::min(kMaxChunksize, num_remaining_samples);
- 
-     if (format_ == WavFormat::kWavFormatPcm) {
-+#ifndef WEBRTC_ARCH_BIG_ENDIAN
-       RTC_CHECK(
-           file_.Write(&samples[i], num_samples_to_write * sizeof(samples[0])));
-+#else
-+      std::array<int16_t, kMaxChunksize> converted_samples;
-+      TranslateEndianness(converted_samples.data(), &samples[i],
-+                          num_samples_to_write);
-+      RTC_CHECK(
-+          file_.Write(converted_samples.data(),
-+                      num_samples_to_write * sizeof(converted_samples[0])));
-+#endif
-     } else {
-       RTC_CHECK_EQ(format_, WavFormat::kWavFormatIeeeFloat);
-       std::array<float, kMaxChunksize> converted_samples;
-       for (size_t j = 0; j < num_samples_to_write; ++j) {
-         converted_samples[j] = S16ToFloat(samples[i + j]);
-       }
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(converted_samples.data(), num_samples_to_write);
-+#endif
-       RTC_CHECK(
-           file_.Write(converted_samples.data(),
-                       num_samples_to_write * sizeof(converted_samples[0])));
-@@ -243,10 +289,6 @@ void WavWriter::WriteSamples(const int16_t* samples, size_t num_samples) {
- }
- 
- void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Need to convert samples to little-endian when writing to WAV file"
--#endif
--
-   for (size_t i = 0; i < num_samples; i += kMaxChunksize) {
-     const size_t num_remaining_samples = num_samples - i;
-     const size_t num_samples_to_write =
-@@ -257,6 +299,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
-       for (size_t j = 0; j < num_samples_to_write; ++j) {
-         converted_samples[j] = FloatS16ToS16(samples[i + j]);
-       }
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(converted_samples.data(), num_samples_to_write);
-+#endif
-       RTC_CHECK(
-           file_.Write(converted_samples.data(),
-                       num_samples_to_write * sizeof(converted_samples[0])));
-@@ -266,6 +311,9 @@ void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
-       for (size_t j = 0; j < num_samples_to_write; ++j) {
-         converted_samples[j] = FloatS16ToFloat(samples[i + j]);
-       }
-+#ifdef WEBRTC_ARCH_BIG_ENDIAN
-+      TranslateEndianness(converted_samples.data(), num_samples_to_write);
-+#endif
-       RTC_CHECK(
-           file_.Write(converted_samples.data(),
-                       num_samples_to_write * sizeof(converted_samples[0])));
---- a/webrtc/common_audio/wav_header.cc
-+++ b/webrtc/common_audio/wav_header.cc
-@@ -14,6 +14,8 @@
- 
- #include "common_audio/wav_header.h"
- 
-+#include <endian.h>
-+
- #include <cstring>
- #include <limits>
- #include <string>
-@@ -26,10 +28,6 @@
- namespace webrtc {
- namespace {
- 
--#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
--#error "Code not working properly for big endian platforms."
--#endif
--
- #pragma pack(2)
- struct ChunkHeader {
-   uint32_t ID;
-@@ -174,6 +172,8 @@ bool FindWaveChunk(ChunkHeader* chunk_header,
-     if (readable->Read(chunk_header, sizeof(*chunk_header)) !=
-         sizeof(*chunk_header))
-       return false;  // EOF.
-+    chunk_header->Size = le32toh(chunk_header->Size);
-+
-     if (ReadFourCC(chunk_header->ID) == sought_chunk_id)
-       return true;  // Sought chunk found.
-     // Ignore current chunk by skipping its payload.
-@@ -187,6 +187,13 @@ bool ReadFmtChunkData(FmtPcmSubchunk* fmt_subchunk, WavHeaderReader* readable) {
-   if (readable->Read(&(fmt_subchunk->AudioFormat), kFmtPcmSubchunkSize) !=
-       kFmtPcmSubchunkSize)
-     return false;
-+  fmt_subchunk->AudioFormat = le16toh(fmt_subchunk->AudioFormat);
-+  fmt_subchunk->NumChannels = le16toh(fmt_subchunk->NumChannels);
-+  fmt_subchunk->SampleRate = le32toh(fmt_subchunk->SampleRate);
-+  fmt_subchunk->ByteRate = le32toh(fmt_subchunk->ByteRate);
-+  fmt_subchunk->BlockAlign = le16toh(fmt_subchunk->BlockAlign);
-+  fmt_subchunk->BitsPerSample = le16toh(fmt_subchunk->BitsPerSample);
-+
-   const uint32_t fmt_size = fmt_subchunk->header.Size;
-   if (fmt_size != kFmtPcmSubchunkSize) {
-     // There is an optional two-byte extension field permitted to be present
-@@ -214,19 +221,22 @@ void WritePcmWavHeader(size_t num_channels,
-   auto header = rtc::MsanUninitialized<WavHeaderPcm>({});
-   const size_t bytes_in_payload = bytes_per_sample * num_samples;
- 
--  header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F');
--  header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size);
--  header.riff.Format = PackFourCC('W', 'A', 'V', 'E');
--  header.fmt.header.ID = PackFourCC('f', 'm', 't', ' ');
--  header.fmt.header.Size = kFmtPcmSubchunkSize;
--  header.fmt.AudioFormat = MapWavFormatToHeaderField(WavFormat::kWavFormatPcm);
--  header.fmt.NumChannels = static_cast<uint16_t>(num_channels);
--  header.fmt.SampleRate = sample_rate;
--  header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample);
--  header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample);
--  header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample);
--  header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
--  header.data.header.Size = static_cast<uint32_t>(bytes_in_payload);
-+  header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F'));
-+  header.riff.header.Size =
-+      htole32(RiffChunkSize(bytes_in_payload, *header_size));
-+  header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E'));
-+  header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' '));
-+  header.fmt.header.Size = htole32(kFmtPcmSubchunkSize);
-+  header.fmt.AudioFormat =
-+      htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatPcm));
-+  header.fmt.NumChannels = htole16(num_channels);
-+  header.fmt.SampleRate = htole32(sample_rate);
-+  header.fmt.ByteRate =
-+      htole32(ByteRate(num_channels, sample_rate, bytes_per_sample));
-+  header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample));
-+  header.fmt.BitsPerSample = htole16(8 * bytes_per_sample);
-+  header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a'));
-+  header.data.header.Size = htole32(bytes_in_payload);
- 
-   // Do an extra copy rather than writing everything to buf directly, since buf
-   // might not be correctly aligned.
-@@ -245,24 +255,26 @@ void WriteIeeeFloatWavHeader(size_t num_channels,
-   auto header = rtc::MsanUninitialized<WavHeaderIeeeFloat>({});
-   const size_t bytes_in_payload = bytes_per_sample * num_samples;
- 
--  header.riff.header.ID = PackFourCC('R', 'I', 'F', 'F');
--  header.riff.header.Size = RiffChunkSize(bytes_in_payload, *header_size);
--  header.riff.Format = PackFourCC('W', 'A', 'V', 'E');
--  header.fmt.header.ID = PackFourCC('f', 'm', 't', ' ');
--  header.fmt.header.Size = kFmtIeeeFloatSubchunkSize;
-+  header.riff.header.ID = htole32(PackFourCC('R', 'I', 'F', 'F'));
-+  header.riff.header.Size =
-+      htole32(RiffChunkSize(bytes_in_payload, *header_size));
-+  header.riff.Format = htole32(PackFourCC('W', 'A', 'V', 'E'));
-+  header.fmt.header.ID = htole32(PackFourCC('f', 'm', 't', ' '));
-+  header.fmt.header.Size = htole32(kFmtIeeeFloatSubchunkSize);
-   header.fmt.AudioFormat =
--      MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat);
--  header.fmt.NumChannels = static_cast<uint16_t>(num_channels);
--  header.fmt.SampleRate = sample_rate;
--  header.fmt.ByteRate = ByteRate(num_channels, sample_rate, bytes_per_sample);
--  header.fmt.BlockAlign = BlockAlign(num_channels, bytes_per_sample);
--  header.fmt.BitsPerSample = static_cast<uint16_t>(8 * bytes_per_sample);
--  header.fmt.ExtensionSize = 0;
--  header.fact.header.ID = PackFourCC('f', 'a', 'c', 't');
--  header.fact.header.Size = 4;
--  header.fact.SampleLength = static_cast<uint32_t>(num_channels * num_samples);
--  header.data.header.ID = PackFourCC('d', 'a', 't', 'a');
--  header.data.header.Size = static_cast<uint32_t>(bytes_in_payload);
-+      htole16(MapWavFormatToHeaderField(WavFormat::kWavFormatIeeeFloat));
-+  header.fmt.NumChannels = htole16(num_channels);
-+  header.fmt.SampleRate = htole32(sample_rate);
-+  header.fmt.ByteRate =
-+      htole32(ByteRate(num_channels, sample_rate, bytes_per_sample));
-+  header.fmt.BlockAlign = htole16(BlockAlign(num_channels, bytes_per_sample));
-+  header.fmt.BitsPerSample = htole16(8 * bytes_per_sample);
-+  header.fmt.ExtensionSize = htole16(0);
-+  header.fact.header.ID = htole32(PackFourCC('f', 'a', 'c', 't'));
-+  header.fact.header.Size = htole32(4);
-+  header.fact.SampleLength = htole32(num_channels * num_samples);
-+  header.data.header.ID = htole32(PackFourCC('d', 'a', 't', 'a'));
-+  header.data.header.Size = htole32(bytes_in_payload);
- 
-   // Do an extra copy rather than writing everything to buf directly, since buf
-   // might not be correctly aligned.
-@@ -391,6 +403,7 @@ bool ReadWavHeader(WavHeaderReader* readable,
-     return false;
-   if (ReadFourCC(header.riff.Format) != "WAVE")
-     return false;
-+  header.riff.header.Size = le32toh(header.riff.header.Size);
- 
-   // Find "fmt " and "data" chunks. While the official Wave file specification
-   // does not put requirements on the chunks order, it is uncommon to find the
diff --git a/specs/w/webrtc-audio-processing/arches.patch b/specs/w/webrtc-audio-processing/arches.patch
deleted file mode 100644
index eaf43dfb457..00000000000
--- a/specs/w/webrtc-audio-processing/arches.patch
+++ /dev/null
@@ -1,86 +0,0 @@
---- webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h	2023-09-05 10:19:47.000000000 -0500
-+++ webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h	2024-02-12 10:04:12.114812565 -0600
-@@ -15,8 +15,9 @@
- #define RTC_BASE_SYSTEM_ARCH_H_
- 
- // Processor architecture detection.  For more info on what's defined, see:
--//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
--//   http://www.agner.org/optimize/calling_conventions.pdf
-+//   https://docs.microsoft.com/en-us/cpp/preprocessor/predefined-macros
-+//   https://www.agner.org/optimize/calling_conventions.pdf
-+//   https://sourceforge.net/p/predef/wiki/Architectures/
- //   or with gcc, run: "echo | gcc -E -dM -"
- #if defined(_M_X64) || defined(__x86_64__)
- #define WEBRTC_ARCH_X86_FAMILY
-@@ -27,29 +28,50 @@
- #define WEBRTC_ARCH_ARM_FAMILY
- #define WEBRTC_ARCH_64_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
--#elif defined(__riscv) || defined(__riscv__)
--#define WEBRTC_ARCH_LITTLE_ENDIAN
--#if __riscv_xlen == 64
--#define WEBRTC_ARCH_64_BITS
--#else
--#define WEBRTC_ARCH_32_BITS
--#endif
- #elif defined(_M_IX86) || defined(__i386__)
- #define WEBRTC_ARCH_X86_FAMILY
- #define WEBRTC_ARCH_X86
- #define WEBRTC_ARCH_32_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
--#elif defined(__ARMEL__)
-+#elif defined(_M_ARM) || defined(__ARMEL__)
- #define WEBRTC_ARCH_ARM_FAMILY
- #define WEBRTC_ARCH_32_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
--#elif defined(__MIPSEL__)
-+#elif defined(__MIPSEL__) || defined(__MIPSEB__)
- #define WEBRTC_ARCH_MIPS_FAMILY
- #if defined(__LP64__)
- #define WEBRTC_ARCH_64_BITS
- #else
- #define WEBRTC_ARCH_32_BITS
- #endif
-+#if defined(__MIPSEL__)
-+#define WEBRTC_ARCH_LITTLE_ENDIAN
-+#else
-+#define WEBRTC_ARCH_BIG_ENDIAN
-+#endif
-+#elif defined(__PPC__)
-+#if defined(__PPC64__)
-+#define WEBRTC_ARCH_64_BITS
-+#else
-+#define WEBRTC_ARCH_32_BITS
-+#endif
-+#if defined(__LITTLE_ENDIAN__)
-+#define WEBRTC_ARCH_LITTLE_ENDIAN
-+#else
-+#define WEBRTC_ARCH_BIG_ENDIAN
-+#endif
-+#elif defined(__sparc) || defined(__sparc__)
-+#if __SIZEOF_LONG__ == 8
-+#define WEBRTC_ARCH_64_BITS
-+#else
-+#define WEBRTC_ARCH_32_BITS
-+#endif
-+#define WEBRTC_ARCH_BIG_ENDIAN
-+#elif defined(__riscv) && __riscv_xlen == 64
-+#define WEBRTC_ARCH_64_BITS
-+#define WEBRTC_ARCH_LITTLE_ENDIAN
-+#elif defined(__riscv) && __riscv_xlen == 32
-+#define WEBRTC_ARCH_32_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
- #elif defined(__pnacl__)
- #define WEBRTC_ARCH_32_BITS
---- webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h~	2024-02-12 10:14:11.277835532 -0600
-+++ webrtc-audio-processing-1.3/webrtc/rtc_base/system/arch.h	2024-02-12 10:25:11.558554823 -0600
-@@ -79,6 +79,9 @@
- #elif defined(__EMSCRIPTEN__)
- #define WEBRTC_ARCH_32_BITS
- #define WEBRTC_ARCH_LITTLE_ENDIAN
-+#elif defined(__s390x__)
-+#define WEBRTC_ARCH_64_BITS
-+#define WEBRTC_ARCH_BIG_ENDIAN
- #else
- #error Please add support for your architecture in rtc_base/system/arch.h
- #endif
diff --git a/specs/w/webrtc-audio-processing/sources b/specs/w/webrtc-audio-processing/sources
deleted file mode 100644
index b25567ac2ed..00000000000
--- a/specs/w/webrtc-audio-processing/sources
+++ /dev/null
@@ -1 +0,0 @@
-SHA512 (webrtc-audio-processing-1.3.tar.xz) = daabaed06ff9d1d4076b2abba14efbbebeb8930b14a99fb47974399d2812f3f851e3d6a691b09fbcfb1a3535c6ade967bac4c17a3728f3138b302e3b844c5c67
diff --git a/specs/w/webrtc-audio-processing/webrtc-audio-processing-1.3-gcc15.patch b/specs/w/webrtc-audio-processing/webrtc-audio-processing-1.3-gcc15.patch
deleted file mode 100644
index b69ac5cfbbd..00000000000
--- a/specs/w/webrtc-audio-processing/webrtc-audio-processing-1.3-gcc15.patch
+++ /dev/null
@@ -1,11 +0,0 @@
-diff -Naur webrtc-audio-processing-1.3-original/webrtc/api/task_queue/task_queue_base.h webrtc-audio-processing-1.3/webrtc/api/task_queue/task_queue_base.h
---- webrtc-audio-processing-1.3-original/webrtc/api/task_queue/task_queue_base.h	2023-09-05 11:19:47.000000000 -0400
-+++ webrtc-audio-processing-1.3/webrtc/api/task_queue/task_queue_base.h	2025-02-04 14:24:16.239045712 -0500
-@@ -10,6 +10,7 @@
- #ifndef API_TASK_QUEUE_TASK_QUEUE_BASE_H_
- #define API_TASK_QUEUE_TASK_QUEUE_BASE_H_
- 
-+#include <cstdint>
- #include <memory>
- 
- #include "api/task_queue/queued_task.h"
diff --git a/specs/w/webrtc-audio-processing/webrtc-audio-processing.spec b/specs/w/webrtc-audio-processing/webrtc-audio-processing.spec
deleted file mode 100644
index 993b4250d4b..00000000000
--- a/specs/w/webrtc-audio-processing/webrtc-audio-processing.spec
+++ /dev/null
@@ -1,219 +0,0 @@
-# This spec file has been modified by azldev to include build configuration overlays.
-# Do not edit manually; changes may be overwritten.
-
-Name:           webrtc-audio-processing
-Version:        1.3
-Release: 10%{?dist}
-Summary:        Library for echo cancellation
-
-License:        BSD-3-Clause
-URL:            http://www.freedesktop.org/software/pulseaudio/webrtc-audio-processing/
-Source0:        http://freedesktop.org/software/pulseaudio/webrtc-audio-processing/%{name}-%{version}.tar.xz
-
-Patch0:         arches.patch
-Patch1:         65f002e.patch
-# Add missing #include for GCC 15
-#
-# Downstream-only because the package is significantly behind upstream, and the
-# code in question has changed quite a bit in the latest release.
-Patch2:         webrtc-audio-processing-1.3-gcc15.patch
-
-BuildRequires: meson
-BuildRequires: gcc gcc-c++
-BuildRequires: abseil-cpp-devel
-#BuildRequires: neon-devel
-
-%description
-%{name} is a library derived from Google WebRTC project that 
-provides echo cancellation functionality. This library is used by for example
-PulseAudio to provide echo cancellation.
-
-%package        devel
-Summary:        Development files for %{name}
-Requires:       %{name}%{?_isa} = %{version}-%{release}
-
-%description    devel
-The %{name}-devel package contains libraries and header
-files for developing applications that use %{name}.
-
-%prep
-%autosetup -p1
-
-%build
-%meson
-%meson_build \
-#%%ifarch %%{arm} aarch64
-#  -Dneon=no \
-#%endif
-
-
-%install
-%meson_install
-
-
-%ldconfig_scriptlets
-
-%files
-%doc NEWS AUTHORS README.md
-%license COPYING
-%{_libdir}/libwebrtc-audio-coding-1.so.3*
-%{_libdir}/libwebrtc-audio-processing-1.so.3*
-
-%files devel
-%{_libdir}/libwebrtc-audio-coding-1.so
-%{_libdir}/libwebrtc-audio-processing-1.so
-%{_libdir}/pkgconfig/webrtc-audio-coding-1.pc
-%{_libdir}/pkgconfig/webrtc-audio-processing-1.pc
-%{_includedir}/webrtc-audio-processing-1/
-
-
-%changelog
-* Mon Sep 08 2025 Benjamin A. Beasley <code@musicinmybrain.net> - 1.3-9
-- Rebuilt for abseil-cpp 20250814.0
-
-* Fri Jul 25 2025 Fedora Release Engineering <releng@fedoraproject.org> - 1.3-8
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_43_Mass_Rebuild
-
-* Mon May 26 2025 Benjamin A. Beasley <code@musicinmybrain.net> - 1.3-7
-- Rebuilt for abseil-cpp 20250512.0
-
-* Tue Feb 25 2025 Benjamin A. Beasley <code@musicinmybrain.net> - 1.3-6
-- Rebuilt for abseil-cpp-20250127.0
-
-* Tue Feb 04 2025 Benjamin A. Beasley <code@musicinmybrain.net> - 1.3-5
-- Patch for GCC 15 (fix RHBZ#2341533)
-
-* Sun Jan 19 2025 Fedora Release Engineering <releng@fedoraproject.org> - 1.3-4
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_42_Mass_Rebuild
-
-* Sun Aug 25 2024 Benjamin A. Beasley <code@musicinmybrain.net> - 1.3-3
-- Rebuilt for abseil-cpp-20240722.0
-
-* Sat Jul 20 2024 Fedora Release Engineering <releng@fedoraproject.org> - 1.3-2
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_41_Mass_Rebuild
-
-* Fri Feb 09 2024 Gwyn Ciesla <gwync@protonmail.com> - 1.3-1
-- 1.3
-
-* Sat Jan 27 2024 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-12
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_40_Mass_Rebuild
-
-* Sat Jul 22 2023 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-11
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_39_Mass_Rebuild
-
-* Sat Jan 21 2023 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-10
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_38_Mass_Rebuild
-
-* Sat Jul 23 2022 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-9
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_37_Mass_Rebuild
-
-* Sat Jan 22 2022 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-8
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_36_Mass_Rebuild
-
-* Fri Jul 23 2021 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-7
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_35_Mass_Rebuild
-
-* Wed Jan 27 2021 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-6
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild
-
-* Wed Jul 29 2020 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-5
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild
-
-* Fri Jan 31 2020 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-4
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild
-
-* Sat Jul 27 2019 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-3
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild
-
-* Sun Feb 03 2019 Fedora Release Engineering <releng@fedoraproject.org> - 0.3.1-2
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild
-
-* Wed Jul 25 2018 Rex Dieter <rdieter@fedoraproject.org> - 0.3.1-1
-- 0.3.1
-- use %%make_build %%make_install %%ldconfig_scriptlets
-
-* Mon Jul 23 2018 Debarshi Ray <rishi@fedoraproject.org> 0.3-9
-- Update License
-
-* Sat Jul 14 2018 Fedora Release Engineering <releng@fedoraproject.org> - 0.3-8
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild
-
-* Sat Mar  3 2018 Peter Robinson <pbrobinson@fedoraproject.org> 0.3-7
-- Add gcc/gcc-c++ build requires
-- Add aarch64 to NEON exception
-
-* Fri Feb 09 2018 Fedora Release Engineering <releng@fedoraproject.org> - 0.3-6
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild
-
-* Mon Jan 22 2018 Rex Dieter <rdieter@fedoraproject.org> - 0.3-5
-- pull in upstream fixes, use %%autosetup
-
-* Thu Aug 03 2017 Fedora Release Engineering <releng@fedoraproject.org> - 0.3-4
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild
-
-* Thu Jul 27 2017 Fedora Release Engineering <releng@fedoraproject.org> - 0.3-3
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild
-
-* Sat Feb 11 2017 Fedora Release Engineering <releng@fedoraproject.org> - 0.3-2
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild
-
-* Mon Jul 18 2016 Rex Dieter <rdieter@fedoraproject.org> - 0.3-1
-- 0.3
-
-* Fri May 27 2016 Rex Dieter <rdieter@fedoraproject.org> - 0.2-7
-- better/upstreamable x86_msse2.patch
-
-* Fri May 27 2016 Rex Dieter <rdieter@fedoraproject.org> - 0.2-6
-- simpler/upstreamable no_undefined.patch (fdo#96244)
-
-* Wed May 25 2016 Than Ngo <than@redhat.com> - 0.2-5
-- add url to upstream bug report
-
-* Tue May 24 2016 Than Ngo <than@redhat.com> - 0.2-4
-- add support big endian
-
-* Mon May 16 2016 Rex Dieter <rdieter@fedoraproject.org> - 0.2-3
-- ExclusiveArch primary archs, FTBFS on big endian arches (#1336466)
-
-* Mon May 16 2016 Rex Dieter <rdieter@fedoraproject.org> - 0.2-2
-- link w/ --no-undefined
-- fix x86 sse2 runtime detection
-
-* Thu May 12 2016 Rex Dieter <rdieter@fedoraproject.org> - 0.2-1
-- webrtc-audio-processing-0.2 (#1335536)
-- %%files: track ABI/API closer
-
-* Fri Feb 05 2016 Fedora Release Engineering <releng@fedoraproject.org> - 0.1-11
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild
-
-* Fri Jun 19 2015 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.1-10
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild
-
-* Sat May 02 2015 Kalev Lember <kalevlember@gmail.com> - 0.1-9
-- Rebuilt for GCC 5 C++11 ABI change
-
-* Mon Aug 18 2014 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.1-8
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild
-
-* Sun Jun 08 2014 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.1-7
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild
-
-* Tue Jan 28 2014 Kyle McMartin <kyle@fedoraproject.org> - 0.1-6
-- webrtc-fix-typedefs-on-other-arches.patch: fix ftbfs on non-x86/arm due to
-  a build #error in typedefs.h, however, the defines are not used anywhere in
-  the code. Fixes build on ppc{,64}, s390x, and aarch64.
-
-* Sun Aug 04 2013 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.1-5
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild
-
-* Thu Jul 11 2013 Debarshi Ray <rishi@fedoraproject.org> 0.1-4
-- Rebuilt to fix broken binary possibly caused by broken toolchain
-
-* Fri Feb 15 2013 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 0.1-3
-- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild
-
-* Tue Oct 9 2012 Dan Horák <dan[at]danny.cz> 0.1-2
-- set ExclusiveArch x86 and ARM for now
-
-* Fri Oct 5 2012 Christian Schaller <christian.schaller@gmail.com> 0.1-1
-- Initial Fedora spec.