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tubestation/third_party/libwebrtc/moz-patch-stack/0031.patch
Michael Froman 00bb744764 Bug 1942428 - Vendor libwebrtc from 882b32d00f 
Upstream commit: https://webrtc.googlesource.com/src/+/882b32d00f6330da948d8e578d259dec0078e7ed
    Reland "Use PayloadTypePicker for video PT assignment"

    This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.

    Reason for revert: Revised codec matching to fix issue.

    Changes also back out some changes that should not have been
    included (using PayloadTypePicker for codec list merging).

    Original change's description:
    > Revert "Use PayloadTypePicker for video PT assignment"
    >
    > This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
    >
    > Reason for revert: Broke internal tests.
    >
    > Original change's description:
    > > Use PayloadTypePicker for video PT assignment
    > >
    > > This includes changes that change the order of codecs.
    > > It is preparatory to doing late assignment of video PTs.
    > >
    > > Bug: webrtc:360058654
    > > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
    > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
    > > Reviewed-by: Henrik Boström <hbos@webrtc.org>
    > > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
    > > Cr-Commit-Position: refs/heads/main@{#43489}
    >
    > Bug: webrtc:360058654
    > Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
    > No-Presubmit: true
    > No-Tree-Checks: true
    > No-Try: true
    > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
    > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
    > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
    > Cr-Commit-Position: refs/heads/main@{#43490}

    Bug: webrtc:360058654
    Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
    Commit-Queue: Harald Alvestrand <hta@webrtc.org>
    Reviewed-by: Artem Titov <titovartem@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#43554}
2025-02-07 15:42:21 -06:00

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From: Dan Minor <dminor@mozilla.com>
Date: Thu, 5 Nov 2020 07:47:00 +0000
Subject: Bug 1654112 - Tweak upstream gn files for Firefox build. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D130075
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/127ace4d8887f11abb201d300a849772a2b519f8
Bug 1820869 - avoid building unreachable files. r=ng,webrtc-reviewers
Differential Revision: https://phabricator.services.mozilla.com/D171922
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/88b3cc6bbece7c53d00e124713330f3d34d2789d
Bug 1822194 - (fix-acabb3641b) Break the new SetParametersCallback stuff into stand-alone files.
acabb3641b from upstream added a callback mechanism to allow failures to be
propagated back to RTCRtpSender.setParameters. Unfortunately, this callback
mechanism was (needlessly) tightly coupled to libwebrtc's implementation of
RTCRtpSender, and also their media channel code. This introduced a lot of
unnecessary dependencies throughout libwebrtc, that spilled into our code as
well.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/59232687efa00e5f7b7bd3d6befca129149e2bf5
Bug 1828517 - (fix-794d599741) account for moved files in BUILD.gn that we don't want to build.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/4a969f6709183d4f55215adaffb8a52b790a8492
Bug 1839451 - (fix-186ebdc1b0) remove BUILD.gn refs to gone files delayable.h, media_channel.h
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0f4d1733cb1a2d8189097af4b5537118ebc95a6
Bug 1839451 - (fix-f6eae959bf) s/rtc_encoder_simulcast_proxy/rtc_simulcast_encoder_adapter/ BUILD ref.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/876b3f5821cd5c30564a82c1da7d057d79d17b01
Bug 1828517 - (fix-a138c6c8a5) handle file moves in BUILD.gn
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cf7e333da17689b3c115a6ffd07fab042bc5f086
Bug 1817024 - (fix-0e2cf6cc01) Skip library create_peer_connection_quality_test_frame_generator. r?mjf!
Differential Revision: https://phabricator.services.mozilla.com/D170887
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fbbc1bf963fda30bca26ae6aac0c3459b8ebea6f
Bug 1826428 - remove libwebrtc's jvm_android.cc from build r=ng,webrtc-reviewers
Based on info from John Lin and previous try runs, we're almost
certainly not using this. Let's try removing it from the build
and landing it. If no problems emerge, we'll be able to remove
our custom changes to upstream code in jvm_android.cc.
Differential Revision: https://phabricator.services.mozilla.com/D174793
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/dca1b97525487ae57d43ced1ebdb4a2d9c9dae89
Bug 1774628 - re-enable support for Windows.Graphics.Capture APIs in libwebrtc. r=pehrsons,webrtc-reviewers
Differential Revision: https://phabricator.services.mozilla.com/D186862
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/08567f4539a12b54202aecbf554ec6540fb99ab2
Bug 1876843 - (fix-082cb56ee7) remove mozilla dependency on pc:media_factory.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/136b3fc0377be6dcaa302469d27968f445e0355e
Bug 1876843 - (fix-b29ff000da) remove mozilla dependency on api:enable_media
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7f403ee038e9797a1aff6161fc70a2d92769851f
Bug 1883116 - (fix-3d9c3687a4) Supporting change of call_factory.cc to create_call.cc.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b86cb7278bc4e557104cec0313d83511b9c8f40d
---
.gn | 2 +
BUILD.gn | 46 ++++++++++++++++++-
api/BUILD.gn | 37 ++++++++++++++-
api/rtp_sender_interface.h | 4 +-
api/rtp_sender_setparameters_callback.cc | 27 +++++++++++
api/rtp_sender_setparameters_callback.h | 28 +++++++++++
api/task_queue/BUILD.gn | 2 +
api/transport/BUILD.gn | 2 +
call/BUILD.gn | 8 +++-
call/audio_send_stream.h | 2 +-
call/video_send_stream.h | 2 +-
common_audio/BUILD.gn | 4 --
common_audio/fir_filter_avx2.cc | 2 +
common_audio/intrin.h | 8 ++++
media/BUILD.gn | 39 +++++++++++++++-
media/base/media_channel.h | 4 --
media/base/media_channel_impl.cc | 13 ------
modules/audio_coding/BUILD.gn | 2 +-
modules/audio_device/BUILD.gn | 17 +++++--
modules/audio_processing/aec3/BUILD.gn | 13 ++----
.../aec3/adaptive_fir_filter_avx2.cc | 2 +-
.../audio_processing/agc2/rnn_vad/BUILD.gn | 2 +-
modules/desktop_capture/BUILD.gn | 29 +-----------
modules/portal/BUILD.gn | 24 ++++++++++
modules/utility/BUILD.gn | 4 ++
modules/video_capture/BUILD.gn | 11 +----
rtc_base/BUILD.gn | 26 ++++++++++-
rtc_base/system/BUILD.gn | 2 +-
test/BUILD.gn | 10 ++++
video/BUILD.gn | 4 +-
webrtc.gni | 32 ++++++++-----
31 files changed, 311 insertions(+), 97 deletions(-)
create mode 100644 api/rtp_sender_setparameters_callback.cc
create mode 100644 api/rtp_sender_setparameters_callback.h
create mode 100644 common_audio/intrin.h
diff --git a/.gn b/.gn
index 12d5e3d4fc..6c011fd453 100644
--- a/.gn
+++ b/.gn
@@ -72,6 +72,8 @@ default_args = {
# Prevent jsoncpp to pass -Wno-deprecated-declarations to users
jsoncpp_no_deprecated_declarations = false
+ use_custom_libcxx = false
+
# Fixes the abi-revision issue.
# TODO(https://bugs.webrtc.org/14437): Remove this section if general
# Chromium fix resolves the problem.
diff --git a/BUILD.gn b/BUILD.gn
index c2e035325c..43b8935b6e 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -33,7 +33,7 @@ if (is_android) {
import("//third_party/jni_zero/jni_zero.gni")
}
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
# This target should (transitively) cause everything to be built; if you run
# 'ninja default' and then 'ninja all', the second build should do no work.
group("default") {
@@ -157,6 +157,10 @@ config("common_inherited_config") {
defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
}
+ if (build_with_mozilla) {
+ defines += [ "WEBRTC_MOZILLA_BUILD" ]
+ }
+
if (!rtc_builtin_ssl_root_certificates) {
defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
}
@@ -496,9 +500,11 @@ config("common_config") {
}
}
+if (is_mac) {
config("common_objc") {
frameworks = [ "Foundation.framework" ]
}
+}
if (!rtc_build_ssl) {
config("external_ssl_library") {
@@ -562,6 +568,34 @@ if (!build_with_chromium) {
"sdk",
"video",
]
+ if (build_with_mozilla) {
+ deps -= [
+ "api:create_peerconnection_factory",
+ "api:enable_media",
+ "api:rtc_error",
+ "api:transport_api",
+ "api/crypto",
+ "api/rtc_event_log:rtc_event_log_factory",
+ "api/task_queue",
+ "api/task_queue:default_task_queue_factory",
+ "api/test/metrics",
+ "api/video_codecs:video_decoder_factory_template",
+ "api/video_codecs:video_decoder_factory_template_dav1d_adapter",
+ "api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
+ "api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
+ "api/video_codecs:video_decoder_factory_template_open_h264_adapter",
+ "api/video_codecs:video_encoder_factory_template",
+ "api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
+ "api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
+ "api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
+ "api/video_codecs:video_encoder_factory_template_open_h264_adapter",
+ "logging:rtc_event_log_api",
+ "p2p:rtc_p2p",
+ "pc:libjingle_peerconnection",
+ "pc:rtc_pc",
+ "sdk",
+ ]
+ }
if (rtc_include_builtin_audio_codecs) {
deps += [
@@ -574,6 +608,16 @@ if (!build_with_chromium) {
deps += [
"api/video:video_frame",
"api/video:video_rtp_headers",
+ "test:rtp_test_utils",
+ ]
+ # Added when we removed deps in other places to avoid building
+ # unreachable sources. See Bug 1820869.
+ deps += [
+ "api/video_codecs:video_codecs_api",
+ "api/video_codecs:rtc_software_fallback_wrappers",
+ "media:rtc_simulcast_encoder_adapter",
+ "modules/video_coding:webrtc_vp8",
+ "modules/video_coding:webrtc_vp9",
]
} else {
deps += [
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 8ec98dfc30..5188fab1f0 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -44,6 +44,9 @@ rtc_source_set("enable_media") {
"environment",
"//third_party/abseil-cpp/absl/base:nullability",
]
+ if (build_with_mozilla) {
+ deps -= [ "../pc:media_factory" ]
+ }
}
rtc_source_set("enable_media_with_defaults") {
@@ -71,7 +74,7 @@ rtc_source_set("enable_media_with_defaults") {
]
}
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
rtc_library("create_peerconnection_factory") {
visibility = [ "*" ]
allow_poison = [ "environment_construction" ]
@@ -223,6 +226,10 @@ rtc_source_set("ice_transport_interface") {
}
rtc_library("dtls_transport_interface") {
+# Previously, Mozilla has tried to limit including this dep, but as
+# upstream changes, it requires whack-a-mole. Making it an empty
+# definition has the same effect, but only requires one change.
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
@@ -239,6 +246,7 @@ rtc_library("dtls_transport_interface") {
"//third_party/abseil-cpp/absl/base:core_headers",
]
}
+}
rtc_library("dtmf_sender_interface") {
visibility = [ "*" ]
@@ -251,6 +259,10 @@ rtc_library("dtmf_sender_interface") {
}
rtc_library("rtp_sender_interface") {
+# Previously, Mozilla has tried to limit including this dep, but as
+# upstream changes, it requires whack-a-mole. Making it an empty
+# definition has the same effect, but only requires one change.
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
@@ -265,6 +277,7 @@ rtc_library("rtp_sender_interface") {
":ref_count",
":rtc_error",
":rtp_parameters",
+ ":rtp_sender_setparameters_callback",
":scoped_refptr",
"../rtc_base:checks",
"../rtc_base/system:rtc_export",
@@ -273,8 +286,23 @@ rtc_library("rtp_sender_interface") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
+
+rtc_library("rtp_sender_setparameters_callback") {
+ visibility = [ "*" ]
+
+ sources = [
+ "rtp_sender_setparameters_callback.cc",
+ "rtp_sender_setparameters_callback.h",
+ ]
+ deps = [
+ ":rtc_error",
+ "//third_party/abseil-cpp/absl/functional:any_invocable",
+ ]
+}
rtc_library("libjingle_peerconnection_api") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
cflags = []
sources = [
@@ -396,6 +424,7 @@ rtc_library("libjingle_peerconnection_api") {
"../rtc_base/system:rtc_export",
]
}
+}
rtc_source_set("frame_transformer_interface") {
visibility = [ "*" ]
@@ -595,6 +624,7 @@ rtc_source_set("peer_network_dependencies") {
}
rtc_source_set("peer_connection_quality_test_fixture_api") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
testonly = true
sources = [ "test/peerconnection_quality_test_fixture.h" ]
@@ -640,6 +670,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_source_set("frame_generator_api") {
visibility = [ "*" ]
@@ -762,6 +793,7 @@ rtc_library("create_frame_generator") {
]
}
+if (!build_with_mozilla) {
rtc_library("create_peer_connection_quality_test_frame_generator") {
visibility = [ "*" ]
testonly = true
@@ -779,6 +811,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
"units:time_delta",
]
}
+}
rtc_source_set("libjingle_logging_api") {
visibility = [ "*" ]
@@ -963,6 +996,7 @@ rtc_source_set("refcountedbase") {
]
}
+if (!build_with_mozilla) {
rtc_library("ice_transport_factory") {
visibility = [ "*" ]
sources = [
@@ -987,6 +1021,7 @@ rtc_library("ice_transport_factory") {
"rtc_event_log:rtc_event_log",
]
}
+}
rtc_library("neteq_simulator_api") {
visibility = [ "*" ]
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 91d7b3c561..4595c28140 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -34,6 +34,8 @@
#include "api/video_codecs/video_encoder_factory.h"
#include "rtc_base/system/rtc_export.h"
+#include "api/rtp_sender_setparameters_callback.h"
+
namespace webrtc {
class RtpSenderObserverInterface {
@@ -46,8 +48,6 @@ class RtpSenderObserverInterface {
virtual ~RtpSenderObserverInterface() {}
};
-using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
-
class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface,
public FrameTransformerHost {
public:
diff --git a/api/rtp_sender_setparameters_callback.cc b/api/rtp_sender_setparameters_callback.cc
new file mode 100644
index 0000000000..99728ef95e
--- /dev/null
+++ b/api/rtp_sender_setparameters_callback.cc
@@ -0,0 +1,27 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// File added by mozilla, to decouple this from libwebrtc's implementation of
+// RTCRtpSender.
+
+#include "api/rtp_sender_setparameters_callback.h"
+
+namespace webrtc {
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+ RTCError error) {
+ if (callback) {
+ std::move(callback)(error);
+ callback = nullptr;
+ }
+ return error;
+}
+
+} // namespace webrtc
diff --git a/api/rtp_sender_setparameters_callback.h b/api/rtp_sender_setparameters_callback.h
new file mode 100644
index 0000000000..45194f5ace
--- /dev/null
+++ b/api/rtp_sender_setparameters_callback.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// File added by mozilla, to decouple this from libwebrtc's implementation of
+// RTCRtpSender.
+
+#ifndef API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
+#define API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
+
+#include "api/rtc_error.h"
+#include "absl/functional/any_invocable.h"
+
+namespace webrtc {
+
+using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+ RTCError error);
+} // namespace webrtc
+
+#endif // API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
diff --git a/api/task_queue/BUILD.gn b/api/task_queue/BUILD.gn
index 9f10f0afc6..c1c917abb2 100644
--- a/api/task_queue/BUILD.gn
+++ b/api/task_queue/BUILD.gn
@@ -29,6 +29,7 @@ rtc_library("task_queue") {
]
}
+if (rtc_include_tests) {
rtc_library("task_queue_test") {
visibility = [ "*" ]
testonly = true
@@ -74,6 +75,7 @@ rtc_library("task_queue_test") {
]
}
}
+}
rtc_library("default_task_queue_factory") {
visibility = [ "*" ]
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 7769526e07..f56ce3ce54 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -107,6 +107,7 @@ rtc_source_set("sctp_transport_factory_interface") {
}
rtc_source_set("stun_types") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"stun.cc",
@@ -129,6 +130,7 @@ rtc_source_set("stun_types") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
if (rtc_include_tests) {
rtc_source_set("test_feedback_generator_interface") {
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 179d4c7efb..9e4c4d1d96 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -48,7 +48,7 @@ rtc_library("call_interfaces") {
"../api:ref_count",
"../api:rtp_headers",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
@@ -379,6 +379,12 @@ rtc_library("call") {
"//third_party/abseil-cpp/absl/functional:bind_front",
"//third_party/abseil-cpp/absl/strings:string_view",
]
+ if (build_with_mozilla) { # See Bug 1820869.
+ deps -= [
+ ":fake_network",
+ ":simulated_network",
+ ]
+ }
}
rtc_library("payload_type_picker") {
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 768ba0b19e..2d2cc89b08 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -27,7 +27,7 @@
#include "api/frame_transformer_interface.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_sender_interface.h"
+#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "api/units/time_delta.h"
#include "call/audio_sender.h"
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 3bf13c5e74..2f5ff4ad3c 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -23,7 +23,7 @@
#include "api/crypto/crypto_options.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_sender_interface.h"
+#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index 4845bc71ef..0b5f6cb858 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -267,14 +267,10 @@ if (current_cpu == "x86" || current_cpu == "x64") {
"resampler/sinc_resampler_avx2.cc",
]
- if (is_win) {
- cflags = [ "/arch:AVX2" ]
- } else {
cflags = [
"-mavx2",
"-mfma",
]
- }
deps = [
":fir_filter",
diff --git a/common_audio/fir_filter_avx2.cc b/common_audio/fir_filter_avx2.cc
index 9cb0f770ca..0031392f8a 100644
--- a/common_audio/fir_filter_avx2.cc
+++ b/common_audio/fir_filter_avx2.cc
@@ -15,6 +15,8 @@
#include <string.h>
#include <xmmintrin.h>
+#include "common_audio/intrin.h"
+
#include "rtc_base/checks.h"
#include "rtc_base/memory/aligned_malloc.h"
diff --git a/common_audio/intrin.h b/common_audio/intrin.h
new file mode 100644
index 0000000000..f6ff7f218f
--- /dev/null
+++ b/common_audio/intrin.h
@@ -0,0 +1,8 @@
+#if defined (__SSE__)
+ #include <immintrin.h>
+ #if defined (__clang__)
+ #include <avxintrin.h>
+ #include <avx2intrin.h>
+ #include <fmaintrin.h>
+ #endif
+#endif
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 69de926516..ce56fbb2bf 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -76,7 +76,7 @@ rtc_library("rtc_media_base") {
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
@@ -127,6 +127,12 @@ rtc_library("rtc_media_base") {
"../video/config:encoder_config",
"//third_party/abseil-cpp/absl/base:core_headers",
]
+ if (build_with_mozilla) {
+ sources -= [
+ "base/adapted_video_track_source.cc",
+ "base/adapted_video_track_source.h",
+ ]
+ }
}
rtc_library("adapted_video_track_source") {
@@ -151,6 +157,9 @@ rtc_library("adapted_video_track_source") {
rtc_source_set("audio_source") {
sources = [ "base/audio_source.h" ]
+ if (build_with_mozilla) {
+ sources -= [ "base/audio_source.h" ]
+ }
}
rtc_library("video_adapter") {
@@ -254,9 +263,16 @@ rtc_library("media_engine") {
"../rtc_base/system:file_wrapper",
"//third_party/abseil-cpp/absl/algorithm:container",
]
+ if (build_with_mozilla) {
+ sources -= [
+ "base/media_engine.cc",
+ "base/media_engine.h",
+ ]
+ }
}
rtc_library("media_channel_impl") {
+if (!build_with_mozilla) {
sources = [
"base/media_channel_impl.cc",
"base/media_channel_impl.h",
@@ -303,6 +319,7 @@ rtc_library("media_channel_impl") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
@@ -382,6 +399,7 @@ rtc_library("codec") {
}
rtc_library("rtp_utils") {
+if (!build_with_mozilla) {
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
@@ -398,8 +416,10 @@ rtc_library("rtp_utils") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("stream_params") {
+if (!build_with_mozilla) {
sources = [
"base/stream_params.cc",
"base/stream_params.h",
@@ -412,6 +432,7 @@ rtc_library("stream_params") {
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
+}
rtc_library("media_constants") {
sources = [
@@ -422,6 +443,7 @@ rtc_library("media_constants") {
}
rtc_library("turn_utils") {
+if (!build_with_mozilla) {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
@@ -432,14 +454,17 @@ rtc_library("turn_utils") {
"../rtc_base/system:rtc_export",
]
}
+}
rtc_library("rid_description") {
+if (!build_with_mozilla) {
sources = [
"base/rid_description.cc",
"base/rid_description.h",
]
deps = []
}
+}
rtc_library("rtc_simulcast_encoder_adapter") {
visibility = [ "*" ]
@@ -521,6 +546,11 @@ rtc_library("rtc_internal_video_codecs") {
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/strings",
]
+ if (build_with_mozilla) {
+ deps -= [
+ "../test:fake_video_codecs",
+ ]
+ }
if (enable_libaom) {
defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
@@ -540,6 +570,13 @@ rtc_library("rtc_internal_video_codecs") {
"engine/internal_encoder_factory.cc",
"engine/internal_encoder_factory.h",
]
+ if (build_with_mozilla) {
+ sources -= [
+ "engine/fake_video_codec_factory.cc",
+ "engine/fake_video_codec_factory.h",
+ "engine/internal_encoder_factory.cc", # See Bug 1820869.
+ ]
+ }
}
rtc_library("rtc_audio_video") {
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 55ec4ff2e9..e286d39f5d 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -67,10 +67,6 @@ class Timing;
namespace webrtc {
class VideoFrame;
-
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error);
-
} // namespace webrtc
namespace cricket {
diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc
index e354db54a0..d2855cbfb0 100644
--- a/media/base/media_channel_impl.cc
+++ b/media/base/media_channel_impl.cc
@@ -31,19 +31,6 @@
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "rtc_base/checks.h"
-namespace webrtc {
-
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error) {
- if (callback) {
- std::move(callback)(error);
- callback = nullptr;
- }
- return error;
-}
-
-} // namespace webrtc
-
namespace cricket {
using webrtc::FrameDecryptorInterface;
using webrtc::FrameEncryptorInterface;
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 3511b8ffed..d671360baf 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -358,7 +358,7 @@ rtc_library("webrtc_opus_wrapper") {
deps += [ rtc_opus_dir ]
public_configs = [ "//third_party/opus:opus_config" ]
} else if (build_with_mozilla) {
- include_dirs = [ getenv("DIST") + "/include/opus" ]
+ public_configs = [ "//third_party/opus:opus_config" ]
}
}
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index aa4d216b28..55c39a530f 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -30,6 +30,7 @@ rtc_source_set("audio_device_default") {
}
rtc_source_set("audio_device") {
+if (!build_with_mozilla) { # See Bug 1820869.
visibility = [ "*" ]
public_deps += [ # no-presubmit-check TODO(webrtc:8603)
":audio_device_api",
@@ -40,6 +41,7 @@ rtc_source_set("audio_device") {
":audio_device_impl",
]
}
+}
rtc_source_set("audio_device_api") {
visibility = [ "*" ]
@@ -55,6 +57,7 @@ rtc_library("audio_device_config") {
}
rtc_library("audio_device_buffer") {
+if (!build_with_mozilla) { # See Bug 1820869.
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
@@ -80,6 +83,7 @@ rtc_library("audio_device_buffer") {
"../../system_wrappers:metrics",
]
}
+}
rtc_library("audio_device_generic") {
sources = [
@@ -250,6 +254,7 @@ if (!build_with_chromium) {
# Contains default implementations of webrtc::AudioDeviceModule for Windows,
# Linux, Mac, iOS and Android.
rtc_library("audio_device_impl") {
+if (!build_with_mozilla) { # See Bug 1820869.
visibility = [ "*" ]
deps = [
":audio_device_buffer",
@@ -295,9 +300,9 @@ rtc_library("audio_device_impl") {
sources = [ "include/fake_audio_device.h" ]
if (build_with_mozilla) {
- sources += [
- "opensl/single_rw_fifo.cc",
- "opensl/single_rw_fifo.h",
+ sources -= [
+ "include/test_audio_device.cc",
+ "include/test_audio_device.h",
]
}
@@ -402,6 +407,7 @@ rtc_library("audio_device_impl") {
sources += [ "dummy/file_audio_device_factory.h" ]
}
}
+}
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
@@ -419,6 +425,7 @@ if (is_mac) {
}
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
@@ -436,8 +443,10 @@ rtc_source_set("mock_audio_device") {
"../../test:test_support",
]
}
+}
-if (rtc_include_tests && !build_with_chromium) {
+# See Bug 1820869 for !build_with_mozilla.
+if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) {
rtc_library("audio_device_unittests") {
testonly = true
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 12a32b8929..39e24c9324 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -260,14 +260,11 @@ if (current_cpu == "x86" || current_cpu == "x64") {
"vector_math_avx2.cc",
]
- if (is_win) {
- cflags = [ "/arch:AVX2" ]
- } else {
- cflags = [
- "-mavx2",
- "-mfma",
- ]
- }
+ cflags = [
+ "-mavx",
+ "-mavx2",
+ "-mfma",
+ ]
deps = [
":adaptive_fir_filter",
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
index b6eda9f117..8d6e1cf3d7 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <immintrin.h>
+#include "common_audio/intrin.h"
#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
#include "rtc_base/checks.h"
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 025794d262..a23a7c15ce 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -121,7 +121,7 @@ rtc_source_set("vector_math") {
if (current_cpu == "x86" || current_cpu == "x64") {
rtc_library("vector_math_avx2") {
sources = [ "vector_math_avx2.cc" ]
- if (is_win) {
+ if (is_win && !build_with_mozilla) {
cflags = [ "/arch:AVX2" ]
} else {
cflags = [
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 4ea31b9db7..e39700419d 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -346,37 +346,12 @@ rtc_library("desktop_capture") {
]
deps += [ ":desktop_capture_objc" ]
}
-
- if (build_with_mozilla) {
- sources += [
- "desktop_device_info.cc",
- "desktop_device_info.h",
- ]
- if (is_win) {
- sources += [
- "app_capturer_win.cc",
- "win/desktop_device_info_win.cc",
- "win/win_shared.cc",
- ]
- }
- }
if (rtc_use_x11_extensions || rtc_use_pipewire) {
sources += [
"mouse_cursor_monitor_linux.cc",
"screen_capturer_linux.cc",
"window_capturer_linux.cc",
]
-
- if (build_with_mozilla && (is_linux || is_chromeos)) {
- sources += [
- "app_capturer_linux.cc",
- "linux/x11/app_capturer_x11.cc",
- "linux/x11/desktop_device_info_linux.cc",
- "linux/x11/desktop_device_info_linux.h",
- "linux/x11/shared_x_util.cc",
- "linux/x11/shared_x_util.h",
- ]
- }
}
if (rtc_use_x11_extensions) {
@@ -535,9 +510,7 @@ rtc_library("desktop_capture") {
deps += [ "../../rtc_base:sanitizer" ]
}
- if (!build_with_mozilla) {
- deps += [ "//third_party/libyuv" ]
- }
+ deps += [ "//third_party/libyuv" ]
if (use_desktop_capture_differ_sse2) {
deps += [ ":desktop_capture_differ_sse2" ]
diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
index e2b2717c89..aff8609b54 100644
--- a/modules/portal/BUILD.gn
+++ b/modules/portal/BUILD.gn
@@ -11,6 +11,7 @@ import("//tools/generate_stubs/rules.gni")
import("../../webrtc.gni")
if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+if (!build_with_mozilla) {
pkg_config("gio") {
packages = [
"gio-2.0",
@@ -88,6 +89,12 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
defines += [ "WEBRTC_USE_GIO" ]
}
}
+} else {
+ config("pipewire_all") {
+ }
+ config("pipewire_config") {
+ }
+}
rtc_library("portal") {
sources = [
@@ -120,5 +127,22 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
deps += [ ":pipewire_stubs" ]
}
+
+ if (build_with_mozilla) {
+ configs -= [
+ ":gio",
+ ":pipewire",
+ ":pipewire_config",
+ ]
+ deps -= [ ":pipewire_stubs" ]
+ defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ]
+ public_deps = [
+ "//third_party/pipewire",
+ "//third_party/drm",
+ "//third_party/gbm",
+ "//third_party/libepoxy"
+ ]
+ }
}
}
+
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index 8cefe5653c..b8d75865f7 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -25,5 +25,9 @@ rtc_source_set("utility") {
"../../rtc_base:platform_thread",
"../../rtc_base/system:arch",
]
+
+ if (build_with_mozilla) {
+ sources -= [ "source/jvm_android.cc" ]
+ }
}
}
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index 29a7bea9d9..a8994aaa68 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -129,21 +129,12 @@ if (!build_with_chromium || is_linux || is_chromeos) {
"strmiids.lib",
"user32.lib",
]
-
- if (build_with_mozilla) {
- sources += [
- "windows/BaseFilter.cpp",
- "windows/BaseInputPin.cpp",
- "windows/BasePin.cpp",
- "windows/MediaType.cpp",
- ]
- }
}
if (is_fuchsia) {
sources += [ "video_capture_factory_null.cc" ]
}
- if (build_with_mozilla && is_android) {
+ if (!build_with_mozilla && is_android) {
include_dirs = [
"/config/external/nspr",
"/nsprpub/lib/ds",
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index d3f3659d91..c8d27ab328 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -312,6 +312,7 @@ rtc_library("sample_counter") {
]
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_library("timestamp_aligner") {
visibility = [ "*" ]
sources = [
@@ -325,6 +326,7 @@ rtc_library("timestamp_aligner") {
"system:rtc_export",
]
}
+}
rtc_library("zero_memory") {
visibility = [ "*" ]
@@ -806,7 +808,9 @@ rtc_library("rtc_json") {
":stringutils",
"//third_party/abseil-cpp/absl/strings:string_view",
]
+if (!build_with_mozilla) {
all_dependent_configs = [ "//third_party/jsoncpp:jsoncpp_config" ]
+}
if (rtc_build_json) {
deps += [ "//third_party/jsoncpp" ]
} else {
@@ -1159,6 +1163,7 @@ if (!build_with_chromium) {
}
rtc_library("network") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"network.cc",
@@ -1196,16 +1201,20 @@ rtc_library("network") {
deps += [ ":win32" ]
}
}
+}
rtc_library("socket_address_pair") {
+if (!build_with_mozilla) {
sources = [
"socket_address_pair.cc",
"socket_address_pair.h",
]
deps = [ ":socket_address" ]
}
+}
rtc_library("net_helper") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"net_helper.cc",
@@ -1216,8 +1225,10 @@ rtc_library("net_helper") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("socket_adapters") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"socket_adapters.cc",
@@ -1236,6 +1247,7 @@ rtc_library("socket_adapters") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("network_route") {
sources = [
@@ -1250,6 +1262,7 @@ rtc_library("network_route") {
}
rtc_library("async_tcp_socket") {
+if (!build_with_mozilla) {
sources = [
"async_tcp_socket.cc",
"async_tcp_socket.h",
@@ -1267,8 +1280,10 @@ rtc_library("async_tcp_socket") {
"network:sent_packet",
]
}
+}
rtc_library("async_udp_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_udp_socket.cc",
@@ -1293,8 +1308,10 @@ rtc_library("async_udp_socket") {
"system:no_unique_address",
]
}
+}
rtc_library("async_packet_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_packet_socket.cc",
@@ -1317,6 +1334,7 @@ rtc_library("async_packet_socket") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
if (rtc_include_tests) {
rtc_library("async_packet_socket_unittest") {
@@ -1389,6 +1407,7 @@ rtc_library("data_rate_limiter") {
}
rtc_library("unique_id_generator") {
+if (!build_with_mozilla) {
sources = [
"unique_id_generator.cc",
"unique_id_generator.h",
@@ -1403,6 +1422,7 @@ rtc_library("unique_id_generator") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("crc32") {
sources = [
@@ -1437,6 +1457,7 @@ rtc_library("stream") {
}
rtc_library("rtc_certificate_generator") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"rtc_certificate_generator.cc",
@@ -1451,6 +1472,7 @@ rtc_library("rtc_certificate_generator") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
rtc_source_set("ssl_header") {
visibility = [ "*" ]
@@ -1507,6 +1529,7 @@ rtc_library("crypto_random") {
}
rtc_library("ssl") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"openssl_key_pair.cc",
@@ -1580,6 +1603,7 @@ rtc_library("ssl") {
deps += [ ":win32" ]
}
}
+}
rtc_library("ssl_adapter") {
visibility = [ "*" ]
@@ -2188,7 +2212,7 @@ if (rtc_include_tests) {
}
}
-if (is_android) {
+if (is_android && !build_with_mozilla) {
rtc_android_library("base_java") {
visibility = [ "*" ]
sources = [
diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn
index b18114f107..3ca6994c81 100644
--- a/rtc_base/system/BUILD.gn
+++ b/rtc_base/system/BUILD.gn
@@ -101,7 +101,7 @@ if (is_mac || is_ios) {
rtc_source_set("warn_current_thread_is_deadlocked") {
sources = [ "warn_current_thread_is_deadlocked.h" ]
deps = []
- if (is_android && !build_with_chromium) {
+ if (is_android && (!build_with_chromium && !build_with_mozilla)) {
sources += [ "warn_current_thread_is_deadlocked.cc" ]
deps += [
"..:logging",
diff --git a/test/BUILD.gn b/test/BUILD.gn
index bfe3cc3279..9be8d14cbd 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -230,6 +230,7 @@ rtc_library("audio_test_common") {
]
}
+if (!build_with_mozilla) {
if (!build_with_chromium) {
if (is_mac || is_ios) {
rtc_library("video_test_mac") {
@@ -283,8 +284,12 @@ if (!build_with_chromium) {
}
}
}
+}
rtc_library("rtp_test_utils") {
+ if (build_with_mozilla) {
+ sources = []
+ } else {
testonly = true
sources = [
"rtcp_packet_parser.cc",
@@ -294,6 +299,7 @@ rtc_library("rtp_test_utils") {
"rtp_file_writer.cc",
"rtp_file_writer.h",
]
+ }
deps = [
"../api:array_view",
@@ -516,7 +522,9 @@ rtc_library("video_frame_writer") {
]
if (!is_ios) {
+ if (!build_with_mozilla) {
deps += [ "//third_party:jpeg" ]
+ }
sources += [ "testsupport/jpeg_frame_writer.cc" ]
} else {
sources += [ "testsupport/jpeg_frame_writer_ios.cc" ]
@@ -1297,6 +1305,7 @@ if (!build_with_chromium) {
}
}
+if (!build_with_mozilla) {
if (!build_with_chromium && is_android) {
rtc_android_library("native_test_java") {
testonly = true
@@ -1339,6 +1348,7 @@ if (!build_with_chromium && is_android) {
sources = [ "android/org/webrtc/native_test/NativeTestWebrtc.java" ]
}
}
+}
rtc_library("call_config_utils") {
testonly = true
diff --git a/video/BUILD.gn b/video/BUILD.gn
index fdd12cf2a7..b767851b13 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -17,7 +17,7 @@ rtc_library("video_stream_encoder_interface") {
"../api:fec_controller_api",
"../api:rtc_error",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api/adaptation:resource_adaptation_api",
"../api/units:data_rate",
@@ -408,7 +408,7 @@ rtc_library("video_stream_encoder_impl") {
":video_stream_encoder_interface",
"../api:field_trials_view",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:sequence_checker",
"../api/adaptation:resource_adaptation_api",
"../api/environment",
diff --git a/webrtc.gni b/webrtc.gni
index 8c1aeab911..4d3d5a8289 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -35,6 +35,11 @@ if (is_mac) {
import("//build/config/mac/rules.gni")
}
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
+
if (is_fuchsia) {
import("//build/config/fuchsia/config.gni")
}
@@ -46,6 +51,11 @@ if (build_with_chromium) {
# This declare_args is separated from the next one because args declared
# in this one, can be read from the next one (args defined in the same
# declare_args cannot be referenced in that scope).
+declare_args() {
+ # Enable to use the Mozilla internal settings.
+ build_with_mozilla = true
+}
+
declare_args() {
# Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h)
# expand to code that will manage symbols visibility.
@@ -92,7 +102,7 @@ declare_args() {
# will tell the pre-processor to remove the default definition of the
# SystemTimeNanos() which is defined in rtc_base/system_time.cc. In
# that case a new implementation needs to be provided.
- rtc_exclude_system_time = build_with_chromium
+ rtc_exclude_system_time = build_with_chromium || build_with_mozilla
# Setting this to false will require the API user to pass in their own
# SSLCertificateVerifier to verify the certificates presented from a
@@ -115,7 +125,7 @@ declare_args() {
# Used to specify an external OpenSSL include path when not compiling the
# library that comes with WebRTC (i.e. rtc_build_ssl == 0).
- rtc_ssl_root = ""
+ rtc_ssl_root = "unused"
# Enable when an external authentication mechanism is used for performing
# packet authentication for RTP packets instead of libsrtp.
@@ -129,13 +139,13 @@ declare_args() {
rtc_exclude_audio_processing_module = false
# Set this to false to skip building examples.
- rtc_build_examples = true
+ rtc_build_examples = false
# Set this to false to skip building tools.
- rtc_build_tools = true
+ rtc_build_tools = false
# Set this to false to skip building code that requires X11.
- rtc_use_x11 = ozone_platform_x11
+ rtc_use_x11 = use_x11
# Set this to use PipeWire on the Wayland display server.
# By default it's only enabled on desktop Linux (excludes ChromeOS) and
@@ -146,9 +156,6 @@ declare_args() {
# Set this to link PipeWire and required libraries directly instead of using the dlopen.
rtc_link_pipewire = false
- # Enable to use the Mozilla internal settings.
- build_with_mozilla = false
-
# Experimental: enable use of Android AAudio which requires Android SDK 26 or above
# and NDK r16 or above.
rtc_enable_android_aaudio = false
@@ -284,7 +291,7 @@ declare_args() {
rtc_build_json = !build_with_mozilla
rtc_build_libsrtp = !build_with_mozilla
rtc_build_libvpx = !build_with_mozilla
- rtc_libvpx_build_vp9 = !build_with_mozilla
+ rtc_libvpx_build_vp9 = true
rtc_build_opus = !build_with_mozilla
rtc_build_ssl = !build_with_mozilla
@@ -293,7 +300,7 @@ declare_args() {
# Chromium uses its own IO handling, so the internal ADM is only built for
# standalone WebRTC.
- rtc_include_internal_audio_device = !build_with_chromium
+ rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla
# Set this to true to enable the avx2 support in webrtc.
# TODO: Make sure that AVX2 works also for non-clang compilers.
@@ -333,6 +340,9 @@ declare_args() {
rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac)
}
+# Enable liboam only on non-mozilla builds.
+enable_libaom = !build_with_mozilla
+
# Make it possible to provide custom locations for some libraries (move these
# up into declare_args should we need to actually use them for the GN build).
rtc_libvpx_dir = "//third_party/libvpx"
@@ -1198,7 +1208,7 @@ if (is_mac || is_ios) {
}
}
-if (is_android) {
+if (is_android && !build_with_mozilla) {
template("rtc_android_library") {
android_library(target_name) {
forward_variables_from(invoker,
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