firefox-main/third_party/libwebrtc/moz-patch-stack/s0027.patch

Source code

Revision control

Copy as Markdown

Other Tools

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
Bug 1820869 - avoid building unreachable files. r=ng,webrtc-reviewers
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.
Bug 1828517 - (fix-794d599741) account for moved files in BUILD.gn that we don't want to build.
Bug 1839451 - (fix-186ebdc1b0) remove BUILD.gn refs to gone files delayable.h, media_channel.h
Bug 1839451 - (fix-f6eae959bf) s/rtc_encoder_simulcast_proxy/rtc_simulcast_encoder_adapter/ BUILD ref.
Bug 1828517 - (fix-a138c6c8a5) handle file moves in BUILD.gn
Bug 1817024 - (fix-0e2cf6cc01) Skip library create_peer_connection_quality_test_frame_generator. r?mjf!
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.
Bug 1774628 - re-enable support for Windows.Graphics.Capture APIs in libwebrtc. r=pehrsons,webrtc-reviewers
Bug 1876843 - (fix-082cb56ee7) remove mozilla dependency on pc:media_factory.
Bug 1876843 - (fix-b29ff000da) remove mozilla dependency on api:enable_media
Bug 1883116 - (fix-3d9c3687a4) Supporting change of call_factory.cc to create_call.cc.
---
.gn | 2 +
BUILD.gn | 45 ++++++++++++++++++-
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 | 2 +-
call/audio_send_stream.h | 2 +-
call/video_send_stream.h | 2 +-
common_audio/BUILD.gn | 4 --
common_audio/fir_filter_avx2.cc | 2 +
media/BUILD.gn | 39 +++++++++++++++-
media/base/media_channel.h | 3 --
media/base/media_channel_impl.cc | 9 ----
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 ++++++++-----
30 files changed, 296 insertions(+), 92 deletions(-)
create mode 100644 api/rtp_sender_setparameters_callback.cc
create mode 100644 api/rtp_sender_setparameters_callback.h
diff --git a/.gn b/.gn
index 3b92d0ad12..c0365bd7a4 100644
--- a/.gn
+++ b/.gn
@@ -71,6 +71,8 @@ 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 9b17a0b627..5c51495ddc 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") {
@@ -160,6 +160,10 @@ 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" ]
}
@@ -502,9 +506,11 @@ config("common_config") {
}
}
+if (is_mac) {
config("common_objc") {
frameworks = [ "Foundation.framework" ]
}
+}
if (!rtc_build_ssl) {
config("external_ssl_library") {
@@ -568,6 +574,33 @@ if (!build_with_chromium) {
"sdk",
"video",
]
+ if (build_with_mozilla) {
+ deps -= [
+ "api:create_modular_peer_connection_factory",
+ "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/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",
+ "pc:libjingle_peerconnection",
+ "pc:rtc_pc",
+ "sdk",
+ ]
+ }
if (rtc_include_builtin_audio_codecs) {
deps += [
@@ -580,6 +613,16 @@ 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",
]
}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 351464779d..637d36f900 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -44,6 +44,9 @@ rtc_library("enable_media") {
"environment",
"//third_party/abseil-cpp/absl/base:nullability",
]
+ if (build_with_mozilla) {
+ deps -= [ "../pc:media_factory" ]
+ }
}
rtc_library("enable_media_with_defaults") {
@@ -89,7 +92,7 @@ rtc_library("create_modular_peer_connection_factory") {
]
}
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
rtc_library("create_peerconnection_factory") {
visibility = [ "*" ]
allow_poison = [ "environment_construction" ]
@@ -246,6 +249,10 @@ 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 = [
@@ -262,6 +269,7 @@ rtc_library("dtls_transport_interface") {
"//third_party/abseil-cpp/absl/base:core_headers",
]
}
+}
rtc_library("dtmf_sender_interface") {
visibility = [ "*" ]
@@ -274,6 +282,10 @@ 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 = [
@@ -288,6 +300,7 @@ 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",
@@ -296,6 +309,20 @@ 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("sctp_transport_interface") {
visibility = [ "*" ]
@@ -546,6 +573,7 @@ rtc_library("video_track_source_proxy_factory") {
}
rtc_library("libjingle_peerconnection_api") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
cflags = []
sources = [
@@ -667,6 +695,7 @@ rtc_library("libjingle_peerconnection_api") {
"../rtc_base/system:rtc_export",
]
}
+}
rtc_library("frame_transformer_interface") {
visibility = [ "*" ]
@@ -890,6 +919,7 @@ 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" ]
@@ -934,6 +964,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("frame_generator_api") {
visibility = [ "*" ]
@@ -1041,6 +1072,7 @@ rtc_library("create_frame_generator") {
]
}
+if (!build_with_mozilla) {
rtc_library("create_peer_connection_quality_test_frame_generator") {
visibility = [ "*" ]
testonly = true
@@ -1058,6 +1090,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
"units:time_delta",
]
}
+}
rtc_source_set("data_channel_event_observer_interface") {
visibility = [ "*" ]
@@ -1251,6 +1284,7 @@ rtc_source_set("refcountedbase") {
}
rtc_library("ice_transport_factory") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"ice_transport_factory.cc",
@@ -1273,6 +1307,7 @@ rtc_library("ice_transport_factory") {
"rtc_event_log",
]
}
+}
rtc_library("neteq_simulator_api") {
visibility = [ "*" ]
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index db04b9580c..56030c1b33 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 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 ebc6ec0177..d0c8e6d843 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
@@ -75,6 +76,7 @@ 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 6f7613452c..cd16754077 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -97,6 +97,7 @@ rtc_source_set("sctp_transport_factory_interface") {
}
rtc_library("stun_types") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"stun.cc",
@@ -119,6 +120,7 @@ rtc_library("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 aa25942030..ae1657bee2 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",
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 84341b5cb1..2ae1742f91 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 c5a24d2c48..6ae8932ca9 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -24,7 +24,7 @@
#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/units/data_rate.h"
#include "api/video/encoded_image.h"
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index e9b469f258..9176dbcc47 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -266,14 +266,10 @@ 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 88890be798..c9aefc297b 100644
--- a/common_audio/fir_filter_avx2.cc
+++ b/common_audio/fir_filter_avx2.cc
@@ -16,6 +16,8 @@
#include <cstdint>
#include <cstring>
+#include "common_audio/intrin.h"
+
#include "rtc_base/checks.h"
#include "rtc_base/memory/aligned_malloc.h"
diff --git a/media/BUILD.gn b/media/BUILD.gn
index c8ddd58965..e923ca0dc0 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -77,7 +77,7 @@ 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",
@@ -128,6 +128,12 @@ 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") {
@@ -154,6 +160,9 @@ 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") {
@@ -263,9 +272,16 @@ 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",
@@ -313,6 +329,7 @@ rtc_library("media_channel_impl") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
@@ -412,6 +429,7 @@ rtc_library("codec_list") {
}
rtc_library("rtp_utils") {
+if (!build_with_mozilla) {
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
@@ -428,8 +446,10 @@ 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",
@@ -443,6 +463,7 @@ rtc_library("stream_params") {
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
+}
rtc_library("media_constants") {
sources = [
@@ -453,6 +474,7 @@ rtc_library("media_constants") {
}
rtc_library("turn_utils") {
+if (!build_with_mozilla) {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
@@ -463,14 +485,17 @@ 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 = [ ":codec" ]
}
+}
rtc_library("rtc_simulcast_encoder_adapter") {
visibility = [ "*" ]
@@ -551,6 +576,11 @@ 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" ]
@@ -570,6 +600,13 @@ 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 ceb5c48256..1d331532f1 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -68,9 +68,6 @@ namespace webrtc {
class VideoFrame;
struct VideoFormat;
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error);
-
class VideoMediaSendChannelInterface;
class VideoMediaReceiveChannelInterface;
class VoiceMediaSendChannelInterface;
diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc
index 2cb3560682..01fcd82dc8 100644
--- a/media/base/media_channel_impl.cc
+++ b/media/base/media_channel_impl.cc
@@ -35,15 +35,6 @@
namespace webrtc {
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error) {
- if (callback) {
- std::move(callback)(error);
- callback = nullptr;
- }
- return error;
-}
-
VideoOptions::VideoOptions()
: content_hint(VideoTrackInterface::ContentHint::kNone) {}
VideoOptions::~VideoOptions() = default;
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 323224ae99..4a24d6a75a 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -362,7 +362,7 @@ 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 bad43a5c91..6091e49b7b 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 = [ "*" ]
@@ -51,6 +53,7 @@ rtc_source_set("audio_device_api") {
}
rtc_library("audio_device_buffer") {
+if (!build_with_mozilla) { # See Bug 1820869.
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
@@ -79,6 +82,7 @@ rtc_library("audio_device_buffer") {
"../../system_wrappers:metrics",
]
}
+}
rtc_library("audio_device_generic") {
sources = [
@@ -258,6 +262,7 @@ 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",
@@ -304,9 +309,9 @@ 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",
]
}
@@ -409,6 +414,7 @@ rtc_library("audio_device_impl") {
sources += [ "dummy/file_audio_device_factory.h" ]
}
}
+}
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
@@ -426,6 +432,7 @@ if (is_mac) {
}
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
@@ -444,8 +451,10 @@ 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 1f5a2717c9..e5c2bb9714 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -275,14 +275,11 @@ 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 f7ba1fda86..121f13e79f 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 <algorithm>
#include <array>
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 075e2a8110..1f3a66667a 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 82c10e4c5f..693e45ff8b 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -348,37 +348,12 @@ 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) {
@@ -541,9 +516,7 @@ 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 de8a81be55..e8367393a3 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",
@@ -89,6 +90,12 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
defines += [ "WEBRTC_USE_GIO" ]
}
}
+} else {
+ config("pipewire_all") {
+ }
+ config("pipewire_config") {
+ }
+}
rtc_library("portal") {
sources = [
@@ -122,5 +129,22 @@ 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 167bce9157..89203fbcb4 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -26,5 +26,9 @@ rtc_library("utility") {
"../../rtc_base:platform_thread_types",
"../../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 39aa39a41c..b26e30e8bb 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -137,21 +137,12 @@ 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 7ae9389189..d4db206648 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -336,6 +336,7 @@ rtc_library("sample_counter") {
]
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_library("timestamp_aligner") {
visibility = [ "*" ]
sources = [
@@ -349,6 +350,7 @@ rtc_library("timestamp_aligner") {
"system:rtc_export",
]
}
+}
rtc_library("zero_memory") {
visibility = [ "*" ]
@@ -822,7 +824,9 @@ rtc_library("rtc_json") {
"//third_party/abseil-cpp/absl/strings",
"//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 {
@@ -1212,6 +1216,7 @@ if (!build_with_chromium) {
}
rtc_library("network") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"network.cc",
@@ -1254,16 +1259,20 @@ 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",
@@ -1274,8 +1283,10 @@ rtc_library("net_helper") {
"//third_party/abseil-cpp/absl/strings",
]
}
+}
rtc_library("socket_adapters") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"socket_adapters.cc",
@@ -1297,6 +1308,7 @@ rtc_library("socket_adapters") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("network_route") {
sources = [
@@ -1311,6 +1323,7 @@ rtc_library("network_route") {
}
rtc_library("async_tcp_socket") {
+if (!build_with_mozilla) {
sources = [
"async_tcp_socket.cc",
"async_tcp_socket.h",
@@ -1333,8 +1346,10 @@ rtc_library("async_tcp_socket") {
"//third_party/abseil-cpp/absl/memory",
]
}
+}
rtc_library("async_udp_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_udp_socket.cc",
@@ -1360,8 +1375,10 @@ rtc_library("async_udp_socket") {
"//third_party/abseil-cpp/absl/base:nullability",
]
}
+}
rtc_library("async_packet_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_packet_socket.cc",
@@ -1383,6 +1400,7 @@ rtc_library("async_packet_socket") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
if (rtc_include_tests) {
rtc_library("async_packet_socket_unittest") {
@@ -1472,6 +1490,7 @@ rtc_library("data_rate_limiter") {
}
rtc_library("unique_id_generator") {
+if (!build_with_mozilla) {
sources = [
"unique_id_generator.cc",
"unique_id_generator.h",
@@ -1489,6 +1508,7 @@ rtc_library("unique_id_generator") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("crc32") {
sources = [
@@ -1519,6 +1539,7 @@ rtc_library("stream") {
}
rtc_library("rtc_certificate_generator") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"rtc_certificate_generator.cc",
@@ -1533,6 +1554,7 @@ rtc_library("rtc_certificate_generator") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
rtc_source_set("ssl_header") {
visibility = [ "*" ]
@@ -1589,6 +1611,7 @@ rtc_library("crypto_random") {
}
rtc_library("ssl") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"openssl_key_pair.cc",
@@ -1661,6 +1684,7 @@ rtc_library("ssl") {
deps += [ ":win32" ]
}
}
+}
rtc_library("ssl_adapter") {
visibility = [ "*" ]
@@ -2399,7 +2423,7 @@ 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 c1181618e9..4a772795ed 100644
--- a/rtc_base/system/BUILD.gn
+++ b/rtc_base/system/BUILD.gn
@@ -101,7 +101,7 @@ if (is_mac || is_ios) {
rtc_library("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 46a410ba6d..7047b53377 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -279,6 +279,7 @@ rtc_library("audio_test_common") {
]
}
+if (!build_with_mozilla) {
if (!build_with_chromium) {
if (is_mac || is_ios) {
rtc_library("video_test_mac") {
@@ -333,8 +334,12 @@ if (!build_with_chromium) {
}
}
}
+}
rtc_library("rtp_test_utils") {
+ if (build_with_mozilla) {
+ sources = []
+ } else {
testonly = true
sources = [
"rtcp_packet_parser.cc",
@@ -344,6 +349,7 @@ rtc_library("rtp_test_utils") {
"rtp_file_writer.cc",
"rtp_file_writer.h",
]
+ }
deps = [
"../api:array_view",
@@ -523,7 +529,9 @@ 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" ]
@@ -1372,6 +1380,7 @@ if (!build_with_chromium) {
}
}
+if (!build_with_mozilla) {
if (!build_with_chromium && is_android) {
rtc_android_library("native_test_java") {
testonly = true
@@ -1414,6 +1423,7 @@ 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 9973cdfa4d..4bd70867d5 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",
@@ -443,7 +443,7 @@ rtc_library("video_stream_encoder_impl") {
"../api:rtc_error",
"../api:rtp_packet_info",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api/adaptation:resource_adaptation_api",
diff --git a/webrtc.gni b/webrtc.gni
index fd7e9775b1..9e968a4114 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.
@@ -79,7 +89,7 @@ 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
@@ -102,7 +112,7 @@ 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.
@@ -116,13 +126,13 @@ 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
@@ -133,9 +143,6 @@ 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
@@ -277,7 +284,7 @@ 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
@@ -286,7 +293,7 @@ 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.
@@ -332,6 +339,9 @@ 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"
@@ -1231,7 +1241,7 @@ 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,