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trisquel-icecat/icecat/third_party/libwebrtc/moz-patch-stack/0030.patch

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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 IceCat 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 | 36 ++++++++++++++-
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 | 14 +++++-
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 | 30 +++++++++++-
rtc_base/system/BUILD.gn | 2 +-
test/BUILD.gn | 10 ++++
video/BUILD.gn | 4 +-
webrtc.gni | 32 ++++++++-----
31 files changed, 319 insertions(+), 98 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 b886ea86aa..19a4b72c2a 100644
--- a/.gn
+++ b/.gn
@@ -73,6 +73,8 @@ 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 571049f3e4..f393179bbb 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") {
@@ -158,6 +158,10 @@ 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" ]
}
@@ -480,9 +484,11 @@ config("common_config") {
}
}
+if (is_mac) {
config("common_objc") {
frameworks = [ "Foundation.framework" ]
}
+}
if (!build_with_chromium) {
# Target to build all the WebRTC production code.
@@ -533,6 +539,34 @@ 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 += [
@@ -545,6 +579,16 @@ 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 16a55f36b4..6b68cfc9ff 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -40,6 +40,9 @@ rtc_source_set("enable_media") {
"../rtc_base/system:rtc_export",
"environment",
]
+ if (build_with_mozilla) {
+ deps -= [ "../pc:media_factory" ]
+ }
}
rtc_source_set("enable_media_with_defaults") {
@@ -66,7 +69,7 @@ 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" ]
@@ -221,6 +224,10 @@ rtc_source_set("ice_transport_interface") {
}
rtc_library("dtls_transport_interface") {
+# Previously, GNU 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 = [
@@ -237,6 +244,7 @@ rtc_library("dtls_transport_interface") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("dtmf_sender_interface") {
visibility = [ "*" ]
@@ -249,6 +257,10 @@ rtc_library("dtmf_sender_interface") {
}
rtc_library("rtp_sender_interface") {
+# Previously, GNU 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 = [
@@ -263,16 +275,31 @@ 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",
"crypto:frame_encryptor_interface",
"video_codecs:video_codecs_api",
]
+}
+}
+
+rtc_library("rtp_sender_setparameters_callback") {
+ visibility = [ "*" ]
+
+ sources = [
+ "rtp_sender_setparameters_callback.cc",
+ "rtp_sender_setparameters_callback.h",
+ ]
+ deps = [
+ ":rtc_error",
+ ]
absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
}
rtc_library("libjingle_peerconnection_api") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
cflags = []
sources = [
@@ -391,6 +418,7 @@ rtc_library("libjingle_peerconnection_api") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_source_set("frame_transformer_interface") {
visibility = [ "*" ]
@@ -564,6 +592,7 @@ 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" ]
@@ -612,6 +641,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_source_set("frame_generator_api") {
visibility = [ "*" ]
@@ -731,6 +761,7 @@ rtc_library("create_frame_generator") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+if (!build_with_mozilla) {
rtc_library("create_peer_connection_quality_test_frame_generator") {
visibility = [ "*" ]
testonly = true
@@ -747,6 +778,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_source_set("libjingle_logging_api") {
visibility = [ "*" ]
@@ -926,6 +958,7 @@ rtc_source_set("refcountedbase") {
]
}
+if (!build_with_mozilla) {
rtc_library("ice_transport_factory") {
visibility = [ "*" ]
sources = [
@@ -949,6 +982,7 @@ 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 7b8ab135c8..7090c233cd 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -32,9 +32,9 @@
#include "api/video_codecs/video_encoder_factory.h"
#include "rtc_base/system/rtc_export.h"
-namespace webrtc {
+#include "api/rtp_sender_setparameters_callback.h"
-using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
+namespace webrtc {
class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface {
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 d2e96de4d0..8697f87433 100644
--- a/api/task_queue/BUILD.gn
+++ b/api/task_queue/BUILD.gn
@@ -31,6 +31,7 @@ rtc_library("task_queue") {
]
}
+if (rtc_include_tests) {
rtc_library("task_queue_test") {
visibility = [ "*" ]
testonly = true
@@ -79,6 +80,7 @@ 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 c06234e612..ea6cb0efb5 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -98,6 +98,7 @@ rtc_source_set("sctp_transport_factory_interface") {
}
rtc_source_set("stun_types") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"stun.cc",
@@ -118,6 +119,7 @@ rtc_source_set("stun_types") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
if (rtc_include_tests) {
rtc_source_set("test_feedback_generator_interface") {
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 57c6f3a516..a9d69b93ef 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -44,7 +44,7 @@ rtc_library("call_interfaces") {
"../api:network_state_predictor_api",
"../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",
@@ -343,6 +343,16 @@ rtc_library("call") {
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
+ if (build_with_mozilla) { # See Bug 1820869.
+ sources -= [
+ "create_call.cc",
+ "degraded_call.cc",
+ ]
+ deps -= [
+ ":fake_network",
+ ":simulated_network",
+ ]
+ }
}
rtc_source_set("receive_stream_interface") {
@@ -370,7 +380,7 @@ rtc_library("video_stream_api") {
"../api:frame_transformer_interface",
"../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 9c2fad652f..f9e49db574 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -25,7 +25,7 @@
#include "api/crypto/frame_encryptor_interface.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 "call/audio_sender.h"
#include "call/rtp_config.h"
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 2c3fe55f8d..ca900823c8 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 2ae6d32710..a45214f754 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 a1a3603bc7..4a715821cf 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -79,7 +79,7 @@ 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",
@@ -138,6 +138,12 @@ rtc_library("rtc_media_base") {
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
+ if (build_with_mozilla) {
+ sources -= [
+ "base/adapted_video_track_source.cc",
+ "base/adapted_video_track_source.h",
+ ]
+ }
}
rtc_library("adapted_video_track_source") {
@@ -163,6 +169,9 @@ 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" ]
+ }
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
deps = []
}
@@ -261,9 +270,16 @@ rtc_library("media_engine") {
"../rtc_base/system:file_wrapper",
]
absl_deps = [ "//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/types:optional",
]
}
+}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
@@ -390,6 +407,7 @@ rtc_library("codec") {
}
rtc_library("rtp_utils") {
+if (!build_with_mozilla) {
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
@@ -406,8 +424,10 @@ rtc_library("rtp_utils") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("stream_params") {
+if (!build_with_mozilla) {
sources = [
"base/stream_params.cc",
"base/stream_params.h",
@@ -420,6 +440,7 @@ rtc_library("stream_params") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
}
+}
rtc_library("media_constants") {
sources = [
@@ -430,6 +451,7 @@ rtc_library("media_constants") {
}
rtc_library("turn_utils") {
+if (!build_with_mozilla) {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
@@ -440,14 +462,17 @@ 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 = [ "*" ]
@@ -523,6 +548,11 @@ rtc_library("rtc_internal_video_codecs") {
"../system_wrappers:field_trial",
"../test:fake_video_codecs",
]
+ if (build_with_mozilla) {
+ deps -= [
+ "../test:fake_video_codecs",
+ ]
+ }
if (enable_libaom) {
defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
@@ -546,6 +576,13 @@ 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 f941823323..1fe86f9588 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -65,10 +65,6 @@ 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 1c08382969..ff69ea62dc 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 8b23955d5b..a49df7e7d2 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -553,7 +553,7 @@ 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 359867e50c..711b62006c 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 = [ "*" ]
@@ -63,6 +65,7 @@ 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",
@@ -89,6 +92,7 @@ rtc_library("audio_device_buffer") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("audio_device_generic") {
sources = [
@@ -265,6 +269,7 @@ 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_api",
@@ -313,9 +318,9 @@ 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",
]
}
@@ -420,6 +425,7 @@ rtc_library("audio_device_impl") {
sources += [ "dummy/file_audio_device_factory.h" ]
}
}
+}
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
@@ -437,6 +443,7 @@ if (is_mac) {
}
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
@@ -454,8 +461,10 @@ rtc_source_set("mock_audio_device") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
-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 f5eb5d5951..3e11a245a1 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -264,14 +264,11 @@ 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 9093a68cf3..3003a585bd 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -122,7 +122,7 @@ 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 fcf1836930..344c8a1531 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -343,37 +343,12 @@ 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) {
@@ -539,9 +514,7 @@ 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 d0756f269b..d7768b2323 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 b8345841fa..7289dbe0a8 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -125,21 +125,12 @@ 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 abc8916e29..db3481e694 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -327,6 +327,7 @@ rtc_library("sample_counter") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_library("timestamp_aligner") {
visibility = [ "*" ]
sources = [
@@ -340,6 +341,7 @@ rtc_library("timestamp_aligner") {
"system:rtc_export",
]
}
+}
rtc_library("zero_memory") {
visibility = [ "*" ]
@@ -858,7 +860,9 @@ rtc_library("rtc_json") {
"strings/json.h",
]
deps = [ ":stringutils" ]
+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",
@@ -1250,16 +1255,20 @@ 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",
@@ -1268,8 +1277,10 @@ rtc_library("net_helper") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
deps = [ "system:rtc_export" ]
}
+}
rtc_library("socket_adapters") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"socket_adapters.cc",
@@ -1289,6 +1300,7 @@ rtc_library("socket_adapters") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("network_route") {
sources = [
@@ -1303,6 +1315,7 @@ rtc_library("network_route") {
}
rtc_library("async_tcp_socket") {
+if (!build_with_mozilla) {
sources = [
"async_tcp_socket.cc",
"async_tcp_socket.h",
@@ -1320,8 +1333,10 @@ rtc_library("async_tcp_socket") {
"network:sent_packet",
]
}
+}
rtc_library("async_udp_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_udp_socket.cc",
@@ -1345,8 +1360,10 @@ rtc_library("async_udp_socket") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("async_packet_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_packet_socket.cc",
@@ -1366,6 +1383,7 @@ rtc_library("async_packet_socket") {
"third_party/sigslot",
]
}
+}
if (rtc_include_tests) {
rtc_library("async_packet_socket_unittest") {
@@ -1394,6 +1412,7 @@ rtc_library("dscp") {
}
rtc_library("proxy_info") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"proxy_info.cc",
@@ -1404,6 +1423,7 @@ rtc_library("proxy_info") {
":socket_address",
]
}
+}
rtc_library("file_rotating_stream") {
sources = [
@@ -1432,6 +1452,7 @@ rtc_library("data_rate_limiter") {
}
rtc_library("unique_id_generator") {
+if (!build_with_mozilla) {
sources = [
"unique_id_generator.cc",
"unique_id_generator.h",
@@ -1446,6 +1467,7 @@ rtc_library("unique_id_generator") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("crc32") {
sources = [
@@ -1473,6 +1495,7 @@ rtc_library("stream") {
}
rtc_library("rtc_certificate_generator") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"rtc_certificate_generator.cc",
@@ -1490,8 +1513,10 @@ rtc_library("rtc_certificate_generator") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_library("ssl") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"helpers.cc",
@@ -1592,6 +1617,7 @@ rtc_library("ssl") {
deps += [ ":win32" ]
}
}
+}
rtc_library("crypt_string") {
sources = [
@@ -1601,6 +1627,7 @@ rtc_library("crypt_string") {
}
rtc_library("http_common") {
+if (!build_with_mozilla) {
sources = [
"http_common.cc",
"http_common.h",
@@ -1617,6 +1644,7 @@ rtc_library("http_common") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_source_set("gtest_prod") {
sources = [ "gtest_prod_util.h" ]
@@ -2176,7 +2204,7 @@ 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 77f5139a2f..486b37590c 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 d98ff0a33d..6151a1a487 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -240,6 +240,7 @@ rtc_library("audio_test_common") {
absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
}
+if (!build_with_mozilla) {
if (!build_with_chromium) {
if (is_mac || is_ios) {
rtc_library("video_test_mac") {
@@ -293,8 +294,12 @@ if (!build_with_chromium) {
}
}
}
+}
rtc_library("rtp_test_utils") {
+ if (build_with_mozilla) {
+ sources = []
+ } else {
testonly = true
sources = [
"rtcp_packet_parser.cc",
@@ -304,6 +309,7 @@ rtc_library("rtp_test_utils") {
"rtp_file_writer.cc",
"rtp_file_writer.h",
]
+ }
deps = [
"../api:array_view",
@@ -558,7 +564,9 @@ rtc_library("video_test_support") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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" ]
@@ -1312,6 +1320,7 @@ if (!build_with_chromium) {
}
}
+if (!build_with_mozilla) {
if (!build_with_chromium && is_android) {
rtc_android_library("native_test_java") {
testonly = true
@@ -1325,6 +1334,7 @@ if (!build_with_chromium && is_android) {
]
}
}
+}
rtc_library("call_config_utils") {
testonly = true
diff --git a/video/BUILD.gn b/video/BUILD.gn
index eeabef8b77..7c49ceb849 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",
@@ -410,7 +410,7 @@ 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 277c12c054..c84464e84a 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")
}
@@ -42,6 +47,11 @@ if (is_fuchsia) {
# 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 GNU 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
@@ -118,7 +128,7 @@ 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.
@@ -135,13 +145,13 @@ declare_args() {
rtc_enable_bwe_test_logging = 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
@@ -152,9 +162,6 @@ declare_args() {
# Set this to link PipeWire and required libraries directly instead of using the dlopen.
rtc_link_pipewire = false
- # Enable to use the GNU 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
@@ -280,7 +287,7 @@ 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
@@ -299,7 +306,7 @@ 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.
@@ -343,6 +350,9 @@ 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"
@@ -1146,7 +1156,7 @@ 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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