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From: Dan Minor <dminor@mozilla.com>
Date: Thu, 5 Nov 2020 07:47:00 +0000
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/127ace4d8887f11abb201d300a849772a2b519f8
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/88b3cc6bbece7c53d00e124713330f3d34d2789d
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
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/4a969f6709183d4f55215adaffb8a52b790a8492
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
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!
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fbbc1bf963fda30bca26ae6aac0c3459b8ebea6f
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.
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
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/08567f4539a12b54202aecbf554ec6540fb99ab2
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/136b3fc0377be6dcaa302469d27968f445e0355e
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7f403ee038e9797a1aff6161fc70a2d92769851f
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 | 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 +
common_audio/intrin.h | 8 ++++
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 ++++++++-----
31 files changed, 305 insertions(+), 92 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 d29e7e3ced..7a29809c83 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.
# Chromium fix resolves the problem.
diff --git a/BUILD.gn b/BUILD.gn
index 4ad7f3e00c..0e27d825e6 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" ]
}
@@ -500,9 +504,11 @@ config("common_config") {
}
}
+if (is_mac) {
config("common_objc") {
frameworks = [ "Foundation.framework" ]
}
+}
if (!rtc_build_ssl) {
config("external_ssl_library") {
@@ -566,6 +572,34 @@ 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 += [
@@ -578,6 +612,16 @@ 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
+ 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 d7d103de5b..b76d8cc965 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") {
@@ -70,7 +73,7 @@ rtc_library("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" ]
@@ -225,6 +228,10 @@ 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 = [
@@ -241,6 +248,7 @@ rtc_library("dtls_transport_interface") {
"//third_party/abseil-cpp/absl/base:core_headers",
]
}
+}
rtc_library("dtmf_sender_interface") {
visibility = [ "*" ]
@@ -253,6 +261,10 @@ 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 = [
@@ -267,6 +279,7 @@ 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",
@@ -275,8 +288,23 @@ 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 = [
@@ -400,6 +428,7 @@ rtc_library("libjingle_peerconnection_api") {
"../rtc_base/system:rtc_export",
]
}
+}
rtc_library("frame_transformer_interface") {
visibility = [ "*" ]
@@ -607,6 +636,7 @@ 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" ]
@@ -651,6 +681,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("frame_generator_api") {
visibility = [ "*" ]
@@ -773,6 +804,7 @@ rtc_library("create_frame_generator") {
]
}
+if (!build_with_mozilla) {
rtc_library("create_peer_connection_quality_test_frame_generator") {
visibility = [ "*" ]
testonly = true
@@ -790,6 +822,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
"units:time_delta",
]
}
+}
rtc_source_set("data_channel_event_observer_interface") {
visibility = [ "*" ]
@@ -978,6 +1011,7 @@ rtc_source_set("refcountedbase") {
]
}
+if (!build_with_mozilla) {
rtc_library("ice_transport_factory") {
visibility = [ "*" ]
sources = [
@@ -1002,6 +1036,7 @@ 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 478e15d570..2b3f7d9cac 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 11b03496be..3cc8835a5e 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -108,6 +108,7 @@ rtc_source_set("sctp_transport_factory_interface") {
}
rtc_library("stun_types") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"stun.cc",
@@ -130,6 +131,7 @@ 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 a70f751d71..0cf0e2fcf0 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 d1b3e64ba3..fa465f7d89 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 906d79598b..2c46bb4641 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/units/data_rate.h"
#include "api/video/video_content_type.h"
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index d40cf4e2b5..0dd84213a9 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -268,14 +268,10 @@ 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 47782316a6..d918ec7d13 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",
@@ -129,6 +129,12 @@ 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") {
@@ -155,6 +161,9 @@ 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") {
@@ -261,9 +270,16 @@ 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",
@@ -311,6 +327,7 @@ rtc_library("media_channel_impl") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
@@ -408,6 +425,7 @@ rtc_library("codec_list") {
}
rtc_library("rtp_utils") {
+if (!build_with_mozilla) {
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
@@ -424,8 +442,10 @@ 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",
@@ -439,6 +459,7 @@ rtc_library("stream_params") {
"//third_party/abseil-cpp/absl/algorithm:container",
]
}
+}
rtc_library("media_constants") {
sources = [
@@ -449,6 +470,7 @@ rtc_library("media_constants") {
}
rtc_library("turn_utils") {
+if (!build_with_mozilla) {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
@@ -459,14 +481,17 @@ 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 = [ "*" ]
@@ -547,6 +572,11 @@ 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" ]
@@ -566,6 +596,13 @@ 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",
+ ]
+ }
}
rtc_library("rtc_audio_video") {
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index b1f026372a..3eefc07cd9 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -67,9 +67,6 @@ 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 b88007703c..a33efe7f8d 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 3a8d0db866..13c3258e8f 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -364,7 +364,7 @@ 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 9bf81c315f..782bd7c99f 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") {
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") {
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 = [
@@ -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") {
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) {
}
}
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
@@ -443,8 +450,10 @@ rtc_source_set("mock_audio_device") {
"../../test:test_support",
]
}
+}
-if (rtc_include_tests && !build_with_chromium) {
+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 a0bda5f494..65aa6a9542 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -261,14 +261,11 @@ 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 9f720a5abf..75d94dd704 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 ddec89972d..154ac79205 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 a49c24fe46..9b1b39ac3e 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -345,37 +345,12 @@ 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) {
@@ -538,9 +513,7 @@ 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 b30cb0c6d4..b59f8b5b2c 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -25,5 +25,9 @@ rtc_library("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 c3e0e5af71..1f17788573 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -132,21 +132,12 @@ 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 794e8c2725..848ff98f20 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -325,6 +325,7 @@ rtc_library("sample_counter") {
]
}
rtc_library("timestamp_aligner") {
visibility = [ "*" ]
sources = [
@@ -338,6 +339,7 @@ rtc_library("timestamp_aligner") {
"system:rtc_export",
]
}
+}
rtc_library("zero_memory") {
visibility = [ "*" ]
@@ -807,7 +809,9 @@ 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 {
@@ -1171,6 +1175,7 @@ if (!build_with_chromium) {
}
rtc_library("network") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"network.cc",
@@ -1212,16 +1217,20 @@ 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",
@@ -1232,8 +1241,10 @@ 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",
@@ -1254,6 +1265,7 @@ rtc_library("socket_adapters") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("network_route") {
sources = [
@@ -1268,6 +1280,7 @@ rtc_library("network_route") {
}
rtc_library("async_tcp_socket") {
+if (!build_with_mozilla) {
sources = [
"async_tcp_socket.cc",
"async_tcp_socket.h",
@@ -1285,8 +1298,10 @@ rtc_library("async_tcp_socket") {
"network:sent_packet",
]
}
+}
rtc_library("async_udp_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_udp_socket.cc",
@@ -1310,8 +1325,10 @@ rtc_library("async_udp_socket") {
"system:no_unique_address",
]
}
+}
rtc_library("async_packet_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_packet_socket.cc",
@@ -1334,6 +1351,7 @@ rtc_library("async_packet_socket") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
if (rtc_include_tests) {
rtc_library("async_packet_socket_unittest") {
@@ -1406,6 +1424,7 @@ rtc_library("data_rate_limiter") {
}
rtc_library("unique_id_generator") {
+if (!build_with_mozilla) {
sources = [
"unique_id_generator.cc",
"unique_id_generator.h",
@@ -1420,6 +1439,7 @@ rtc_library("unique_id_generator") {
"//third_party/abseil-cpp/absl/strings:string_view",
]
}
+}
rtc_library("crc32") {
sources = [
@@ -1454,6 +1474,7 @@ rtc_library("stream") {
}
rtc_library("rtc_certificate_generator") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"rtc_certificate_generator.cc",
@@ -1468,6 +1489,7 @@ rtc_library("rtc_certificate_generator") {
"//third_party/abseil-cpp/absl/functional:any_invocable",
]
}
+}
rtc_source_set("ssl_header") {
visibility = [ "*" ]
@@ -1524,6 +1546,7 @@ rtc_library("crypto_random") {
}
rtc_library("ssl") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"openssl_key_pair.cc",
@@ -1597,6 +1620,7 @@ rtc_library("ssl") {
deps += [ ":win32" ]
}
}
+}
rtc_library("ssl_adapter") {
visibility = [ "*" ]
@@ -2340,7 +2364,7 @@ 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 85e65b3785..6173ab3fc5 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -232,6 +232,7 @@ rtc_library("audio_test_common") {
]
}
+if (!build_with_mozilla) {
if (!build_with_chromium) {
if (is_mac || is_ios) {
rtc_library("video_test_mac") {
@@ -285,8 +286,12 @@ if (!build_with_chromium) {
}
}
}
+}
rtc_library("rtp_test_utils") {
+ if (build_with_mozilla) {
+ sources = []
+ } else {
testonly = true
sources = [
"rtcp_packet_parser.cc",
@@ -296,6 +301,7 @@ rtc_library("rtp_test_utils") {
"rtp_file_writer.cc",
"rtp_file_writer.h",
]
+ }
deps = [
"../api:array_view",
@@ -511,7 +517,9 @@ 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" ]
@@ -1313,6 +1321,7 @@ if (!build_with_chromium) {
}
}
+if (!build_with_mozilla) {
if (!build_with_chromium && is_android) {
rtc_android_library("native_test_java") {
testonly = true
@@ -1355,6 +1364,7 @@ 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 bf77b139ca..32200bb360 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 4fed37040d..ec55e54b9d 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
@@ -295,7 +302,7 @@ 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
@@ -304,7 +311,7 @@ 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.
@@ -347,6 +354,9 @@ 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"
@@ -1220,7 +1230,7 @@ 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,