Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ )
Reason for revert:
Breaks lots of downstream projects.
Original issue's description:
> Remove remains of webrtc/base
>
> All downstream code have been updated to the new location.
>
> In PRESUBMIT.py:
> * Remove webrtc/rtc_base from CPP_BLACKLIST
> * Add webrtc/rtc_base to LEGACY_API_DIRS
>
> Fix some duplicated paths in
> webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>
> BUG=webrtc:7634
> TBR=kwiberg@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2973183002
> Cr-Commit-Position: refs/heads/master@{#18948}
> Committed:
https://chromium.googlesource.com/external/webrtc/+/9483b49bafc681a8360dff7217e7651a74dea71d
TBR=kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7634
Review-Url: https://codereview.webrtc.org/2976633002
Cr-Commit-Position: refs/heads/master@{#18949}
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index c413f28..249411b 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -28,7 +28,7 @@
":audio_mixer_api",
":transport_api",
"..:webrtc_common",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"audio_codecs:audio_codecs_api",
]
}
@@ -83,8 +83,8 @@
deps = [
":rtc_stats_api",
"..:webrtc_common",
- "../rtc_base:rtc_base",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base",
+ "../base:rtc_base_approved",
"audio_codecs:audio_codecs_api",
]
@@ -143,7 +143,7 @@
]
deps = [
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
@@ -153,8 +153,8 @@
]
deps = [
+ "../base:rtc_base_approved",
"../modules:module_api",
- "../rtc_base:rtc_base_approved",
]
}
@@ -178,7 +178,7 @@
]
deps = [
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
"../system_wrappers",
]
@@ -206,7 +206,7 @@
]
deps = [
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
}
@@ -235,7 +235,7 @@
]
deps = [
":libjingle_peerconnection_api",
- "../rtc_base:rtc_base_approved",
+ "../base:rtc_base_approved",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
diff --git a/webrtc/api/audio_codecs/BUILD.gn b/webrtc/api/audio_codecs/BUILD.gn
index 2174fb1..416ccbb 100644
--- a/webrtc/api/audio_codecs/BUILD.gn
+++ b/webrtc/api/audio_codecs/BUILD.gn
@@ -27,7 +27,7 @@
]
deps = [
"../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
+ "../../base:rtc_base_approved",
]
}
@@ -38,8 +38,8 @@
]
deps = [
":audio_codecs_api",
+ "../../base:rtc_base_approved",
"../../modules/audio_coding:builtin_audio_decoder_factory_internal",
- "../../rtc_base:rtc_base_approved",
]
}
@@ -50,7 +50,7 @@
]
deps = [
":audio_codecs_api",
+ "../../base:rtc_base_approved",
"../../modules/audio_coding:builtin_audio_encoder_factory_internal",
- "../../rtc_base:rtc_base_approved",
]
}
diff --git a/webrtc/api/audio_codecs/g722/BUILD.gn b/webrtc/api/audio_codecs/g722/BUILD.gn
index 2c1349a..d2470a2 100644
--- a/webrtc/api/audio_codecs/g722/BUILD.gn
+++ b/webrtc/api/audio_codecs/g722/BUILD.gn
@@ -26,8 +26,8 @@
deps = [
":audio_encoder_g722_config",
"..:audio_codecs_api",
+ "../../../base:rtc_base_approved",
"../../../modules/audio_coding:g722",
- "../../../rtc_base:rtc_base_approved",
]
}
@@ -39,7 +39,7 @@
deps = [
"..:audio_codecs_api",
"../../..:webrtc_common",
+ "../../../base:rtc_base_approved",
"../../../modules/audio_coding:g722",
- "../../../rtc_base:rtc_base_approved",
]
}
diff --git a/webrtc/api/audio_codecs/ilbc/BUILD.gn b/webrtc/api/audio_codecs/ilbc/BUILD.gn
index 6ef8856..bba2662 100644
--- a/webrtc/api/audio_codecs/ilbc/BUILD.gn
+++ b/webrtc/api/audio_codecs/ilbc/BUILD.gn
@@ -26,8 +26,8 @@
deps = [
":audio_encoder_ilbc_config",
"..:audio_codecs_api",
+ "../../../base:rtc_base_approved",
"../../../modules/audio_coding:ilbc",
- "../../../rtc_base:rtc_base_approved",
]
}
@@ -39,7 +39,7 @@
deps = [
"..:audio_codecs_api",
"../../..:webrtc_common",
+ "../../../base:rtc_base_approved",
"../../../modules/audio_coding:ilbc",
- "../../../rtc_base:rtc_base_approved",
]
}
diff --git a/webrtc/api/audio_codecs/opus/BUILD.gn b/webrtc/api/audio_codecs/opus/BUILD.gn
index 29a68ff..c7f7ac8 100644
--- a/webrtc/api/audio_codecs/opus/BUILD.gn
+++ b/webrtc/api/audio_codecs/opus/BUILD.gn
@@ -18,7 +18,7 @@
"audio_encoder_opus_config.h",
]
deps = [
- "../../../rtc_base:rtc_base_approved",
+ "../../../base:rtc_base_approved",
]
defines = []
if (rtc_opus_variable_complexity) {
@@ -35,9 +35,9 @@
deps = [
":audio_encoder_opus_config",
"..:audio_codecs_api",
+ "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed?
+ "../../../base:rtc_base_approved",
"../../../modules/audio_coding:webrtc_opus",
- "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed?
- "../../../rtc_base:rtc_base_approved",
]
}
@@ -49,7 +49,7 @@
deps = [
"..:audio_codecs_api",
"../../..:webrtc_common",
+ "../../../base:rtc_base_approved",
"../../../modules/audio_coding:webrtc_opus",
- "../../../rtc_base:rtc_base_approved",
]
}
diff --git a/webrtc/api/audio_codecs/test/BUILD.gn b/webrtc/api/audio_codecs/test/BUILD.gn
index 4a0c878..32cef2d 100644
--- a/webrtc/api/audio_codecs/test/BUILD.gn
+++ b/webrtc/api/audio_codecs/test/BUILD.gn
@@ -21,8 +21,8 @@
]
deps = [
"..:audio_codecs_api",
- "../../../rtc_base:protobuf_utils", # TODO(kwiberg): Why is this needed?
- "../../../rtc_base:rtc_base_approved",
+ "../../../base:protobuf_utils", # TODO(kwiberg): Why is this needed?
+ "../../../base:rtc_base_approved",
"../../../test:audio_codec_mocks",
"../../../test:test_support",
"../g722:audio_decoder_g722",
diff --git a/webrtc/api/video_codecs/BUILD.gn b/webrtc/api/video_codecs/BUILD.gn
index 5e27c78..d435534 100644
--- a/webrtc/api/video_codecs/BUILD.gn
+++ b/webrtc/api/video_codecs/BUILD.gn
@@ -21,7 +21,7 @@
deps = [
"..:video_frame_api",
"../..:webrtc_common",
+ "../../base:rtc_base_approved",
"../../common_video",
- "../../rtc_base:rtc_base_approved",
]
}