Refactor webrtc/modules/audio_processing for GN check
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/audio_processing/*",
BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2558813003
Cr-Commit-Position: refs/heads/master@{#15475}
diff --git a/.gn b/.gn
index c44ccc8..cdb8794 100644
--- a/.gn
+++ b/.gn
@@ -27,6 +27,7 @@
"//webrtc/modules/audio_coding/*",
"//webrtc/modules/audio_device/*",
"//webrtc/modules/audio_mixer/*",
+ "//webrtc/modules/audio_processing/*",
"//webrtc/stats:rtc_stats",
"//webrtc/voice_engine",
"//webrtc/voice_engine:level_indicator",
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index 86d5c21..7a64845 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -228,42 +228,6 @@
return result
-def _CheckNoRtcBaseDeps(input_api, gn_files, output_api):
- pattern = input_api.re.compile(r'base:rtc_base\s*"')
- violating_files = []
- for f in gn_files:
- gn_exceptions = (
- os.path.join('audio_device', 'BUILD.gn'),
- os.path.join('base_tests', 'BUILD.gn'),
- os.path.join('desktop_capture', 'BUILD.gn'),
- os.path.join('p2p', 'BUILD.gn'),
- os.path.join('sdk', 'BUILD.gn'),
- os.path.join('webrtc_test_common', 'BUILD.gn'),
- os.path.join('webrtc_tests', 'BUILD.gn'),
-
- # TODO(ehmaldonado): Clean up references to rtc_base in these files.
- # See https://bugs.chromium.org/p/webrtc/issues/detail?id=3806
- os.path.join('webrtc', 'BUILD.gn'),
- os.path.join('xmllite', 'BUILD.gn'),
- os.path.join('xmpp', 'BUILD.gn'),
- os.path.join('modules', 'BUILD.gn'),
- os.path.join('audio_device', 'BUILD.gn'),
- os.path.join('pc', 'BUILD.gn'),
- )
- if f.LocalPath().endswith(gn_exceptions):
- continue
- contents = input_api.ReadFile(f)
- if pattern.search(contents):
- violating_files.append(f)
- if violating_files:
- return [output_api.PresubmitError(
- 'Depending on rtc_base is not allowed. Change your dependency to '
- 'rtc_base_approved and possibly sanitize and move the desired source '
- 'file(s) to rtc_base_approved.\nChanged GN files:',
- items=violating_files)]
- return []
-
-
def _CheckNoSourcesAbove(input_api, gn_files, output_api):
# Disallow referencing source files with paths above the GN file location.
source_pattern = input_api.re.compile(r' +sources \+?= \[(.*?)\]',
@@ -332,7 +296,6 @@
result = []
if gn_files:
- result.extend(_CheckNoRtcBaseDeps(input_api, gn_files, output_api))
result.extend(_CheckNoSourcesAbove(input_api, gn_files, output_api))
result.extend(_CheckNoMixingCAndCCSources(input_api, gn_files, output_api))
return result
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 5ba9f3e..6ad7775 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -161,6 +161,7 @@
deps = [
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
+ "../../base:gtest_prod",
"../audio_coding:isac",
]
public_deps = [
@@ -265,6 +266,7 @@
}
deps = [
+ "../..:webrtc_common",
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers",
@@ -283,6 +285,10 @@
if (current_cpu == "x86" || current_cpu == "x64") {
rtc_static_library("audio_processing_sse2") {
+ # TODO(ehmaldonado): Remove (bugs.webrtc.org/6828)
+ # Errors on cyclic dependency with :audio_processing if enabled.
+ check_includes = false
+
sources = [
"aec/aec_core_sse2.cc",
"utility/ooura_fft_sse2.cc",
@@ -303,6 +309,10 @@
if (rtc_build_with_neon) {
rtc_static_library("audio_processing_neon") {
+ # TODO(ehmaldonado): Remove (bugs.webrtc.org/6828)
+ # Errors on cyclic dependency with :audio_processing if enabled.
+ check_includes = false
+
sources = [
"aec/aec_core_neon.cc",
"aecm/aecm_core_neon.cc",
@@ -362,6 +372,10 @@
"-ffat-lto-objects",
]
}
+ deps = [
+ ":audio_processing_c",
+ "../../base:rtc_base_approved",
+ ]
}
}
@@ -397,6 +411,8 @@
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
+ "../..:webrtc_common",
+ "../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers:system_wrappers_default",
"//third_party/gflags:gflags",
@@ -420,6 +436,8 @@
":audioproc_debug_proto",
":audioproc_protobuf_utils",
":audioproc_test_utils",
+ "../../base:rtc_base_approved",
+ "../../common_audio:common_audio",
"../../system_wrappers",
"../../system_wrappers:system_wrappers_default",
"../../test:test_support",
@@ -442,8 +460,10 @@
]
deps = [
+ ":audio_processing",
"../../base:rtc_base_approved",
"../../common_audio",
+ "../../system_wrappers:system_wrappers",
]
}
@@ -456,7 +476,10 @@
]
deps = [
":audio_processing",
+ "../..:webrtc_common",
+ "../../common_audio:common_audio",
"../../system_wrappers:metrics_default",
+ "../../system_wrappers:system_wrappers",
"../../test:test_support",
"//testing/gtest",
"//third_party/gflags",
@@ -472,7 +495,9 @@
]
deps = [
":audio_processing",
+ "../..:webrtc_common",
"../../system_wrappers:metrics_default",
+ "../../system_wrappers:system_wrappers",
]
}
@@ -484,6 +509,8 @@
deps = [
":audio_processing",
":audioproc_test_utils",
+ "../../base:rtc_base_approved",
+ "../../common_audio:common_audio",
"../../system_wrappers:metrics_default",
"//third_party/gflags",
]
@@ -522,6 +549,8 @@
deps = [
":audioproc_debug_proto",
+ "../..:webrtc_common",
+ "../../base:rtc_base_approved",
]
}
}