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Unified Diff: webrtc/api/BUILD.gn

Issue 2059703002: Move webrtc/audio_*.h to webrtc/api/call (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changed to webrtc/api/call Created 4 years, 6 months ago
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Index: webrtc/api/BUILD.gn
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index cb3066eae69a3b0a580e4658a14b6cf76044d799..6df9f3f391542adc607bfd66d3e9724a9dbaeb0f 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -15,6 +15,25 @@ group("api") {
]
}
+source_set("call_api") {
+ sources = [
+ "call/audio_receive_stream.h",
+ "call/audio_send_stream.h",
+ "call/audio_sink.h",
+ "call/audio_state.h",
+ ]
+
+ configs += [ "..:common_config" ]
+ public_configs = [ "..:common_inherited_config" ]
+
+ deps = [
+ # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+ "..:webrtc_common",
+ "../base:rtc_base_approved",
+ "../modules/audio_coding:audio_encoder_interface",
+ ]
+}
+
config("libjingle_peerconnection_warnings_config") {
# GN orders flags on a target before flags from configs. The default config
# adds these flags so to cancel them out they need to come from a config and
@@ -115,6 +134,7 @@ source_set("libjingle_peerconnection") {
}
deps = [
+ ":call_api",
"../call",
"../media",
"../pc",
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