Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 50819f1..569d214 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -1569,8 +1569,11 @@
":isac_fix",
":webrtc_opus",
"../..:webrtc_common",
+ "../../api:libjingle_peerconnection_api",
"../../rtc_base:rtc_base_approved",
+ "../../system_wrappers:metrics_default",
"../../system_wrappers:system_wrappers_default",
+ "../../test:field_trial",
"../../test:test_main",
"../audio_processing",
"//testing/gtest",
@@ -1702,6 +1705,8 @@
"../..:webrtc_common",
"../../common_audio",
"../../rtc_base:rtc_base_approved",
+ "../../system_wrappers:metrics_default",
+ "../../test:field_trial",
]
configs += [ ":RTPencode_config" ]