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" ]