Move RtpExtension to api/ directory and config.h/.cc to call/.

webrtc/modules/audio_coding/BUILD.gn

Index: webrtc/modules/audio_coding/BUILD.gn
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index a3964c90804e9aa0618e65f7dfe0a6fc3887bde1..449655da7555df8ecec9094beeceb6d976280170 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -1389,6 +1389,7 @@ if (rtc_include_tests) {
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
@@ -1419,6 +1420,7 @@ if (rtc_include_tests) {
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
@@ -1466,6 +1468,7 @@ if (rtc_include_tests) {
       ":neteq_tools",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs/opus:audio_encoder_opus",
+      "../../call:call",
       "../../common_audio",
       "../../rtc_base:protobuf_utils",
       "../../test:test_main",
@@ -1517,6 +1520,7 @@ if (rtc_include_tests) {
         ":neteq",
         ":neteq_test_tools",
         "../..:webrtc_common",
+        "../../call:call",
         "../../rtc_base:rtc_base_approved",
         "../../system_wrappers:system_wrappers_default",
         "../../test:test_support",
@@ -1572,8 +1576,12 @@ if (rtc_include_tests) {
       ":isac_fix",
       ":webrtc_opus",
       "../..:webrtc_common",
+      "../../api:libjingle_peerconnection_api",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
+      "../../system_wrappers:metrics_default",
       "../../system_wrappers:system_wrappers_default",
+      "../../test:field_trial",
       "../../test:test_main",
       "../audio_processing",
       "//testing/gtest",
@@ -1704,8 +1712,12 @@ if (rtc_include_tests) {
       ":pcm16b",
       ":webrtc_opus",
       "../..:webrtc_common",
+      "../../api:libjingle_peerconnection_api",
+      "../../call:call",
       "../../common_audio",
       "../../rtc_base:rtc_base_approved",
+      "../../system_wrappers:metrics_default",
+      "../../test:field_trial",
     ]
 
     configs += [ ":RTPencode_config" ]
@@ -1748,6 +1760,7 @@ if (rtc_include_tests) {
     ]
 
     deps = [
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:rtp_test_utils",
@@ -1793,6 +1806,7 @@ if (rtc_include_tests) {
       ":neteq",
       ":neteq_test_tools",
       ":pcm16b",
+      "../../call:call",
       "../../system_wrappers:system_wrappers_default",
       "//testing/gtest",
       "//third_party/gflags:gflags",
@@ -1816,6 +1830,7 @@ if (rtc_include_tests) {
       ":neteq_quality_test_support",
       ":neteq_tools",
       ":webrtc_opus",
+      "../../call:call",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
@@ -1833,6 +1848,7 @@ if (rtc_include_tests) {
       ":neteq",
       ":neteq_test_support",
       "../..:webrtc_common",
+      "../../call:call",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "//third_party/gflags",
@@ -1852,6 +1868,7 @@ if (rtc_include_tests) {
       ":neteq_quality_test_support",
       ":neteq_tools",
       "../..:webrtc_common",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_main",
@@ -1871,6 +1888,7 @@ if (rtc_include_tests) {
       ":isac_fix",
       ":neteq",
       ":neteq_quality_test_support",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
@@ -1889,6 +1907,7 @@ if (rtc_include_tests) {
       ":g711",
       ":neteq",
       ":neteq_quality_test_support",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
@@ -2041,6 +2060,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":webrtc_opus",
+      "../../call:call",
       "../../common_audio",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_main",