Stop using public_deps in the call module.

Bug: webrtc:8603
Change-Id: I048127bc86f213e638e6814ac8a86761cb8a64db
Reviewed-on: https://webrtc-review.googlesource.com/28624
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21072}

call/BUILD.gn

@@ -121,12 +121,6 @@ rtc_static_library("call") {
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
-  public_deps = [
-    ":call_interfaces",
-    "../api:call_api",
-    "../api:libjingle_peerconnection_api",
-  ]
-
   deps = [
     ":bitrate_allocator",
     ":call_interfaces",
@@ -196,6 +190,7 @@ if (rtc_include_tests) {
     deps = [
       ":bitrate_allocator",
       ":call",
+      ":call_interfaces",
       ":mock_rtp_interfaces",
       ":rtp_interfaces",
       ":rtp_receiver",

logging/BUILD.gn

@@ -177,7 +177,9 @@ if (rtc_enable_protobuf) {
       deps = [
         ":rtc_event_log_impl",
         ":rtc_event_log_parser",
+        "../api:libjingle_peerconnection_api",
         "../call",
+        "../call:call_interfaces",
         "../modules/audio_coding:audio_network_adaptor",
         "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
         "../modules/rtp_rtcp",

media/BUILD.gn

@@ -128,6 +128,7 @@ rtc_static_library("rtc_audio_video") {
   libs = []
   deps = [
     "../api:video_frame_api_i420",
+    "../call:call_interfaces",
     "../modules/video_coding:video_coding_utility",
   ]
   sources = [

test/BUILD.gn

@@ -465,6 +465,7 @@ rtc_source_set("direct_transport") {
     "..:webrtc_common",
     "../api:transport_api",
     "../call",
+    "../call:call_interfaces",
     "../modules/rtp_rtcp",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:sequenced_task_checker",
@@ -563,6 +564,7 @@ rtc_source_set("test_common") {
     "../api/video_codecs:video_codecs_api",
     "../audio",
     "../call",
+    "../call:call_interfaces",
     "../call:rtp_sender",
     "../call:video_stream_api",
     "../common_video",