From a0e1a55dc9e3a8945cf307255eb559aad0e1871d Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Mon, 4 Dec 2017 10:50:51 +0100
Subject: [PATCH] Stop using public_deps in the call module.
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

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    | 7 +------
 logging/BUILD.gn | 2 ++
 media/BUILD.gn   | 1 +
 test/BUILD.gn    | 2 ++
 4 files changed, 6 insertions(+), 6 deletions(-)

diff --git a/call/BUILD.gn b/call/BUILD.gn
index 0b83d117e0..280d19e3c6 100644
--- a/call/BUILD.gn
+++ b/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",
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index aa8aadba07..9588778a7a 100644
--- a/logging/BUILD.gn
+++ b/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",
diff --git a/media/BUILD.gn b/media/BUILD.gn
index b78f55da49..1ac5485b57 100644
--- a/media/BUILD.gn
+++ b/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 = [
diff --git a/test/BUILD.gn b/test/BUILD.gn
index e54228dd45..efb1978201 100644
--- a/test/BUILD.gn
+++ b/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",