From 245660a33d9879d50c8270d298fafc1529cf9ba9 Mon Sep 17 00:00:00 2001
From: Mirko Bonadei <mbonadei@webrtc.org>
Date: Tue, 10 Oct 2017 14:38:11 +0200
Subject: [PATCH] Fix Gn untracked headers in webrtc/call.

This CL is the same CL we had at https://codereview.webrtc.org/3014543002/.
Since we cannot land it with Rietveld anymore let's move the discussion
to Gerrit.

BUG=webrtc:7641
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I5662bec318544b07f476c12ecada997d726e7361
Reviewed-on: https://webrtc-review.googlesource.com/7981
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20224}
---
 audio/BUILD.gn |  1 +
 call/BUILD.gn  | 22 +++++++++++++++++++++-
 video/BUILD.gn |  1 +
 3 files changed, 23 insertions(+), 1 deletion(-)

diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 6917713dfa..a9ca0d55ab 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -40,6 +40,7 @@ rtc_static_library("audio") {
     "../api:optional",
     "../api/audio_codecs:audio_codecs_api",
     "../api/audio_codecs:builtin_audio_encoder_factory",
+    "../call:bitrate_allocator",
     "../call:call_interfaces",
     "../call:rtp_interfaces",
     "../common_audio",
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 706d502b21..d41dd70dfe 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -88,9 +88,25 @@ rtc_source_set("rtp_sender") {
   ]
 }
 
-rtc_static_library("call") {
+rtc_source_set("bitrate_allocator") {
   sources = [
     "bitrate_allocator.cc",
+    "bitrate_allocator.h",
+  ]
+  deps = [
+    "../modules/bitrate_controller",
+    "../rtc_base:rtc_base_approved",
+    "../rtc_base:sequenced_task_checker",
+    "../system_wrappers",
+  ]
+  if (!build_with_chromium && is_clang) {
+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
+rtc_static_library("call") {
+  sources = [
     "call.cc",
     "callfactory.cc",
     "callfactory.h",
@@ -110,6 +126,7 @@ rtc_static_library("call") {
   ]
 
   deps = [
+    ":bitrate_allocator",
     ":call_interfaces",
     ":rtp_interfaces",
     ":rtp_receiver",
@@ -168,6 +185,7 @@ if (rtc_include_tests) {
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
+      "fake_rtp_transport_controller_send.h",
       "flexfec_receive_stream_unittest.cc",
       "rtcp_demuxer_unittest.cc",
       "rtp_demuxer_unittest.cc",
@@ -175,6 +193,7 @@ if (rtc_include_tests) {
       "rtx_receive_stream_unittest.cc",
     ]
     deps = [
+      ":bitrate_allocator",
       ":call",
       ":mock_rtp_interfaces",
       ":rtp_interfaces",
@@ -187,6 +206,7 @@ if (rtc_include_tests) {
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
+      "../modules/congestion_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/pacing:mock_paced_sender",
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 202a7066cb..0833e52a3f 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -58,6 +58,7 @@ rtc_static_library("video") {
     "../api:optional",
     "../api:transport_api",
     "../api/video_codecs:video_codecs_api",
+    "../call:bitrate_allocator",
     "../call:call_interfaces",
     "../call:rtp_interfaces",
     "../call:video_stream_api",