Low-bandwidth audio testing

webrtc/test/BUILD.gn

Index: webrtc/test/BUILD.gn
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 267d1e2d0598386297caf0cd2b4a71fc04e4da24..4920a5c6f2cc56599e38d63fdb29cae9205f36e8 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -254,6 +254,7 @@ if (is_ios) {
 rtc_test("test_support_unittests") {
   deps = []
   sources = [
+    "fake_audio_device_unittest.cc",
     "fake_network_pipe_unittest.cc",
     "frame_generator_unittest.cc",
     "rtp_file_reader_unittest.cc",
@@ -408,6 +409,9 @@ rtc_source_set("test_common") {
     "//testing/gmock",
     "//testing/gtest",
   ]
+  if (!is_android) {
+    deps += [ "../modules/video_capture:video_capture_internal_impl" ]

[kjellander_webrtc, 2017/03/21 08:31:49]

I'm still curious about this. Does it come from :video_test_common? Can it be
moved in there instead? We need to try hard not making things more complicated
in the dependency tree.

[kjellander_webrtc, 2017/03/21 08:43:41]

Let's keep this as is if you're willing to just have a quick look in a separate
CL. OK?

[oprypin_webrtc, 2017/03/23 09:37:01]

OK

+  }
 }
 
 config("test_renderer_exported_config") {