Index: webrtc/modules/audio_coding/BUILD.gn |
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn |
index 96c1ea981983f0e26a34040b15fae38c4dcbc628..7b5d2d8b966de957364291fc787660c853f5c02c 100644 |
--- a/webrtc/modules/audio_coding/BUILD.gn |
+++ b/webrtc/modules/audio_coding/BUILD.gn |
@@ -1209,6 +1209,19 @@ if (rtc_include_tests) { |
} |
} |
+ executable("rtp_to_text") { |
aleloi
2016/08/01 15:26:33
This seems unused or deprecated. There is a 'retur
kjellander_webrtc
2016/08/01 19:03:43
Let's see what Henrik L has to say. I believe he a
|
+ testonly = true |
+ |
+ sources = [ |
+ "neteq/test/rtp_to_text.cc", |
+ ] |
+ |
+ deps = [ |
+ ":neteq_test_tools", |
+ "../../system_wrappers", |
+ ] |
+ } |
+ |
executable("rtp_analyze") { |
testonly = true |