Chromium Code Reviews| 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 |