Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(712)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/voice_engine/BUILD.gn

Issue 2685673003: Define RtpTransportControllerSendInterface. (Closed)
Patch Set: Comment improvements. Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« webrtc/call/rtp_transport_controller.h ('K') | « webrtc/video/video_send_stream.cc ('k') | webrtc/voice_engine/channel.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/voice_engine/BUILD.gn
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 31cdff8d5c43facf4efe76000be050dd033966fe..15f6c0d7a22babab1f70f9d7ec6d7dd9568c156a 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -140,6 +140,10 @@ rtc_static_library("voice_engine") {
"../api/audio_codecs:builtin_audio_decoder_factory",
"../audio/utility:audio_frame_operations",
"../base:rtc_base_approved",
+
+ # TODO(nisse): Delete when declaration of RtpTransportController
+ # and related interfaces move to api/.
+ "../call:call_interfaces",
"../common_audio",
"../logging:rtc_event_log_api",
"../modules/audio_coding:audio_format_conversion",
« webrtc/call/rtp_transport_controller.h ('K') | « webrtc/video/video_send_stream.cc ('k') | webrtc/voice_engine/channel.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698