Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(556)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/call/audio_state.h

Issue 2550273003: Moved call.h and most of api/call/* into call/ (Closed)
Patch Set: Separated headers into :call_interfaces; update deps Created 4 years 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/BUILD.gn ('K') | « webrtc/call/audio_send_stream.cc ('k') | webrtc/call/bitrate_estimator_tests.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/call/audio_state.h
diff --git a/webrtc/api/call/audio_state.h b/webrtc/call/audio_state.h
similarity index 93%
rename from webrtc/api/call/audio_state.h
rename to webrtc/call/audio_state.h
index b8dca3fb4ec67f9144e69ccb8deb08ad7765b427..2c26a1749b5b62df61cef8aba5d606d23d1b92a9 100644
--- a/webrtc/api/call/audio_state.h
+++ b/webrtc/call/audio_state.h
@@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
-#define WEBRTC_API_CALL_AUDIO_STATE_H_
+#ifndef WEBRTC_CALL_AUDIO_STATE_H_
+#define WEBRTC_CALL_AUDIO_STATE_H_
#include "webrtc/api/audio/audio_mixer.h"
#include "webrtc/base/refcount.h"
@@ -46,4 +46,4 @@ class AudioState : public rtc::RefCountInterface {
};
} // namespace webrtc
-#endif // WEBRTC_API_CALL_AUDIO_STATE_H_
+#endif // WEBRTC_CALL_AUDIO_STATE_H_
« webrtc/call/BUILD.gn ('K') | « webrtc/call/audio_send_stream.cc ('k') | webrtc/call/bitrate_estimator_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698