Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(126)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: webrtc/video/BUILD.gn

Issue 2528343002: H.264 packetization mode 0 (try 3) (Closed)
Patch Set: Modified initialization after discussions with magjed 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/modules/rtp_rtcp/source/rtp_format_h264.cc ('K') | « webrtc/test/fake_encoder.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/video/BUILD.gn
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 66191efb8a054d9767e53b560f3f4be2845e35ea..9709e45c2a2752af179a11dd77f20b90f0e45799 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -144,6 +144,7 @@ if (rtc_include_tests) {
# TODO(pbos): Rename test suite.
rtc_source_set("video_tests") {
testonly = true
+ defines = []
sources = [
"call_stats_unittest.cc",
"encoder_rtcp_feedback_unittest.cc",
@@ -171,7 +172,7 @@ if (rtc_include_tests) {
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
if (rtc_use_h264) {
- defines = [ "WEBRTC_USE_H264" ]
+ defines += [ "WEBRTC_USE_H264" ]
}
}
}
« webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc ('K') | « webrtc/test/fake_encoder.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698