History log of /external/webrtc/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
1323fc39babb2638c5795c1e23e97c977c5c29f3 11-Nov-2015 Henrik Kjellander <kjellander@webrtc.org> Remove webrtc/test/channel_transport/include

Move the header file into webrtc/test/channel_transport instead.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel
R=henrika@webrtc.org, henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1431983006 .

Cr-Commit-Position: refs/heads/master@{#10595}
/external/webrtc/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
00b8f6b3643332cce1ee711715f7fbb824d793ca 26-Feb-2015 kwiberg@webrtc.org <kwiberg@webrtc.org> Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
3c0aae17f0e3a70fe90ecc6835926b66a3de18fb 04-Sep-2014 kjellander@webrtc.org <kjellander@webrtc.org> Change gflags and gmock includes to be full paths.

This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc
5b3b6b17844942c41d629f6cd9b44f1644cc0ea2 10-Oct-2013 kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Reorganize GYP targets to make webrtc.gyp more usable.

When WebRTC is built as a part of Chromium, some of
the stuff in webrtc.gyp will not be found. This CL
fixes this.

TEST=trybots passing. I also did some manual builds for Android with the android_builder_webrtc target in build/all_android.gyp of a Chromium checkout.
BUG=chromium:304143
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2353004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4949 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc