The measurement and trace entries had been mixed up in the calls to webrtc::test::PrintResult, resulting in the plotted graphs were named after the metric. The parameter names are quite confusing which probably led to this.
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/1093007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3496 4adac7df-926f-26a2-2b94-8c16560cd09d
This avoids them getting wiped during sync for every build, which saves
build time on Windows.
I also removed no longer present google-visualization-python dir.
BUG=none
TEST=none
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3488 4adac7df-926f-26a2-2b94-8c16560cd09d
This avoids the directory getting wiped before it's synced out again
for every build.
BUG=none
TEST=none
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3486 4adac7df-926f-26a2-2b94-8c16560cd09d
When doing test automation, the prompt in vie_auto_test is not working as expected on Windows when the test is run from a Buildbot. As soon a prompt is presented to the test runner, vie_auto_test exits, assuming the user pressed Ctrl-D.
By adding a third option for the Stop/Modify call prompt that allows running the call indefinitely (and making that the default), no prompt is displayed when the --auto_custom_call flag is used.
BUG=none
TEST=Execution with vie_auto_test.exe --auto_custom_call --override "Enter destination IP.=192.168.3.11" and by running vie_auto_test in interactive mode.
+ Trybots passing.
Review URL: https://webrtc-codereview.appspot.com/1099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3478 4adac7df-926f-26a2-2b94-8c16560cd09d
Attempt number 2. Now with working tests.
This change allows to build fully unbundled GoogleTTS apk that can be deployed
on any >= ICS_MR1 device.
All static libraries under src/* can be build using ndk stl libraries, using
WEBRTC_STL varible. libwebrtc_audio_coding_gnustl_static is static version of
libwebrtc_audio_coding, build using gnustl from ndk.
Change-Id: I41a5163eb434432eab3131f5df23ffd311e6159b
Due to a bug in the RTP module, which appeared during packet loss, we have had too short delay in the Win Large Test. When the bug was fixed we had a regression error that should be fixed with this update.
Review URL: https://webrtc-codereview.appspot.com/1091005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3466 4adac7df-926f-26a2-2b94-8c16560cd09d
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.
BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1078004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
This change allows to build fully unbundled GoogleTTS apk that can be deployed
on any >= ICS_MR1 device.
All static libraries under src/* can be build using ndk stl libraries, using
WEBRTC_STL varible. libwebrtc_audio_coding_gnustl_static is static version of
libwebrtc_audio_coding, build using gnustl from ndk.
Bug: 6397748
Change-Id: Ibf0acb11d3e605a1d4c668bbf98b0a0bb55399bc
A reverse copy is removed. The index to src buffer could be -1, this happens very often. The reverse copy is not needed as the content of the destination is overwritten further down in "WebRtcIlbcfix_CbConstruct()"
Bug=issue281
TEST=manual test over 1600 files TIMIT database, all outputs are bit-exact with the ones generated from head revision. Local run of asan does not generate any warning.
Review URL: https://webrtc-codereview.appspot.com/1063013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3447 4adac7df-926f-26a2-2b94-8c16560cd09d
However, two other "hacks" had to be added to maintain bit-exactness
with legacy.
Note that this change requires a new version of the universal.rtp test
input, although the output reference stays the same.
Moving reference files, and using a new input vector for NetEq4.
The new input vector neteq_universal_new.rtp is identical to the old
neteq_universal.rtp, except that the payload type for CNG packets that
follows a wideband codec is changed to 98.
Update to resources revision 15 where the new reference files are.
Also changing a faulty log error.
Review URL: https://webrtc-codereview.appspot.com/1078009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3442 4adac7df-926f-26a2-2b94-8c16560cd09d
The changes in r1068 that moved over to the webrtc/test/buildbot_tests.py launch script was not properly tested on the real machine for the audio_e2e_test. Due to that it contained a few syntax errors and paths that were not resolved as expected. This CL fixes this and has been tested more thorougly.
BUG=none
TEST=Ran, standing in the checkout dir:
out/Release/buildbot_tests.py -t audio_e2e_test
with successful result.
Review URL: https://webrtc-codereview.appspot.com/1070012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3438 4adac7df-926f-26a2-2b94-8c16560cd09d
