5 Commits

Author SHA1 Message Date
Bjorn Volcker
fb49451014 Disables mic bump-up level if not built with chromium
In http://chromegw.corp.google.com/viewvc/chrome-internal?view=rev&revision=61016 a feature to bump up low input audio levels to a fixed value of 33%. In https://webrtc-codereview.appspot.com/43109004/ a configuration to choose an arbitrary level was added, but still using 33% as default.
The original bump-up feature was added to fix audio issues in chrome, but affected also non-chrome users. This CL disables the feature for non-chrome applications.

Note that the default value is set to 0, but any value up to 12 will do. Zero was selected because it is more clear that the feature is turned off.

BUG=4529
R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43259004

Cr-Commit-Position: refs/heads/master@{#9048}
2015-04-22 04:39:47 +00:00
Jelena Marusic
c317ce5456 VoE: move mock directory 1 level up
Changes:
1. Moved directory voice_engine/include/mock to voice_engine/mock (current recommendation).
2. Updated includes where necessary.

Caution:
We need confirmation that these mocks are indeed used only locally.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48089004

Cr-Commit-Position: refs/heads/master@{#9005}
2015-04-15 10:45:09 +00:00
kjellander@webrtc.org
a33f05e8d7 Re-land "Remove <(webrtc_root) from source file entries."
Changes differing from https://webrtc-codereview.appspot.com/37859004:
* I put the include_tests==1 stuff of audio_coding.gypi in its
  own audio_coding_tests.gypi file, including the Android and isolate
  targets which were incorrectly located in the previous CL
* I moved the bwe utilities in remote_bitrate_estimator.gypi
  into include_tests==1 since they depend on test.gyp after I
  cleaned up the duplicated inclusion of rtp_file_reader.cc

R=stefan@webrtc.org
TBR=tina.legrand@webrtc.org
TESTED=Passing gyp and compile using:
webrtc/build/gyp_webrtc -Dinclude_tests=1
webrtc/build/gyp_webrtc -Dinclude_tests=0
I also setup a Chromium checkout with my checkout mounted in
third_party/webrtc and ran build/gyp_chromium successfully.

BUG=4185

Review URL: https://webrtc-codereview.appspot.com/33159004

Cr-Commit-Position: refs/heads/master@{#8205}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8205 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 14:30:41 +00:00
kjellander@webrtc.org
1ece0cbbec Revert "Remove <(webrtc_root) from source file entries."
And the follow-up fix in r8198 that was not sufficient.
Reason: breaks Chromium bots runhooks (GYP).

I will have to try some more to make sure I don't
include test code, since include_tests==0 in Chromium.

TBR=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37039004

Cr-Commit-Position: refs/heads/master@{#8200}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8200 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:02:42 +00:00
kjellander@webrtc.org
2d2a1f9f05 Remove <(webrtc_root) from source file entries.
This required to move the AGC tools source files
into webrtc/tools and create a new agc_test_utils target.

Since audio_codec_speed_tests.gypi referenced sources above,
the best approach I could come up with was to add an audio_coding.gypi
file at a higher level and move the targets in there (+ the includes from
modules.gyp which is an improvement IMO).

I also added a PRESUBMIT.py check to prevent new source
entries being added with <(webrtc_root) in the path.

BUG=4185
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37859004

Cr-Commit-Position: refs/heads/master@{#8197}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 10:24:44 +00:00