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webrtc_m130/webrtc/voice_engine/test/auto_test/standard
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pkasting@chromium.org 4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
..
audio_processing_test.cc
Re-enable AGC tests:
2014-04-16 11:58:18 +00:00
codec_before_streaming_test.cc
Include files from webrtc/.. paths in voice_engine/
2013-05-21 13:52:32 +00:00
codec_test.cc
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
2014-09-03 12:28:06 +00:00
dtmf_test.cc
Reland "Remove DTMF status methods from Voice Engine" r7276
2014-10-01 08:23:21 +00:00
external_media_test.cc
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-05-12 12:19:19 +00:00
file_before_streaming_test.cc
Include files from webrtc/.. paths in voice_engine/
2013-05-21 13:52:32 +00:00
file_test.cc
Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798
2014-08-11 21:06:30 +00:00
hardware_before_initializing_test.cc
Include files from webrtc/.. paths in voice_engine/
2013-05-21 13:52:32 +00:00
hardware_before_streaming_test.cc
Removes parts of the webrtc::VoEHardware sub API (relanding)
2014-05-09 11:43:00 +00:00
hardware_test.cc
Removes parts of the webrtc::VoEHardware sub API (relanding)
2014-05-09 11:43:00 +00:00
mixing_test.cc
This is related to an earlier CL of enabling Opus 48 kHz.
2014-07-18 12:28:28 +00:00
neteq_stats_test.cc
Include files from webrtc/.. paths in voice_engine/
2013-05-21 13:52:32 +00:00
rtp_rtcp_before_streaming_test.cc
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-05-12 08:53:27 +00:00
rtp_rtcp_extensions.cc
Use size_t more consistently for packet/payload lengths.
2014-11-20 22:28:14 +00:00
rtp_rtcp_test.cc
Report total bitrate for all streams in GetStats.
2014-11-14 11:52:04 +00:00
video_sync_test.cc
VoiceEngine(iOS & Android): removed NOT_SUPPORTED
2014-04-01 00:16:35 +00:00
voe_base_misc_test.cc
Ref-counted rewrite of ChannelManager.
2013-08-07 17:57:36 +00:00
volume_test.cc
VoEVolumeTest: Enabled Linux flaky tests
2014-05-20 10:43:42 +00:00