mikhal@webrtc.org 45f2da0920 VCM/JB: Porting jitter_buffer_test to gtest. ...

Tests were not modified, but ported as is.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1391004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3956 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-05-03 22:22:46 +00:00

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audio_coding

Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert().

2013-05-03 18:11:36 +00:00

audio_conference_mixer

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audio_device

Remove 44.1 kHz workaround from AudioDevice on PulseAudio.

2013-05-03 19:01:46 +00:00

audio_processing

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

bitrate_controller

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desktop_capture

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interface

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media_file

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pacing

Adding trace and changing pacing constants

2013-05-02 19:02:17 +00:00

remote_bitrate_estimator

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rtp_rtcp

Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."

2013-05-03 12:02:11 +00:00

utility

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

video_capture

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video_coding

VCM/JB: Porting jitter_buffer_test to gtest.

2013-05-03 22:22:46 +00:00

video_processing/main

Consolidate common_audio into a single target.

2013-04-30 23:43:26 +00:00

video_render

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module_common_types_unittest.cc

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modules.gyp

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