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webrtc_m130/webrtc/modules
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minyue@webrtc.org 194fea7640 The lastest commit on this file was in
https://webrtc-codereview.appspot.com/15529004/

The final patch set should have included this, but was missed.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6755 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 09:55:51 +00:00
..
audio_coding
The lastest commit on this file was in
2014-07-22 09:55:51 +00:00
audio_conference_mixer
This is related to an earlier CL of enabling Opus 48 kHz.
2014-07-18 12:28:28 +00:00
audio_device
Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
2014-07-18 17:13:28 +00:00
audio_processing
AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float
2014-07-18 07:50:29 +00:00
bitrate_controller
…
desktop_capture
Fix an invalid memory access due to typo in win/cursor.cc.
2014-07-15 20:32:03 +00:00
interface
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media_file
Add unit test for MediaFile WAV file writing
2014-07-17 08:11:32 +00:00
pacing
…
remote_bitrate_estimator
…
rtp_rtcp
Remove remains of WEBRTC_NO_STL.
2014-07-22 06:48:58 +00:00
utility
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video_capture
Runtime guard for iOS7 property.
2014-07-18 17:17:59 +00:00
video_coding
r6709 lacks a change in BUILD.gn
2014-07-16 22:18:49 +00:00
video_processing
…
video_render
…
module_common_types_unittest.cc
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modules_java_chromium.gyp
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modules_java.gyp
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modules_tests.isolate
…
modules_unittests.isolate
…
modules.gyp
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OWNERS
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