henrik.lundin@webrtc.org 3ab57c514c Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets ...

This is a relanding of r5725, now with a fix for the failing tests.

BUG=2935
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5738 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-03-20 15:09:38 +00:00

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ACMTest.cc

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ACMTest.h

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APITest.cc

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APITest.h

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Channel.cc

Fix the break caused by r5579.

2014-02-19 23:07:31 +00:00

Channel.h

Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable.

2014-02-19 20:31:17 +00:00

delay_test.cc

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dual_stream_unittest.cc

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EncodeDecodeTest.cc

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EncodeDecodeTest.h

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initial_delay_unittest.cc

Changing the buffer size (slots) to 1.5 seconds @ 30 ms packets

2014-03-20 15:09:38 +00:00

insert_packet_with_timing.cc

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iSACTest.cc

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iSACTest.h

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opus_test.cc

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opus_test.h

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PCMFile.cc

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PCMFile.h

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RTPFile.cc

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RTPFile.h

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SpatialAudio.cc

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SpatialAudio.h

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target_delay_unittest.cc

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TestAllCodecs.cc

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TestAllCodecs.h

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Tester.cc

Disable TestOpusNewACM on Android.

2014-03-11 20:40:59 +00:00

TestFEC.cc

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TestFEC.h

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TestStereo.cc

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TestStereo.h

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TestVADDTX.cc

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TestVADDTX.h

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TimedTrace.cc

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TimedTrace.h

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TwoWayCommunication.cc

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TwoWayCommunication.h

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utility.cc

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utility.h

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