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Commits on Apr 4, 2013

• Fix unit test compilation

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  83ee428

• Fix compilation on gcc 4.7

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  committed
  8098652

• Unison patches

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  21f433b

Commits on Mar 28, 2013

• Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549

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  55742e5

• Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots.

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  46144bb

• Removed CPU APIs from VoEHardware. Code is now only used by test applications.

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  3b6f728

• Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are curre…

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  fa2dd22

Commits on Mar 27, 2013

• Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..."

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  2ffc8bf

• Removed CPU APIs from VoEHardware. Code is now only used by test applications.

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  365ca40

• Fix broken audio.

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  0c0795e

• G722-stereo has been missing when creating AudioDecoder.

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  a0bba27

• NetEq4 fails if the first packets inserted in are out-of-band DTMFs.

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  88a7940

• Fix flakiness in network up/down event tests when running under memcheck.

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  88f12ab

• WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.

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  6fc5215

• Add interface to signal a network down event.

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  dca71b2

Commits on Mar 26, 2013

• Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events).

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  6e34ceb

• Remove VoE's default call in Trace::SetLevelFilter.

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  f386e2b

• Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.

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  aa0fcd7

• Alphabetize include order in fake_voe_external_media.h.

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  31b4448

Commits on Mar 25, 2013

• Restart Android capture after orientation change.

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  a7a643e

• Add some VoE and AudioProcessing mocks.

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  13f66d1

• Refactor unittest trace printouts to a separate class.

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  8826e34

• Enable the below APIs for iOS.

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  0c1f10b

Commits on Mar 22, 2013

• Introduced pause and resume to the pacer

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  9522792

• Updated WebRTC version to 3.27

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  f49577f

• Bugfix for extended RTP/RTCP test

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  7fd368f

• Move the VIE tests to use external transport instead of the built in udp transport

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  c075e25

• Add trace printouts to all unit tests.

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  4c27c03

• Add min and target bitrate to VideoCodec.

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  e1198e6

• Move the VoE tests to use external transport instead of the built in udp transport

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  aa922de

Commits on Mar 21, 2013

• Creating a copy of Udp transport under webrtc/test

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  af6aa7b

• Cleanup nanosleep -> SleepMs

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  495c563

• WebRtc_Word -> stdint in audio_coding/g711/

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  4a48fd6

• Remove incorrect asserts.

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  a2df078

• WebRtc_Word -> stdint in audio_coding/cng/

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  e49f252

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