Bump WebRTC version number. Starting now, we will be setting WebRTC major version numbers to align with Chrome.

R=niklas.enbom@webrtc.org
TBR=niklas.embom

Review URL: https://webrtc-codereview.appspot.com/15219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6942 4adac7df-926f-26a2-2b94-8c16560cd09d
Make sync_chromium use the git-cache when on the bots.

This should help bootstrapping speed, as well as allow better clobbering
support.

R=agable@chromium.org
TBR=agable@chromium.org, kjellander@webrtc.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/17179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6943 4adac7df-926f-26a2-2b94-8c16560cd09d
Use --gclientfile instead of --spec, because windows is THE WORST.

--spec contains newlines, which are interpreted as actual newlines in the
command line, which causes gclient to fall apart at the seams.

TBR=agable@chromium.org, kjellander@webrtc.org, szager@chromium.org
BUG=2863, chromium:339647

Review URL: https://webrtc-codereview.appspot.com/22429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6944 4adac7df-926f-26a2-2b94-8c16560cd09d
Convert nsx_core_neon.S to unified syntax.

That way, it builds with both gcc and clang's integrated assembler.
No intentional behavior change.

BUG=chromium:124610
R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/15199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6945 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix clang -Wformat warnings.

An unsigned int was passed through %lu instead of %u (harmless on 32bit).
More seriously, a wide string was passed through %s, which means only the
first byte in the string got printed (since the 2nd byte is likely 0 in
UCS-2). Use %ls to include the whole string, even though it might not be
renderable in the target 8bit buffer.

BUG=chromium:82385
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6946 4adac7df-926f-26a2-2b94-8c16560cd09d
Refactoring common_audio/signal_processing: Remove unused macro WEBRTC_SPL_MUL_32_32_RSFT32BI

The WEBRTC_SPL_MUL_32_32_RSFT32BI macro was removed in r6169, since it was unused. This CL removes the arm and mips optimizations of it.

BUG=3348, 3353
TESTED=locally and trybots
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6947 4adac7df-926f-26a2-2b94-8c16560cd09d
Use a deterministic input in NetEqBgnTest

This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.

The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)

BUG=3715
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
Add send-side bit-exactness test for AudioCoding Module

This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:

test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest

The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.)  Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.

The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.

BUG=3521
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
Making sure muc members get recorded.

This is an upstream of a change I made; will describe in a separate
email thread.

Essentially, the members map wasn't getting populated in the callclient
example, so it was always empty. Now it will be populated correctly.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6950 4adac7df-926f-26a2-2b94-8c16560cd09d