Add unit tests to rtcp_receiver_test.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6994 4adac7df-926f-26a2-2b94-8c16560cd09d
Disable VideoAdapterTest.BlackOutput on DrMemory.

Reports uninitialized-memory reads that seem to originate from when the
frame is copied. The test passes if we remove CPU optimizations from
libyuv, disabling test until we figure out whether it's an unsupported
instruction in DrMemory, bug in libyuv or bug in the test.

BUG=3754
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6995 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the different samples per channel in aecdump

BUG=webrtc:3359
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6996 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 74235596-> 74297316

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6997 4adac7df-926f-26a2-2b94-8c16560cd09d
common_audio: Removed macro WEBRTC_SPL_DIV

The macro has no built-in divide by zero check. The only thing that is done is casting to int32_t.
In addition a bug was discovered where it was supposed to do a division with rounding, but instead did a division with truncation + addition by 2. This is corrected in this CL.

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6998 4adac7df-926f-26a2-2b94-8c16560cd09d
GN: Implement video_engine, video_capture and video_render.

Also add more from common.gypi to webrtc.gni.

These GN configs are based on GYP files in r6997.

BUG=3441
TEST=Trybots and local compile using:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default

Passed compile from a Chromium checkout with src/third_party/webrtc linked to the webrtc/ dir of a checkout with this patch applied.

R=brettw@chromium.org, glaznev@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6999 4adac7df-926f-26a2-2b94-8c16560cd09d
Add 60 fps video support

R=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7000 4adac7df-926f-26a2-2b94-8c16560cd09d
Add talk owners to migrated talk folders

BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7001 4adac7df-926f-26a2-2b94-8c16560cd09d
Specify an ECDH group for ECDHE.

By default, OpenSSL cannot negotiate ECDHE cipher suites as a server because it
doesn't know what curve to use.

BUG=chromium:406458
TEST=Download Firefox nightly build from 2014-08-12.
  https://ftp.mozilla.org/pub/mozilla.org/firefox/nightly/2014-08-12-mozilla-central-debug/
  Point Firefox to https://apprtc.appspot.com
  Point Chrome on Android to the URL Firefox redirects to (it'll say ?r=NUMBERS at the end)
  After tapping through various permissions prompts on either side, the call goes through.

R=agl@chromium.org, henrike@webrtc.org, jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7002 4adac7df-926f-26a2-2b94-8c16560cd09d
Add CHECK and friends from Chromium.

Replace FATAL_ERROR_IF with the more familiar (to Chromium developers)
CHECK and DCHECK. The full Chromium implementation is fairly elaborate
but I copied enough to get us most of the benefits. I believe the main
missing component is a more advanced stack dump. For this bit I relied
on the V8 implementation.

There are a few minor modifications from the Chromium original:
- The FatalMessage class is specialized for logging fatal error
messages and aborting. Chromium uses the general LogMessage class,
which we could consider moving towards in the future.
- NOTIMPLEMENTED() and NOTREACHED() have been removed, partly because
I don't want to rely on our logging.h until base/ and system_wrappers/
are consolidated.
- FATAL() replaces LOG(FATAL).

Minor modifications from V8's stack dump:
- If parsing of a stack trace symbol fails, just print the unparsed
symbol. (I noticed this happened on Mac.)
- Use __GLIBCXX__ and __UCLIBC__. This is from examining the backtrace
use in Chromium.

UNREACHABLE() has been removed because its behavior is different than
Chromium's NOTREACHED(), which is bound to cause confusion. The few uses
were replaced with FATAL(), matching the previous behavior.

Add a NO_RETURN macro, allowing us to remove unreachable return
statements following a CHECK/FATAL.

TESTED=the addition of dummy CHECK, DCHECK, CHECK_EQ and FATAL did the
did the right things. Stack traces work on Mac, but I don't get symbols
on Linux.

R=henrik.lundin@webrtc.org, kwiberg@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7003 4adac7df-926f-26a2-2b94-8c16560cd09d
Precompile out our standalone CHECK macros in a Chromium build.

As documented, the use of overrides/webrtc/base/logging.h in a Chromium
build reuslts in redefined macro errors. Fortunately, Chromium's macros
can be used as drop-in replacements for the standalone versions.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/17239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7004 4adac7df-926f-26a2-2b94-8c16560cd09d
When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated.

  constraints . SetMandatoryReceiveAudio (false);

The problem is that webrtc::GetTrackIdBySsrc returns false if audio is not available. However it should continue and check for the video track.

BUG=webrtc:3755
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7005 4adac7df-926f-26a2-2b94-8c16560cd09d
Add jiayl to talk OWNERS.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7006 4adac7df-926f-26a2-2b94-8c16560cd09d
win: Replace custom assert() macro with regular assert.h

The current code got added in libjingle r103; I don't see a good reason for it.
Things still build with plain old assert.h.

The custom assert was wrong: __debugbreak() is documented to return void,
so doing `cond ? true : __debugbreak()` shouldn't build (and it doesn't in
clang-cl). It's possible to make it build by writing
`cond ? true : (__debugbreak(), true)`, but just using the regular header
seems like a much better fix.

BUG=chromium:82385
Review URL: https://webrtc-codereview.appspot.com/19139004/


git-svn-id: http://webrtc.googlecode.com/svn/trunk@7007 4adac7df-926f-26a2-2b94-8c16560cd09d