Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate

This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes

The sizes saved in the aecdumps were always the input length, and this is not necessarily true when there is a change in sample rate. But the sample rates dumped are correct, so we can calculate the sizes from them knowing that we use 10ms chunks.

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

Review URL: https://webrtc-codereview.appspot.com/24389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7039 4adac7df-926f-26a2-2b94-8c16560cd09d
Partial revert of r7014 (Android APK refactor)

This reverts selected parts of r7014 to enable
rolling WebRTC in Chromium DEPS.

This works around the problem with GYP includes
being processed in the first pass (i.e. variables
cannot be used for paths). Using a dependency with
a path using a variable that is conditioned for
build_with_chromium being 0 or 1 solves the Chromium
build.

These changes will be restored once I've finished
a major GYP refactoring that will break out all
test related code (at least the parts that includes
the Android APK targets) into a separate chain
of GYP targets that are not processed when generating
projects for Chromium (which is why r7014 is breaking
the Chromium build).

BUG=3741
TESTED=Passing compilation of standalone using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug
Then verified the *_apk  targets are generated and compiled.

Passing compilation from a Chromium checkout with third_party/webrtc
directory removed and a new empty third_party/webrtc mapped to the
standalone checkout using:
sudo mount --bind /path/to/trunk/webrtc third_party/webrtc
Then running build/gyp_chromium
I also verified WebRTC GYP targets exist and are able to compile.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
Audio codecs to include webrtc/typedefs.h

Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

CL Generated with:
$ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7041 4adac7df-926f-26a2-2b94-8c16560cd09d
Set resolution based on incoming VideoFrames.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/17269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove flake in SendsLowerResolutionOnSmallerFrames.

Speculative fix for break on Linux64 Release. It looks like the second
frame is being dropped which is likely because the two frames are sent
too close to eachother. Adding a delay of 33ms in between them to make
sure the second one isn't dropped.

R=minyue@webrtc.org
TBR=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7043 4adac7df-926f-26a2-2b94-8c16560cd09d
Network up/down signaling in Call.

BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 74694022-> 74696326

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7041 " Audio codecs to include webrtc/typedefs.h"

Breaks gn build, see e.g. http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Linux%20GN/builds/1248/steps/compile/logs/stdio

R=turaj@webrtc.org
TBR=andresp@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/19219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7046 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 74696326-> 74723281

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7047 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix leak of NSAutoreleasePool.

This looks like something that's no longer applicable. From what I saw this code path isn't on a static initializer that runs before main. Should be okay to drain (release) pool outside of this scope.

BUG=3659
R=henrike@webrtc.org, jiayl@webrtc.org, noahric@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7048 4adac7df-926f-26a2-2b94-8c16560cd09d
cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error
BUG=3663
TESTED=ninja local build on windows.
R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7049 4adac7df-926f-26a2-2b94-8c16560cd09d
* Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files.

"gcl try" fails to upload these large files so adding them independently.

R=andrew@webrtc.org, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7050 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove ambiguous call to MakeCheckOpString.

BUG=3777
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7051 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix compile errors on webrtc/base.

R=fbarchard@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7052 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix audio/video sync when FEC is enabled.

Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.

BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
Change return value for number of discarded packets to be int.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove HybridVideoEngine.

This is currently unused dead code.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support for multi-channel DTMF tone generation

This CL opens up support for DTMF tones to be played to multi-channel
outputs. The same tones are replicated across all channels. Unit tests
are updated.

Also adding a new method AudioMultiVector::CopyChannel.

BUG=crbug/407114
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7056 4adac7df-926f-26a2-2b94-8c16560cd09d
ACMOpus: Remove useless member variable fec_enabled_

R=henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7057 4adac7df-926f-26a2-2b94-8c16560cd09d