external/webrtc.git
4 weeks agoAdd PRESUBMIT check for GYP files including source files above itself. master
kjellander@webrtc.org [Tue, 27 Jan 2015 13:13:24 +0000 (13:13 +0000)]
Add PRESUBMIT check for GYP files including source files above itself.

This is needed because some tools does not support files
located above the project generated.

BUG=4185
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8166 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoRoll chromium_revision 4664fe0..9070a80 (312733:313233)
kjellander@webrtc.org [Tue, 27 Jan 2015 13:11:10 +0000 (13:11 +0000)]
Roll chromium_revision 4664fe0..9070a80 (312733:313233)

Relevant changes:
* src/third_party/boringssl/src: 5fa3eba..347f025
* src/third_party/libvpx: 8dc6ea9..5da40ca
* src/tools/gyp: adb7d24..b28bd7d
* src/tools/swarming_client: e98dde9..d863df3
Details: https://chromium.googlesource.com/chromium/src/+/4664fe0..9070a80/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8165 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoUpdate StreamDataCounter with FEC bytes.
asapersson@webrtc.org [Tue, 27 Jan 2015 12:17:29 +0000 (12:17 +0000)]
Update StreamDataCounter with FEC bytes.

Add histograms stats for send/receive FEC bitrate:
- "WebRTC.Video.FecBitrateReceivedInKbps"
- "WebRTC.Video.FecBitrateSentInKbps"

Correct media payload bytes in StreamDataCounter to not include FEC bytes.

Fix stats for rtcp packets sent/received per minute (regression from r7910).

BUG=crbug/419657
R=holmer@google.com, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoAEC: Implements a new function for calculating delay metrics
bjornv@webrtc.org [Tue, 27 Jan 2015 11:30:54 +0000 (11:30 +0000)]
AEC: Implements a new function for calculating delay metrics

Two new member variables have been added and the code for calculating the delay metrics have been moved to a function.

BUG=2994
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8163 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoReland of: "Implement elapsed time and capture start NTP time estimation." revision...
magjed@webrtc.org [Tue, 27 Jan 2015 09:57:01 +0000 (09:57 +0000)]
Reland of: "Implement elapsed time and capture start NTP time estimation." revision @8139

Link to original CL: https://review.webrtc.org/36909004/

R=pbos@webrtc.org
TBR=pthatcher@webrtc.org
BUG=4227

Review URL: https://webrtc-codereview.appspot.com/39669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8162 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoSupport VP8 HW decoding on devices with Exynos codec.
glaznev@webrtc.org [Mon, 26 Jan 2015 23:07:19 +0000 (23:07 +0000)]
Support VP8 HW decoding on devices with Exynos codec.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoFix bug in GetREDStatus(): it doesn't actually return the current status.
pkasting@chromium.org [Mon, 26 Jan 2015 22:35:29 +0000 (22:35 +0000)]
Fix bug in GetREDStatus(): it doesn't actually return the current status.

BUG=none
TEST=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8159 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoUpdate AppRTCDemo to use renamed GAE messages.
glaznev@webrtc.org [Mon, 26 Jan 2015 22:22:50 +0000 (22:22 +0000)]
Update AppRTCDemo to use renamed GAE messages.

BUG=4221
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8158 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoAdd an AudioRingBuffer class wrapper for the ring_buffer.h C interface.
andrew@webrtc.org [Mon, 26 Jan 2015 21:23:53 +0000 (21:23 +0000)]
Add an AudioRingBuffer class wrapper for the ring_buffer.h C interface.

Integrate it in Blocker to demonstrate use.

TEST=beamforming sounds good.
R=aluebs@webrtc.org, mgraczyk@chromium.org, sahark@google.com

Review URL: https://webrtc-codereview.appspot.com/36799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8157 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoConsolidate anonymous namespace content and file-static methods to all be in the
pkasting@chromium.org [Mon, 26 Jan 2015 19:59:32 +0000 (19:59 +0000)]
Consolidate anonymous namespace content and file-static methods to all be in the
anonymous namespace, in preparation for refactoring a few of the functions a
little.

No code change.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8155 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoMake it easier to use external libyuv + cleanup GYP files.
kjellander@webrtc.org [Mon, 26 Jan 2015 19:17:26 +0000 (19:17 +0000)]
Make it easier to use external libyuv + cleanup GYP files.

It is now easier to use an external libyuv library.
Fix some GYP errors.
Remove the temporary webrtc_base target (depends on
https://codereview.chromium.org/865603002/ being landed
first).

BUG=4185
R=andresp@webrtc.org, andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8154 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoRefactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16()
bjornv@webrtc.org [Mon, 26 Jan 2015 15:32:47 +0000 (15:32 +0000)]
Refactor common_audio/vad: Removed usage of macro WEBRTC_SPL_MUL_16_16()

BUG=3348,3353
TESTED=Locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8152 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoMove ThreadChecker into rtc_base_approved.
tommi@webrtc.org [Mon, 26 Jan 2015 15:27:29 +0000 (15:27 +0000)]
Move ThreadChecker into rtc_base_approved.

To do this, I'm removing ThreadChecker's dependency on the 'Thread' class, so that the checker works with any thread and doesn't rely on TLS.
Also simplifying CriticalSection's implementation on Windows since a critical section on Windows already knows what thread currently owns the lock.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8151 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoEnable encoder multi-threading for VP9.
marpan@webrtc.org [Mon, 26 Jan 2015 15:21:36 +0000 (15:21 +0000)]
Enable encoder multi-threading for VP9.

R=stefan@webrtc.org
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8150 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoTemporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface...
kwiberg@webrtc.org [Mon, 26 Jan 2015 13:03:32 +0000 (13:03 +0000)]
Temporarily revert r8147 ("Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h")

Some out-of-tree code that uses base/scoped_ptr.h is defining nullptr
to 0, which causes an obvious compilation error and perhaps other
subtle problems. I'm hoping to get that sorted out and re-land this CL
soon.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8149 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoIntroduce rtc::CheckedDivExact
henrik.lundin@webrtc.org [Mon, 26 Jan 2015 11:08:53 +0000 (11:08 +0000)]
Introduce rtc::CheckedDivExact

Use the new method to replace local ones in AudioEncoder{Opus,Isac}.

COAUTHOR:kwiberg@webrtc.org

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoUpdate base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h
kwiberg@webrtc.org [Mon, 26 Jan 2015 08:57:57 +0000 (08:57 +0000)]
Update base/scoped_ptr.h from system_wrappers/interface/scoped_ptr.h

The latter file was more up-to-date. The files are now identical
with the following exceptions:

  * The namespace used (rtc vs. webrtc).

  * The name of the include guard.

  * base/scoped_ptr.h still has two extra methods, accept() and use().

  * base/scoped_ptr.h still includes webrtc/base/common.h even though
    it doesn't need it itself, since several .cc files expect to get
    it for free by incuding base/scoped_ptr.h. This is of course bad
    manners, and the "unused" include will be removed in a future CL.

A later CL will remove system_wrappers/interface/scoped_ptr.h.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8147 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoRemove win_asan trybot from PRESUBMIT.py
kjellander@webrtc.org [Sun, 25 Jan 2015 19:27:03 +0000 (19:27 +0000)]
Remove win_asan trybot from PRESUBMIT.py

Removing it since it no longer exists.
See https://codereview.chromium.org/872263002/

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8146 4adac7df-926f-26a2-2b94-8c16560cd09d

4 weeks agoRoll chromium_revision c086b4e..4664fe0 (312108:312733)
kjellander@webrtc.org [Sun, 25 Jan 2015 19:17:56 +0000 (19:17 +0000)]
Roll chromium_revision c086b4e..4664fe0 (312108:312733)

Mainly to pick up the MIPS changes in
https://codereview.chromium.org/843563002/
for which the changes in
https://webrtc-codereview.appspot.com/41399004/
are included in this CL.

Relevant changes:
* src/third_party/android_tools: 56b3d3e..aaeda3d
* src/third_party/boringssl/src: ca9a538..5fa3eba
* src/third_party/libvpx: 4f9bd1b..8dc6ea9
* src/third_party/openmax_dl: 1a4171c..8f7bf0b
* src/tools/gyp: 194ec65..adb7d24
* src/tools/swarming_client: 0a795bd..e98dde9
Details: https://chromium.googlesource.com/chromium/src/+/c086b4e..4664fe0/DEPS

Clang version was not updated in this roll.

BUG=4214, 4222
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8145 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRevert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
tkchin@webrtc.org [Fri, 23 Jan 2015 21:20:41 +0000 (21:20 +0000)]
Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."

> Remove frame copy in ViEExternalRendererImpl::RenderFrame
>
> Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.
>
> BUG=1128
> R=mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36489004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8144 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRevert 8139 "Implement elapsed time and capture start NTP time e..."
tkchin@webrtc.org [Fri, 23 Jan 2015 21:17:38 +0000 (21:17 +0000)]
Revert 8139 "Implement elapsed time and capture start NTP time e..."

> Implement elapsed time and capture start NTP time estimation.
>
> These two elements are required for end-to-end delay estimation.
>
> BUG=1788
> R=stefan@webrtc.org
> TBR=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36909004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8143 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoReland r7980:
jiayl@webrtc.org [Fri, 23 Jan 2015 17:33:34 +0000 (17:33 +0000)]
Reland r7980:
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908

BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoChange a GYP reference to cpufeatures.gypi
fdegans@chromium.org [Fri, 23 Jan 2015 16:35:17 +0000 (16:35 +0000)]
Change a GYP reference to cpufeatures.gypi

This will allow us to move the remaining GYP file in android_tools
to the chromium repository by removing the direct reference to it.

BUG=webrtc:4115
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8140 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoImplement elapsed time and capture start NTP time estimation.
pbos@webrtc.org [Fri, 23 Jan 2015 14:55:00 +0000 (14:55 +0000)]
Implement elapsed time and capture start NTP time estimation.

These two elements are required for end-to-end delay estimation.

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

Review URL: https://webrtc-codereview.appspot.com/36909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8139 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoDisable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
kjellander@webrtc.org [Fri, 23 Jan 2015 14:34:52 +0000 (14:34 +0000)]
Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness

Disabling the test on all platforms since it's likely it can happen
on any platform, even if it's only been observed on Win x64 Release.

Running tests in parallel is a huge performance benefit to the team,
since it approximately reduces build cycle with 60-75%.

BUG=4219
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8138 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRe-allowing RED in voice engine.
minyue@webrtc.org [Fri, 23 Jan 2015 11:58:42 +0000 (11:58 +0000)]
Re-allowing RED in voice engine.

Path of audio RED packets was blocked in r4692 by accident. It ought be enabled again.

BUG=3619
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8137 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRemove frame copy in ViEExternalRendererImpl::RenderFrame
magjed@webrtc.org [Fri, 23 Jan 2015 11:50:13 +0000 (11:50 +0000)]
Remove frame copy in ViEExternalRendererImpl::RenderFrame

Add new interface for delivering frames to ExternalRenderer. The purpose is to avoid having to extract a packed buffer from I420VideoFrame, which will cause a deep frame copy.

BUG=1128
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8136 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoSwitch to use range based loops in the BWE simulation framework.
stefan@webrtc.org [Fri, 23 Jan 2015 08:29:52 +0000 (08:29 +0000)]
Switch to use range based loops in the BWE simulation framework.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8135 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoLeave BIO_METHOD non-const.
davidben@webrtc.org [Thu, 22 Jan 2015 23:06:17 +0000 (23:06 +0000)]
Leave BIO_METHOD non-const.

This breaks building against OpenSSL upstream, which is still supported on iOS.
This reverts part of https://webrtc-codereview.appspot.com/34649004.

BUG=none
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8132 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoChange GetStreamBySsrc to not copy StreamParams.
tommi@webrtc.org [Thu, 22 Jan 2015 23:00:41 +0000 (23:00 +0000)]
Change GetStreamBySsrc to not copy StreamParams.
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple.  Also, we can use lambdas now :)

BUG=
R=perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoFix a crash in AllocationSequence.
jiayl@webrtc.org [Thu, 22 Jan 2015 21:28:39 +0000 (21:28 +0000)]
Fix a crash in AllocationSequence.
Internal bug 19074679.

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8130 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRevert 8125 "Modify some tests to never use DTX disable mode"
kjellander@webrtc.org [Thu, 22 Jan 2015 19:02:03 +0000 (19:02 +0000)]
Revert 8125 "Modify some tests to never use DTX disable mode"

Broke compile on the Chromium FYI bots:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293

Error:
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35:
../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined]
#define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity))
        ^
../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here
#define LOG(sev) \
        ^
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined]
#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
        ^
../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here
#define LOG_V(sev) \
        ^
2 errors generated.

> Modify some tests to never use DTX disable mode
>
> DTX disable mode will be removed as a part of the ACM redesign work.
>
> COAUTHOR:kwiberg@webrtc.org
>
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34769004

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoChange sprintf use in talk samples to snprintf
jlmiller@webrtc.org [Thu, 22 Jan 2015 18:49:06 +0000 (18:49 +0000)]
Change sprintf use in talk samples to snprintf

BUG=2301
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8128 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoCorrect GetDriveType error handling.
jlmiller@webrtc.org [Thu, 22 Jan 2015 17:44:19 +0000 (17:44 +0000)]
Correct GetDriveType error handling.

BUG=4020
R=brucedawson@google.com, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8127 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoModify some tests to never use DTX disable mode
henrik.lundin@webrtc.org [Thu, 22 Jan 2015 13:30:58 +0000 (13:30 +0000)]
Modify some tests to never use DTX disable mode

DTX disable mode will be removed as a part of the ACM redesign work.

COAUTHOR:kwiberg@webrtc.org

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8125 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoIntegrate send-side BWE into simulation framework.
stefan@webrtc.org [Thu, 22 Jan 2015 10:10:53 +0000 (10:10 +0000)]
Integrate send-side BWE into simulation framework.

BUG=4173
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8123 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoSplit packets/bytes in StreamDataCounter into RtpPacketCounter struct.
asapersson@webrtc.org [Thu, 22 Jan 2015 09:39:59 +0000 (09:39 +0000)]
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
Prepares for adding FEC bytes to the StreamDataCounter.

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

Review URL: https://webrtc-codereview.appspot.com/37579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoFix bug in thresholds for bitrate probing and adjust thresholds to allow a larger...
stefan@webrtc.org [Thu, 22 Jan 2015 09:12:23 +0000 (09:12 +0000)]
Fix bug in thresholds for bitrate probing and adjust thresholds to allow a larger dispersion and concentration for successful probes.

BUG=crbug:425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8121 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoMake iSAC SWB own its decoder
henrik.lundin@webrtc.org [Thu, 22 Jan 2015 08:16:29 +0000 (08:16 +0000)]
Make iSAC SWB own its decoder

A bug in the ACM codec database caused iSAC-swb to behave differently
from iSAC-wb and -fb. With this fix, all iSAC codecs behave the same
with respect to decoder ownership.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8120 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoFix a use-after-free when sending queued messages is aborted for blocked channel.
jiayl@webrtc.org [Thu, 22 Jan 2015 00:55:10 +0000 (00:55 +0000)]
Fix a use-after-free when sending queued messages is aborted for blocked channel.

BUG=4187
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8119 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoFix an unitialized variable warning.
andrew@webrtc.org [Wed, 21 Jan 2015 22:05:12 +0000 (22:05 +0000)]
Fix an unitialized variable warning.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35819004

Patch from Sebastien Marchand <sebmarchand@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8118 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoGN: Prepare to remove webrtc_base target
kjellander@webrtc.org [Wed, 21 Jan 2015 20:22:33 +0000 (20:22 +0000)]
GN: Prepare to remove webrtc_base target

Keep the webrtc_base target temporarily while waiting for
Chromium to pick up this revision. Then we'll update Chromium
and remove the webrtc_base target for real.

This should have been a part of https://code.google.com/p/webrtc/source/detail?r=7140

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8117 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRe-land "Support 48kHz in AEC"
aluebs@webrtc.org [Wed, 21 Jan 2015 19:10:55 +0000 (19:10 +0000)]
Re-land "Support 48kHz in AEC"

Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

Original: https://webrtc-codereview.appspot.com/28319004/
Reverted: https://webrtc-codereview.appspot.com/33949004/

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8116 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoFix TransientDetectorTest in modules_unittests on Android ARM64
aluebs@webrtc.org [Wed, 21 Jan 2015 18:01:28 +0000 (18:01 +0000)]
Fix TransientDetectorTest in modules_unittests on Android ARM64

BUG=webrtc:4200
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8115 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoDisable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.
minyue@webrtc.org [Wed, 21 Jan 2015 14:22:39 +0000 (14:22 +0000)]
Disable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_voip on ARM64.

BUG=4199
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8114 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoChange CreateOrGetReportBlockInformation to have one return path.
asapersson@webrtc.org [Wed, 21 Jan 2015 13:07:04 +0000 (13:07 +0000)]
Change CreateOrGetReportBlockInformation to have one return path.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoSimplify and guard access to WindowsRealTimeClock.
pbos@webrtc.org [Wed, 21 Jan 2015 12:51:13 +0000 (12:51 +0000)]
Simplify and guard access to WindowsRealTimeClock.

Addresses data race in get_time() causing incorrect timer roll-over
detection. This roll-over caused NTP timers to jump by 2^32
milliseconds affecting A/V sync and end-to-end delay calculations.

BUG=4206
R=dvyukov@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8112 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoUpdate StatsReport and by extension StatsCollector to reduce data copying.
tommi@webrtc.org [Wed, 21 Jan 2015 11:36:18 +0000 (11:36 +0000)]
Update StatsReport and by extension StatsCollector to reduce data copying.

Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.

BUG=2822
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRemove unnecessary dependencies from webrtc_all target.
kjellander@webrtc.org [Wed, 21 Jan 2015 10:06:55 +0000 (10:06 +0000)]
Remove unnecessary dependencies from webrtc_all target.

The xmllite and xmpp dependencies are pulled in when include_tests==1
but I need to be able to do a build without processing them
having include_tests==0.

BUG=4185
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8109 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoOnly report fraction of lost packets if report_block_stats has been updated.
asapersson@webrtc.org [Wed, 21 Jan 2015 09:00:19 +0000 (09:00 +0000)]
Only report fraction of lost packets if report_block_stats has been updated.

R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8108 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoIndentation changes.
asapersson@webrtc.org [Wed, 21 Jan 2015 08:22:50 +0000 (08:22 +0000)]
Indentation changes.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoCorrect the class name in peerconnection_jni.cc.
braveyao@webrtc.org [Wed, 21 Jan 2015 07:57:06 +0000 (07:57 +0000)]
Correct the class name in peerconnection_jni.cc.

BUG=4194
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8106 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoUpdate libjingle license statements at top of talk files for consistency
jlmiller@webrtc.org [Tue, 20 Jan 2015 21:36:13 +0000 (21:36 +0000)]
Update libjingle license statements at top of talk files for consistency

BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoBump to version 41.
tnakamura@webrtc.org [Tue, 20 Jan 2015 18:52:01 +0000 (18:52 +0000)]
Bump to version 41.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8104 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoSetting Opus target application.
minyue@webrtc.org [Tue, 20 Jan 2015 16:01:50 +0000 (16:01 +0000)]
Setting Opus target application.

This CL is to allow to set Opus target application at the creation of an encoder.

According to Opus spec, there are three applications:

OPUS_APPLICATION_VOIP
OPUS_APPLICATION_AUDIO
OPUS_APPLICATION_RESTRICTED_LOWDELAY

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

Review URL: https://webrtc-codereview.appspot.com/37479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8103 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoMove internal capture+render to build_with_chromium==0 condition
kjellander@webrtc.org [Tue, 20 Jan 2015 11:40:45 +0000 (11:40 +0000)]
Move internal capture+render to build_with_chromium==0 condition

This will avoid errors related to DirectX not being found
for Chromium bots (mainly GN, but it's safest to do the same
changes for GYP since they also make sense there as GYP generation
go slightly faster without having to process those targets).

Thanks to vchigrin@yandex-team.ru for originally suggesting
this being fixed in
https://webrtc-codereview.appspot.com/37639004/

TESTED=
Successfully ran:
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dbuild_with_chromium=1
and trybots.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8102 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRoll chromium_revision a6eafec..c086b4e
kjellander@webrtc.org [Tue, 20 Jan 2015 11:39:27 +0000 (11:39 +0000)]
Roll chromium_revision a6eafec..c086b4e

Relevant changes:
* src/testing/gtest: 8245545..be18681
* src/tools/gyp: 82b0804..194ec65
* src/tools/swarming_client: c44f572..0a795bd
Details: https://chromium.googlesource.com/chromium/src/+/a6eafec..c086b4e/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8101 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRevert 8080 "Support 48kHz in AEC"
tina.legrand@webrtc.org [Tue, 20 Jan 2015 10:22:49 +0000 (10:22 +0000)]
Revert 8080 "Support 48kHz in AEC"

> Support 48kHz in AEC
>
> Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
> Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.
>
> BUG=webrtc:3146
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28319004

TBR=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8100 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRemove webrtc/base/compile_assert.h
kwiberg@webrtc.org [Tue, 20 Jan 2015 08:46:55 +0000 (08:46 +0000)]
Remove webrtc/base/compile_assert.h

It was previously removed as part of r8058, and reinstated in r8064
because of outside dependencies. Those dependencies have now been
dealt with, so the removal should stick this time.

R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8099 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoCleanup for Rtp Rtcp API test.
changbin.shao@intel.com [Tue, 20 Jan 2015 05:42:52 +0000 (05:42 +0000)]
Cleanup for Rtp Rtcp API test.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8098 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoUpdate StatsCollector's interface in preparation of more changes.
tommi@webrtc.org [Mon, 19 Jan 2015 20:41:26 +0000 (20:41 +0000)]
Update StatsCollector's interface in preparation of more changes.

This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8095

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRevert 8095 "Update StatsCollector's interface in preparation of..."
tommi@webrtc.org [Mon, 19 Jan 2015 17:34:23 +0000 (17:34 +0000)]
Revert 8095 "Update StatsCollector's interface in preparation of..."

> Update StatsCollector's interface in preparation of more changes.
>
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
>
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
>
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
>
> BUG=2822
> R=perkj@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/36829004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoUpdate StatsCollector's interface in preparation of more changes.
tommi@webrtc.org [Mon, 19 Jan 2015 16:49:33 +0000 (16:49 +0000)]
Update StatsCollector's interface in preparation of more changes.

This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoAdd UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000...
stefan@webrtc.org [Mon, 19 Jan 2015 15:44:47 +0000 (15:44 +0000)]
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.

The previous CL was reverted for two reasons:
- Added a static initializer because std::string.
- Landed before the corresponding chromium CL, which has now been landed.

BUG=crbug:425925
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8094 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoFixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.
phoglund@webrtc.org [Mon, 19 Jan 2015 13:57:59 +0000 (13:57 +0000)]
Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test.

The was was really, really difficult to run before because you needed
a custom env with both LD_PRELOAD and library path. Now the script will
set up the correct library path and be more transparent about what it
requires.

BUG=None
TESTED=locally
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8093 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoAdding TRYSERVER_PROJECT to codereview.settings.
kjellander@webrtc.org [Mon, 19 Jan 2015 13:51:59 +0000 (13:51 +0000)]
Adding TRYSERVER_PROJECT to codereview.settings.

Recent infra changes makes this being needed to
trigger tryjobs from Rietveld.

TBR=sergiyb@chromium.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8092 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoAdd /talk/examples/androidtests/{bin,gen} to .gitignore.
kjellander@webrtc.org [Mon, 19 Jan 2015 12:52:43 +0000 (12:52 +0000)]
Add /talk/examples/androidtests/{bin,gen} to .gitignore.

TBR=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8091 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoDisable tests failing on Android ARM64 (Nexus9).
kjellander@webrtc.org [Mon, 19 Jan 2015 12:46:01 +0000 (12:46 +0000)]
Disable tests failing on Android ARM64 (Nexus9).

BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.

Review URL: https://webrtc-codereview.appspot.com/33919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoDisable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.
sprang@webrtc.org [Mon, 19 Jan 2015 12:06:35 +0000 (12:06 +0000)]
Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan.

Tests are flaky on tsan, disabling for now.

BUG=4135
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8089 4adac7df-926f-26a2-2b94-8c16560cd09d

5 weeks agoRemove unused private data member engine_id_
tommi@webrtc.org [Mon, 19 Jan 2015 07:54:29 +0000 (07:54 +0000)]
Remove unused private data member engine_id_

BUG=chromium:447445
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8088 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agorelease the turn allocation by sending a refresh request with lifetime 0
pthatcher@webrtc.org [Sat, 17 Jan 2015 00:58:15 +0000 (00:58 +0000)]
release the turn allocation by sending a refresh request with lifetime 0

BUG=406578

Patch originally from philipp.hancke@googlemail.com

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8087 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRe-enable the messagequeue unittests. These were commented out at one point but never...
decurtis@webrtc.org [Fri, 16 Jan 2015 17:52:53 +0000 (17:52 +0000)]
Re-enable the messagequeue unittests. These were commented out at one point but never reenabled.

R=hellner@chromium.org, henrike@webrtc.org
CC=juberti@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8086 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRevert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500...
stefan@webrtc.org [Fri, 16 Jan 2015 13:52:52 +0000 (13:52 +0000)]
Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps."

TBR=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8085 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRemove unnecessary remote bitrate estimator build rule which serves no purpose.
andresp@webrtc.org [Fri, 16 Jan 2015 07:50:17 +0000 (07:50 +0000)]
Remove unnecessary remote bitrate estimator build rule which serves no purpose.

BUG=4185
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8084 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoAdd stats collection for the data channel.
decurtis@webrtc.org [Thu, 15 Jan 2015 22:55:07 +0000 (22:55 +0000)]
Add stats collection for the data channel.

BUG=1805
R=bemasc@chromium.org, hta@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoFixes reference counting problem when a TransportProxy points to a Transport prior...
decurtis@webrtc.org [Thu, 15 Jan 2015 22:53:49 +0000 (22:53 +0000)]
Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels.

Until the TransportProxy enters the "negotiated" state we only create
ChannelImpls but we don't hook up to them. However, we still neeed to
reserve their spot and increment the reference count.

Once we are negotiated we can hook all the ChannelProxy's to the
corresponding ChannelImpls.

This change is needed to implement maxbundle.

BUG=1574
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8082 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoUpdate AppRTCDemo UI.
tkchin@webrtc.org [Thu, 15 Jan 2015 22:38:21 +0000 (22:38 +0000)]
Update AppRTCDemo UI.

- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoSupport 48kHz in AEC
aluebs@webrtc.org [Thu, 15 Jan 2015 19:52:05 +0000 (19:52 +0000)]
Support 48kHz in AEC

Doing something similar for the band 16-24kHz to what is done for the band 8-16kHz. The only difference is that there is no comfort noise added in this band. Could not test how this sounds because there are no aecdumps with 48kHz sample rate as nfar as I know.
Tested for 32kHz sample rate and the output is bitexact with how it was before this CL.

BUG=webrtc:3146
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8080 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoFix a case where empty candidate id is used
guoweis@webrtc.org [Thu, 15 Jan 2015 18:52:36 +0000 (18:52 +0000)]
Fix a case where empty candidate id is used

BUG=4161
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8071

Review URL: https://webrtc-codereview.appspot.com/35749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8079 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoOnly adapt AGC when the desired signal is present
aluebs@webrtc.org [Thu, 15 Jan 2015 18:07:21 +0000 (18:07 +0000)]
Only adapt AGC when the desired signal is present

Take the 50% quantile of the mask and compare it to certain threshold to determine if the desired signal is present. A hold is applied to avoid fast switching between states.
is_signal_present_ has been plotted and looks as expected. The AGC adaptation sounds promising, specially for the cases when the speaker fades in and out from the beam direction.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28329005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8078 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoAdd UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000...
stefan@webrtc.org [Thu, 15 Jan 2015 14:45:27 +0000 (14:45 +0000)]
Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps.

BUG=crbug:425925
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8076 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoLog configs when creating video streams in Call.
pbos@webrtc.org [Thu, 15 Jan 2015 10:09:39 +0000 (10:09 +0000)]
Log configs when creating video streams in Call.

Adds VideoReceiveStream::Config::ToString and logs configs in both
Call::CreateVideoSendStream and Call::CreateVideoReceiverStream.

R=mflodman@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/41519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8075 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRemove dual stream functionality in ACM
henrik.lundin@webrtc.org [Thu, 15 Jan 2015 09:36:30 +0000 (09:36 +0000)]
Remove dual stream functionality in ACM

This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoClean unnecessary workaround for chromium import.
andresp@webrtc.org [Thu, 15 Jan 2015 09:12:45 +0000 (09:12 +0000)]
Clean unnecessary workaround for chromium import.

BUG=4185
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8073 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoAdd percentage of fec packets and recovered media packets to histogram stats:
asapersson@webrtc.org [Thu, 15 Jan 2015 07:40:20 +0000 (07:40 +0000)]
Add percentage of fec packets and recovered media packets to histogram stats:
- "WebRTC.Video.ReceivedFecPacketsInPercent"
- "WebRTC.Video.RecoveredMediaPacketsInPercentOfFec"

BUG=crbug/419657
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8072 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoFix a case where empty candidate id is used
guoweis@webrtc.org [Thu, 15 Jan 2015 06:53:07 +0000 (06:53 +0000)]
Fix a case where empty candidate id is used

BUG=4161
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoAdd WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.
andrew@webrtc.org [Thu, 15 Jan 2015 02:56:06 +0000 (02:56 +0000)]
Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version.

This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.

Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/36689004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoAdd beamforming to audioproc_float utility.
mgraczyk@chromium.org [Thu, 15 Jan 2015 01:28:36 +0000 (01:28 +0000)]
Add beamforming to audioproc_float utility.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8069 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoMove ring_buffer to common_audio.
andrew@webrtc.org [Thu, 15 Jan 2015 00:09:53 +0000 (00:09 +0000)]
Move ring_buffer to common_audio.

In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.

R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoAdd BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possi...
pthatcher@webrtc.org [Wed, 14 Jan 2015 23:19:06 +0000 (23:19 +0000)]
Add BundlePolicy to RTCConfiguration.  Don't change any behavior.  Just make it possible to make progress in Chromium while we work on the behavior.

R=decurtis@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoFix searching for DirectX SDK during GN build.
kjellander@webrtc.org [Wed, 14 Jan 2015 21:25:25 +0000 (21:25 +0000)]
Fix searching for DirectX SDK during GN build.

Before that GN just checked for DXSDK_DIR environment variable.
GYP does more and checks registry, let's do the same in GN.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37599004

Patch from Vyacheslav Chigrin <vchigrin@yandex-team.ru>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8066 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRemove WebRtcVideoEncoderFactory2.
pbos@webrtc.org [Wed, 14 Jan 2015 17:29:27 +0000 (17:29 +0000)]
Remove WebRtcVideoEncoderFactory2.

This interface is no longer required and just adds complexity.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/33009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRevert removing of compile_assert.h.
turaj@webrtc.org [Wed, 14 Jan 2015 17:17:11 +0000 (17:17 +0000)]
Revert removing of compile_assert.h.

In https://webrtc-codereview.appspot.com/39469004 compile_assert.h is removed and that resulted in some bots to break. There is a pending CL to fix the issue https://chromereviews.googleplex.com/141837013/
, meanwhile I revert this change.

TBR=kwiberg@google.com

Review URL: https://webrtc-codereview.appspot.com/35779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8064 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoExclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.
kjellander@webrtc.org [Wed, 14 Jan 2015 17:00:15 +0000 (17:00 +0000)]
Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.

This test is too slow to execute:
[ RUN      ] EndToEndTest.SendsAndReceivesH264
e:\b\build\slave\win-drmem\build\src\webrtc\video\end_to_end_tests.cc(287): error: Value of: Wait()
  Actual: 3
Expected: kEventSignaled
Which is: 1
Timed out while waiting for enough frames to be decoded.
[  FAILED  ] EndToEndTest.SendsAndReceivesH264 (72812 ms)

BUG=3159
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8063 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoImproved fairness simulation by starting the flows 20 seconds apart.
stefan@webrtc.org [Wed, 14 Jan 2015 16:45:29 +0000 (16:45 +0000)]
Improved fairness simulation by starting the flows 20 seconds apart.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8062 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoImplement SimulcastEncoderAdapter support.
pbos@webrtc.org [Wed, 14 Jan 2015 16:26:23 +0000 (16:26 +0000)]
Implement SimulcastEncoderAdapter support.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/37589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRemove dual stream functionality in VoiceEngine
henrik.lundin@webrtc.org [Wed, 14 Jan 2015 16:07:26 +0000 (16:07 +0000)]
Remove dual stream functionality in VoiceEngine

This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRemove RTX SSRC when deleting the default receive stream.
mflodman@webrtc.org [Wed, 14 Jan 2015 15:07:07 +0000 (15:07 +0000)]
Remove RTX SSRC when deleting the default receive stream.

BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoRemove COMPILE_ASSERT and use static_assert everywhere
kwiberg@webrtc.org [Wed, 14 Jan 2015 10:51:54 +0000 (10:51 +0000)]
Remove COMPILE_ASSERT and use static_assert everywhere

COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.

R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d

6 weeks agoMove system_wrappers.gyp files to the proper directory.
andresp@webrtc.org [Wed, 14 Jan 2015 09:30:52 +0000 (09:30 +0000)]
Move system_wrappers.gyp files to the proper directory.

Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d