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Author SHA1 Message Date
kjellander@webrtc.org
acb8085678 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: 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
 2015-01-25 19:17:56 +00:00
tkchin@webrtc.org
7519de519e 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
 2015-01-23 21:20:41 +00:00
jiayl@webrtc.org
dacdd9403d 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
 2015-01-23 17:33:34 +00:00
fdegans@chromium.org
8919cfe9ce 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
 2015-01-23 16:35:17 +00:00
minyue@webrtc.org
456f01441a 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
 2015-01-23 11:58:42 +00:00
magjed@webrtc.org
182ea46fac 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
 2015-01-23 11:50:13 +00:00
stefan@webrtc.org
73ee4537be 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
 2015-01-23 08:29:52 +00:00
davidben@webrtc.org
36d5c3cb44 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
 2015-01-22 23:06:17 +00:00
tommi@webrtc.org
586f2eda0d 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
 2015-01-22 23:00:41 +00:00
jiayl@webrtc.org
7e5b380437 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
 2015-01-22 21:28:39 +00:00
kjellander@webrtc.org
ff108fe508 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
 2015-01-22 19:02:03 +00:00
jlmiller@webrtc.org
ea1c84285c 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
 2015-01-22 17:44:19 +00:00
henrik.lundin@webrtc.org
043db24767 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
 2015-01-22 13:30:58 +00:00
stefan@webrtc.org
e5251ad63c 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
 2015-01-22 10:10:53 +00:00
asapersson@webrtc.org
cfd82dfc11 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
 2015-01-22 09:39:59 +00:00
stefan@webrtc.org
3dd33a6787 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
 2015-01-22 09:12:23 +00:00
henrik.lundin@webrtc.org
fbd37bd737 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
 2015-01-22 08:16:29 +00:00
andrew@webrtc.org
e65d9d974c 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
 2015-01-21 22:05:12 +00:00
kjellander@webrtc.org
c429b824b3 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
 2015-01-21 20:22:33 +00:00
aluebs@webrtc.org
c78d81ae89 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
 2015-01-21 19:10:55 +00:00
aluebs@webrtc.org
e81c5d6d7e 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
 2015-01-21 18:01:28 +00:00
minyue@webrtc.org
11af039590 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
 2015-01-21 14:22:39 +00:00
asapersson@webrtc.org
df7b65ba01 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
 2015-01-21 13:07:04 +00:00
pbos@webrtc.org
f938922c5c 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
 2015-01-21 12:51:13 +00:00
kjellander@webrtc.org
f66a6b2a00 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
 2015-01-21 10:06:55 +00:00
asapersson@webrtc.org
e7358eabbc 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
 2015-01-21 09:00:19 +00:00
asapersson@webrtc.org
9ffd8fe96b 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
 2015-01-21 08:22:50 +00:00
tnakamura@webrtc.org
cbacd9e3bf 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
 2015-01-20 18:52:01 +00:00
minyue@webrtc.org
7dba7860c7 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
 2015-01-20 16:01:50 +00:00
kjellander@webrtc.org
853049fa30 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
 2015-01-20 11:40:45 +00:00
tina.legrand@webrtc.org
ee0c100d54 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
 2015-01-20 10:22:49 +00:00
kwiberg@webrtc.org
f88f88edde 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
 2015-01-20 08:46:55 +00:00
changbin.shao@intel.com
9691b36995 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
 2015-01-20 05:42:52 +00:00
stefan@webrtc.org
474e36e623 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
 2015-01-19 15:44:47 +00:00
kjellander@webrtc.org
a32d15448d 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
 2015-01-19 12:46:01 +00:00
tommi@webrtc.org
2624b1ed23 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
 2015-01-19 07:54:29 +00:00
pthatcher@webrtc.org
fe672e3839 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
 2015-01-17 00:58:15 +00:00
decurtis@webrtc.org
d7de1209ae 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
 2015-01-16 17:52:53 +00:00
stefan@webrtc.org
a1aea10af2 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
 2015-01-16 13:52:52 +00:00
andresp@webrtc.org
4ba1e44ff0 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
 2015-01-16 07:50:17 +00:00
decurtis@webrtc.org
357469da5a 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
 2015-01-15 22:53:49 +00:00
aluebs@webrtc.org
64d3c4b9ac 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
 2015-01-15 19:52:05 +00:00
guoweis@webrtc.org
89aa276e2e 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
 2015-01-15 18:52:36 +00:00
aluebs@webrtc.org
d82f55d2a7 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
 2015-01-15 18:07:21 +00:00
stefan@webrtc.org
3e42a8a56a 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
 2015-01-15 14:45:27 +00:00
pbos@webrtc.org
32e8528581 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
 2015-01-15 10:09:39 +00:00
henrik.lundin@webrtc.org
1f67b53c88 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
 2015-01-15 09:36:30 +00:00
andresp@webrtc.org
9ce01e6416 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
 2015-01-15 09:12:45 +00:00
asapersson@webrtc.org
0800db74b9 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
 2015-01-15 07:40:20 +00:00
guoweis@webrtc.org
61c1247224 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
 2015-01-15 06:53:07 +00:00