107 Commits

Author SHA1 Message Date
Peter Kasting
dce40cf804 Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
Peter Thatcher
c2ee2c86f9 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well.
R=deadbeef@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1229283003 .

Cr-Commit-Position: refs/heads/master@{#9690}
2015-08-07 23:05:42 +00:00
pbos
8fc7fa798f Base A/V synchronization on sync_labels.
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.

BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1181653002

Cr-Commit-Position: refs/heads/master@{#9586}
2015-07-15 15:03:04 +00:00
Henrik Lundin
441f634731 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)

The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.

Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec."

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1151573021.

Cr-Commit-Position: refs/heads/master@{#9401}
2015-06-09 14:03:23 +00:00
Henrik Lundin
3fbf3f8841 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.

BUG=4696
TBR=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1166463006

Cr-Commit-Position: refs/heads/master@{#9380}
2015-06-05 09:04:20 +00:00
Henrik Lundin
5f4b7e2873 Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec.

BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54659004

Cr-Commit-Position: refs/heads/master@{#9378}
2015-06-05 07:55:40 +00:00
Jelena Marusic
c28a896a7b VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
BUG=4690

Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code

R=solenberg@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56499004

Cr-Commit-Position: refs/heads/master@{#9330}
2015-05-29 13:05:52 +00:00
Tommi
76b62ff1ad Clean up now-unused code that was used for libpeerconnection.[so|dll].
BUG=chromium:463660
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56409004

Cr-Commit-Position: refs/heads/master@{#9240}
2015-05-20 20:36:42 +00:00
Fredrik Solenberg
ccb49e79fd Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51009004

Cr-Commit-Position: refs/heads/master@{#9216}
2015-05-19 09:37:39 +00:00
Henrik Lundin
64dad838e6 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49329004

Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
Henrik Lundin
1f629232d5 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55369004

Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
Henrik Lundin
fd32f35aff Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.

Contains a tentative fix to the chrome build breakage caused by the
original change.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47139004

Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
Henrik Lundin
cdb47a4533 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53399004

Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
Henrik Lundin
208a2294cd Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
Fredrik Solenberg
e444a3dcd3 WebRtcVoiceEngine: Get rid of unnecessary template base class.
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46219004

Cr-Commit-Position: refs/heads/master@{#9155}
2015-05-07 14:42:12 +00:00
Fredrik Solenberg
aaf8ff2e45 WebRtcVoiceEngine: virtual to override + git cl format.
BUG=
R=kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54369004

Cr-Commit-Position: refs/heads/master@{#9154}
2015-05-07 14:05:57 +00:00
Fredrik Solenberg
6179b89e53 Remove unused API on WebRtcVoiceEngine.
BUG=1695
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46209004

Cr-Commit-Position: refs/heads/master@{#9153}
2015-05-07 14:01:30 +00:00
Fredrik Solenberg
4b60c73e74 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49269004

Cr-Commit-Position: refs/heads/master@{#9150}
2015-05-07 12:07:46 +00:00
Karl Wiberg
9478437fde rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
     int8_t*, and char*. Previously, they accepted void*, meaning that
     any kind of pointer was accepted. I think requiring an explicit
     cast in cases where the input array isn't already of a byte-sized
     type is a better compromise between convenience and safety.

  2. data() can now return a uint8_t* instead of a char*, which seems
     more appropriate for a byte array, and is harder to mix up with
     zero-terminated C strings. data<int8_t>() is also available so
     that callers that want that type instead won't have to cast, as
     is data<char>() (which remains the default until all existing
     callers have been fixed).

  3. Constructors, SetData(), and AppendData() now accept arrays
     natively, not just decayed to pointers. The advantage of this is
     that callers don't have to pass the size separately.

  4. There are new constructors that allow setting size and capacity
     without initializing the array. Previously, this had to be done
     separately after construction.

  5. Instead of TransferTo(), Buffer now supports swap(), and move
     construction and assignment, and has a Pass() method that works
     just like std::move(). (The Pass method is modeled after
     scoped_ptr::Pass().)

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

Review URL: https://webrtc-codereview.appspot.com/42989004

Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +00:00
Bjorn Volcker
bf395c1fc0 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.

This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.

The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/

BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699004

Cr-Commit-Position: refs/heads/master@{#8861}
2015-03-25 21:46:10 +00:00
pbos@webrtc.org
8296ec518b Fix heap-use-after-free in WebRtcVideoEngine2.
Found in libjingle_peerconnection_unittest on asan while trying to
default-enable WebRtcVideoEngine2.

BUG=1788
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44779004

Cr-Commit-Position: refs/heads/master@{#8808}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8808 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-20 14:28:31 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths.
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
henrik.lundin@webrtc.org
8038d42749 Follow-up fixes for G722
This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.

BUG=3951
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-11 08:38:24 +00:00
henrik.lundin@webrtc.org
f85dbce041 Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.

BUG=3951
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 12:25:00 +00:00
henrik.lundin@webrtc.org
dced5d7835 Revert "Advertise G722 as 8 kHz rather than 16 kHz"
This reverts r7645.

TBR=pthatcher@webrtc.org
BUG=3951

Review URL: https://webrtc-codereview.appspot.com/24199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 15:27:43 +00:00
henrik.lundin@webrtc.org
1dcca4028f Advertise G722 as 8 kHz rather than 16 kHz
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.

R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.

Review URL: https://webrtc-codereview.appspot.com/27879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 08:55:01 +00:00
minyue@webrtc.org
2623695dfb Renaming bandwidth to bitrate in webrtcvoiceengine.
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.

This is to remove the confusion inside webrtcvoiceengine

BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 02:27:08 +00:00
xians@webrtc.org
3cefbc99f4 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
This also marks all virtual overrides of other classes in the same files. 

This will make a subsequent change I intend to do safer, where I'll change the 
argument types of the base Transport functions, by breaking the compile if I 
miss any overrides. 

This also highlighted a number of unused functions. I've removed some of these. 

TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none 
TEST=none

Review URL: https://webrtc-codereview.appspot.com/28709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
buildbot@webrtc.org
97abeee282 (Auto)update libjingle 77263371-> 77296420
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 22:24:30 +00:00
buildbot@webrtc.org
1f8a23757a (Auto)update libjingle 74235596-> 74297316
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6997 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 10:52:44 +00:00
buildbot@webrtc.org
b4c7b09c13 (Auto)update libjingle 73927775-> 74032598
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-25 12:11:58 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 17:26:08 +00:00
buildbot@webrtc.org
6b21b71068 (Auto)update libjingle 72205295-> 72320533
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 15:08:53 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
buildbot@webrtc.org
0d15159b04 (Auto)update libjingle 69634309-> 69640360
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6512 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 19:02:09 +00:00
buildbot@webrtc.org
8563ef448a (Auto)update libjingle 69587333-> 69588608
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 00:13:01 +00:00
buildbot@webrtc.org
150835ea34 (Auto)update libjingle 66236292-> 66294299
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-06 15:54:38 +00:00
buildbot@webrtc.org
0d34f1446a (Auto)update libjingle 66033941-> 66098243
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-02 16:54:25 +00:00
buildbot@webrtc.org
af6640fce7 (Auto)update libjingle 65729829-> 65752960
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 21:31:51 +00:00
henrike@webrtc.org
79047f99c1 (Auto)update libjingle 62691533-> 62713454
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5653 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 23:46:59 +00:00
mallinath@webrtc.org
67ee6b9a62 Update talk to 60923971
Review URL: https://webrtc-codereview.appspot.com/7909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-03 16:57:16 +00:00
wu@webrtc.org
a8910d2f88 Update talk to 60094938.
Review URL: https://webrtc-codereview.appspot.com/7489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-23 22:12:45 +00:00
sergeyu@chromium.org
4b26e2eee3 Update libjingle to 59676287
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-15 23:15:54 +00:00
wu@webrtc.org
a9890800e0 Update talk to 58127566 together with
https://webrtc-codereview.appspot.com/5309005/.

R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 00:21:03 +00:00
wu@webrtc.org
2018269dc3 Revert 5274 "Update talk to 58113193 together with https://webrt..."
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
> 
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5719004

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:54:25 +00:00
wu@webrtc.org
a129b6cd13 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-12 22:40:39 +00:00
sergeyu@chromium.org
5bc25c41fc Update libjingle to 57692857
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 00:24:06 +00:00
wu@webrtc.org
1d1ffc9ad2 Update talk to 54898858.
TEST=try bots
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2414004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4979 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 18:12:02 +00:00
wu@webrtc.org
4551b793de Update libjingle to 53920541.
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2371004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4945 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-09 15:37:36 +00:00