buildbot@webrtc.org
a85307737c
(Auto)update libjingle 81702493-> 81755413
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 09:01:18 +00:00
kwiberg@webrtc.org
3cd26b677a
Revert r7858 ("DCHECK: Reference condition parameter in release builds")
...
Apparently Visual Studio is cleverer than I am at figuring out what
local variables are actually unused.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7859 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:57:14 +00:00
kwiberg@webrtc.org
3148060e61
DCHECK: Reference condition parameter in release builds
...
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as
int x = ...
DCHECK_EQ(x, 17);
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7858 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 08:45:47 +00:00
henrik.lundin@webrtc.org
ff1a3e36bd
Make an AudioEncoder subclass for comfort noise
...
BUG=3926
R=bjornv@webrtc.org , kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7857 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 07:29:08 +00:00
andrew@webrtc.org
6fd52f36db
Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon.
...
WebRtcSpl_DownsampleFastNeon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platform.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: Ic43f5452eb7e555b998b1d1e79a9e1530be5c948
Review URL: https://webrtc-codereview.appspot.com/24359004
Patch from Yang Zhang <yang.zhang@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7856 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 00:59:48 +00:00
andrew@webrtc.org
ae20d3bbce
Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon.
...
WebRtcSpl_CrossCorrelationNeon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platform.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I84f9fb953448b62da452ab8dd60e2c0628293587
Review URL: https://webrtc-codereview.appspot.com/30189004
Patch from Yang Zhang <yang.zhang@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7855 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 23:58:39 +00:00
tommi@webrtc.org
aa2c342c10
Add back a constructor to fix FYI build.
...
TBR=perkj
Review URL: https://webrtc-codereview.appspot.com/24349005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 20:23:06 +00:00
tommi@webrtc.org
5c32a84620
Attempt to fix FYI bots.
...
The FYI bots went red after https://webrtc-codereview.appspot.com/32179004/ landed.
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7853 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 19:59:27 +00:00
tkchin@webrtc.org
87776a8935
iAppRTCDemo: WebSocket based signaling.
...
Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.
BUG=
R=glaznev@webrtc.org , jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 19:32:35 +00:00
pthatcher@webrtc.org
0babb4a4e6
Fix a comment.
...
R=juberti@webrtc.org , pbos@webrtc.org , sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 19:01:45 +00:00
tommi@webrtc.org
c9d155faeb
Move implementation of types in statstypes. to its cc file.
...
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 18:18:06 +00:00
henrika@webrtc.org
a954c07ee1
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
...
BUG=4034
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +00:00
minyue@webrtc.org
19dd129c69
Revert 7846 "Adding DTX to WebRTC Opus wrapper"
...
> Adding DTX to WebRTC Opus wrapper
>
> This is a step toward adding Opus DTX support in WebRTC.
>
> Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
>
> https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
>
> We transmit the first 1-byte packet to let decoder be in-sync
>
> BUG=webrtc:1014
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/13219004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7848 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 15:11:15 +00:00
asapersson@webrtc.org
f244760827
Add histograms for receive statistics:
...
- decoded frames per second ("WebRTC.Video.DecodedFramesPerSecond")
- percentage of delayed frames to rendered ("WebRTC.Video.DelayedFramesToRenderer")
- average delay (of delayed frames) to renderer ("WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs")
BUG=crbug/419657
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 14:13:26 +00:00
minyue@webrtc.org
4321f175f1
Adding DTX to WebRTC Opus wrapper
...
This is a step toward adding Opus DTX support in WebRTC.
Note that opus_encode() returns 1 byte in case of DTX, then the packet does not need to be transmitted. See
https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__encoder.html
We transmit the first 1-byte packet to let decoder be in-sync
BUG=webrtc:1014
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7846 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 13:27:39 +00:00
tommi@webrtc.org
5c3ee4bce6
Add empty implementation file that will hold statstypes.h implementation.
...
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:47:01 +00:00
minyue@webrtc.org
1784d7cfad
Adding an codec interal CNG test in NetEq.
...
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7843 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:46:39 +00:00
pbos@webrtc.org
9115cde6c9
Merge VP8 changes.
...
R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/35389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7841 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:36:40 +00:00
kwiberg@webrtc.org
e04a93bcf5
Move the AudioDecoder interface out of NetEq
...
It belongs with the codecs, next to the AudioEncoder interface.
R=andrew@webrtc.org , henrik.lundin@webrtc.org , kjellander@webrtc.org
Previously committed here: https://code.google.com/p/webrtc/source/detail?r=7798
and reverted here: https://code.google.com/p/webrtc/source/detail?r=7799
Review URL: https://webrtc-codereview.appspot.com/27309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7839 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 10:12:53 +00:00
asapersson@webrtc.org
97d0489058
Add video send bitrates to histogram stats:
...
- total bitrate ("WebRTC.Video.BitrateSentInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateSentInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps")
- retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps")
Add retransmitted bytes to StreamDataCounters.
Change in UpdateRtpStats to also update counters for retransmitted packet.
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 09:47:53 +00:00
kjellander@webrtc.org
7ba9f27f2b
Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper.
...
This makes it possible to clean up the recipes a lot.
BUG=
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7837 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 06:46:13 +00:00
glaznev@webrtc.org
eef85387ec
Fix AppRTCDemo closing error for KK and JB Android devices.
...
- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.
R=jiayl@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 01:29:17 +00:00
stefan@webrtc.org
86b6d65ef1
Remove no longer used video codec test framework.
...
Moves one test to the vp8 unittests which might still be good to have.
Also does a bit of clean up in vp8 unittests.
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7835 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 00:02:45 +00:00
henrik.lundin@webrtc.org
8911bc52f1
Add AudioEncoder::Max10MsFramesInAPacket
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7834 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:15:55 +00:00
henrik.lundin@webrtc.org
130fef89dd
Bugfix in AudioDecoderTest
...
When the encoded frame size (L ms) was larger than 10 ms, the test would
repeat the first 10 ms L/10 times for each encoded frame. This is now
fixed.
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7833 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 21:07:59 +00:00
stefan@webrtc.org
edeea91803
Change all system clock types to int64_t in bitrate_controller.
...
They are both compared to int64_t types inside the class, and is being called
with int64_t types. Could possibly cause bugs.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 19:46:23 +00:00
henrik.lundin@webrtc.org
fcbe36a1d9
Add const qualifier to WebRtcPcm16b_Encode
...
BUG=909
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7831 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 18:26:49 +00:00
kwiberg@webrtc.org
a1ef7bfa15
ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable.
...
Ideally, this is a stopgap fix until ATTRIBUTE_UNUSED can be given a
proper definition.
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7830 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:53:10 +00:00
andrew@webrtc.org
3b3c406908
Revert 7826 "Change Android PeerConnectionUnittest to build usin..."
...
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575
> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
>
> This also add a new build target to build java PeerConnection using Chromes build macros.
>
> BUG=4031
> R=kjellander@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28189004
TBR=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:21:50 +00:00
kwiberg@webrtc.org
cb858ba397
Make an AudioEncoder subclass for iLBC
...
BUG=3926
R=henrik.lundin@webrtc.org , kjellander@google.com
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7828 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 17:11:44 +00:00
bjornv@webrtc.org
ee43263a50
Cleaned up real_fft APIs due to non-existing NEON code
...
There are NEON APIs that are not used. Cleaning that up for better overview.
BUG=3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7827 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 16:36:22 +00:00
perkj@webrtc.org
ed7824b1c0
Change Android PeerConnectionUnittest to build using Chrome macros.
...
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 15:41:01 +00:00
asapersson@webrtc.org
ba8138ba38
Change type of nack_last_time_sent_full_ from uint32_t to int64_t.
...
Could cause nack requests to be sent too frequently.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7825 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 13:29:02 +00:00
kjellander@webrtc.org
aefe61ae2a
PRESUBMIT: Add check for checkdeps.
...
Several times I've run into the problem with
presubmit crashing when uploading a CL from a checkout
where gclient sync hasn't run yet.
This will print a user friendly error message instead.
BUG=
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7824 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 13:00:30 +00:00
kjellander@webrtc.org
7db359b94a
Roll chromium_revision 24b4c73..8e72e1d
...
Relevant changes:
* src/buildtools: 6ea835d..535aff2
* src/third_party/android_tools: 4c47ef6..4f723e2
* src/third_party/boringssl/src: 69a0160..00505ec
* src/third_party/icu: 866ff69..53ecf0f
* src/third_party/libvpx: 429874c..9fbec81
Details: 24b4c73..8e72e1d
/DEPS
Clang version was not updated in this roll.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7823 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 11:48:35 +00:00
kjellander@webrtc.org
d91d359feb
PRESUBMIT: Add iOS ARM64 trybots to default set.
...
BUG=chromium:436831
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7822 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 07:05:38 +00:00
marpan@webrtc.org
fb01376eca
Adjust some parameters for VP9 tests.
...
Needed for the next/upcoming libvpx roll.
BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7821 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-08 06:25:51 +00:00
glaznev@webrtc.org
e2a9261f3e
Improve AppRTCDemo connection speed by sending all
...
http POST requests asynchronously.
R=jiayl@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 20:11:06 +00:00
kjellander@webrtc.org
bd8cc0b914
Add codereview.settings to the /talk subdirectory
...
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/talk
(which is what you get when working with the repo currently
put in Chrome's src/third_party/libjingle/source/talk).
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7819 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 13:47:37 +00:00
kjellander@webrtc.org
5af8cd77e2
Add codereview.settings to the /webrtc subdirectory
...
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/webrtc
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7818 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 13:43:35 +00:00
kjellander@webrtc.org
599e299b9d
cricket::VideoFrame int64 to int64_t.
...
Needed for successful compile of ios arm64.
BUG=3898
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30359004
Patch from Zeke Chin <tkchin@webrtc.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-05 09:42:57 +00:00
bemasc@webrtc.org
9b5467e88d
Fix assertion failure when closing data channel, and add a unit test.
...
BUG=4066
R=jiayl@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7816 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 23:16:52 +00:00
glaznev@webrtc.org
4b407aa985
Update AppRTCDemo README with information on 3-dot-apprtc server
...
and new command line arguments.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 22:42:59 +00:00
guoweis@webrtc.org
7169afd9d5
With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.
...
BUG=411086
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 17:59:29 +00:00
glaznev@webrtc.org
369746bcb8
Support new WebSocket signaling format.
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- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.
BUG=3937,3995,4041
R=jiayl@webrtc.org , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 17:28:52 +00:00
stefan@webrtc.org
0b38478885
Add support for parsing header only RTP dumps with bwe_rtp_play.
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Also adds support for printing the original_length in rtp_to_text.
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:43:49 +00:00
pbos@webrtc.org
9f79fe684a
Merge remote bitrate estimator changes.
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R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7811 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:34:06 +00:00
minyue@webrtc.org
33ccdfa1f5
Relanding r7807.
...
r7807 was reverted to be excluded from the cause of a failure.
It has been verified and can reland now.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7810 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 12:14:12 +00:00
minyue@webrtc.org
52bc4f4797
Revert 7807 "Removing unused opus wrapper APIs."
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> Removing unused opus wrapper APIs.
>
> WebRtcOpus_DecodeNew(), WebRtcOpus_DecoderInitNew() have become the APIs and are ready to replace old WebRtcOpus_Decode() and WebRtcOpus_DecoderInit().
>
> WebRtcOpus_DecodePlcMaster/Slave() are also removed.
>
> BUG=
> R=henrik.lundin@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/28139004
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 11:00:50 +00:00
kjellander@webrtc.org
c0991fe606
Roll chromium_revision 24b4c73..f27c369
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This enables 64-bit compilation for iOS.
Relevant changes:
* src/buildtools: 6ea835d..ded3294
* src/third_party/boringssl/src: 69a0160..00505ec
* src/third_party/libvpx: 429874c..64bec31
Details: 24b4c73..f27c369
/DEPS
Clang version was not updated in this roll.
BUG=chromium:436831
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7808 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 10:55:50 +00:00