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Author SHA1 Message Date
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. 
- 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. 
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. 
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." 
> 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 
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
minyue@webrtc.org
e54a6342dd 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 08:47:25 +00:00
guoweis@webrtc.org
8c9ff203c5 Redo the change of https://webrtc-codereview.appspot.com/30949004/ 
The previous change causes a build issue as there is subclass of TransportChannel in chromium. To break the circular dependency, a stub of implementation for GetState() is provided and will be removed once the jingle_glue::MockTransportChannel has the function defined.

TBR=pthatcher@webrtc.org

BUG=411086

Review URL: https://webrtc-codereview.appspot.com/34369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 07:56:02 +00:00
guoweis@webrtc.org
fd8422938c Revert "Implement GetState() for channel's connectivity check state." 
This reverts commit ff72f9e692.

TBR=pthatcher@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/33469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 00:51:59 +00:00
guoweis@webrtc.org
ff72f9e692 Implement GetState() for channel's connectivity check state. 
Previously, IceState is considered completed when there is only one connection (and the rest was trimmed). However, since the trimming logic is only done within the scope of network, when IPv6 and IPv4 both exist, the completion event is never fired.

This change adds the GetState() to each channel and it could decide what Completion means. The transport object then aggregates all channels before determining it's completed.

Each channel's IceState will be aggregrated at Transport level for overall Ice state

BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 00:14:49 +00:00
andrew@webrtc.org
fd4acf6d55 Adding WebRtcSpl_MaxAbsValueW16 intrinsics version 
The modification only uses the unique part of the WebRtcSpl_MaxAbsValue
 function. Pass Spltest.MinMaxOperationTest conformance test on both
 ARMv7 and ARM64. And the single function performance is similar with
 original assembly version on different platforms. If not specified, the
 code is compiled by GCC 4.6. The result is the "X version / C version"
 ratio, and the less is better.

| run 100k times             | cortex-a7 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |   (1.7Ghz) |
| CPU target                 |           |            |
|----------------------------+-----------+------------|
| Neon asm                   |       32% |        15% |
| Neon intrinsics (GCC 4.6)  |       36% |        37% |
| Neon intrinsics (GCC 4.8)  |       35% |        18% |

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Ia2f6822ec58774b401cc440b6751a97e540b5048

Review URL: https://webrtc-codereview.appspot.com/30109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 21:59:02 +00:00
andrew@webrtc.org
3a52458237 add WebRtcIsacfix_AutocorrNeon's intrinsics version 
The modification only uses the unique part of the
 WebRtcIsacfix_AutocorrC function. Pass FiltersTest.AutocorrFixTest test
 on both ARMv7 and ARM64, and the single function performance is similar
 with original assembly version on different platforms. If not
 specified, the code is compiled by GCC 4.6. The result is the "X
 version / C version" ratio, and the less is better.

| run 100k times             | cortex-a7 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |   (1.7Ghz) |
| CPU target                 |           |            |
|----------------------------+-----------+------------|
| Neon asm                   |       24% |        23% |
| Neon intrinsics (GCC 4.6)  |       33% |        32% |
| Neon intrinsics (GCC 4.8)  |       27% |        27% |

BUG=3850
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id6cd0671502fadbebd10b1f5493f5b16c988286f

Review URL: https://webrtc-codereview.appspot.com/27999004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 21:58:18 +00:00
henrik.lundin@webrtc.org
8dc21dc238 Rename internal AudioEncoder::Encode method to EncodeInternal 
BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 20:36:03 +00:00
andrew@webrtc.org
d1fac61e8f Remove need for assembly offset generation in aecm and ns module. 
All *neon.S files in aecm and ns modules have been removed. We need no
assembly offset generation now.

Pass byte to byte conformance test for aecm and ns test in audioproc
between new NEON (written in intrinsics) version and C version on both
ARMv7 and ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I05d43d0c04d00bead65ca8c8fda25f0a42394b2b

Review URL: https://webrtc-codereview.appspot.com/32229004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 17:54:38 +00:00
kwiberg@webrtc.org
3800e13a3a Revert r7798 ("Move the AudioDecoder interface out of NetEq") 
Apparently, it caused all sorts of problems I don't have time to
straighten out right now.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 16:28:17 +00:00
kwiberg@webrtc.org
00ba1a7dfd Move the AudioDecoder interface out of NetEq 
It belongs with the codecs, next to the AudioEncoder interface.

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

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 14:23:23 +00:00
pbos@webrtc.org
0fb6ad2004 Check if cpu_monitor_ exists before Stop(). 
R=asapersson@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 13:44:29 +00:00
henrik.lundin@webrtc.org
fa914e283c Adding a duration printout to neteq_rtpplay 
BUG=2692
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 13:28:53 +00:00
asapersson@webrtc.org
d8aed6b321 Verify that cpu_monitor exists before calling Stop(). 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 12:37:47 +00:00
kjellander@webrtc.org
c3e097cdc5 Add Android test runner script for WebRTC. 
The Android test execution toolchain scripts in Chromium
has been causing headaches for us several times. Mostly
because they're tailored at running Chrome tests only.

Wrapping their script in our own avoids the pain of
upstreaming new test names to Chromium and rolling them
in to get them running on our bots.

TESTED=Ran a test on a local device using:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests  --verbose --isolate-file-path webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate --release
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 09:57:08 +00:00
kjellander@webrtc.org
8e5c814ef0 Convert DEPS to only reference Git repos 
Also replace all doublequoted Python strings
with single-quoted ones.

BUG=chromium:412012
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 07:11:44 +00:00
jiayl@webrtc.org
511f8a8ef2 TurnPort should ignore STUN binding reponses when using shared socket. 
BUG=4043
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 02:17:07 +00:00
marpan@webrtc.org
001f3b9818 Adjust parameter in videoprocessor_integration_test for vp9. 
TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 02:00:12 +00:00
aluebs@webrtc.org
a7384a1126 Simplify audio_buffer APIs 
Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 01:06:35 +00:00
marpan@webrtc.org
ceca014b8b Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9. 
BUG=4059

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 01:05:43 +00:00
pthatcher@webrtc.org
eb0954248d Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 00:34:10 +00:00
glaznev@webrtc.org
d01955179a Change minimum video encoder initialization resolution to 
176x144 to ensure HW encoder can be initialized.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 23:41:18 +00:00