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Author SHA1 Message Date
guoweis@webrtc.org
fa603981f2 Add a new class InterfaceAddress inherited from IPAddress to keep track of IPv6 Address flags. 
Skeleton put in place in Network::GetFilterIPs() which will be used to
filter addresses

BUG=3773
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 23:42:40 +00:00
brettw@chromium.org
87ff9c8efa Fix up configs applying to GN build. 
The audio_processing target didn't have the build configs applying to it which led to some logging errors.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/22339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 23:34:56 +00:00
jiayl@webrtc.org
7d4891d3f1 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7068

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:43:15 +00:00
fbarchard@google.com
a941970d4a Change explicit static cast from int to uint16_t to implicit cast of 0u. 
BUG=3663
TESTED=local windows build with VS2013.
R=harryjin@google.com, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:37:27 +00:00
brettw@chromium.org
9fe11010f7 Fix the RTC+Chromium GN build. 
LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets.

Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited).

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 19:15:33 +00:00
fbarchard@google.com
54cf1505e2 ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. 
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 18:34:53 +00:00
jiayl@webrtc.org
22406fcc9b TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7070

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 15:44:05 +00:00
houssainy@google.com
04b853b56a Bot Browser files moved to /bot/browser/ 
because android files will be a different and will need to add more files for Android.

There was a CL to move the browser files form bot/browser/ to /bot/ as i thought it will be the same files used for Android.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 14:50:09 +00:00
mallinath@webrtc.org
3d81b1b22a Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got 
reverted due to some internal compile failures.

In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.

Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093

TBR=juberti@webrtc.org
BUG=1179

Review URL: https://webrtc-codereview.appspot.com/22329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 14:38:10 +00:00
guoweis@webrtc.org
4bbd3c83a8 fix a bug in the logic when new Networks are merged. This happens when 
we have 2 networks with the same key

BUG=410554 in chromium

http://code.google.com/p/chromium/issues/detail?id=410554

Corresponding change in chromium is
https://codereview.chromium.org/536133003/

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19249005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7117 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 13:54:45 +00:00
sprang@webrtc.org
1b088ee316 More suppressions, uninitialized read in cricket::VideoFrame::Validate 
BUG=3789
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7116 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 11:50:19 +00:00
andresp@webrtc.org
4d19e05ab2 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. 
This needs to happen sooner or later as if webrtc/base/checks.h happens to be included transitively here it would collide.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 11:45:44 +00:00
pbos@webrtc.org
b420191743 Expose VideoEncoders with webrtc/video_encoder.h. 
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 10:40:56 +00:00
andrew@webrtc.org
641bda6f9c Initialize ChannelBuffer's memory to avoid uninitialized reads. 
Removed the zero out memset in this change:
https://review.webrtc.org/24469004/

assuming it was unneeded. Dr. Memory taught me that assupmtion was
invalid. linux_memcheck try runs might have caught this, if they
weren't flaking out on unrelated stuff.

TBR=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/28429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7113 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 23:11:44 +00:00
henrike@webrtc.org
8b0b21161a Revert 7093: "Implementing ICE Transports type handling in libjingle transport." 
TBR=mallinath@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/28419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 22:46:28 +00:00
brettw@chromium.org
519c9e207d Convert GN visibility to be a list. 
GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 22:45:18 +00:00
pbos@webrtc.org
7118e61669 Finish work queue in SctpDataMediaChannelTest. 
Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).

R=jiayl@webrtc.org
BUG=3608,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/28399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 21:44:07 +00:00
jiayl@webrtc.org
0e52772aa9 Fix a bot-breaking memory leak from early returning in ParseMediaDescription. 
BUG=3791
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 21:43:43 +00:00
jiayl@webrtc.org
c172320bd2 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. 
This reverts commit r7068.

TBR=kjellander@webrtc.org
BUG=2108

Review URL: https://webrtc-codereview.appspot.com/23539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:44:36 +00:00
andrew@webrtc.org
17454f79dc Add ctors to ChannelBuffer to enable copying on construction. 
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.

R=claguna@google.com

Review URL: https://webrtc-codereview.appspot.com/24469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:27:04 +00:00
buildbot@webrtc.org
fd42f9dd6f (Auto)update libjingle 74955991-> 75042522 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 19:45:36 +00:00
sprang@webrtc.org
1272ee59b3 Suppress uninitialized read warning in cricket::VideoFrame::Validate 
BUG=3789
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 14:00:38 +00:00
henrik.lundin@webrtc.org
c64246f42c Set a default speech type in iSAC wrapper 
If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.

BUG=crbug/411162
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:40:58 +00:00
henrik.lundin@webrtc.org
ed8bcd3ac5 Starting to implement the new ACM API 
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.

This is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:13:19 +00:00
houssainy@google.com
9600519147 Adding the ability to test on Chrome for Android. 
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 13:01:40 +00:00
bjornv@webrtc.org
37c39f3784 audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16 
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.

Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 11:21:56 +00:00
bjornv@webrtc.org
0d394f3609 video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16 
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 11:19:39 +00:00
houssainy@google.com
c77e4d6aef - Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices. 
- Select BotType using nodeJs terminal command.

- ping_pong.js test added.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 10:36:11 +00:00
kjellander@webrtc.org
142bb9d870 Roll chromium_revision 94532b1..ea769fd 
Summary of changes (git diff 94532b1..ea769fd DEPS):
* buildtools 2328da4..ea4dc0e
* third_party/android_tools 3186999..7fc902d
* third_party/boringssl 6c7aed0..7bdec13
* third_party/libjpeg_turbo 2ed5319..3963fbc
* third_party/libvpx 982d147..ceebbcc (r291730:291805)
* third_party/nss 90c5f9a..7b5b6ec
* third_party/usrsctp/usrsctplib e6e1833..8975bd5

BUG=3608
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 10:06:37 +00:00
stefan@webrtc.org
fe16167507 Fix RTT calculations for send-only channels. 
As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.

BUG=3781
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 08:45:25 +00:00
sprang@webrtc.org
c30e9e2300 Ignore FEC packet in stats, if it is first packet on ssrc. 
BUG=chrome:410456
R=mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 08:20:18 +00:00
kjellander@webrtc.org
6d08ca6379 GN: Prefix WebRTC specific variables with "rtc_" 
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 17:36:10 +00:00
kjellander@webrtc.org
f68cf93e1b Add video_capture_tests_apk_target 
In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.

BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 17:35:51 +00:00
mallinath@webrtc.org
7256d31d28 Implementing ICE Transports type handling in libjingle transport. 
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 04:08:44 +00:00
kjellander@webrtc.org
a781f68712 Fix rm command for class cleanup in r7091 
In https://webrtc-codereview.appspot.com/20339004
the rm command was missing 'r' for recursive mode.

TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-06 22:11:28 +00:00
kjellander@webrtc.org
9510022e1f Cleanup temporary class files for OpenSlDemo 
I've seen tryjobs failing when they shouldn't on
the Android trybots and I suspect this might have
something to do with it.

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-06 18:03:45 +00:00
thorcarpenter@google.com
cc060563f3 Remove unnecessary include from testutils.cc. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 21:19:00 +00:00
buildbot@webrtc.org
992febb997 (Auto)update libjingle 74873066-> 74873164 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:39:08 +00:00
thorcarpenter@google.com
a3344cfda4 Fix webrtcvideoframe tests. 
R=fbarchard@google.com, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:34:13 +00:00
jiayl@webrtc.org
ddb85ab85b Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added

BUG=3592
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:31:56 +00:00
henrik.lundin@webrtc.org
8f073c5054 Create a new interface for AudioCodingModule 
This is a first draft of the interface, and is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:16:23 +00:00
buildbot@webrtc.org
af5fa95258 (Auto)update libjingle 74857067-> 74860820 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:03:50 +00:00
buildbot@webrtc.org
7e3bd3d7de (Auto)update libjingle 74851128-> 74857067 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 11:45:42 +00:00
buildbot@webrtc.org
bc6fa1876e (Auto)update libjingle 74825992-> 74851128 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 11:08:01 +00:00
pbos@webrtc.org
287e9614b3 Disable TestDrain test on memcheck bots. 
P2PTransportChannelMultihomedTest.TestDrain is flaky on memcheck bots,
likely the test timeout is insufficient for memcheck which incurs a
serious slowdown.

BUG=2409,3447
R=minyue@webrtc.org
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 10:11:24 +00:00
pbos@webrtc.org
cdb48dbc23 Enable VideoAdapterTest.BlackOutput on DrMemory. 
DrMemory r2061 fixes how the instruction psrlw's shadow is mirrored ->
this false positive is now gone.

R=kjellander@webrtc.org
BUG=3754

Review URL: https://webrtc-codereview.appspot.com/25399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 09:46:34 +00:00
kjellander@webrtc.org
fed47dc205 Drop buildbot_tests.py script 
This is no longer used since the buildbots have moved
over to recipes (where these arguments are configured).
See https://code.google.com/p/chromium/codesearch#chromium/tools/build/scripts/slave/recipe_modules/webrtc/api.py&l=73
for details.

This is essentially a revert of
https://webrtc-codereview.appspot.com/1021006

BUG=None
TESTED=None
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 08:25:38 +00:00
kjellander@webrtc.org
a2da031dc0 Remove use_relative_paths from DEPS 
This makes it possible for us to migrate to using the bot_update step
on our buildbots. That would mean they'd use a Git checkout, which
brings stability, speed and best of all: re-enables the
DEPS-second-sync capability on our trybots that we've been lacking.

bot_update currently doesn't support the use_relative_paths variable
so the synced deps end up in the wrong path with it enabled.

Since Chromium doesn't use it, and it doesn't pollute our
DEPS file that much, I think we should switch.

NOTICE: Any custom_deps entries for the solution in .gclient have to be
updated to support this change, which includes the entry normally present
for Valgrind binaries. The bots will need to be updated as well at the
same time as landing this.

BUG=3534
TESTED=Verified a local sync works.
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 08:25:24 +00:00
henrik.lundin@webrtc.org
bcf75e32a3 Modifying audio_coding/codecs/OWNERS 
Adding myself.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7077 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 07:18:50 +00:00
bjornv@webrtc.org
c2c4117477 common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing 
Affected components:
* AECMobile
  - Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)

BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 06:01:53 +00:00