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Author SHA1 Message Date
tommi@webrtc.org
43e54e36bf Revert 8095 "Update StatsCollector's interface in preparation of..." 
> Update StatsCollector's interface in preparation of more changes.
> 
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
> 
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
> 
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
> 
> BUG=2822
> R=perkj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36829004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 17:34:23 +00:00
tommi@webrtc.org
5b76fd79df Update StatsCollector's interface in preparation of more changes. 
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 16:49:33 +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
phoglund@webrtc.org
f9d3555ec3 Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test. 
The was was really, really difficult to run before because you needed
a custom env with both LD_PRELOAD and library path. Now the script will
set up the correct library path and be more transparent about what it
requires.

BUG=None
TESTED=locally
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 13:57:59 +00:00
kjellander@webrtc.org
ce3ac53757 Adding TRYSERVER_PROJECT to codereview.settings. 
Recent infra changes makes this being needed to
trigger tryjobs from Rietveld.

TBR=sergiyb@chromium.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 13:51:59 +00:00
kjellander@webrtc.org
018c087a6d Add /talk/examples/androidtests/{bin,gen} to .gitignore. 
TBR=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 12:52:43 +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
sprang@webrtc.org
ff9462eb54 Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan. 
Tests are flaky on tsan, disabling for now.

BUG=4135
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 12:06:35 +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
487a444215 Add stats collection for the data channel. 
BUG=1805
R=bemasc@chromium.org, hta@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 22:55:07 +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
tkchin@webrtc.org
ef2a5dd398 Update AppRTCDemo UI. 
- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 22:38:21 +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
andrew@webrtc.org
6c3855258d Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version. 
This intrinsics version gives bit-exact result as the current C
code. And the performance is 14% better than current assembly
neon version, 3.4 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.

Change-Id: Icce5eaf2e17790ce44513d52b53b9f600cc16f96

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/36689004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 02:56:06 +00:00
mgraczyk@chromium.org
5a92b78e86 Add beamforming to audioproc_float utility. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 01:28:36 +00:00
andrew@webrtc.org
6b6301588e Move ring_buffer to common_audio. 
In preparation for adding a C++ wrapper in common_audio. Also, change
the return type of Init to void and call it from Create.

R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 00:09:53 +00:00
pthatcher@webrtc.org
fd630a50d2 Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior. 
R=decurtis@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 23:19:06 +00:00
kjellander@webrtc.org
693e01c910 Fix searching for DirectX SDK during GN build. 
Before that GN just checked for DXSDK_DIR environment variable.
GYP does more and checks registry, let's do the same in GN.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37599004

Patch from Vyacheslav Chigrin <vchigrin@yandex-team.ru>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 21:25:25 +00:00
pbos@webrtc.org
f1c8b90520 Remove WebRtcVideoEncoderFactory2. 
This interface is no longer required and just adds complexity.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/33009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 17:29:27 +00:00
turaj@webrtc.org
e5a31e1bf5 Revert removing of compile_assert.h. 
In https://webrtc-codereview.appspot.com/39469004 compile_assert.h is removed and that resulted in some bots to break. There is a pending CL to fix the issue https://chromereviews.googleplex.com/141837013/
, meanwhile I revert this change.

TBR=kwiberg@google.com

Review URL: https://webrtc-codereview.appspot.com/35779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 17:17:11 +00:00
kjellander@webrtc.org
85fa94dff5 Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory. 
This test is too slow to execute:
[ RUN      ] EndToEndTest.SendsAndReceivesH264
e:\b\build\slave\win-drmem\build\src\webrtc\video\end_to_end_tests.cc(287): error: Value of: Wait()
  Actual: 3
Expected: kEventSignaled
Which is: 1
Timed out while waiting for enough frames to be decoded.
[  FAILED  ] EndToEndTest.SendsAndReceivesH264 (72812 ms)

BUG=3159
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 17:00:15 +00:00
stefan@webrtc.org
387841ac5c Improved fairness simulation by starting the flows 20 seconds apart. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:45:29 +00:00
pbos@webrtc.org
f18fba2f7b Implement SimulcastEncoderAdapter support. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/37589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:26:23 +00:00
henrik.lundin@webrtc.org
8315d7de85 Remove dual stream functionality in VoiceEngine 
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:07:26 +00:00
mflodman@webrtc.org
b4e5d1b34e Remove RTX SSRC when deleting the default receive stream. 
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 15:07:07 +00:00
kwiberg@webrtc.org
2ebfac5649 Remove COMPILE_ASSERT and use static_assert everywhere 
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.

R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 10:51:54 +00:00
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory. 
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

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

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:30:52 +00:00
kjellander@webrtc.org
a35f741bb0 Add .classpath + talk/app/webrtc/androidtests to .gitignore 
TBR=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:05:38 +00:00
pbos@webrtc.org
f7a5893f80 Combine RegKeyTests to prevent parallel execution. 
Executing these tests in parallel causes failures due to conflicting
registry keys, combining them to unblock launching a parallel win32 bot.
Ideally these keys would be generated differently per-process and not
conflict at all (so it can be run in parallel repeatedly alongside itself).

BUG=4162
R=kjellander@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:03:16 +00:00
phoglund@webrtc.org
ef090927f4 No longer asserting in mocks, split first test case in two methods. 
This way assertions will be caught in the test runner instead of crashing other Android threads.

BUG=None
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 08:56:06 +00:00
kjellander@webrtc.org
69f47381fb Roll chromium_revision 3dd2edf..a6eafec (310717:311223) 
Relevant changes:
* src/third_party/android_tools: 8fe116f..56b3d3e
* src/third_party/boringssl/src: aac2f6a..ca9a538
* src/third_party/icu: 51c1a4c..4e3266f
* src/third_party/libvpx: d3f3dce..4f9bd1b
Details: 3dd2edf..a6eafec/DEPS

The following were moved into src/buildtools:
* src/third_party/libc++/trunk
* src/third_party/libc++abi/trunk

Clang version was not updated in this roll.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 06:06:04 +00:00
mgraczyk@chromium.org
d6e84d9d13 Always copy processed audio to output buffer in ProcessStream. 
In the old AudioFrame ProcessStream API, input and output buffers were shared.
Now that the buffers are distinct, the input must be copied to the
output even when no processing occurred.

R=andrew@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=78de5010d167d1e375e05d26177aad43c2e2de08

Review URL: https://webrtc-codereview.appspot.com/41459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 01:33:54 +00:00
aluebs@webrtc.org
c0da63c707 Optimize minimum delay in blocker 
Could not hear any difference when running the beamformer_test, although sample-wise it changes because of the non-linear character of the processing.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 22:28:35 +00:00
kwiberg@webrtc.org
af9d56f38c Unify the two copies of template_util.h 
This patch basically deletes webrtc/base/template_util.h (which is the
more outdated copy, although there are only cosmetical differences)
and moves webrtc/system_wrappers/source/template_util.h to take its
place.

The reunified header uses the rtc namespace like the old
webrtc/base/template_util.h, rather than the webrtc namespace like
webrtc/system_wrappers/source/template_util.h.

R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 20:32:04 +00:00
pbos@webrtc.org
0b0c24177b Only return Rtx mode in RTXSendStatus(). 
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38569004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 14:15:15 +00:00
kwiberg@webrtc.org
3df38b442f Unify the two copies of compile_assert.h 
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.

R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 11:37:48 +00:00
kjellander@webrtc.org
58a1ba6ffc Roll chromium_revision 271c6cc..3dd2edf (309333:310717) 
Relevant changes:
* src/buildtools: 23a4e2f..451dcd0
* src/third_party/boringssl/src: 306e520..aac2f6a
* src/third_party/openmax_dl: 0164270..1a4171c
* src/tools/gyp: fe00999..82b0804
* src/tools/swarming_client: 119b084..c44f572
Details: 271c6cc..3dd2edf/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 10:59:21 +00:00
andrew@webrtc.org
46323b3786 Remove useless AudioProcessing::Create() overload. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 06:48:06 +00:00