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Author SHA1 Message Date
pbos@webrtc.org
a5f6fb53ba Permit single-stream max bitrates above 2000k. 
BUG=4463
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49509004

Cr-Commit-Position: refs/heads/master@{#8839}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 22:30:11 +00:00
jiayl@webrtc.org
a197a5eed6 Update libsrtp includes in preparation of roll into Chromium. 
This CL is in preparation to roll the libsrtp update which landed in
https://codereview.chromium.org/936663005/ into Chromium.

BUG=https://code.google.com/p/chromium/issues/detail?id=328475
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40209004

Cr-Commit-Position: refs/heads/master@{#8838}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 22:12:19 +00:00
tommi@webrtc.org
a3ffc56cee Allow setting thread priorities in Chromium on all but linux platforms. 
The previous check was overly broad, so narrowing it down to linux only.

R=pbos@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43929004

Cr-Commit-Position: refs/heads/master@{#8837}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 20:11:45 +00:00
henrik.lundin@webrtc.org
39fc1d3d48 Disable PeerConnectionClientTest.testLoopbackVp9 
The test is flaky on Nexus 9.

BUG=4430
TBR=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44839004

Cr-Commit-Position: refs/heads/master@{#8836}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 19:57:52 +00:00
henrik.lundin@webrtc.org
0b44b58a3c Limit disabling of PeerConnectionEndToEndTest.Call to Windows 
The test seems to be flaky only on Windows.

BUG=4464
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44829004

Cr-Commit-Position: refs/heads/master@{#8835}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 19:48:19 +00:00
tkchin@webrtc.org
64eb2ff0b9 iOS library build script 
Script for building iOS fat libraries with armv7/arm64/x86_64.

BUG=4119
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51429004

Cr-Commit-Position: refs/heads/master@{#8834}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 19:08:15 +00:00
tommi@webrtc.org
9509fbfc30 Split EventWrapper in twain. 
I'm splitting the timer functions in EventWrapper into a separate interface.
- Users of the timer functions have different needs than users of a generic event
- Providing a default implementation for EventWrapper that simply uses rtc::Event.

This means that clients of WebRTC that don't use the relatively few classes, typically rendering classes, that depend on the event timer functionality, also don't pull in dependencies on multimedia timers.

R=mflodman@webrtc.org, mflodman
BUG=

Review URL: https://webrtc-codereview.appspot.com/48599004

Cr-Commit-Position: refs/heads/master@{#8833}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 16:25:46 +00:00
henrik.lundin@webrtc.org
82e8ae4ee8 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest 
The test has been flaky recently.

BUG=4464
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46689004

Cr-Commit-Position: refs/heads/master@{#8832}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 14:25:50 +00:00
pbos@webrtc.org
2b4ce3a501 Convert webrtc/video/ abort/assert to CHECK/DCHECK. 
Also replaces NULL with nullptr. This gives nicer error messages and
keeps style consistent.

BUG=1756
R=magjed@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42879004

Cr-Commit-Position: refs/heads/master@{#8831}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 13:13:15 +00:00
minyue@webrtc.org
41d2befe9f Limit RED audio payload to narrow band. 
In SDP, RED audio codec has its own sample rate. Currently, we offer RED/8000 (8 kHz). But the actual send codec can violate this sample rate. The way to solve it is to introduce more RED payload types, e.g., RED/16000, RED/32000.

As a first step towards that, we, in this CL, limit the current RED (RED/8000) to work only with 8 kHz codecs.

BUG=3619
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43849004

Cr-Commit-Position: refs/heads/master@{#8830}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:58:17 +00:00
tommi@webrtc.org
1596a4f88b Temporarily disable SetPriority when building with Chromium. 
This is due to errors we were hitting with Chromium's sandbox policy for pthread_setschedparam.

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

Review URL: https://webrtc-codereview.appspot.com/45839004

Cr-Commit-Position: refs/heads/master@{#8829}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:39:38 +00:00
magjed@webrtc.org
d4e7d49628 Scaler: Recycle allocations using buffer pool. 
BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49489004

Cr-Commit-Position: refs/heads/master@{#8828}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:28:14 +00:00
henrik.lundin@webrtc.org
09b6ff9460 Disable PLC for iSAC 
A codec's packet-loss concealer is called once from NetEq before
decoding the first packet after a packet loss. The purpose is not to
use the PLC output, but to prepare the state of the decoder such that
it may recover faster after the loss. However, this effect is not
achieved by calling iSAC's PLC. Also, there are some problems with the
fixed-point implementation of the PLC (see the associated bug).

BUG=4423
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42849004

Cr-Commit-Position: refs/heads/master@{#8827}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:24:14 +00:00
kjellander@webrtc.org
ee0c5af314 Remove unused version.py script. 
This should have been removed back in
https://webrtc-codereview.appspot.com/4899004

TBR=andrew@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/48629004

Cr-Commit-Position: refs/heads/master@{#8826}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 12:20:18 +00:00
jmarusic@webrtc.org
aa0bbab8ec Fix build failure 
There was a build failure due to including checks.h. Removed the include.
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48639004

Cr-Commit-Position: refs/heads/master@{#8825}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 11:43:14 +00:00
jmarusic@webrtc.org
a4bef3e6c0 AcmReceiver: use std::map instead of an array to keep the list of decoders 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50419004

Cr-Commit-Position: refs/heads/master@{#8824}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 11:20:31 +00:00
pbos@webrtc.org
3335a4ffc8 Prevent asserting on unset start bitrate. 
Quick fix for crash in WebRtcBrowserTest.NegotiateOfferWithBLine.
Follow-up CL from stefan@ that changes this path is under review.

BUG=1788,1667
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51419005

Cr-Commit-Position: refs/heads/master@{#8823}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 09:49:12 +00:00
kjellander@webrtc.org
50ed0d9630 Roll chromium_revision 6311617..da9a1c0 (321517:321718) 
Relevant changes:
* src/third_party/icu: d319ad9..7c81740
Details: 6311617..da9a1c0/DEPS

Clang version changed 223108:231690
Details: 6311617..da9a1c0/tools/clang/scripts/update.sh

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44809004

Cr-Commit-Position: refs/heads/master@{#8822}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 07:13:21 +00:00
kjellander@webrtc.org
e5e92bd556 Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows (fix) 
In https://webrtc-codereview.appspot.com/43899004/ I managed to get some
kind of weird whitespace character in there that completely breaks Goma
and local compilation. This fixes that.

BUG=4452
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43909004

Cr-Commit-Position: refs/heads/master@{#8821}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 16:28:18 +00:00
kjellander@webrtc.org
cfde27eeb3 Disable WebRtcVideoMediaChannelTest.AddRemoveRecvStreamAndRender on Windows. 
The test is flaky:
http://build.chromium.org/p/client.webrtc/builders/Win64%20Release/builds/4179

BUG=4452
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43899004

Cr-Commit-Position: refs/heads/master@{#8820}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 16:09:47 +00:00
tommi@webrtc.org
38492c5b6f Re-land 8810 "- Add a SetPriority method to ThreadWr..." 
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
> 
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> > 
> > BUG=
> > R=magjed@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44729004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/48609004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459005

Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:42:46 +00:00
tommi@webrtc.org
90a1cb4630 Revert 8810 "- Add a SetPriority method to ThreadWrapper" 
Seeing if this is causing roll issues.

> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
> 
> BUG=
> R=magjed@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44729004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48609004

Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:34:46 +00:00
tommi@webrtc.org
b789f6271a Re-land 8809 "Set WebRtcVideoEngine2 as the WebRtcMe..." 
I've kicked of a roll into Chromium with out the WebRtcVideoEngine2 change, to see if it was causing the roll problems, but re-landing in the meantime.

> Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine."
> content_browsertests started failing around the time the change landed and rolls are failing now.
> I'm going to try rolling this back, start a roll, and then re-land.
> 
> > Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
> > 
> > Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> > has shown no major regressions on Chrome Canary/Dev that haven't been
> > addressed, so enabling it in time before feature freeze.
> > 
> > BUG=1788
> > R=mflodman@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44759004
> 
> TBR=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/43889004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459004

Cr-Commit-Position: refs/heads/master@{#8817}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 12:50:44 +00:00
tommi@webrtc.org
0c3400168a Revert 8809 "Set WebRtcVideoEngine2 as the WebRtcMediaEngine." 
content_browsertests started failing around the time the change landed and rolls are failing now.
I'm going to try rolling this back, start a roll, and then re-land.

> Set WebRtcVideoEngine2 as the WebRtcMediaEngine.
> 
> Removes the experiment launching WebRTC-NewVideoAPI. This field trial
> has shown no major regressions on Chrome Canary/Dev that haven't been
> addressed, so enabling it in time before feature freeze.
> 
> BUG=1788
> R=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44759004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43889004

Cr-Commit-Position: refs/heads/master@{#8816}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 12:45:44 +00:00
braveyao@webrtc.org
346a64b9b5 Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. 
So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places.
Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency.

BUG=4240
TEST=Manual Test
R=andrew@webrtc.org, henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36029004

Cr-Commit-Position: refs/heads/master@{#8815}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-21 01:06:14 +00:00
wtc@chromium.org
4553941d32 Document the 'int' return value of Resampler methods. 
Remove an obsolete TODO comment.

R=andrew@webrtc.org
BUG=none

Review URL: https://webrtc-codereview.appspot.com/48589004

Cr-Commit-Position: refs/heads/master@{#8814}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 23:28:39 +00:00
andrew@webrtc.org
3200a64b3c Minor fix for MIPS Android build. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

Cr-Commit-Position: refs/heads/master@{#8813}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 22:55:43 +00:00
glaznev@webrtc.org
4ddc9387bd Support VP8 hardware encoding and decoding on IA devices. 
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42829004

Cr-Commit-Position: refs/heads/master@{#8812}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 21:21:17 +00:00
pbos@webrtc.org
b9557a9bb7 Fix code to handle crashes for non-VP8. 
Unit tests will be submitted Monday, submitting this part to get the
Android bots green.

BUG=1667, 1788
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44789004

Cr-Commit-Position: refs/heads/master@{#8811}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 19:53:15 +00:00
tommi@webrtc.org
b6817d793f - Add a SetPriority method to ThreadWrapper 
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional

BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44729004

Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 15:52:43 +00:00
pbos@webrtc.org
66df3cf7ab Set WebRtcVideoEngine2 as the WebRtcMediaEngine. 
Removes the experiment launching WebRTC-NewVideoAPI. This field trial
has shown no major regressions on Chrome Canary/Dev that haven't been
addressed, so enabling it in time before feature freeze.

BUG=1788
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44759004

Cr-Commit-Position: refs/heads/master@{#8809}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 15:45:17 +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
pbos@webrtc.org
a3209a2b27 Release buffer pool in Vp8DecoderImpl::Release(). 
Permits reusing an external VP8DecoderImpl instance from another
VideoReceiveStream without a thread-checker DCHECK blowing up. Also
releases buffers that would've been kept in memory even though the
decoder isn't configured.

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

Review URL: https://webrtc-codereview.appspot.com/50449004

Cr-Commit-Position: refs/heads/master@{#8807}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 13:36:25 +00:00
pbos@webrtc.org
8904290aca Make screenshare target bitrate experiment always on 
BUG=4083
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44699004

Patch from sprang@webrtc.org <sprang@webrtc.org>.

Cr-Commit-Position: refs/heads/master@{#8806}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 12:50:34 +00:00
kjellander@webrtc.org
d9c5024ee7 Roll chromium_revision bd49b12..6311617 (320783:321517) 
Relevant changes:
* src/buildtools: d4dd4f7..3b302fe
* src/third_party/android_tools: 98a4345..8b18ef7
* src/third_party/boringssl/src: bf0df92..642f149
* src/third_party/icu: eda9e75..d319ad9
* src/third_party/libvpx: f80cf58..00cf1b1
Details: bd49b12..6311617/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47709004

Cr-Commit-Position: refs/heads/master@{#8805}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 12:35:18 +00:00
perkj@webrtc.org
9f9ea7e5ab Clean up webrtc external capture. 
This cl removes the dependency to the external capture module if external capturing is used in webrtc.
It also removes two external capture methods that is not needed.
Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input.

R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43879004

Cr-Commit-Position: refs/heads/master@{#8804}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 10:55:39 +00:00
pbos@webrtc.org
443ad403f5 Remove FullStackTest frame pointer handles. 
Simplifies code, speculative fix for a DCHECK crash in ForemanCifPlr5.

BUG=4451
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45809005

Cr-Commit-Position: refs/heads/master@{#8803}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 07:34:38 +00:00
pbos@webrtc.org
6231fb6dac Prevent crashes when copying a zero-size frame. 
BUG=4451
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44749004

Cr-Commit-Position: refs/heads/master@{#8802}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 07:33:11 +00:00
bjornv@webrtc.org
6069032ebb Refactor audio_coding/isac: removed usage of macro WEBRTC_SPL_LSHIFT_W32 
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.

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

Review URL: https://webrtc-codereview.appspot.com/44659004

Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 07:03:41 +00:00
bjornv@webrtc.org
4ab23d0e8f Refactor audio_coding/ilbc: removes usage of macro WEBRTC_SPL_LSHIFT_W32 
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.

Also removes unnecessary casts to int32_t from int16_t.

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

Review URL: https://webrtc-codereview.appspot.com/48519004

Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 06:01:43 +00:00
andrew@webrtc.org
bd8c865f43 Remove build-time beamformer flags. 
RealFourier is now unconditionally enabled since we can fall back to the
Ooura FFT. We no longer need to condition users on rtc_use_openmax_dl.

R=aluebs@webrtc.org, mgraczyk@google.com

Review URL: https://webrtc-codereview.appspot.com/50439004

Cr-Commit-Position: refs/heads/master@{#8799}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 00:28:42 +00:00
andrew@webrtc.org
04c50981f8 Add the Ooura FFT to RealFourier. 
We are using the Ooura FFT in a few places:
- AGC
- Transient suppression
- Noise suppression

The optimized OpenMAX DL FFT is considerably faster, but currently does
not compile everywhere, notably on iOS. This change will allow us to use
Openmax when possible and otherwise fall back to Ooura.

(Unfortunately, noise suppression won't be able to take advantage of it
since it's not C++. Upgrade time?)

R=aluebs@webrtc.org, mgraczyk@chromium.org

Review URL: https://webrtc-codereview.appspot.com/45789004

Cr-Commit-Position: refs/heads/master@{#8798}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 20:07:43 +00:00
kjellander@webrtc.org
ba86031b34 Whitespace change to trigger new Git pollers (2). 
TBR=

Review URL: https://webrtc-codereview.appspot.com/50429004

Cr-Commit-Position: refs/heads/master@{#8797}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 18:11:35 +00:00
kjellander@webrtc.org
cf3fb9b3ba Whitespace change to trigger new Git pollers. 
TBR=
BUG=

Review URL: https://webrtc-codereview.appspot.com/47669004

Cr-Commit-Position: refs/heads/master@{#8796}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 17:51:21 +00:00
henrika@webrtc.org
80d9aeeda5 Adds full-duplex unit test to AudioDeviceTest on Android 
BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42709004

Cr-Commit-Position: refs/heads/master@{#8795}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 15:28:42 +00:00
tommi@webrtc.org
361981faa8 Use scoped_ptr for ThreadWrapper::CreateThread. 
BUG=
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45799004

Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:45:42 +00:00
tina.legrand@webrtc.org
c7d5a733b0 Disable flaky test on DrMemory bots 
BUG=4454
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47699004

Cr-Commit-Position: refs/heads/master@{#8793}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:44:13 +00:00
tommi@webrtc.org
27c0be9dfe Remove ThreadObj #define and kThreadMaxNameLength from thread_wrapper. 
BUG=
R=hbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47679004

Cr-Commit-Position: refs/heads/master@{#8792}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:36:43 +00:00
tina.legrand@webrtc.org
0c26299739 Disabling two flaky tests in libjingle_media_unittest. 
BUG=4452,4453
R=kjellander@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44739004

Cr-Commit-Position: refs/heads/master@{#8791}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 13:28:20 +00:00
magjed@webrtc.org
17c64d1c96 Revert "Put ViEFrameProviderBase::DeliverFrame back in the critical section in ViECapturer::DeliverI420Frame" 
This reverts commit r8724.

Reason for revert: This was not the cause of the tsan issues.

BUG=1128
R=mflodman@webrtc.org, pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50389004

Cr-Commit-Position: refs/heads/master@{#8790}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 10:58:17 +00:00