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Author SHA1 Message Date
henrike@webrtc.org
b8395ebe14 (Auto)update libjingle 62293974-> 62364298 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-28 21:57:22 +00:00
henrike@webrtc.org
eec3843596 TSAN only disable of two of libjingle's tests for atomic ops as they are failing for TSAN-bot. 
BUG=2981
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9239005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5622 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-28 18:06:42 +00:00
jiayl@webrtc.org
9fd8d87ff5 Adds APIs for reporting pacer queuing delay. 
BUG=2775
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 22:32:40 +00:00
andrew@webrtc.org
56e4a05053 Remove ProcessingComponent's dependence on AudioProcessingImpl. 
- Move needed accessors to AudioProcessing.
- Inject the crit directly as a dependency.
- Remove the now unneeded EchoCancellationImplWrapper.

BUG=2894
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5620 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 22:23:17 +00:00
henrike@webrtc.org
806768a6ca (Auto)update libjingle 62281784-> 62293974 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 21:03:09 +00:00
henrike@webrtc.org
704bf9ebec (Auto)update libjingle 62063505-> 62278774 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5617 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 17:52:04 +00:00
jiayl@webrtc.org
f0fc72f70e Call PrintWindow for the first time of capturing to capture the window frames correctly. 
This will fix artifacts on the captured window frames, especially for cmd, which
sometimes leaks glimpss of other window's content.

BUG=
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 16:43:12 +00:00
andrew@webrtc.org
00073aafa8 Clean up CPU detection defines in SincResampler a little. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-27 04:12:34 +00:00
jiayl@webrtc.org
0231e801d6 Invalidate the whole screen when the frame size is changed. 
Otherwise we'll compare frames of different sizes and read into invalid
memory.

BUG=https://code.google.com/p/chromium/issues/detail?id=345498
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/9149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5614 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 18:54:57 +00:00
andrew@webrtc.org
2038920a2b Use scoped_ptr<T[]> in SincResampler to avoid .get()[] weirdness. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5613 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 18:14:54 +00:00
henrik.lundin@webrtc.org
c0e9aebe8f Add SetConfig method to FakeNetworkPipe and to DirectTransport 
This method allow the user to change the network configuration
during run-time. This is useful when testing how components react
to changing bandwidth.

BUG=2636
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5612 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 13:34:52 +00:00
braveyao@webrtc.org
eaadecaf98 iOS, AppRTCDemo: Fixes exception due to JSON for ice using "urls" instead of "url", which is introduced by r5599. 
BUG=2962
TEST=
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5610 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-26 04:16:02 +00:00
marpan@webrtc.org
90173e188f Roll libvpx 248011:251850 
R=andrew@webrtc.org
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/9119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5609 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-25 22:09:02 +00:00
aluebs@webrtc.org
bc1d22461b Add experimental noise suppression flag to audioproc test 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-25 16:50:22 +00:00
sprang@webrtc.org
050892a95b Missing include in experiments.h 
webrtc/typedefs.h should be included in webrtc/experiments.h since the
type uint32_t is being used and it is not indirectly included from this
file.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-25 09:17:43 +00:00
wu@webrtc.org
7f52a6ef2b Split the implementation of VP8Encoder|Decoder::Create into a seperated file 
(vp8_factory.cc).

R=fischman@webrtc.org, marpan@google.com, marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5606 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 23:56:39 +00:00
henrike@webrtc.org
79a1cff65a Android, AppRTCDemo: Fixes java exception due to JSON for ice using "urls" instead of "url". 
BUG=2952
TEST=Manual
TBR=braveyao

Review URL: https://webrtc-codereview.appspot.com/9099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 23:22:18 +00:00
fischman@webrtc.org
bf88eccf33 Added turn-prober.sh: a super-simple prober for TURN servers & candidates. 
BUG=2187
R=juberti@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 21:52:59 +00:00
wu@webrtc.org
78ea3d50e0 Check pcConfig (which can be null) before use. 
BUG=

TEST=manully with pc1.html
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/9079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5603 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 21:51:58 +00:00
henrike@webrtc.org
91cbaa477c (Auto)update libjingle 61966318-> 62063505 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 16:51:32 +00:00
asapersson@webrtc.org
23caa2d8d6 Fix to get total number of sent and received rtcp packets. 
BUG=2638
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8979005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 09:27:38 +00:00
braveyao@webrtc.org
4f0801bd39 AviRecorder is missing a critical section. 
BUG=2885
TEST=AUTOTEST
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 09:19:36 +00:00
braveyao@webrtc.org
bc0470f559 AppRTC Sample: Switch AppRTC to use RTCIceServer.urls. 
BUG=2832
TEST=Manual Test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/7739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5599 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-24 03:43:03 +00:00
kjellander@webrtc.org
55fcd716f3 Disable libjingle_peerconnection_java_unittest 
Broken by libjingle roll in r5590.

TBR=henrike@webrtc.org
BUG=2960
TEST=git try --bot=linux_baremetal --revision=5597

Review URL: https://webrtc-codereview.appspot.com/9029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5598 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-23 18:47:27 +00:00
bjornv@webrtc.org
33af96c5c2 Removed unused mock methods in audio_processing 
TESTED=trybots,modules_unittests
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 23:56:05 +00:00
henrike@webrtc.org
d43aa9de7a Update libjingle 61901702->61966318 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 23:43:24 +00:00
henrike@webrtc.org
a7b981843f Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). 
BUG=N/A
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5595 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 15:51:43 +00:00
tina.legrand@webrtc.org
125a66adc2 Memory and Tsan tests: Turn off the new-ACM tests 
In this CL https://webrtc-codereview.appspot.com/8829004/, I splitted the tests so that new-ACM runs in separate tests. Almost all of these tests are too slow for the memory and tsan bots, and were already excluded from them. This CL turns off the new-ACM tests from these bots.

BUG=https://code.google.com/p/webrtc/issues/detail?id=2951
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 14:12:00 +00:00
xians@webrtc.org
ef2215110c Revert 5590 "description" 
> description

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5593 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 10:31:29 +00:00
asapersson@webrtc.org
0f2809a5ac Add RTCP packet class. 
Adds packet types: sr, rr, bye, fir.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 08:14:45 +00:00
andrew@webrtc.org
c0907eff42 MIPS optimizations for AEC audio processing module 
The resulting output streams obtained by testing with audioproc test application
are bit-exact with generic C code output streams.

Performance gain achieved:
- mips32 ~ 17%
- mips32r2 ~ 20%
- mipsdsp & mipsdspr2 ~ 21%

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7359004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 00:13:31 +00:00
henrike@webrtc.org
2643805a20 description 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5590 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 22:32:53 +00:00
elham@webrtc.org
3f170dd309 Updated WebRTC version to 3.50 
TBR= wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5589 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 22:31:07 +00:00
andrew@webrtc.org
d617a44a4f Add an AlignedFreeDeleter and remove scoped_ptr_malloc. 
- Transition scoped_ptr_mallocs to scoped_ptr.
- AlignedFreeDeleter matches Chromium's version.

TESTED=try bots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 21:08:36 +00:00
turaj@webrtc.org
d4d5be8781 Minor improvement in RoundToInt16 implementation. 
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 20:55:21 +00:00
asapersson@webrtc.org
a0a6df3910 Modified overuse detection thresholds. 
BUG=1577
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 17:37:37 +00:00
henrik.lundin@webrtc.org
04a691adac Removing a variable that was never read 
In NetEq4, the local variable discard_count in
PacketBuffer::DiscardOldPackets() was incremented but never read.
Removing it.

TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 15:27:00 +00:00
fbarchard@google.com
66061992fb ifdef the alsa code based on macro USE_X11 
BUG=none
TEST=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-20 03:05:05 +00:00
henrike@webrtc.org
056176b962 Presubmit script that prohibits cls to both trunk/webrtc and trunk/talk. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7999006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 23:18:19 +00:00
turaj@webrtc.org
78f0db4710 Fix the break caused by r5579. 
TBR=tlegrand@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/8939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 23:07:31 +00:00
henrike@webrtc.org
571df2dca9 Update libjingle 61759961->61834300 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 23:04:26 +00:00
turaj@webrtc.org
c2d69d3229 Resolves memcheck issue in AudioCodingModuleTest. The issue is coditional jumnp based on uninitialized variable. 
BUG=2944
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 20:31:17 +00:00
jiayl@webrtc.org
97e7a640d8 Make WindowCapturerLinux handling window resize events. 
We need to re-initialize the XServerPixelBuffer to the new size
when a window resize event is received.

BUG=https://code.google.com/p/chromium/issues/detail?id=339953
R=sergeyu@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/8679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 17:28:41 +00:00
andresp@webrtc.org
242102517d Added architecture for compiling under chrome NaCl. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 13:55:02 +00:00
tina.legrand@webrtc.org
056287eee0 This CL separate all ACM tests with new and old implementation of ACM and NetEq. The reason is to debug an issue with failure on Android try bots. We need to see if the error only occurs with the new ACM/NetEq, or if it is a flakiness that affects both. 
BUG=issue2874
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 13:45:54 +00:00
asapersson@webrtc.org
8098e07478 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). 
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().

BUG=2638
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 11:59:02 +00:00
henrika@webrtc.org
b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. 
Reasons for removing:

- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.

BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 08:58:08 +00:00
sergeyu@chromium.org
e384104166 Fix DesktopAndCursorComposer not to crash 
DesktopAndCursorComposer was crashing when screen/window
capturer returns a NULL frame due to an error.

BUG=crbug.com/344093
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 23:26:34 +00:00
henrike@webrtc.org
5cf3e8f0f0 (Auto)update libjingle $LAST_P10_REVISION-> $NEW_P10_REVISION 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 22:28:52 +00:00
andrew@webrtc.org
27c6980239 Move the volume quantization workaround from VoE to AGC. 
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 20:24:56 +00:00