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Author SHA1 Message Date
henrike@webrtc.org
065247b5b7 Rebase webrtc/base with r6863 version of talk/base: 
cls integrated: r6809
svn diff -r 6808:6809 http://webrtc.googlecode.com/svn/trunk/talk/base > 6809.diff
patch -p0 -i 6809.diff

BUG=3379
TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:32:13 +00:00
henrik.lundin@webrtc.org
1c8391205e Use test::Packet test::PacketSource classes in neteq_rtpplay 
This change replaces the old NETEQTEST_RTPpacket and
NETEQTEST_DummyRTPpacket with the new test::Packet class. Note that the
Packet class automatically handles "dummy" packets (i.e., packets for
which only the header and a length field was stored to file)
automatically. There is no need to explicitly signal this to the
application any longer. The RTP input file is now handled as a
test::PacketSource object.

Also adding a new ConvertHeader method to the Packet class. This is
needed to extract the header information as an alternative data type.

Finally, some dead code was deleted from rtp_analyze.cc (unrelated to
the reset of this change).

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:29:38 +00:00
bjornv@webrtc.org
96d8b0e69f Revert 6860 "SSE2 version of SubbandCoherence()" 
> SSE2 version of SubbandCoherence()
> 
> The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
> reported by audioproc is ~3.3%
> 
> The output is bit exact.
> 
> R=bjornv@webrtc.org, cd@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18779004
> 
> Patch from Scott LaVarnway <slavarnw@gmail.com>.

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 12:09:13 +00:00
bjornv@webrtc.org
0db82f337f SSE2 version of SubbandCoherence() 
The performance gain on a x86 laptop (Intel(R) Core(TM) i5-2520M CPU @ 2.50GHz)
reported by audioproc is ~3.3%

The output is bit exact.

R=bjornv@webrtc.org, cd@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18779004

Patch from Scott LaVarnway <slavarnw@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 10:38:31 +00:00
henrike@webrtc.org
3763b9bda0 webrtc/base: removes linkage of crypto 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 21:26:18 +00:00
stefan@webrtc.org
59a2f9f584 Remove the old H264 code now that a new H.264 packetizer has been implemented. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:09:24 +00:00
stefan@webrtc.org
9d74f7ce8c Fix single nalu packetization bug. 
Nalus which had the same size as the max payload size would cause the payload size accounting to wrap.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6846 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 15:02:16 +00:00
pbos@webrtc.org
e8c84bf4de Fix so video_replay logs aren't spammed. 
Add unknown-SSRC counters instead and log number of unknown packets at
end of session.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/13119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 14:42:45 +00:00
minyue@webrtc.org
1d956dd1a7 Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal. 
It is advisable to set the packet loss rate of FEC conservatively. Say, if the estimated loss rate is 5%, we can set it to 1%. The risk of degradation in quality is small and the overall performance is good.

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 12:31:36 +00:00
henrik.lundin@webrtc.org
ea25784107 Change how background noise mode in NetEq is set 
This change prepares for switching default background noise (bgn) mode
from on to off. The actual switch will be done later.

In this change, the bgn mode is included as a setting in NetEq's config
struct. We're also removing the connection between playout modes and
bgn modes in ACM. In practice this means that bgn mode will change from
off to on for streaming mode, but since the playout modes are not used
it does not matter.

BUG=3519
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6843 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 12:27:37 +00:00
pbos@webrtc.org
4b5625e5ac RTP video playback tool using Call APIs. 
Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 16:26:56 +00:00
stefan@webrtc.org
1ccff349ee Fix crashing fake network pipe tests. 
These tests are not included in bots, this will be fixed in a follow-up by pbos@webrtc.org.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 15:41:58 +00:00
minyue@webrtc.org
2a8df7c375 Fixing two bugs in voe_cmd_test. 
I am trying to add a new functionality to voe_cmd_test, and I found two bugs:

1. in r5928, a functionality was removed but the item in the menu was not. Functionalities after it are offset.

r5928: https://code.google.com/p/webrtc/source/detail?r=5928&path=/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

2. in r6736, opus are set to output 48 kHz audio. When mixing Opus output with an audio file, channel.cc may go wrong.

r6736: https://code.google.com/p/webrtc/source/detail?r=6736

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 10:05:19 +00:00
stefan@webrtc.org
79c3359e67 Add end-to-end H.264 packetization test. 
Also correctly wires up H.264 packetization in the new Call api.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 09:24:53 +00:00
stefan@webrtc.org
8b033adb19 Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant. 
R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 08:06:53 +00:00
fbarchard@google.com
d7b4dea801 initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 
BUG=3660
TESTED=set DEPOT_TOOLS_WIN_TOOLCHAIN=0 & set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 &ninja -C out\Debug
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:46:42 +00:00
pbos@webrtc.org
dde16f19e3 Fix some code styles. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22009004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:35:43 +00:00
fbarchard@google.com
e086af0fa3 Fix implicite cast from signed int to unsigned int in unittest.cc 
BUG=3636
TESTED=set GYP_DEFINES=target_arch=ia32 & call python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Debug
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 17:10:52 +00:00
stefan@webrtc.org
fdcb42dac4 Fix potential crash when depacketizing VP8. 
Caused by a missing check for H264 when reading the RTPVideoTypeHeader union.

R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 13:21:18 +00:00
henrike@webrtc.org
d6542852f3 Unbreaks linux.cc in Chromium. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 21:51:14 +00:00
minyue@webrtc.org
0040a6ef97 This is a setup to solve 
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).

BUG=
R=henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 14:41:57 +00:00
asapersson@webrtc.org
84b9e1e9d9 Fix for retransmission. Base layer packets were not retransmitted. 
Issue introduced in r6669.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 11:59:42 +00:00
stefan@webrtc.org
e1c9caf6ee Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804. 
TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 15:07:59 +00:00
stefan@webrtc.org
2ec560606b Add H.264 packetization. 
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 14:59:24 +00:00
stefan@webrtc.org
bfe6e08195 Add simulation of network effects to video_loopback tool. 
Also add support for uniform random packet loss to the fake network pipe.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 12:30:18 +00:00
andrew@webrtc.org
fdbe1442c5 Use C functions in aec for MIPS 
With GCC 4.9, the MIPS NDK toolchain has been changed to only support 16 spregs by default - the even-numbered ones. This has been changed to support the R6 MIPS architecture. While the old behaviour could be restored by adding "-modd-spreg", this would come with a performance hit because the kernel would emulate odd-numbered spregs and missing R2 instructions.
As a result of this change, the functions removed in this CL no longer compile as there are no longer enough spregs for the compiler to assign. So we are removing these functions and they will use the C implementation until the MIPS code is rewritten.

R=andrew@webrtc.org, ljubomir.papuga@gmail.com, pasko@chromium.org

Review URL: https://webrtc-codereview.appspot.com/16159005

Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 14:39:10 +00:00
asapersson@webrtc.org
e75d78d32d Integrate rtcp packet class to rtcp receiver tests. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 08:21:50 +00:00
henrike@webrtc.org
1e7d60e451 merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:". 
TBR=andrew@webrtc.org
BUG=b/15773179

Review URL: https://webrtc-codereview.appspot.com/16999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 04:45:23 +00:00
henrike@webrtc.org
961293d469 webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 21:58:50 +00:00
sergeyu@chromium.org
af9e7943d1 Fix compilation on windows with clang, indentation cleanups 
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6779

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 19:42:19 +00:00
henrike@webrtc.org
c7b8f39e2e Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python. 
R=pbos@webrtc.org
TBR=andrew@webrtc.org
BUG=b/15773179

Review URL: https://webrtc-codereview.appspot.com/18929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 18:36:55 +00:00
turaj@webrtc.org
1ebd2e96df Remove timestamp retreival warning/error. 
An error reported while retreiving playout timestamp if no RTP packet received, yet. This causes an overflow of errors/warnings in applications where few channel are created but only one is actively engaged in a conversation. Therefore, we don't find such logging informative (there is no check upon correctness of timestamp computaion only if a packet already received).

BUG=3545
TEST=manual with voe_cmd_test,try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:50:10 +00:00
sergeyu@chromium.org
2386882266 Revert "Fix compilation on windows with clang, indentation cleanups" 
This reverts commit f628eaedfeea97e13c63c78dd42f2b1c76723619.

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:37:12 +00:00
sergeyu@chromium.org
a44fce5920 Fix compilation on windows with clang, indentation cleanups 
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:28:25 +00:00
mflodman@webrtc.org
f9460688a6 Make sure padding is sent on the first sending RTP module. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-24 16:41:25 +00:00
stefan@webrtc.org
f24c4a3b8d Fix flaky ramp-up test. 
Don't require the first estimate to be less than the target bitrate. There are other tests verifying that BWE works, so it's enough for this test to measure the
time it takes to ramp-up.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 10:27:41 +00:00
minyue@webrtc.org
194fea7640 The lastest commit on this file was in 
https://webrtc-codereview.appspot.com/15529004/

The final patch set should have included this, but was missed.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 09:55:51 +00:00
andresp@webrtc.org
b0c8228755 Remove no longer used SkipEncodingUnusedStreams. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 07:17:17 +00:00
andresp@webrtc.org
5ab7616983 Remove remains of WEBRTC_NO_STL. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 06:48:58 +00:00
andrew@webrtc.org
ceafa8cce9 MIPS optimizations for ISAC (patch #2) 
Implemented functions:
- WebRtcIsacfix_CalculateResidualEnergy
- WebRtcIsacfix_Spec2Time
- WebRtcIsacfix_Time2Spec
- WebRtcIsacfix_HighpassFilterFixDec32
- WebRtcIsacfix_PCorr2Q32

Gain achieved: aprox. further 5% on top of patch#1 on ISAC encoding path.
The optimizations are bit-exact to the C code, with the excception of the
MIPS DSPr2 variant of the WebRtcIsacfix_Time2Spec function (the accuracy of
the WebRtcIsacfix_Time2Spec MIPS DSPr2 variant is same or better than C
variant). Code verification and improvement achieved have been determined
using the iSACFixtest application.

R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19749004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 16:43:13 +00:00
pbos@webrtc.org
3c10758b3b Check before send/receive rtp header extensions. 
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13949004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-20 15:27:35 +00:00
minyue@webrtc.org
f563e85ab0 This is to re-open an earlier CL 
https://webrtc-codereview.appspot.com/16619005/

which is reverted due to an issue in audio conference mixer.

This issue has been solved in
https://webrtc-codereview.appspot.com/20779004/

BUG=webrtc:3155
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:11:27 +00:00
tkchin@webrtc.org
ff50debd37 Runtime guard for iOS7 property. 
BUG=3487
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 17:17:59 +00:00
tkchin@webrtc.org
9343cf67a9 Fix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS. 
BUG=3581
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 17:13:28 +00:00
minyue@webrtc.org
026859b983 This is related to an earlier CL of enabling Opus 48 kHz. 
https://webrtc-codereview.appspot.com/16619005/

It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz.

TEST=locally solved https://webrtc-codereview.appspot.com/16619005/

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 12:28:28 +00:00
pbos@webrtc.org
e9e4253a3c Sleep in ThreadTest thread functions. 
Prevents busy loops that really mess up Valgrind's thread scheduling,
this brings runtimes from up to minutes down to milliseconds.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 10:12:50 +00:00
kwiberg@webrtc.org
e364ac902f AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float 
Specifically, when someone asks for a const pointer to the int16
version of the array, there's no need to invalidate the float version
of that array, and vice versa. (But obviously, invalidation still has
to happen when someone asks for a non-const pointer.)

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

Review URL: https://webrtc-codereview.appspot.com/18809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 07:50:29 +00:00
andrew@webrtc.org
c145668dc8 Reduce runtime of RingBufferTest by a factor of 100. 
This test was needlessly long.

TBR=pbos

Review URL: https://webrtc-codereview.appspot.com/15029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6724 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 23:16:44 +00:00
wu@webrtc.org
4f5da030f1 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. 
There are two audioFrameLists. The previous check wouldn't work correctly if each list had a single member.

TEST=chrome https://apprtc.appspot.com/?debug=loopback&video=false and verify e2e delay stats
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6723 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 22:19:21 +00:00
stefan@webrtc.org
8b94e3da0f Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. 
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 16:10:14 +00:00