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Author SHA1 Message Date
glaznev@webrtc.org
46ffc70878 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. 
BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 17:11:36 +00:00
pbos@webrtc.org
963b979510 Remove potential deadlock in WebRtcVideoEngine2. 
Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.

R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999

Review URL: https://webrtc-codereview.appspot.com/26729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 14:27:27 +00:00
kjellander@webrtc.org
a9e363e721 Roll chromium_revision c264a05..fc668e2 (297113:298195) 
Mainly to pick up https://codereview.chromium.org/619723006/
to fix our MSan bot.

Summary of changes (git diff c264a05..fc668e2 DEPS):
* third_party/boringssl 01fe820..c7dd5f30
* third_party/usrsctp/usrsctplib 8975bd5..d5685d4
* tools/swarming_client 79940aee..33d442a

Clang updated 216630:217949 (git diff c264a05..fc668e2 tools/clang/scripts/update.sh)
This caused TSan v2 to hit an assert in libjingle_peerconnection_unittest
and libjingle_media_unittest, which is why the dlclose call
had to be disabled for now (webrtc:3895).

BUG=3895
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 12:49:34 +00:00
pbos@webrtc.org
77d5a57e5c Revert "Only configure the SSL library in one place." 
This reverts commit r7378, which broke Windows compile targets
elsewhere.

R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=chromium:413497

Review URL: https://webrtc-codereview.appspot.com/28679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 11:43:03 +00:00
kjellander@webrtc.org
6ed1cf49f0 Isolate: Remove use of --ignore_broken_items 
BUG=chromium:395700
R=jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/30659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 09:17:35 +00:00
henrik.lundin@webrtc.org
9103953b58 Fix neteq_rtpplay so that empty SSRC is valid 
In r7380, the command line flag --ssrc was added to neteq_rtpplay.
However, it was not possible to omit that flag, since the validation
did not accept an empty string. This CL fixes that.

TBR=kwiberg@webrtc.org
BUG=2692

Review URL: https://webrtc-codereview.appspot.com/24869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 07:18:36 +00:00
henrik.lundin@webrtc.org
7cbc4f969a Set NetEq playout mode through the Config struct 
This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.

BUG=3520
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 06:37:39 +00:00
henrik.lundin@webrtc.org
8b65d511a0 Add an SSRC filter to neteq_rtpplay 
This allows the user to set the desired SSRC if the input file
contains multiple streams.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 05:30:04 +00:00
turaj@webrtc.org
532ed43e85 Prevent reading outside iSAC bitstream, if the stream is corrupted. 
BUG=chrome_373312(#24)
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 00:21:02 +00:00
henrike@webrtc.org
8234fa6f0e Only configure the SSL library in one place. 
Build settings now use use_openssl in both Chromium and standalone builds. It
moves all the platform-specific SSL-related build checks to be conditioned on
this flag as appropriate.

This is to avoid colliding with Chromium's transition away from NSS.

BUG=chromium:413497
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 22:30:46 +00:00
henrike@webrtc.org
2fe5893748 Mac: adds missing _DEBUG flag to mac debug builds. 
BUG=3836
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 22:04:11 +00:00
henrike@webrtc.org
528fc650d8 Fixing build issue with L-sdk 
Based on https://codereview.appspot.com/153000043/

BUG=https://code.google.com/p/chromium/issues/detail?id=420293
R=niklas.enbom@webrtc.org, serya@chromium.org, yfriedman@chromium.org

Review URL: https://webrtc-codereview.appspot.com/29659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:56:43 +00:00
henrike@webrtc.org
9a742b4840 talk: removes empty directories base and sound. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:52:59 +00:00
houssainy@google.com
5d3e7ac1a3 Check on the existence of report directory 
Reports will be written at rtcBot/test/reports/<report_name>

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:21:27 +00:00
pbos@webrtc.org
42684be21b Wire up CPU adaptation in WebRtcVideoEngine2. 
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.

BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-03 11:25:45 +00:00
henrike@webrtc.org
31b75eae05 Moves xmllite's unittests to rtc_unittest. 
BUG=3836
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 18:43:47 +00:00
glaznev@webrtc.org
25cc745d6b Switch to SW video decoder on Android after getting 2 or more 
critical errors from HW decoder.

BUG=410730
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 16:58:05 +00:00
henrik.lundin@webrtc.org
4b133da5fd Let RtpFileSource use RtpFileReader 
RtpFileSource used to implement it's own RTP dump file reader, but
with this change it simply uses RtpFileReader. One benefit is that
pcap files are now also supported.

All NetEq test tools that use RtpFileSource are updated.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 08:19:38 +00:00
bjornv@webrtc.org
348eac641e audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >> 
A trivial macro that is replaced. Affected components:
* AGC
* NSx

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

Review URL: https://webrtc-codereview.appspot.com/29599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 08:07:05 +00:00
sergeyu@chromium.org
5fa8c458d8 Remove mouse cursor capturer from the ScreenCapturer interface 
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7363

Review URL: https://webrtc-codereview.appspot.com/31529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 01:47:10 +00:00
sergeyu@chromium.org
6138f0f89d Revert "Remove mouse cursor capturer from the ScreenCapturer interface" 
This reverts commit 0adc4953512ee0a57cf7f3c0591b024c2316554a. It broke
FYI bots

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 01:36:43 +00:00
sergeyu@chromium.org
1fced0f2aa Remove mouse cursor capturer from the ScreenCapturer interface 
Mouse can be captured using MouseCursorMonitor and all code in chromium
already uses it instead of ScreenCapturer.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 00:18:10 +00:00
sergeyu@chromium.org
76819d315d Add error trap for XFixesGetCursorImage() 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3245
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 23:07:12 +00:00
andrew@webrtc.org
325cff01b4 Import LappedTransform and friends. 
Add code for doing block-based frequency domain processing. Developed
and reviewed in isolation. Corresponding export CL:
https://chromereviews.googleplex.com/95187013/

R=bercic@google.com, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 17:42:18 +00:00
henrike@webrtc.org
593c3a0868 rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 16:33:03 +00:00
henrike@webrtc.org
4530b2ca48 Revert 7355 "Fix parallelization in libjingle_p2p_unittest." 
Breaks waterfall.

TBR=pbos@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/22909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 15:43:55 +00:00
henrike@webrtc.org
36b0c1afae Adds PRESUBMIT.py dispensation for depending on rtc_base. 
Dispensation for: a few test suites, desktop capture and libjingle.

BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 14:40:58 +00:00
pbos@webrtc.org
fd29205e6e Fix parallelization in libjingle_p2p_unittest. 
Adding VirtualSocketServers to SessionTest and RelayServerTest to avoid
contention on real ports.

R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 out/Debug/libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/26679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 12:31:31 +00:00
pbos@webrtc.org
c86e45d7c4 Fix parallelizability in modules_tests. 
R=henrik.lundin@webrtc.org
BUG=3873
TEST=third_party/gtest-parallel/gtest-parallel -r 10 -w 64 out/Debug/modules_tests

Review URL: https://webrtc-codereview.appspot.com/24799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 10:05:40 +00:00
henrik.lundin@webrtc.org
4cebd84c79 Reland "Remove DTMF status methods from Voice Engine" r7276 
This reverts r7277.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 08:23:21 +00:00
kjellander@webrtc.org
4e4fe4f9ae Add support for MSan 
Add third_party/instrumented_libraries to setup_links.py
Add tools/msan/blacklist.txt which is the default location used
by MSan.

These changes are prerequisites to be able to use MSan with WebRTC.
To use it, one must also run:
sudo third_party/instrumented_libraries/install-build-deps.sh
to get the instrumented libraries installed (requires
/etc/apt/sources.list to be setup with deb-src entries).

NOTICE: Compilation is not yet working, but with this we can setup
a FYI bot to work with.

BUG=chromium:416871
TESTED=gclient sync + generate projects using:
GYP_DEFINES='msan=1 use_instrumented_libraries=1 instrumented_libraries_jobs=20' webrtc/build/gyp_webrtc
Built successfully in Release and ran a couple of tests (some crashed, some passed).

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 08:03:19 +00:00
kjellander@webrtc.org
afefed5c93 Update checkdeps.py rules in DEPS 
The initial rules didn't allow including
source from third_party, which is incorrect.
Cleanup irrelevant rules for directories that
are ignored, since WebRTC don't have any source
code in those locations.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 06:03:47 +00:00
henrike@webrtc.org
83fe69da95 Added presubmit protecting against inclusion of rtc_base, while allowing rtc_base_approved. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 21:54:26 +00:00
kjellander@webrtc.org
3037bc3447 GN: Add common configs to tools and test. 
Similar changes as in https://review.webrtc.org/28589004/
were missed in https://review.webrtc.org/25569004/.
This should fix the Chromium WebRTC FYI bots that currently
are broken due to lack of include paths.

BUG=3441
TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 19:07:58 +00:00
kjellander@webrtc.org
b8caf6a504 GN: Enable libvpx, add link target and convert some test targets 
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).

I also converted a few test targets and made a GN file for
third_party/gflags.

BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 18:05:02 +00:00
andrew@webrtc.org
d05756f0a2 Changed mips_arch_variant variable value corresponding to Chromium code changes. 
Chromium commit URL: https://crrev.com/c8a5da7455b57b2399e4a69e8100c098d9870052

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23809004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:53:24 +00:00
xians@webrtc.org
79a7148108 Revert 7337 "Reland 28629004: adding new AEC dump start interfac..." 
> Reland 28629004: adding new AEC dump start interface for chrome
> 
> adding new AEC dump start interface for chrome.
> 
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
> 
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org, kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/27639004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:29:13 +00:00
xians@webrtc.org
7aad5e5cce Revert 7338 "Fixed the android build by making the interface pur..." 
> Fixed the android build by making the interface pure virtual.
> 
> TBR=asapersson@webrtc.org, bjornv@webrtc.org,
> 
> Review URL: https://webrtc-codereview.appspot.com/24789004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:26:15 +00:00
houssainy@google.com
d0bb5862f5 Collecting stats every fixed time in webrtc_video_streaming.js test 
and prepare the format these collected stats to be plotted using one of
external dev-tools.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7340 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:20:15 +00:00
andrew@webrtc.org
db75a66b0f Minor code change to fix some warnings in MIPS build. 
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26619004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:17:50 +00:00
xians@webrtc.org
90d1979d77 Fixed the android build by making the interface pure virtual. 
TBR=asapersson@webrtc.org, bjornv@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/24789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:15:22 +00:00
xians@webrtc.org
14092e00f1 Reland 28629004: adding new AEC dump start interface for chrome 
adding new AEC dump start interface for chrome.

This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

Chromium bug:crbug/415935
TEST=bots
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 14:35:15 +00:00
henrike@webrtc.org
792d1a0541 Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest. 
BUG=N/A
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 14:21:10 +00:00
xians@webrtc.org
875206196c Revert 7334 "adding new AEC dump start interface for chrome." 
> adding new AEC dump start interface for chrome.
> 
> This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
> http://msdn.microsoft.com/en-us/library/ms235460.aspx
> 
> Chromium bug:crbug/415935
> TEST=bots
> R=bjornv@webrtc.org, kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28629004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 13:30:05 +00:00
xians@webrtc.org
2e417d6428 adding new AEC dump start interface for chrome. 
This is required because we are not allow to pass CRT objects across dll boundaries, that says, when we pass a file descriptor from chrome dll to libpeerconnection dll, the file descriptor will become invalid immediate, more information can be found here:
http://msdn.microsoft.com/en-us/library/ms235460.aspx

Chromium bug:crbug/415935
TEST=bots
R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 13:11:27 +00:00
henrik.lundin@webrtc.org
38c121c484 Minor modifications to test::RtpFileReader 
Adding original_length to the Packet struct. This is populated with
the plen value from the RTP dump file. In the case of reading a
pcap file, original_length will be equal to length.

Also increasing the maximum packet size to 3500 bytes. This is to
accomodate some test files that contain PCM16b audio encoding.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 11:08:44 +00:00
pbos@webrtc.org
1795c358fc Add default implementation of Add/RemoveObserver. 
Needed to remove Add/RemoveObserver from RTCVideoEncoderFactory in
Chromium before removing these completely. This is done to keep the
chromium.webrtc.fyi bots happy and to make rolling webrtc revisions
easier.

R=stefan@webrtc.org
BUG=3876

Review URL: https://webrtc-codereview.appspot.com/23839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:45:25 +00:00
bjornv@webrtc.org
65e56dba53 audio_processing/aecm: Added help function for calculating log of energy 
The same operation of calculating log of the energy was executed four times. This CL adds a help function LogOfEnergyInQ8() for that.

BUG=N/A
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:31:28 +00:00
bjornv@webrtc.org
23ec8372a6 audio_processing: Removed usage of macro WEBRTC_SPL_MUL 
WEBRTC_SPL_MUL is a trivial multiplication after casting to int32_t. This is already taken care of by the compiler which makes the macro unnecessary.

Affected components:
* AGC
* NSx

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

Review URL: https://webrtc-codereview.appspot.com/25429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:29:28 +00:00
bjornv@webrtc.org
750423c722 audio_processing: Replaced trivial macro WEBRTC_SPL_LSHIFT_W32 with << 
Affected components:
* AECM
* AGC
* HPF
* NSx

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

Review URL: https://webrtc-codereview.appspot.com/27629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:26:36 +00:00