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Author SHA1 Message Date
tnakamura@webrtc.org
a367aeab82 Bump to version 40 
TBR=niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/26109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7683 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 16:23:15 +00:00
magjed@webrtc.org
f7c5d4fac7 Revert 7679 "webrtc::Scaler: Preserve aspect ratio" 
> webrtc::Scaler: Preserve aspect ratio
> 
> BUG=3936
> R=glaznev@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28969004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7682 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 13:12:09 +00:00
kjellander@webrtc.org
525baea03f Add PROJECT to codereview.settings 
This is needed once we move over to Chromium's
Rietveld instance at codereview.chromium.org.

BUG=3884
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7681 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 10:00:47 +00:00
kjellander@webrtc.org
944fb57140 Roll chromium_revision 375f736..91f1781 
Relevant changes:
* buildtools: 51ca1f2..c27f95b
* tools/gyp: b13d8f2..487c0b6
* tools/swarming_client: 41036ec..1f8ba35
Details: 375f736..91f1781/DEPS

Clang version was not updated in this roll.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7680 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 09:57:19 +00:00
magjed@webrtc.org
809986b95f webrtc::Scaler: Preserve aspect ratio 
BUG=3936
R=glaznev@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7679 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 09:51:30 +00:00
asapersson@webrtc.org
cd621a8657 Add thread annotations to overuse_frame_detector class. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 09:40:19 +00:00
henrik.lundin@webrtc.org
8038d42749 Follow-up fixes for G722 
This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.

BUG=3951
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 08:38:24 +00:00
turaj@webrtc.org
1431e4dd1c Revert 7675 "Make an AudioEncoder subclass for iSAC" 
Above CL did not compile on Android. Followings are links to Android builds

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320

> Make an AudioEncoder subclass for iSAC
> 
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/25019004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 01:44:13 +00:00
kwiberg@webrtc.org
05feff013e Make an AudioEncoder subclass for iSAC 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 23:53:08 +00:00
henrike@webrtc.org
33045ab2c1 Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003". 
BUG=3379
R=tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/27119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 19:43:11 +00:00
henrike@webrtc.org
43e033e778 Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted." 
BUG=3379
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7673 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 19:40:29 +00:00
andrew@webrtc.org
4ffc7341ca replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon by intrinsics. 
The modification only uses the unique part of the ResetAdaptiveChannel
 function. Pass byte to byte conformance test both on ARM32 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6. The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |       15% |       30% |        12% |
| Neon inline                |       21% |       30% |        12% |
| Neon intrinsics (GCC 4.6)  |       19% |       32% |        12% |
| Neon intrinsics (GCC 4.8)  |       20% |       32% |        12% |
| Neon intrinsics (LLVM 3.4) |       19% |       30% |        12% |

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29019004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7672 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 17:27:53 +00:00
andrew@webrtc.org
d024f759a8 clear asm code and unused functions in audio processing module 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25119004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7671 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 17:19:57 +00:00
henrike@webrtc.org
c4922316b4 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. 
TBR=niklas.enbom@webrtc.org
BUG=3379

Review URL: https://webrtc-codereview.appspot.com/30959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 15:31:24 +00:00
pbos@webrtc.org
d819803d45 Wire up DSCP support in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 14:41:43 +00:00
stefan@webrtc.org
83d4804a50 Put send-side bwe probing under finch experiment. 
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 13:55:16 +00:00
pbos@webrtc.org
957e802fe0 Refactor SetDefaultEncoderConfig to work on existing codecs. 
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/32009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 12:36:11 +00:00
pbos@webrtc.org
a5d29fcd59 Add unit to dropped frames. 
Missing unit causes less dropped frames to be reported as a regression
and not an improvement.

R=stefan@webrtc.org
BUG=chromium:429206

Review URL: https://webrtc-codereview.appspot.com/25139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7666 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 09:54:19 +00:00
kjellander@webrtc.org
bd495fab27 .gitignore updates 
Update after Chromium roll in https://review.webrtc.org/24179004/
and Android project updates in https://review.webrtc.org/25029004/

BUG=
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 06:51:34 +00:00
buildbot@webrtc.org
3c1970f9f3 (Auto)update libjingle 79414100-> 79428003 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 17:58:41 +00:00
andresp@webrtc.org
188d3b2245 Enable VP9 video codec support on webrtcvideoengine behind a field trial. 
BUG=chromium:431285
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 13:21:04 +00:00
henrik.lundin@webrtc.org
f85dbce041 Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" 
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.

BUG=3951
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 12:25:00 +00:00
perkj@webrtc.org
d105cc81dc Change dummy address to use 0.0.0.0 instead of :: 
This is to not break compatiblity with FF.

https://code.google.com/p/chromium/issues/detail?id=430333

TBR=pthatcher@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 11:22:06 +00:00
pbos@webrtc.org
d42a3adf42 Remove partially defined WebRtcRTPHeader from Parse(). 
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change.
To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader.

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

Review URL: https://webrtc-codereview.appspot.com/28919004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 11:02:12 +00:00
pbos@webrtc.org
a2ef4fe9c3 Prevent a lot of VideoSendStream reconfigures. 
Checking whether we're setting the same configuration or not.
Experimentally this brings down underlying reconfigures from ~20 to
about 4-5.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 10:54:43 +00:00
andresp@webrtc.org
82775b1396 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. 
This will allow to plugin VP9 based on a field trial.

R=pbos@webrtc.org, pbos, pthatcher

Review URL: https://webrtc-codereview.appspot.com/27949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 09:37:54 +00:00
henrika@webrtc.org
5e160660a6 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). 
Second attempt to land https://webrtc-codereview.appspot.com/32399004/

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 20:35:13 +00:00
pkasting@chromium.org
332331fb01 Use uint16s for port numbers in webrtc/p2p/base. 
This is a necessary precursor to using uint16s for port numbers more
consistently in Chromium code.

This also makes some minor formatting changes in surrounding code (function declaration wrapping, virtual -> override).

BUG=chromium:81439
TEST=none
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7656 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 20:19:22 +00:00
henrike@webrtc.org
d89b69aade Fix WebRTC Win64 + BoringSSL build. 
There were many size_t to int conversions. RAND_poll and RAND_seed no longer do
anything in BoringSSL, so fix that one by removing it. Use a checked_cast for
the remaining ones.

BUG=chromium:429039
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7655 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 17:23:09 +00:00
henrika@webrtc.org
dd43bbed8f Volume buttons in AppRTCDemo should affect output audio volume (part II). 
See https://webrtc-codereview.appspot.com/32399004/ for part I.

BUG=3279
TEST=AppRTC demo
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 15:48:05 +00:00
henrik.lundin@webrtc.org
dced5d7835 Revert "Advertise G722 as 8 kHz rather than 16 kHz" 
This reverts r7645.

TBR=pthatcher@webrtc.org
BUG=3951

Review URL: https://webrtc-codereview.appspot.com/24199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 15:27:43 +00:00
buildbot@webrtc.org
34bda43fa6 (Auto)update libjingle 79326895-> 79329222 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:44:55 +00:00
henrika@webrtc.org
e5421e9602 Volume buttons in AppRTCDemo should affect output audio volume. 
BUG=3279
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:19:19 +00:00
perkj@webrtc.org
fd0efb694a Remove deprecated PeerConnection APIs. 
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.

R=tommi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:16:36 +00:00
andresp@webrtc.org
19b4741004 Removing unused method GetDefaultVideoEncoderConfig. 
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 11:16:32 +00:00
pbos@webrtc.org
931e3da8f2 Log formatting fix for VideoEncoderConfig. 
R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:35:08 +00:00
buildbot@webrtc.org
0ef890a4ba (Auto)update libjingle 79285346-> 79320771 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:22:08 +00:00
mcasas@webrtc.org
6340acde68 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. 
Also removed some unused "summary" ListPreference
fields.

The looks of it can be found in [1] (lowest row).

[1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:05:48 +00:00
henrik.lundin@webrtc.org
1dcca4028f Advertise G722 as 8 kHz rather than 16 kHz 
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.

R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.

Review URL: https://webrtc-codereview.appspot.com/27879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 08:55:01 +00:00
kwiberg@webrtc.org
8b2058e733 Remove the state_ member from AudioDecoder 
The subclasses that need a state pointer should declare them---with
the right type, not void*, to get rid of all those casts.

Two small but not quite trivial cleanups are included because they
blocked the state_ removal:

  - AudioDecoderG722Stereo now inherits directly from AudioDecoder
    instead of being a subclass of AudioDecoderG722.

  - AudioDecoder now has a CngDecoderInstance member function, which
    is implemented only by AudioDecoderCng. This replaces the previous
    practice of calling AudioDecoder::state() and casting the result
    to a CNG_dec_inst*. It still isn't pretty, but now the blemish is
    plainly visible in the AudioDecoder class declaration.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24169005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7644 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 07:54:31 +00:00
kjellander@webrtc.org
32022c6fb1 Revert 7642 "Fix memcheck and dr memory after flakiness dashboar..." 
Reason: Turns out this does not solve the problem as the buildbots
run into another error later on during collecting gtest output.
The problem is solved by excluding these bots from flakiness dashboard
data instead, in https://codereview.chromium.org/705913002/

> Fix memcheck and dr memory after flakiness dashboard deployment.
> 
> I deployed buildbot configuration for uploading data to the
> flakiness dashboard but I didn't predict our Dr Memory and
> Memcheck bots would fail due to the new flag passed to the test.
> Adding the --gtest_output flag to the script will avoid the build
> to fail as a workaround.
> 
> TBR=andrew@webrtc.org
> TESTED=Passing test run using:
> src/tools/valgrind-webrtc/webrtc_tests.sh --test audio_decoder_unittests --tool memcheck --target Release --build-dir src/out --gtest_output=xml:audio_decoder_unittests.xml
> 
> BUG=
> 
> Review URL: https://webrtc-codereview.appspot.com/28999004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7643 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 06:26:34 +00:00
kjellander@webrtc.org
724fbaf473 Fix memcheck and dr memory after flakiness dashboard deployment. 
I deployed buildbot configuration for uploading data to the
flakiness dashboard but I didn't predict our Dr Memory and
Memcheck bots would fail due to the new flag passed to the test.
Adding the --gtest_output flag to the script will avoid the build
to fail as a workaround.

TBR=andrew@webrtc.org
TESTED=Passing test run using:
src/tools/valgrind-webrtc/webrtc_tests.sh --test audio_decoder_unittests --tool memcheck --target Release --build-dir src/out --gtest_output=xml:audio_decoder_unittests.xml

BUG=

Review URL: https://webrtc-codereview.appspot.com/28999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7642 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 06:04:09 +00:00
marpan@webrtc.org
7e4a05ec29 Exclude SendsAndReceivesVP9 for linux-memcheck. 
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7641 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 05:47:59 +00:00
andrew@webrtc.org
53bed75104 Change DrMemory exclusion to match changed test name. 
Due to:
https://code.google.com/p/webrtc/source/detail?r=7574

TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7640 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 05:33:01 +00:00
marpan@webrtc.org
f6b7c7e6a6 Exclude SendsAndReceivesVP9 for WinDrMemory. 
https://code.google.com/p/webm/issues/detail?id=872

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 05:09:26 +00:00
marpan@webrtc.org
e1745cbb7c Adjust parameter in vp9 rate control test. 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7638 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 02:55:53 +00:00
marpan@webrtc.org
5f1e2e42a8 Increase speed setting for VP9 (from 5 to 6) and re-enable end_to_end test. 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 02:02:28 +00:00
tkchin@webrtc.org
ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in 
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 22:01:53 +00:00
pbos@webrtc.org
6a364fe11b Remove uses of build date/time. 
Uses of __DATE__ and __TIME__ are blocking deterministic Chromium
builds. We're not really making use of these, and if anything they're
likely to be misleading as it's impossible to distinguish between a new
revision and a freshly-built old branch.

R=mflodman@webrtc.org, tnakamura@webrtc.org
BUG=3983

Review URL: https://webrtc-codereview.appspot.com/27039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7635 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 17:40:28 +00:00
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2. 
Adds stats to verify bandwidth and pacer stats.

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

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 14:05:29 +00:00