Commit Graph

6935 Commits

Author SHA1 Message Date
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
buildbot@webrtc.org
a22a628356 (Auto)update libjingle 79205306-> 79244016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7633 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 13:25:48 +00:00
kjellander@webrtc.org
72fd339352 Restore old behavior for Android in fileutils.cc
From r7014 the Android APK tests are designed to be
build from a standalone WebRTC checkout instead of a
Chromium checkout. Because of that, the special handling
for both cases can be removed.

I also don't think we need to use the
base::android::GetExternalStorageDirectory() method since
all devices has a symlink at /sdcard that points
to /storage/emulated/legacy on the Android device.

This essentially reverts the changes in
https://webrtc-codereview.appspot.com/1754005/
plus some minor changes.

BUG=webrtc:3741
TEST=Locally running test_support_unittests APK test on an
Android device using:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s test_support_unittests --verbose --isolate-file-path=webrtc/test/test_support_unittests.isolate
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7632 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 06:28:50 +00:00
kjellander@webrtc.org
f6e1600a7d Roll chromium_revision d3db2ff..375f736
Mainly to bring in https://codereview.chromium.org/636083002
updates to opus.

Relevant changes (d3db2ff..375f736/DEPS)
* buildtools: 6bc51af..1ca1f25
* third_party/libsrtp: 98284c8..6446144
* tools/gyp: a61e860..b13d8f2
* tools/swarming_client: bcb3bc3..41036ec

Clang version was not updated in this roll.

BUG=
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/24179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7631 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 02:09:20 +00:00
glaznev@webrtc.org
dc8662435b Fix android_clang build.
BUG=
R=kjellander@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7630 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 01:15:10 +00:00
niklas.enbom@webrtc.org
368215dacb Revert 7623 "Remove the state_ member from AudioDecoder"
Breaks Chrome compile:
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(131) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3867: 'webrtc::NetEqImpl::InsertPacketInternal': function call missing argument list; use '&webrtc::NetEqImpl::InsertPacketInternal' to create a pointer to member
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(152) : error C3861: 'LOG_FERR1': identifier not found
e:\b\build\slave\win_builder\build\src\third_party\webrtc\modules\audio_coding\neteq\neteq_impl.cc(169) : error C3867: 'webrtc::NetEqImpl::GetAudioInternal': function call missing argument list; use '&webrtc::NetEqImpl::GetAudioInternal' to create a pointer to member
...

> 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

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30879005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7629 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:45:58 +00:00
niklas.enbom@webrtc.org
8a232f65dd Revert 7625 "Don't use DCHECK when you need the side effects..."
Reverting since 7623 might depend on this one

> Don't use DCHECK when you need the side effects...
> 
> R=pbos@webrtc.org
> TBR=henrik.lundin@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/32369004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7628 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:43:59 +00:00
buildbot@webrtc.org
795d003770 (Auto)update libjingle 79200114-> 79205306
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 00:14:02 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface.
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 23:06:15 +00:00
kwiberg@webrtc.org
b8425bc4f3 Don't use DCHECK when you need the side effects...
R=pbos@webrtc.org
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7625 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 22:10:18 +00:00
buildbot@webrtc.org
45ecf4c092 (Auto)update libjingle 79169148-> 79192489
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 21:48:54 +00:00
kwiberg@webrtc.org
9e525585fd 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@7623 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 21:18:47 +00:00
stefan@webrtc.org
7c29e8c2f3 Add support for VP9 in webrtc::Call and video_loopback.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7622 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:41:15 +00:00
stefan@webrtc.org
d839e0ab52 Reduce to 2 probes when probing for initial bandwidth.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23359005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:33:55 +00:00
stefan@webrtc.org
db26247a9b Add UMA for measuring the diff between the BWE at 2 seconds compared to the BWE at 20 seconds when the BWE should have converged.
BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7620 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:32:10 +00:00
mcasas@webrtc.org
8944c9d08b AppRTCDemoActivity: use differnet Themes for different API levels
The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.

I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)

Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 17:26:22 +00:00
kjellander@webrtc.org
d367321a3f Add kjellander as PRESUBMIT.py OWNER
BUG=
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31949005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7618 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 17:06:31 +00:00
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

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

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:27:16 +00:00
pbos@webrtc.org
fad9aecbf5 Remove protected files from talk/PRESUBMIT.py.
All files may now be committed to.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:06:35 +00:00
pbos@webrtc.org
88ef632286 Falling back on single-stream on multiple SSRC.
Instead of failing, use one stream. Also clamp video min bitrate.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 15:29:29 +00:00