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Author SHA1 Message Date
marpan@webrtc.org
ceca014b8b Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9. 
BUG=4059

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 01:05:43 +00:00
pthatcher@webrtc.org
eb0954248d Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 00:34:10 +00:00
glaznev@webrtc.org
d01955179a Change minimum video encoder initialization resolution to 
176x144 to ensure HW encoder can be initialized.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 23:41:18 +00:00
andrew@webrtc.org
1751ee7d32 Remove -flax-vector-conversions flag for ARM NEON building. 
Pass compilation on both ARMv7 and ARM64. The generated
binary (audioproc) is byte to byte (with symbol striped) same as
before. The output of audioproc -aecm is also byte to byte same between
C and NEON version on ARMv7 and ARM64.

Change-Id: Ibdf40fe085f6bad1311f59bf9318bbcf37dd7ce5

BUG=3850
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/30239004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 19:36:14 +00:00
andrew@webrtc.org
ac68ef9ad4 Clear 2 unused functions in audio processing aecm module. 
unused functions:
WebRtcAecm_WindowAndFFTNeon
WebRtcAecm_InverseFFTAndWindowNeon

BUG=3580
R=andrew@webrtc.org

Change-Id: I12c50a8706d40f9ea98208b5733c00ede7b1f435

Review URL: https://webrtc-codereview.appspot.com/30269004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 18:33:52 +00:00
perkj@webrtc.org
beee9cec22 Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. 
The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.

TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 14:38:18 +00:00
henrik.lundin@webrtc.org
7f1dfa5b61 Adding a payload type to AudioEncoder objects 
The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 12:08:39 +00:00
kwiberg@webrtc.org
0cd5558f2b AudioEncoder subclass for G722 
BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 11:45:51 +00:00
kjellander@webrtc.org
84515f841d Roll chromium_revision 309cf65..24b4c73 
Two VP9 tests needed to be disabled (see webrtc:4059) to make all tests pass.

Relevant changes:
* src/third_party/android_tools: ea50ccc..4c47ef6
* src/third_party/icu: dd72764..866ff69
* src/third_party/libvpx: 2e5ced5..429874c
* src/third_party/nss: 258342e..bb4e75a
* src/third_party/yasm/source/patched-yasm: c960eb1..4671120
* src/tools/gyp: 0a381c0..fe00999
* src/tools/swarming_client: 5b827c9..1d4965c
Details: 309cf65..24b4c73/DEPS

Clang version was not updated in this roll.

BUG=4059
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 08:48:08 +00:00
pthatcher@webrtc.org
5950b645b9 Use c++11 features in webrtc/base/network.cc as a test to see if we can use them. 
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25209005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 23:18:27 +00:00
pthatcher@webrtc.org
146e0fd30f Fix the build by putting in a typecast to avoid a comparison between 
signed and unsigned ints introduced in cl/81073932.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:07:52 +00:00
glaznev@webrtc.org
dea5173edf Add start bitrate and vp8 hw acceleration option to 
Android AppRTCDemo.

- Add an option to set VP8 encoder start bitrate
usig x-google-start-bitrate line in remote SDP.
- Allow to enabled/disable VP8 hw decoder and
encoder acceleration using appRTC settings.

BUG=4046
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:02:13 +00:00
buildbot@webrtc.org
32ec0dd032 (Auto)update libjingle 81063831-> 81073932 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 17:57:36 +00:00
andresp@webrtc.org
7f722492f1 Set simulcastIdx field to zero even if it has no meaning. 
Helps to be able to memcmp between 2 parses of the same packet.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:29:29 +00:00
pbos@webrtc.org
273a414b0e Report encoded frame size in VideoSendStream. 
Implements reporting transmitted frame size in WebRtcVideoEngine2.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033

Review URL: https://webrtc-codereview.appspot.com/33399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:23:21 +00:00
henrik.lundin@webrtc.org
1db20a4180 Adding EncodedInfo struct to AudioEncoder::Encode 
This struct will be expanded in future changes.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 14:44:50 +00:00
henrik.lundin@webrtc.org
20446e7e56 Move and rename neteq/test/RTPcat to neteq/tools/rtpcat 
BUG=2692
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 14:23:01 +00:00
henrik.lundin@webrtc.org
c93437ef96 Add test NetEqDecodingTest.CngFirst 
This CL adds a test to verify that it is ok to start the stream with
a comfort noise packet.

BUG=4021
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 11:42:42 +00:00
henrik.lundin@webrtc.org
83317146ba Adding a new test helper RtpFileWriter and use it in RTPcat 
This new helper class writes RTP packets to file in rtpdump format.
A unit test is also included.

The new test class is used while re-writing the test tool RTPcat.

BUG=2692
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 11:25:04 +00:00
kjellander@webrtc.org
4796301c0e Whitespace change to force builds. 
TBR=buildbot@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 09:10:38 +00:00
kjellander@webrtc.org
e75f2cea5f Add FORCE_HTTPS_COMMIT_URL to codereview.settings. 
This will make it possible to use a https URL when committing
to SVN from from git-svn checkouts created with 'fetch webrtc'
(i.e. from a pure Git mirror in Chrome infrastructure).

This will have effect only after
https://codereview.chromium.org/760903004/ is landed.

BUG=chromium:412012
TESTED=This CL will be committed using git cl dcommit from
a checkout created with 'fetch webrtc', combined
with depot_tools patched with https://codereview.chromium.org/760903004/

Review URL: https://webrtc-codereview.appspot.com/32569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 09:09:07 +00:00
kjellander@webrtc.org
cc7755becd Whitespace change 
TBR=buildbot@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-29 16:47:53 +00:00
kjellander@webrtc.org
74499efc05 Add whitespace.txt file. 
This is useful as a recommended way to trigger now builds
with a noop change.
I believe it's going to be used more frequently as we're closing
in on the Git switch, to test committing and pushing.

TBR=phoglund@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7764 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-29 15:42:29 +00:00
tommi@webrtc.org
2c13f659c7 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-28 10:37:31 +00:00
asapersson@webrtc.org
83b5200f95 Add framerate for complete received frames to histogram stats: 
"WebRTC.Video.CompleteFramesReceivedPerSecond".

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-28 10:17:13 +00:00
aluebs@webrtc.org
cc144deaab Make bands vector in SplittingFilter Analysis const 
BUG=webrtc:3146
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-28 00:26:27 +00:00
aluebs@webrtc.org
8789376cd3 Move ChannelBuffer class to channel_buffer file 
No change in functionallity.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 23:40:25 +00:00
pbos@webrtc.org
d87213af49 Remove unused RtpStatistics struct. 
This unused struct is basically a copy of RtcpStatistics in
webrtc/common_types.h. I expect this wasn't properly removed when that
one was added.

R=tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/25239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7758 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 13:48:35 +00:00
kjellander@webrtc.org
7d4e6d012c Roll chromium_revision d8c9041..309cf65 
Relevant changes:
* testing/gtest 4650552..8245545
* testing/gmock 896ba0e..2976396
* third_party/boringssl 2f3ba91..69a0160
* third_party/icu: 6242e2f..dd72764
* third_party/libyuv: 5a09c3e..d204db6
* tools/gyp: b13d8f2..0a381c0

Details: d8c9041..309cf65/DEPS

Clang version was not updated in this roll.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 10:41:04 +00:00
asapersson@webrtc.org
d952c40c7e Add receive bitrates to histogram stats: 
- total bitrate ("WebRTC.Video.BitrateReceivedInKbps")
- media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps")
- rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps")
- padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps")

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 07:38:56 +00:00
tkchin@webrtc.org
3e9ad26112 Refactor iOS AppRTC parsing code. 
Moved parsing code to JSON categories for the relevant objects.
No longer prefer ISAC as audio codec.

BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31989005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 00:52:38 +00:00
aluebs@webrtc.org
79b9eba3ab Implement 3 band splitting filter bank by upsampling and splitting twice into 2 bands 
Implemented the 3 bands splitting filter bank by:
1. Upsample by 4/3.
2. Split twice into 2 bands.
3. Discard upper most band, because it is empty anyway.

A unittest was also implemented:
1. Generate a signal from presence or absence of sine waves of different frequencies.
2. Split into 3 bands and check their presence or absence.
3. Recombine the bands.
4. Calculate delay (as it is an IIR it depends on frequency).
5. Check that the cross correlation of input and output is high enough at that delay.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 20:21:38 +00:00
jiayl@webrtc.org
7806d8fe40 Fix an ASSERT that fires in a browser test for renegotiation. 
See https://code.google.com/p/chromium/issues/detail?id=293125#c33

BUG=crbug/293125
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7753 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 19:58:50 +00:00
sprang@webrtc.org
a71bb6033b Revert 7750 "Don't reset sequence number for a stream on deactiv..." 
> Don't reset sequence number for a stream on deactivate/reactivate.
>
> BUG=chromium:431908
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/32199004

TBR=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 19:33:15 +00:00
andrew@webrtc.org
a56a2c57cf Enabling building with NEON on ARM64 
This patch enables NEON on ARM64 platform. Passed building both on
Android ARMv7 and Android ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/25069004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 17:01:40 +00:00
sprang@webrtc.org
31f7a0e710 Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 16:55:52 +00:00
henrik.lundin@webrtc.org
91d928e737 Rename RtpFileReader::Packet to RtpPacket and move out of RtpFileReader 
This is in preparation for creating a new class RtpFileWriter which
will use the same RtpPacket struct.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7749 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 15:50:30 +00:00
perkj@webrtc.org
2faf7eea6f Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" 
This reverts commit 308e7ff613.

Original cl description:

This adds an Android apk for running tests on the Java layer of PeerConnection.

The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 07:35:37 +00:00
glaznev@webrtc.org
58edb83fd4 Add video encoder fps and bitrate statistics to 
Android AppRTCDemo UI.

BUG=4045
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 00:39:42 +00:00
pbos@webrtc.org
008731868a Implement settable min/start/max bitrates in Call. 
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

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

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 14:03:34 +00:00
pbos@webrtc.org
b951eb12c9 Add back EXPECT_TRUEs. 
These shouldn't fail, but EXPECT_TRUE gives nicer error messages that
work in Release. These changes got through unreviewed in r7726.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 11:13:28 +00:00
pbos@webrtc.org
ba253473da Reenable GetStats test. 
Also increasing start bitrate to have the test go significantly faster
on average. Hopefully an assert hit in the jitter buffer while running
this test was fixed in r7735.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/26239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 09:39:04 +00:00
glaznev@webrtc.org
dab5d92df6 Use mirror image for Android AppRTCDemo local preview. 
Similar to Chrome apprtc using mirror image for camera
local preview provides better experience when device
is rotated.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 17:31:01 +00:00
henrik.lundin@webrtc.org
03499a0e95 Add wav output capability to neteq_rtpplay 
This CL makes neteq_rtpplay capable of writing to wav files as well as
pcm files. This is done through the new class OutputWavFile, which
wraps a WavWriter object in an AudioSink interface.

BUG=2692
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 14:50:53 +00:00
henrik.lundin@webrtc.org
aff1751c96 Add new test for VP8 packetizer to test tight partitions 
It was discovered that if remaining_bytes is an exact multiple of
max_payload_len in RtpPacketizerVp8::CalcNextSize, then the packetizer
will produce too many packets (i.e., split the payload into more
packets than needed).

This CL adds a test to trigger the problem.

BUG=4019
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 12:36:58 +00:00
kjellander@webrtc.org
dde19a6f60 sync_chromium.py: Check for chromium/src 
Make sure the script alwyas downloads Chromium
if there's no current download. This case can happen
if a user is removing the 'src' folder but doesn't know
to remove the .last_sync_chromium file.

BUG=
TESTED=Renamed chromium/src and ran a sync keeping the .last_sync_chromium file, verified it started downloading.
TBR=iannucci@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 10:08:03 +00:00
kjellander@webrtc.org
3398a4ac15 PRESUBMIT: Only notify GN changes for GYP files in webrtc/* 
We don't maintain a BUILD.gn file for talk/ since it's a part
of Chromium:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjingle/BUILD.gn
Because of this, it's confusing to get warnings about updating
a GYP file in talk/ from the PRESUBMIT check.

TESTED=Successsfully ran git cl presubmit with this change
applied on top of a CL containing changes in .gyp files.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 10:05:37 +00:00
kjellander@webrtc.org
8562f23acb OWNERS: Remove tomasl@ and mallinath@ 
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport

R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 10:05:05 +00:00
pbos@webrtc.org
4f16c874c6 Simplifying VideoReceiver and JitterBuffer. 
Removing frame_buffers_ array and dual-receiver mechanism. Also adding
some thread annotations to VCMJitterBuffer.

R=stefan@webrtc.org
BUG=4014

Review URL: https://webrtc-codereview.appspot.com/27239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 09:06:48 +00:00
pbos@webrtc.org
9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 08:25:50 +00:00