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Author SHA1 Message Date
buildbot@webrtc.org
dd4742a9ef (Auto)update libjingle 66388864-> 66406192 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 14:50:35 +00:00
buildbot@webrtc.org
ed97bb0eb4 (Auto)update libjingle 66340694-> 66388864 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 11:15:20 +00:00
braveyao@webrtc.org
94f1d4cd55 Fix odd codes in video_capture on Mac. 
BUG=3272
TEST=vie_auto_test
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 02:57:13 +00:00
buildbot@webrtc.org
f9277a9381 (Auto)update libjingle 66326258-> 66340694 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 00:29:05 +00:00
fischman@webrtc.org
b1eb43142e video_render.gypi: clean up some libraries directives to be more specific. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-07 00:09:30 +00:00
buildbot@webrtc.org
861d4b0de9 (Auto)update libjingle 66322380-> 66326258 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 22:11:02 +00:00
henrike@webrtc.org
3129e684a3 openmax_dl was not added to .gitignore in r6037. 
BUG=N/A
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 21:47:45 +00:00
buildbot@webrtc.org
0581f0ba0a (Auto)update libjingle 66303009-> 66322380 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 21:36:31 +00:00
buildbot@webrtc.org
a18b4c96af (Auto)update libjingle 66301332-> 66303009 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:48:14 +00:00
buildbot@webrtc.org
e65c9a6e67 (Auto)update libjingle 66299810-> 66301332 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:28:28 +00:00
buildbot@webrtc.org
0b53bd29af (Auto)update libjingle 66294299-> 66299810 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 17:12:36 +00:00
buildbot@webrtc.org
150835ea34 (Auto)update libjingle 66236292-> 66294299 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 15:54:38 +00:00
tina.legrand@webrtc.org
cfb18dd7a3 Rolling new version of Opus.gyp 
The new version enables optimizations on iOS.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 11:57:01 +00:00
kjellander@webrtc.org
e5e16d7515 Update svn:ignore for resources and third_party. 
This prevents the following from getting wiped on each build:
resources/*.dat
third_party/binutils
third_party/libc++
third_party/libc++abi
third_party/openmax_dl



git-svn-id: http://webrtc.googlecode.com/svn/trunk@6059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 08:14:38 +00:00
wu@webrtc.org
ed4cb56575 Remove timestamp_extrapolator's dependency to Clock and vcm defines. 
TEST=existing tests
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-06 04:50:49 +00:00
buildbot@webrtc.org
5ee0f05d5f (Auto)update libjingle 66138442-> 66236292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 20:18:08 +00:00
andrew@webrtc.org
382c0c209d Allow the RTP level indicator computation to work at any sample rate. 
Break out the computation to a separate class, and call directly into
this from channel.cc rather than going through AudioProcessing. This
circumvents AudioProcessing's sample rate limitations.

We now compute the RMS over all samples rather than downmixing to a
single channel. This makes the call point in channel.cc easier, is
more "correct" and should have similar (negligible) complexity.

This caused slight changes in the RMS output, so the ApmTest.Process
reference has been updated. Snippet of the failing output:

[ RUN      ] ApmTest.Process
Running test 4 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 5 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 6 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 10 of 12...
Value of: rms_level
  Actual: 27
Expected: test->rms_level()
Which is: 28
Running test 11 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27
Running test 12 of 12...
Value of: rms_level
  Actual: 26
Expected: test->rms_level()
Which is: 27

BUG=3290
TESTED=Chrome assert is avoided and both voe_cmd_test and apprtc
produce reasonable printed out results from RMS().

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 18:22:21 +00:00
andrew@webrtc.org
a0edf4cb04 Remove ALLOW_UNUSED. 
Turns out Chromium won't be applying this to COMPILE_ASSERT. We don't
need it at all then.

R=thakis@chromium.org
TBR=thakis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13469005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 18:18:02 +00:00
wu@webrtc.org
0224c20fa6 * Add 100ms network delay to test CaptureNtpTimeWithNetworkJitter. 
* Re-enable test CaptureNtpTimeWithNetworkJitter.
* Use 100ms as the threadhold as a FYI since this is a performance test.

BUG=3271
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 17:42:43 +00:00
jiayl@webrtc.org
4220434d37 Implement the Windows screen capturer using the Magnification API. 
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.

BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org, wez@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6048

Review URL: https://webrtc-codereview.appspot.com/12149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 16:08:47 +00:00
tina.legrand@webrtc.org
7dccce3948 Revert 6048 "Implement the Windows screen capturer using the Mag..." 
> Implement the Windows screen capturer using the Magnification API.
> The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.
> 
> BUG=2789
> TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
> R=sergeyu@chromium.org, wez@chromium.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12149004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 11:17:26 +00:00
braveyao@webrtc.org
633aff6bd0 WebRTCDemo: correct set trace filter operation. 
BUG=3285
TEST=Manul Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 04:24:47 +00:00
andrew@webrtc.org
9f453b1a1b Add ALLOW_UNUSED and update COMPILE_ASSERT to Chromium's latest. 
Fixes building with gcc 4.8.

TBR=fdegans@google.com
BUG=chromium:321833

Review URL: https://webrtc-codereview.appspot.com/12439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-04 03:04:26 +00:00
buildbot@webrtc.org
41451d4e55 (Auto)update libjingle 66106643-> 66138442 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-03 05:39:45 +00:00
jiayl@webrtc.org
b235c56017 Implement the Windows screen capturer using the Magnification API. 
The original ScreenCapturerWin is renamed ScreenCapturerWinGdi.

BUG=2789
TESTED=full desktop cast and single monitor cast works on win7 and win8 desktop mode. Have to use GDI capturer on win8 metro mode. Changing display configuration work on the fly.
R=sergeyu@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-03 00:16:29 +00:00
kjellander@webrtc.org
cdaf2b9b73 Add svn:ignore to resources 
This will prevent re-download of the files in
resources/remote_bitrate_estimator for each new build.



git-svn-id: http://webrtc.googlecode.com/svn/trunk@6047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 19:40:15 +00:00
buildbot@webrtc.org
cc06c75f28 (Auto)update libjingle 66100938-> 66106643 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 18:51:11 +00:00
buildbot@webrtc.org
13d6776c46 (Auto)update libjingle 66098243-> 66100938 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 17:33:29 +00:00
buildbot@webrtc.org
0d34f1446a (Auto)update libjingle 66033941-> 66098243 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 16:54:25 +00:00
henrika@webrtc.org
7f3a041d23 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. 
BUG=3289
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 13:59:58 +00:00
asapersson@webrtc.org
9205c87820 Pointers were not dereferenced in GetRtpStatistics. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 13:24:42 +00:00
stefan@webrtc.org
24bd364d3e Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. 
This fixes an issue where the user doesn't know which channels are "active" and therefore can't properly sum the estimates for all channels.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 12:35:37 +00:00
fischman@webrtc.org
e3a628997f Roll libvpx 264320:267596 
BUG=3038
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 00:04:20 +00:00
fischman@webrtc.org
14ea7e8922 AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation. 
- tap display to toggle visibility
- increased getStats frequency to 1hz.

R=glaznev@google.com

Review URL: https://webrtc-codereview.appspot.com/19419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 20:57:55 +00:00
fischman@webrtc.org
dd92feb6dd AppRTCDemo(android): send the created SDP, not the local description after setting it 
This is required to allow explicit filtering of ICE candidates.

BUG=3288
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 19:06:18 +00:00
turaj@webrtc.org
560dce5d48 Pull openmax into third_party. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6037 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:53:51 +00:00
jiayl@webrtc.org
9c16c39e61 Sets the SCTP port codec in the native SessionDescription. 
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.

BUG=3141
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:30:30 +00:00
jiayl@webrtc.org
53d82350c5 Ignore identical remote fingerprint in DtlsTransportChannelWrapper::SetRemoteFingerprint. 
Trying to set the same remote fingerprint could happen during renegotiation and should not fail.

BUG=crbug/362431
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 00:00:19 +00:00
andrew@webrtc.org
e44a84d851 Only clamp to 16 kHz when AECM is enabled. 
Otherwise we could needlessly downsample to 16 kHz (rather than 32 kHz)
when HW AEC is used.

BUG=3259
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 18:58:23 +00:00
tkchin@webrtc.org
ff2733204d Implement ObjC DataChannel wrapper 
R=fischman@webrtc.org
BUG=3112

Review URL: https://webrtc-codereview.appspot.com/16369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 18:32:33 +00:00
andrew@webrtc.org
65f933899b Fix constness of AudioBuffer accessors. 
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.

Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/

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

Review URL: https://webrtc-codereview.appspot.com/15379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 16:44:13 +00:00
buildbot@webrtc.org
740e6b339a (Auto)update libjingle 65843899-> 65880186 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 15:33:45 +00:00
fischman@webrtc.org
7c82adae61 AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial. 
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16379004

Patch from Bridger Maxwell <bridgeyman@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 00:17:47 +00:00
turaj@webrtc.org
9bd49becc1 Fix a data race in ACM1 when audio is pulled. 
BUG=chromium:348511
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 20:27:45 +00:00
fischman@webrtc.org
a86c42c424 libjingle_unittest now compiles and passes on iOS! (reland of r5986) 
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && \
  ~/src/ios-deploy/ios-deploy -d -u -v -b \
    ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app

Note that the test's use of signals means that lldb will break in the middle
of the suite.  To ignore these signals tell lldb:

pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue

BUG=3241
R=kjellander@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 18:37:29 +00:00
buildbot@webrtc.org
681f787cc4 (Auto)update libjingle 65752960-> 65813736 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:55:26 +00:00
henrike@webrtc.org
f2aafe4355 Added include of assert.h for files calling assert but missing the include. 
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19409005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:54:17 +00:00
fischman@webrtc.org
f04a6ea733 MediaCodecVideoEncoder: limit MediaCodec bitrate to 95% of requested to avoid overshoot. 
BUG=3194
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/17379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:53:30 +00:00
henrike@webrtc.org
82d3cb68cd Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 
BUG=N/A
R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:50:47 +00:00
henrike@webrtc.org
ceffdbc371 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() 
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:11:07 +00:00