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
pbos@webrtc.org
0300939484
Disable failing GoogleWifiTrace3Mbps.
...
Disables BweFeedbackTest.GoogleWifiTrace3Mbps instead of
BweSimulation.GoogleWifiTrace3Mbps.
TBR=stefan@webrtc.org
BUG=3277
Review URL: https://webrtc-codereview.appspot.com/20389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6017 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 15:25:59 +00:00
pbos@webrtc.org
9353e6bc55
Disable GoogleWifiTrace3Mbps.
...
Breaks bots, according to stefan@ there's a missing file for this test
to run.
TBR=stefan@webrtc.org
BUG=3277
Review URL: https://webrtc-codereview.appspot.com/13409005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6016 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:49:56 +00:00
stefan@webrtc.org
dfe2a1c995
Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6015 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 14:21:42 +00:00
mflodman@webrtc.org
f223746521
Upping start bitrate to min, if set to a lower value i SetSendCodec.
...
BUG=3276
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21379005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 12:38:42 +00:00
pbos@webrtc.org
f5433753b8
Suppress DrMemory allocator mismatch errors.
...
Bot is flooded with these errors, suppressing them to get back to a
normal state, contacting DrMemory team.
R=kjellander@webrtc.org
BUG=3275
TEST=git try -b win_drmemory_light -r 6012
Review URL: https://webrtc-codereview.appspot.com/13419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6013 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 12:24:45 +00:00
kjellander@webrtc.org
97e67cb476
Fix iOS assembly compile error.
...
In the roll of
https://webrtc-codereview.appspot.com/13369007
the fix in transform_neon.S was incorrectly removed
assuming it was only affecting Android when rolling to
265795. This CL fixes the iOS build when rolled to
266514.
Error looks like:
[893/2157] CC obj/webrtc/modules/audio_coding/codecs/isac/main/source/iSAC.entropy_coding.o
FAILED: /Volumes/data/b/build/goma/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang -MMD -MF obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o.d -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DDISABLE_NACL -DCHROMIUM_BUILD -DUSE_LIBJPEG_TURBO=1 -DENABLE_CONFIGURATION_POLICY -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DENABLE_EGLIMAGE=1 -DCLD_VERSION=1 -DENABLE_SPELLCHECK=1 -DDISABLE_FTP_SUPPORT=1 -DWEBRTC_RESTRICT_LOGGING -DWEBRTC_MODULE_UTILITY_VIDEO -DWEBRTC_ARCH_ARM -DWEBRTC_ARCH_ARM_V7 -DWEBRTC_ARCH_ARM_NEON -DWEBRTC_POSIX -DWEBRTC_MAC -DWEBRTC_IOS -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DNDEBUG -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DNS_BLOCK_ASSERTIONS=1 -D_FORTIFY_SOURCE=2 -I../.. -I../.. -I../../webrtc -I../../webrtc/common_audio/resampler/include -I../../webrtc/common_audio/signal_processing/include -I../../webrtc/common_audio/vad/include -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk -Os -gdwarf-2 -fvisibility=hidden -Werror -Wnewline-eof -miphoneos-version-min=6.0 -arch armv7 -Wall -Wendif-labels -Wextra -Wno-unused-parameter -Wno-missing-field-initializers -Wheader-hygiene -Wno-c++11-narrowing -Wno-char-subscripts -Wno-unneeded-internal-declaration -Wno-covered-switch-default -Wstring-conversion -Wno-deprecated-register -Wno-absolute-value -Wno-selector-type-mismatch -std=c99 -fcolor-diagnostics -c ../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S -o obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:45:11: error: immediate expression for mov requires :lower16: or :upper16
mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
^
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:458:11: error: immediate expression for mov requires :lower16: or :upper16
mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
in
http://build.chromium.org/p/client.webrtc/builders/iOS%20Release/builds/911/steps/compile/logs/stdio
TBR=ajm
TEST=ios trybots passing tryjob based on r6010.
BUG=
Review URL: https://webrtc-codereview.appspot.com/12439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 10:25:30 +00:00