e6c3966530
Fix some chromium-style warnings in webrtc/test/
...
BUG=163
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1907004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4428 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-30 13:08:38 +00:00
a6f56acc53
Fix some chromium-style warnings in webrtc/tools/
...
BUG=163
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1908004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4427 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-30 12:50:59 +00:00
096515b070
Fix some chromium-style warnings in webrtc/modules/audio_device/
...
BUG=163
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1897005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4426 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-30 12:32:59 +00:00
d818dcb939
Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData.
...
R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1841004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4424 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 21:48:11 +00:00
d6134c7cfd
PeerConnectionTest.java: make the test work for the bots' v4l2loopback.
...
- Make the test agnostic to the actual resolution used, since v4l2_file_player
is playing a non-640x480 file (go/httfw)
- Teach DeviceInfoLinux::FillCapabilityMap() about I420 since that's what
v4l2_file_player is feeding.
Requires https://gist.github.com/fischman/2e9a9b2efd2ad363ef82 be applied to the
v4l2loopback driver code.
BUG=1796
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1891004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4422 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 20:43:15 +00:00
7694562805
Land http://webrtc-codereview.appspot.com/1632005/
...
TBR=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1895004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4420 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 18:37:32 +00:00
c0aa29c98c
Updated WebRTC version to 3.37
...
TBR=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1894004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4417 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 16:57:21 +00:00
8400246fce
Improved error messages when binaries are missing. Also stderr = stdout now.
...
Now that these scripts are called from browser tests, we need to print everything on stdout since the tests will throw away stderr when invoking programs. I chose to assign sys.stderr to sys.stdout. Otherwise I would have missed stuff like parser.error, which print to stderr.
The error message will get improved because the old code did not catch the case when the binary was missing, which lead to a very confusing error when that was the case. This gets fixed now.
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1886004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4416 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 11:01:03 +00:00
f87177a757
To fix a bug in InverseFFTAndWindow() function in AECM.
...
It's a bufer overwritting issue, and thus Android AppRTCDemo app was broken (reported by Ami).
Tested with audioproc offline test. Bit-exact.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4415 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-26 23:43:33 +00:00
b6a6a24fda
Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c , line 594 "static void PrepareSpectrumC()".
...
Tested with audioproc. Bit exact.
R=andrew@webrtc.org , johannkoenig@google.com
Review URL: https://webrtc-codereview.appspot.com/1859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4411 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-26 16:24:34 +00:00
b6433b7a1e
Access receiving_ under receive_cs critical section
...
Note: InsertRTPPacket/InsertRTCPPacket could be merged into
ReceivedRTPPacket, as there are no other callers.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4410 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-26 09:02:46 +00:00
abab1d8456
Don't set clang_use_chrome_plugins in common.gypi
...
This caused a failure on chrome os ASAN bots (where that flag is disabled):
http://build.chromium.org/p/chromium.memory/builders/Chromium%20OS%20%28x86%29%20ASAN/builds/5491
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1882004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4408 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-26 00:55:46 +00:00
14c966c706
Fixes resources and data path in modules_unittests.isolate.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4407 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 22:44:04 +00:00
b86fbaf1d4
Downstream latest Chromium SincResampler changes.
...
Replace the BlockSize() workaround we were using previously to support
the push wrapper with the upstream request_frames interface. This
requires a bit of a trick to ensure we don't add more delay than
necessary. On the first pass we use a dummy Resample() call in order to
prime the buffer such that all later calls only require a single input
request through Run().
Notably, this brings in an optimized loop condition, improving
performance by ~2% - 3% on tested platforms and avoids a 20% performance
hit with clang. This addresses issue2041.
Only negligible changes to the PushSincResamplerTest SNR thresholds, due
to a fractional sample adjustment in output delay.
This still retains the per-instance CPU detection, as webrtc lacks a
LazyInstance helper for static initialization.
Ideally, we would adopt SetRatio() in PushSincResampler's
InitializeIfNeeded() for on-the-fly changes, but this will require a way
to update request_frames.
The diff against Chromium upstream is available here:
https://codereview.chromium.org/19470003
BUG=2041
TESTED=unit tests, voe_cmd_test in loopback running through all codecs
with 44.1 kHz and 48 kHz device formats using a stereo mic.
R=dalecurtis@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1838004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4406 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 22:04:30 +00:00
099b8c9e8e
Update include paths in device_info_external.cc
...
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1875004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4401 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 18:41:43 +00:00
61e596fc49
Add a Config class interface to AudioProcessing for passing options.
...
Pass the Config down to all AudioProcessing components.
Also add an EchoCancellationImplWrapper to optionally create different
EchoCancellationImpls.
BUG=2117
TBR=turaj@webrtc.org
TESTED=git try
Review URL: https://webrtc-codereview.appspot.com/1843004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4400 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 18:28:29 +00:00
8e3bbedacd
Fix include path in video_capture_external.cc
...
Fix build error introduced in r4337
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1873004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4397 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-25 16:55:58 +00:00
fc8aaf02e1
Formalized Real 16-bit FFT for APM.
...
It also prepares for introducing Real 16-bit FFT Neon code from Openmax to SPL. CL https://webrtc-codereview.appspot.com/1819004/ takes care of that, but this CL is a prerequisite of that one.
Tested audioproc with an offline file. Bit exact.
R=andrew@webrtc.org , rtoy@google.com
Review URL: https://webrtc-codereview.appspot.com/1830004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4390 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-24 17:38:23 +00:00
d102e66ef9
Fix ScreenCapturerLinux not to use XDamage when requested.
...
When moving this code to webrtc I added line "use_x_damage=true" for
debugging and forgot to remove it when landing this code, so the
capturer always tries to use XDamage.
BUG=crbug.com/263003
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1854004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4387 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 20:05:42 +00:00
678cf29d8b
webrtc/common_types.h: Document bitrate fields' units.
...
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1847004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4386 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 18:32:10 +00:00
8d27a1c723
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
...
BUG=1932
TESTED=git try
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1851004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4385 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 18:15:11 +00:00
6879c8adad
Hooking up first simple CPU adaptation version.
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1767004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 11:35:00 +00:00
5c280ecd57
Revert 4382 "Makes webrtc and libjingle build from the same gyp-..."
...
Failures: breaks build bots. Will have to disable Android NDK build for libjingle. The TSAN issues are in webrtc which should be unaffected. Flakey? Here are the failing tests:
http://chromegw/i/internal.client.webrtc/builders/Android%20NDK/builds/303 and http://chromegw/i/internal.client.webrtc/builders/Linux%20Tsan/builds/284
> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
>
> BUG=1932
> TESTED=git try
> R=andrew@webrtc.org , fischman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1836004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1834005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4383 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 03:30:32 +00:00
5fcddf2334
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
...
BUG=1932
TESTED=git try
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1836004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4382 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-23 00:27:43 +00:00
390fcb7a20
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
...
BUG=2091
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1833004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4381 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-22 22:32:50 +00:00
129afc29fb
Correctly rebuild WebRTCDemo after jni/ source file changes
...
BUG=1980
TEST=Modify source file under jni/ and WebRTCDemo will rebuild
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1831004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4377 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-20 04:43:08 +00:00
0df5b8dfa6
Revert 4372 "Makes webrtc and libjingle build from the same gyp-..."
...
> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
>
> TESTED=git try
> BUG=1932
> R=andrew@webrtc.org , fischman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1804004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1835004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4373 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-18 18:58:29 +00:00
4e4bf4db8b
Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
...
TESTED=git try
BUG=1932
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1804004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4372 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-18 18:33:55 +00:00
c6d5b50b41
AppRTCDemo: build fixes for iOS build in webrtc
...
BUG=1421,1450,1451
TESTED=git try, also the same patch (along with a bunch of other, non-webrtc changes) in a libjingle checkout allows building iOS AppRTCDemo
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4371 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-18 02:02:07 +00:00
d2102afa2a
Undo libvpx include changes in r4348 to fix build.
...
A longer term fix is needed, but this at least quickly unblocks the build.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1816005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4367 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-17 18:48:24 +00:00
a3f30143b7
Default constructor for RtcpAppHandler.
...
Whenever this test (RtcpApplicationDefinedPacketsCanBeSentAndReceived) fails
because it's being run on a slower system (such as one running under valgrind),
valgrind reports a lot of undefined-value errors. Initializing the data
makes sure that, while the EXPECT_EQs trigger, they don't cause any errors in
valgrind.
BUG=
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1822004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4363 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-17 14:25:45 +00:00
64e2cbf184
clean up incomplete revert in r4357
...
Also revert r4319, will follow up with pbos
Reason for recent series of reverts: video freezes when testing with packet loss
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1817004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 21:52:59 +00:00
aa4d96a134
Revert r4301
...
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 19:25:04 +00:00
42581545eb
Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.
...
Libjingle sets the variables here: https://code.google.com/p/libjingle/source/browse/trunk/talk/supplement.gypi
BUG=N/A
R=andrew@webrtc.org , fischman@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1787005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4354 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 16:37:22 +00:00
3d8647f17d
Include files from webrtc/.. paths in signal_processing/.
...
BUG=1662
R=andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1784004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4352 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:32:03 +00:00
0c4e05afbb
Include files from webrtc/.. paths in media_file/.
...
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1784005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4351 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:05:40 +00:00
9b82dced8d
Make sure first RTP packet counts as in-order.
...
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1811004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 13:03:35 +00:00
2e10b8e4a0
Include files from webrtc/.. paths in bitrate_controller/.
...
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1787004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4349 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:54:53 +00:00
a4407329d4
Include files from webrtc/.. paths in video_coding/.
...
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
4a44ea21d7
Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc"
...
TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1803004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:46:06 +00:00
4888fd4827
Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered"
...
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1790006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4345 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:21:48 +00:00
b7eda43810
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on
...
several SSRCs"
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1774006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 21:08:27 +00:00
6f5707e184
Revert r4328
...
R=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1774005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 20:59:52 +00:00
8543c1c77c
Updated WebRTC version to 3.36
...
TBR=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1780005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4341 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-15 17:19:45 +00:00
df119c9a45
Remove dead video_capture for QuickTime.
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4339 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 18:08:13 +00:00
a9b74ad716
Include files from webrtc/.. paths in video_capture/.
...
BUG=1662
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1788004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4337 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 10:03:52 +00:00
8b06200802
Include files from webrtc/.. paths in utility/.
...
BUG=1662
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1786004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:28:10 +00:00
0ed57c51a3
Remove dead code testAPI.cc.
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BUG=
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4335 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:23:05 +00:00
5aa3f1b4c0
Include files from webrtc/.. paths in video_render/.
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BUG=1662
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1782006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4334 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-12 08:12:08 +00:00
5b10d8fb18
Fix some voe_auto_test uninitialised-value errors.
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BUG=
R=tommi@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4332 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 15:50:07 +00:00
