3145a642b7
Correctly rebuild WebRTCDemo-debug.apk after modules/ source file changes.
...
BUG=1980
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26 20:20:05 +00:00
e6168f5f41
Adding a first simple version of overuse detection, but not hooked up.
...
BUG=
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1717004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4268 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26 11:23:01 +00:00
1c986e7c89
Removed ViE file API.
...
R=asapersson@webrtc.org , niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1723004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4267 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26 09:12:49 +00:00
a5fd2f1348
Do basic parsing of RTCP headers in PcapFileReader to enable log filtering.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1697004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-26 08:36:07 +00:00
91811e2b04
Remove unused multi stream bandwidth estimator.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1712004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-25 20:36:14 +00:00
a4c5abb52a
Make sure padding packets are sent.
...
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1717006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-25 15:46:16 +00:00
3348ae2b97
mac: Mark kCGLPFAFullScreen as allowed in a 10.6-only function.
...
kCGLPFAFullScreen is marked deprecated starting with 10.6 in the 10.9 SDK,
but it's functional on 10.6 and this code only runs on 10.6 and will go away
when support for 10.6 is dropped.
BUG=webrtc:1958
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1710004
Patch from Nico Weber <thakis@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4255 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-21 23:33:10 +00:00
6eb53f71d6
Fix memory bot failure
...
Exit the method with critical setting held. This should make
the memory bot happy.
TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1704005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4251 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 23:01:39 +00:00
2e402ce873
Enqueue packet in pacer if sending fails
...
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.
BUG=1930
R=pwestin@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1693004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 20:18:31 +00:00
9ca7360b97
VCM: removing max jitter estimate
...
BUG= 1921
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1690004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4249 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-20 20:13:07 +00:00
0851df8d60
Remove unneeded *_NOT_SUPPORTED from VoEAudioProcessing.
...
* Remove ANDROID_NOT_SUPPORTED from a bunch of echo metrics calls
where it actually is supported.
* No error to call GetTypingDetectionStatus.
* Consolidate typing detection disablement to reduce boilerplate.
R=niklas.enbom@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1683004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4247 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19 17:03:47 +00:00
8ccb9f9716
Fixes some pacer/padding issues found while testing.
...
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.
BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1682004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-19 14:13:42 +00:00
d7148c86c5
Use 3 threads for higher than 720p resolutions
...
BUG=1893
TEST=untested
R=ajm@google.com , andrew@webrtc.org , dingkai@google.com , marpan@google.com , marpan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1684004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4243 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18 22:06:42 +00:00
30fb7b83d5
Add a log message to see video delay break down
...
Shows video delay in terms of:
1. Min playout delay
2. Jitter delay
3. Max decode time
4. Render delay
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1674004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4242 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-18 21:37:09 +00:00
a20eb91154
Make ScreenCapturerMac work in versions of OSX before Lion.
...
The screen capturer was broken when moving code to webrtc: width
and height parameters for glReadPixels were swapped by mistkake.
BUG=crbug.com/244102
R=alexeypa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1678005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4239 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 22:22:40 +00:00
9e182795a9
Enable ScreenCapturer unittests
...
previously ScreenCapturer unittests were disabled by mistake
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4238 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 21:14:36 +00:00
a590b41c9a
Use intptr_t to represent window IDs on all platforms.
...
Previously void* was used on windows which makes it harder to work
with the IDs in cross-platform code.
R=alexeypa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1672004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4237 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 20:02:21 +00:00
508a84b255
Wire up pacer-based padding.
...
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.
Padding will for now only be generated by the first sending RTP module.
BUG=1837
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1612005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 12:53:37 +00:00
50fb4afade
Revert r4145 "Revert 4127 "Switch frame list implementation to std::map.""
...
TBR=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1678004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4233 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 07:33:58 +00:00
c8b29a2feb
Revert r4146 "Revert 4104 "Refactor jitter buffer to use separate lists for de...""
...
TBR=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1677004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4232 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-17 07:13:16 +00:00
7262ad1385
Fix AV sync issue
...
r4229 introduced an AV sync issue due to an error.
This is a one linear fix and provides the correct
current video delay for synchronization.
TBR=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1675004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4231 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-15 06:51:27 +00:00
9b23ecb939
Log current and target AV delay in ViESyncModule
...
R=mikhal@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1668006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4229 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 23:30:58 +00:00
63e988856e
Merge more tests into modules_{unit,integration}tests.
...
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests
A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests
I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.
Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests
Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).
Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).
BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1656004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 20:09:44 +00:00
f27389ca9f
WebRTCDemo: ensures that using front and back camera work as expected.
...
I.e. egress: Real world up is stream up.
Ingress: stream up is app up.
Local (preview): Real world up is app up.
BUG=1763
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1642004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-14 05:37:13 +00:00
d4ed1a3e2c
Fixes linker issue with no op trace.
...
BUG=N/A
R=henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4226 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-13 09:34:54 +00:00
fee739c224
Risk of division by zero.
...
bug=b9338699
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1634004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 20:10:06 +00:00
dd97ef4e28
Revert 4211 "Build all java files into jar for each module on An..."
...
Reason for revert: behold the meltdown of the "trunk" bots on http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
Turns out that include in gyp is fraught with peril: https://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
> Build all java files into jar for each module on Android
>
> BUG=
> R=fischman@webrtc.org , niklas.enbom@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1636004
>
> Patch from Jeremy Mao <yujie.mao@intel.com >.
TBR=fischman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/1660005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4222 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 17:39:29 +00:00
20a993f88a
Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test.
...
Take two of http://review.webrtc.org/1657004/
This time with execution on trybots.
BUG=1925
TEST=win,win_rel,mac,mac_rel,linux,linux_rel trybots passing.
R=mflodman
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1658004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4221 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 14:38:01 +00:00
935d705370
Disable ViEExtendedIntegrationTest.RunsCodecTestWithoutErrors test.
...
Disable on Windows due to failures on bots.
BUG=1925
TEST=compile on Linux and Windows.
R=mflodman
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/1657004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 13:59:57 +00:00
04996cd5e5
Fix breakage due to test_fec conversion to gtest.
...
In my attempt to commit a subset of http://review.webrtc.org/1647005/
instead of all of it, I forgot to add the gtest dependency to the
test_fec.gypi. This CL fixes that.
TEST=local compile + win_rel,mac_rel,linux_rel trybots
BUG=1916
R=marpan
TBR=marpan
Review URL: https://webrtc-codereview.appspot.com/1655004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4219 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 12:15:33 +00:00
22bbbdfa68
Convert test_fec to gtest
...
All tests needs to be gtest tests in order to be executed
with the upcoming isolate/swarm framework.
TEST=trybots passing
BUG=1916
R=andrew@webrtc.org , marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/1647005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4218 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 11:55:05 +00:00
7124dd8561
Disable ViEStandardIntegrationTest.RunsRtpRtcpTestWithoutErrors test.
...
BUG=1790
TEST=Just local compilation.
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1654004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4217 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 08:28:09 +00:00
b097670264
G722_1/G722_1C codecs won't instantiate
...
BUG=issue1890
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1650004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4215 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 07:41:42 +00:00
6c35e0b0f7
Reorganize test targets in WebRTC
...
This CL will lower the number of test targets in WebRTC by:
Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006 ):
* resampler_unittests
* signal_processing_unittests
* vad_unittests
Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests
Merge into test_support_unittests:
* channel_transport_unittests
channel_transport.gyp was also removed in favor for test.gyp.
I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.
Buildbot configuration update will be synced with the commit of this CL.
TEST=trybots
BUG=1843
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-11 08:29:17 +00:00
1374965680
Build all java files into jar for each module on Android
...
BUG=
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1636004
Patch from Jeremy Mao <yujie.mao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4211 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 23:34:27 +00:00
4af0878e57
Allow the screen capturer to capture oversized cursors and cursors without alpha channel (Windows).
...
Changes in this CL:
- CaptureCursor() scans the cursor to verify that it has alpha channel.
- The AND mask of the cursor is used to reconstruct transparency if the cursor does not have alpha channel.
- CaptureCursor() always outlines the cursor when a "screen reverse" pixel detected. Previously it was only done for black and while cursors.
Added desktop_capture_unittest.MouseCursorShapeTest to test the cursor conversion code.
BUG=chromium:223147
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1627004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4210 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 22:29:17 +00:00
5e03f8ab67
Landing binary cursor image files to be used in a follow up CL.
...
See https://webrtc-codereview.appspot.com/1627004/ for more details. TBR since that CL has been reviewed and LGTMed.
TBR=sergeyu@chromium.org
BUG=chromium:223147
Review URL: https://webrtc-codereview.appspot.com/1647004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4209 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 21:07:31 +00:00
5137b9752f
Updated WebRTC version to 3.33
...
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1645004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4204 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 17:03:51 +00:00
509754c4c9
Making no NACK mode work again in VideoEngine.
...
BUG=1910
TEST=ViE autotest loopback with no protection and some percent packet loss
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1631004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4203 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 15:50:12 +00:00
1819fd711a
RW lock access to ssrc maps in VideoCall.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1640004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4202 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 13:48:26 +00:00
adb51f5709
Add back the WEBRTC_DIRECT_TRACE flag.
...
BUG=
R=andresp@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1596004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4201 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 09:03:41 +00:00
83a062cc5f
AudioDeviceAndroidOpenSLES: NULL variables might be referenced in StopPlayout()
...
BUG=1891
Test=ManualTest
R=fischman@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1622004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4200 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-10 08:09:05 +00:00
569fdef732
Revert some variables to uint32_t to fix compile errors on Mac gcc.
...
TBR=xians
Review URL: https://webrtc-codereview.appspot.com/1633004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4199 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-08 00:43:25 +00:00
6f69eb78dd
Allow audio devices with up to 64 channels on Mac.
...
Does not increase memory requirements. Adds an additional check to ensure
configurations requiring more memory per IO block than the input ring buffer
contains are rejected.
BUG=1904
TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality.
Selecting a higher sample rate (96 kHz), which would otherwise give choppy
audio, instead results in an error.
R=henrika@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1628004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4198 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 17:56:50 +00:00
1064cf06b0
Fixed Rtp/Rtcp tests
...
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1627005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4196 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 16:03:19 +00:00
6367fe885a
Fix relative path to .gitignore and other minor changes.
...
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1624005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4195 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 15:43:04 +00:00
3ba883f0fc
Removing functionality for inserting pre-encoded frames instead of raw
...
video frames. The functionality hasn't been used for a long time and
should be done properly if used in the future.
This is a pre-step for implementing CPU overload control.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1630004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4194 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 13:57:57 +00:00
b69cc15467
Add script for appending entries to .gitignore.
...
TBR=kjellander
Review URL: https://webrtc-codereview.appspot.com/1629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4193 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 02:04:12 +00:00
da710448b2
Fix size_t to int conversion error on Win64.
...
TBR=pwestin
Review URL: https://webrtc-codereview.appspot.com/1626005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4192 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-07 01:43:12 +00:00
7e4ff354e3
Remove fake screen capturer because it's not used anywhere.
...
R=alexeypa@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1625004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4191 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-06 23:11:33 +00:00
