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1309 Commits

Author SHA1 Message Date
solenberg@webrtc.org
46db413e22 Fix failing tests on 32 bit Linux. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1534004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:53:42 +00:00
turaj@webrtc.org
e46c8d3875 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. 
TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1384005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 20:39:43 +00:00
solenberg@webrtc.org
561990fd73 - Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp. 
- Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying.
- Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions).
- Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().

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

Review URL: https://webrtc-codereview.appspot.com/1521004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 19:04:19 +00:00
sergeyu@chromium.org
6ec25073e3 Disable WindowCapturer tests on OSX and Linux 
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1533004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 18:47:07 +00:00
sergeyu@chromium.org
6ebfd346ae Add direct_dependent_settings in common.gypi. 
When building chromium targets that depend on webrtc, compiler settings must
have the include path to webrtc and webrtc-specific defines that the headers
may depend on. Added direct_dependent_settings in common.gyp, so that all
webrtc target propagate these settings to dependencies.

R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1371005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-22 18:22:21 +00:00
mikhal@webrtc.org
2eaf98b38b Refactor VCM/Timing. 
No changes in functionality.

R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1514004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 17:58:43 +00:00
stefan@webrtc.org
3417eb49f6 Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted. 
TEST=trybots
BUG=1799
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 15:25:53 +00:00
hclam@chromium.org
0d540c3762 Log timestamp of the frame when it's dropped from the render module 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1515005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4075 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 00:16:01 +00:00
solenberg@webrtc.org
c0352d566a Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. 
BUG=
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1510004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-20 20:55:07 +00:00
sergeyu@chromium.org
b10ccbec02 Window capturer implementation for Windows. 
R=alexeypa@chromium.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1477004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-19 07:02:48 +00:00
fischman@webrtc.org
8d6eb56085 Avoid NPE crash on Android platforms that don't support getting preview framerate. 
- catch Camera.setParameters() signaling errors through RuntimeException (!)
- make video_demo_apk rebuild when .java sources change

BUG=1778
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1493004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 17:33:31 +00:00
stefan@webrtc.org
9f557c140e Improve wraparound handling in the render time extrapolator. 
This was actually working as intended, but as r3970 changed when render timestamps were extrapolated to when a frame was taken out for decoding, the wraparound could have happened in the Update() step before it had happened in the ExtrapolateLocalTime() step. This causes render timestamps to be generated 13 hours into the future.

TEST=trybots
BUG=1787
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1497004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 12:55:07 +00:00
phoglund@webrtc.org
14d7700d00 Moved command line parsing to internal tools and moved back the mic volume thingie. 
BUG=
R=henrika@webrtc.org, kjellander@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1491004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-17 11:52:08 +00:00
turaj@webrtc.org
8630cfe016 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. 
BUG=issue1770
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1485004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 23:54:54 +00:00
hclam@chromium.org
fe307e1332 Add one unit test for NACKing a key frame 
Adding a test case that wasn't covered. This new test is passing.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 21:19:59 +00:00
hclam@chromium.org
b3e5acfb66 Cleanup traces in WebRTC 
Remove some unused traces and add a trace counter for encoded video size.

R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1476004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 21:13:02 +00:00
pbos@webrtc.org
b9bb3d1e7d Avoid resetting encoder on identical settings. 
BUG=1681
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1481005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 18:40:48 +00:00
marpan@webrtc.org
890f6092e6 Bugfix: VCM would report wrong sentBitrate 
issue: https://code.google.com/p/webrtc/issues/detail?id=1755

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1484004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 15:38:44 +00:00
phoglund@webrtc.org
9919ad5caf Formatted FEC stuff. 
Unfortunately I had to pull in quite a bit of stuff due to use of unencapsulated public member variables.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1401004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 15:06:28 +00:00
stefan@webrtc.org
2038214c77 Log too long non-decodable duration events. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1488004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 11:39:06 +00:00
solenberg@webrtc.org
7ebbea14a9 Add handling of the absolute send time header extension to the rtp_rtcp module. 
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1480004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 11:10:31 +00:00
mikhal@webrtc.org
6cfa3907c8 Updating NACK RTX test 
BUG=1513
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1274006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 20:17:43 +00:00
mikhal@webrtc.org
cb20a5b2d7 VCM/JB: Bug fix in ExtractAndSetDecode 
BUG=1771
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1466005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 17:10:44 +00:00
solenberg@webrtc.org
5add4ad09c RemoteBitrateEstimatorTest::TestRateIncreaseReordering sent in arrival timestamps in non monotonically increasing order. Fixed. 
BUG=
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1481004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4034 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 13:49:57 +00:00
braveyao@webrtc.org
c93b1d038d CoreAudio Win: release resources safely under certain rare circumstance in GTalkplugin 
BUG=
TEST=voe_auto_test
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 10:14:56 +00:00
niklas.enbom@webrtc.org
e2a800644c Linux support for typing detection 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 21:33:11 +00:00
turaj@webrtc.org
4ce838934c Address sanitizer out of bounds read in iSAC 
BUG=issue1770
TBR=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/1472006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 17:42:22 +00:00
andresp@webrtc.org
29b2219914 Adding a factory to remote bitrate estimator and allow it to be set via config. 
Additionally:
 - clean api to set remote bitrate estimator mode.
 - clean api to set over use detector options.

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

Review URL: https://webrtc-codereview.appspot.com/1448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 12:10:58 +00:00
stefan@webrtc.org
1673481ed7 Fixes a bug where the render buffer size (and indirectly the non-continuous duration) was computed incorrectly. 
BUG=1769
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1473004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 12:00:47 +00:00
mflodman@webrtc.org
bb984f516e Removed Mac capture crash and memory leak. 
BUG=1697,1761
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1465005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 10:47:19 +00:00
phoglund@webrtc.org
527f6c62fc Reformatted FEC tables. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 09:25:01 +00:00
andresp@webrtc.org
185bae4b6f Replace ExtraCodecOptions with new Config class that supports multiple settings at once. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1452004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4017 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 08:02:25 +00:00
justinlin@chromium.org
7bfb3a3227 Add more tracing for key frames. 
R=mallinath@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 22:59:00 +00:00
phoglund@webrtc.org
43bf6ce322 Revert 4008 "Avoid resetting video encoder for similar configs." 
> Avoid resetting video encoder for similar configs.
> 
> BUG=1681
> R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1442006

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1431005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4010 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 15:39:26 +00:00
pbos@webrtc.org
aa4efd1535 Avoid resetting video encoder for similar configs. 
BUG=1681
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1442006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 11:27:16 +00:00
fbarchard@google.com
1e3c794688 Use 2 threads for HD, or 1 for VGA or less. 
BUG=1739
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1438005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-09 18:43:38 +00:00
phoglund@webrtc.org
315d39866e Formatted dtmf_queue. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 10:04:06 +00:00
stefan@webrtc.org
d98e784f5f Fix VCMProcessTimer::TimeUntilProcess() unsigned-integer underflow problem. 
BUG=1665
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1341004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 06:38:53 +00:00
niklas.enbom@webrtc.org
3be565b502 Refactoring for typing detection 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 21:04:24 +00:00
stefan@webrtc.org
ef14488d03 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. 
BUG=1663
R=mikhal@webrtc.org, ronghuawu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 19:16:33 +00:00
mikhal@webrtc.org
8f86cc8712 VCM/Receiver: Return null when can't extract frame. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1435004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 18:05:21 +00:00
mikhal@webrtc.org
474e915272 Relanding 3962: VCM/JB: Porting jitter_buffer_test to gtest 
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:55:03 +00:00
mikhal@webrtc.org
759b041019 Relanding r3952: VCM: Updating receiver logic 
BUG=r1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1433004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:36:00 +00:00
mikhal@webrtc.org
9c7685f9a6 VCM/JB: Break and skip to key if possible 
BUG=1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1421004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:07:52 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
stefan@webrtc.org
d3a1959678 Fix jitter buffer unittest. 
TBR=mflodman@webrtc.org
BUG=1737

Review URL: https://webrtc-codereview.appspot.com/1430005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:35:58 +00:00
stefan@webrtc.org
a5dee33639 Correctly add packets to nack list when sequence number wraps. 
BUG=1737
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1427004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 11:11:17 +00:00
pwestin@webrtc.org
0f29810288 Fix crash in pacer. 
BUG=1731
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1410006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 16:37:22 +00:00
stefan@webrtc.org
4ce19b1664 Revert r3952 "VCM: Updating receiver logic" 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1410005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 13:16:51 +00:00
stefan@webrtc.org
273759048c Revert r3956 "VCM/JB: Porting jitter_buffer_test to gtest." 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1408005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 13:12:58 +00:00