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63a509858d
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977 Commits

Author SHA1 Message Date
andrew@webrtc.org
63a509858d Rename AudioFrame members. 
BUG=
TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/542005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 23:56:37 +00:00
leozwang@webrtc.org
7fdb909339 Reformat and add more debug info into ViESurfaceRenderer 
BUG=
TEST=test on android
Review URL: https://webrtc-codereview.appspot.com/546004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 16:45:55 +00:00
hta@webrtc.org
404843e6e5 Timeout tests for TMMBR 
Added a test that injects 3 packets and then times out.

Rewrote the packet construction in test to use a builder format.
This makes tests a lot more readable.

Odd behaviour of timeout found; documented as test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/553004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 09:56:45 +00:00
hta@webrtc.org
3c0df7d376 Fixed a build break: I'd forgotten to remove a DELETE statement. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/555004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 08:22:21 +00:00
hta@webrtc.org
47059b5dfb Adds unit tests for RTCP receiver, focusing on TMMBR handling. 
This is the first part of a plan:

- Get basic unit tests for rtcp_receiver.
- Get an unit test for some code inside rtcp_receiver
  that touches the TMMBRSet class in hard-to-decipher ways
  (rtcp_receiver_help, GetTMMBRSet function, use of memmove()).
- Refactor the TMMBRSet class.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/547005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 07:46:22 +00:00
andrew@webrtc.org
ecac9b721e Add tests for downmixing and no processing. 
BUG=issue419
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/532001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-05-02 00:04:10 +00:00
leozwang@webrtc.org
e7ac5fde72 Minor changes to remove dead code in opensl es 
BUG=
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 14:42:17 +00:00
hta@webrtc.org
65a4e4ed56 Minor refactoring: RTCPReceiver::BoundingSet 
Remove ability to modify a pointer argument.

Added a test for transmitting a non-empty TMMBN

BUG=
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/542004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 11:23:41 +00:00
hta@webrtc.org
c2d985257b untabify 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 08:25:10 +00:00
hta@webrtc.org
9d54cd12ab TMMBN sender test passes, fixed receiver flag bug 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-30 08:24:55 +00:00
andrew@webrtc.org
5c0c18d823 Fix coverity issues in ACM. 
Fixes: Big parameter passed by value (PASS_BY_VALUE)
Passing parameter codec of size 52 bytes by value.

BUG=
TEST=audio_coding_module_tests, trybots

Review URL: https://webrtc-codereview.appspot.com/529002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 17:06:48 +00:00
marpan@webrtc.org
2d0223286b VPM: fix to coverity issues 10255-10258 (unintended sign extension). 
Review URL: https://webrtc-codereview.appspot.com/532002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 15:56:02 +00:00
pwestin@webrtc.org
b1313aac7c Fix missing h file change. 
Review URL: https://webrtc-codereview.appspot.com/535001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:42:05 +00:00
pwestin@webrtc.org
49888ce428 Breaking out send side bitrate controll cont. 
Review URL: https://webrtc-codereview.appspot.com/475004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-27 05:25:53 +00:00
mallinath@webrtc.org
e611619f60 Fixing the header file path in gypi file. 
BUG=473
Review URL: https://webrtc-codereview.appspot.com/529001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 23:03:15 +00:00
tommi@webrtc.org
a990e122c4 * Change the reference counting implementation for VoE to be per object and 
not per interface. This simplifies things a bit, reduces code and makes it
  possible to implement reference counting (if we ever want) without the
  static Delete() method.  (Reference counted objects are traditionally
  implicitly deleted via the last Release())

* Since the reference counting code is now simpler, there's no need for the
  RefCount class so I'm removing it.

* Also removing the optional |ignoreRefCounters| variable from the VoiceEngine::Delete
  method.  The justification is that it's no longer used and the reason it was there
  in the first place was to avoid bugs in third party code, so it's an indication that
  something is wrong elsewhere.

* Fix a crash in voe_network_impl and voe_extended_test that would happen on machines with IPv6 support.

* Fix an assert (handle the case) in the extended audio tests when SetCNAME is called with a NULL pointer.

* As a side effect of this change, hopefully the footprint of VoE will be slightly smaller :)

BUG=10445 (Coverity)
Review URL: https://webrtc-codereview.appspot.com/521003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2127 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 15:28:22 +00:00
tina.legrand@webrtc.org
bc1b43b297 Refactoring of audio_coding_module_impl 
First patch set: pure formatting.

Review URL: https://webrtc-codereview.appspot.com/522001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 08:53:45 +00:00
tina.legrand@webrtc.org
a6ecd1ebb5 Refactoring one of the ACM tests: TestStereo, to follow the style guide. 
(First patch: formatting the test file)

TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/507001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 07:54:30 +00:00
mflodman@webrtc.org
1868780c81 Disabled UnremovedSocketsGetCollectedAtManagerDeletion in UdpSocketManager unittest. 
TBR= hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/520004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2122 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-26 06:40:00 +00:00
hta@webrtc.org
ad929899c7 Tests for udp_socket_manager. 
These tests basically check that socket allocation does not leak memory.

BUG=
TEST=unittested, ran under valgrind.

Review URL: https://webrtc-codereview.appspot.com/519003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-25 14:06:50 +00:00
asapersson@webrtc.org
d18dd6dc7e Made OnPacketLossStatisticsUpdate function virtual (needed for ViCE). 
Review URL: https://webrtc-codereview.appspot.com/520002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-25 07:19:02 +00:00
andrew@webrtc.org
369166a179 Add API for disabling the high pass filter. 
BUG=issue419
TEST=manually with voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/509003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-24 18:38:03 +00:00
elham@webrtc.org
5f49dba1a1 Hi Magnus, I added some of the changes that you suggested before. Let me know what you think. 
Review URL: https://webrtc-codereview.appspot.com/507004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 21:24:02 +00:00
andrew@webrtc.org
4e423b3b1e Handle master/slave timestamp wrap. 
BUG=issue410
TEST=neteq_unittests, audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/506001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2098 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 18:59:00 +00:00
vikasmarwaha@webrtc.org
99ac3f7be5 Fixed trunacated trace problem in WebRTC. http://b.corp.google.com/issue?id=5607856 
Review URL: https://webrtc-codereview.appspot.com/508004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 17:04:35 +00:00
pwestin@webrtc.org
ddab60be56 Wire up pading. 
Review URL: https://webrtc-codereview.appspot.com/509002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2094 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 14:52:15 +00:00
hta@webrtc.org
bf9f469a13 Lifetime management for UdpSocketManager 
Make tests use Create/Destroy *or* new/delete for UdpSocketManager.
Move responsibility for calling Destroy on UdpSocketManager from transport
destructor to transport Destroy function.

This all ends up in not leaking memory in InitializeSourcePorts test.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/512001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 13:19:30 +00:00
asapersson@webrtc.org
92591adc67 Fixes link issues in google3 (change by tomasl). 
Review URL: https://webrtc-codereview.appspot.com/509001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 13:10:55 +00:00
asapersson@webrtc.org
83ed0a4359 Try to resend next packet in nack list even if previous packet is not found. 
Review URL: https://webrtc-codereview.appspot.com/515001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 12:43:05 +00:00
pwestin@webrtc.org
fcbbe1f341 Removed unused callback code from video coding test. 
Review URL: https://webrtc-codereview.appspot.com/504003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2086 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 08:33:18 +00:00
pwestin@webrtc.org
a2cd732139 Fix for calling OnNetworkChanged too often. 
Review URL: https://webrtc-codereview.appspot.com/508006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2085 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-23 08:32:47 +00:00
marpan@webrtc.org
88ad06b999 VCM: Added condition(s) for setting FEC protection factor to zero at low bitrates. 
Review URL: https://webrtc-codereview.appspot.com/494001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 16:05:24 +00:00
asapersson@webrtc.org
63a34f4f29 Fix in SendPadData. Do not increase sequence number if packet is "empty" and not sent. 
Review URL: https://webrtc-codereview.appspot.com/508001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 13:20:27 +00:00
phoglund@webrtc.org
bb77000123 Added a virtual destructor to get the test to compile on all platforms. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/507003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 13:03:12 +00:00
hta@webrtc.org
bbd6b561cf Memory leak fix: Deleting a factory 
Also expanded some documentation.
Bug found by Valgrind bot.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/507002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2080 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 09:51:26 +00:00
bjornv@webrtc.org
bcde776340 Changed Delay Estimator create call 
Unit tests updated and runs.
Change made w.r.t. issue 441.

BUG=Issue441
TEST=None

Review URL: https://webrtc-codereview.appspot.com/498001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 09:35:20 +00:00
hta@webrtc.org
0abe535e16 Refactored udp_transport to take socket manager as dependency injection 
This avoids having to deal with the socket manager in the unittest.

Extended tests to cover one case where sockets got allocated.

BUG=
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/496001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 08:23:16 +00:00
andrew@webrtc.org
b61f1fa675 Reset slave when switching to a stereo codec. 
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/494003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-20 01:10:14 +00:00
tommi@webrtc.org
e84373c38f Atomic32Wrapper -> Atomic32 
This change does the following:

* Remove the Atomic32Wrapper and AtomicImpl classes and provide the Atomic32
  implementation directly via platform specific source files.
* Move/rename/delete all source files accordingly
* The atomic value itself is now volatile. Previously it was only the pointer to
  the memory that was volatile, but not the actual value.
* No additional heap allocations are now done for the atomic value.

In a follow up cl I plan to start using Atomic32 in the RefCount class in order
to fix issues reported by Coverity.
Review URL: https://webrtc-codereview.appspot.com/490004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-19 14:28:45 +00:00
pwestin@webrtc.org
1cd1162c7a Break out of send side bandwidth estimation and controll. 
Review URL: https://webrtc-codereview.appspot.com/490010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-19 12:13:52 +00:00
asapersson@webrtc.org
a768970bac Parse out ssrcs in REMB message (needed for ViCE) . 
Review URL: https://webrtc-codereview.appspot.com/486003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-19 07:01:29 +00:00
tina.legrand@webrtc.org
faa0ab85d7 NetEQ stereo sync 
This CL allows NetEQ to do expand at startup, to make master and slave always go in sync. Before it could happen that master did merge, while slave performed an expand, leading to sync-problems between the channels.

Updating DEPS for new reference files for unittest.

BUG=410
TEST=neteq_unittests

Review URL: https://webrtc-codereview.appspot.com/487005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-18 17:59:53 +00:00
leozwang@webrtc.org
16f6bb35b6 Fix a minor compilation error on android 
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/479014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-18 16:49:06 +00:00
bjornv@webrtc.org
4ade5506eb Delay Estimator Unit tests 
This CL includes unit tests to verify correct behavior of the delay estimator used in AEC and AECM.

Tested with audioproc_unittest

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/491009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-18 09:42:16 +00:00
bjornv@webrtc.org
2e729762c7 New _CreateBinaryDelayEstimator() and removed _history_size() 
Changed create function to match malloc() in functionality.
Removed unused function.

Tested with audioproc_unittest

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/491010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-18 08:30:29 +00:00
bjornv@webrtc.org
180f83f8e2 File name change to follow style 
A unit test should end with *_unittest.* Otherwise the test itself will be evaluated for line coverage.

Tested with audioproc_unittest

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/493008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-18 07:01:10 +00:00
andrew@webrtc.org
1bc98bc873 Remove erroneous error trace. 
It appears this trace is informational, not an error.
Discovered in a plugin log from a ChromeOS device. Just
decided to remove it.

BUG=http://code.google.com/p/chromium-os/issues/detail?id=29356
TEST=build on Linux

Review URL: https://webrtc-codereview.appspot.com/479012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-17 20:10:47 +00:00
tommi@webrtc.org
7ab51497a7 Remove usage of Atomic32Wrapper from a few places. 
In these places, it doesn't make much sense to use an atomic variable we were using
Atomic32Wrapper::operator= anyway (which does not use atomic operations).
Review URL: https://webrtc-codereview.appspot.com/492005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-17 13:33:28 +00:00
hta@webrtc.org
52c0fec34c Added UDP socket factory function to UdpTransportImpl constructor 
This is a refactoring in preparation for creating small unit tests for the
udp_transport module.

BUG=
TEST=unittest

Review URL: https://webrtc-codereview.appspot.com/482004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-17 12:39:04 +00:00
marpan@webrtc.org
3e2e7038e6 VPM: Allow for option to compute the content metrics every nth frame. 
Review URL: https://webrtc-codereview.appspot.com/492006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2034 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-04-16 15:58:14 +00:00