Commit Graph

1117 Commits

Author SHA1 Message Date
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
mflodman@webrtc.org
b1fbf016b5 Added timestamp logs, i.e. only tracing.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/482001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2030 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-16 12:58:49 +00:00
mflodman@webrtc.org
3c611fd4fd Removed NetEQ Test compile error.
BUG=443
TEST=Compiles using clang version 3.1 (trunk 153589)

Review URL: https://webrtc-codereview.appspot.com/493005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2029 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-16 10:57:27 +00:00
xians@webrtc.org
aef0a61117 webrtc: OTHER_CPLUSPLUSFLAGS should be a list, not a string.
Review URL: https://webrtc-codereview.appspot.com/492007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2028 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-16 09:57:16 +00:00
bjornv@webrtc.org
bfda85f2ee Safe handling of invalid inputs in delay_estimator.
The delay_estimator crash on invalid create inputs when running new unit tests. This can't occur on a higher level unless corresponding enum and defines are incorrectly changed.

The create and free functions are now more like malloc() and free() in design. The complete change to that will be done in a seperate CL.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/492003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2027 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-16 07:28:29 +00:00
mflodman@webrtc.org
a014ecc049 Changed CriticalSectionScoped constructor usage for ADM.
Only search and replace for the input to the constructor, no other changes.

BUG=184
TEST=Compiles on all platforms.

Review URL: https://webrtc-codereview.appspot.com/483001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2015 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-12 12:15:51 +00:00
bjornv@webrtc.org
7056908774 System delay unit tests
Added a system delay test class. Noticed I don't need the ApmTest class at all, which simplified the implementation.

Start at patch set 3. The others are not complete.

BUG=None
TEST=

Review URL: https://webrtc-codereview.appspot.com/475003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2014 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-12 12:13:50 +00:00
tina.legrand@webrtc.org
16b6b90a82 Split of stereo packets moved
In this CL I have rewritten the way we handle stereo packets in VoE.
Before this change we split the packets in the RTP module and added two packets to ACM, one for the left channel and one for the right. This lead to timing problems caused when a different thread called RecOut in between the two calls to add stereo packet to ACM. (RecOut is called to pull audio data, decode packets, on the receiving side).

While doing the change I also took the opportunity to changed some functions so that the data stream is uint8 everywhere.

The list of files in this CL is long, but should be fairly easy to review. It is difficult to see what has been changed  in some of the tests, but I can explain offline.

Reviewers:
Björn - /src/modules/audio_coding
Patrik - /src/modules/rtp_rtcp
Patrik -/src/modules/utility
Henrik A - /src/voice_engine

BUG=410
TEST=voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/473003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2012 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-12 11:02:38 +00:00
pwestin@webrtc.org
ce33035dee Cleanup encode call.
Review URL: https://webrtc-codereview.appspot.com/491003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2011 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-12 06:59:14 +00:00
stefan@webrtc.org
c35f5ced92 Enable multi-frame FEC by default for temporal layers <= 2. For two temporal layers we currently only protect the base layer.
This also introduces zero column insertion into packet masks when some sequence numbers deliberately haven't been given to the FEC generator.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/477001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2005 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-11 07:42:25 +00:00
cd@webrtc.org
85b4a1b715 Special version of 'bitrv2' when len is 128:
* 5.5% AEC overall speedup for the straight C path.
* 8.6% AEC overall speedup for the SSE2 path.
Review URL: https://webrtc-codereview.appspot.com/480001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2004 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-10 21:25:17 +00:00
leozwang@webrtc.org
91ed80e5c3 Correct wrong trace level
Review URL: https://webrtc-codereview.appspot.com/487002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2002 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-10 14:42:20 +00:00
tommi@webrtc.org
98ad0ff1b0 Move the COMPILE_ASSERT macro to its own header file.
TEST=n/a
BUG=none
Review URL: https://webrtc-codereview.appspot.com/492002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2001 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-10 11:53:07 +00:00
andrew@webrtc.org
e713fd0eee Enable the "unused variable" warning on Windows.
- Break out direct_show_base_classes to its own gyp file to have it
  treated as third party code.
- Fix the resulting warnings (courtesy of Tommi).

BUG=
TEST=build on Windows (vie_auto_test currently failing at HEAD)

Review URL: https://webrtc-codereview.appspot.com/489001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2000 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-10 07:13:46 +00:00
pwestin@webrtc.org
cac787842c New attempt to cleanup TMMBR.
Review URL: https://webrtc-codereview.appspot.com/472007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1990 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-05 08:30:10 +00:00
cd@webrtc.org
70ed0a6f91 No casting and call lower precision trigonometric functions:
* 2.2% AEC overall speedup for the straight C path.
* 2.4% AEC overall speedup for the SSE2 path.
Review URL: https://webrtc-codereview.appspot.com/477002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1989 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-05 00:38:55 +00:00
leozwang@webrtc.org
f3dc22f7d1 Reformat android related code
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/472004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1988 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-04 17:15:42 +00:00
pwestin@webrtc.org
ae19720982 Fixed assert strings where string literals are used as booleans.
Review URL: https://webrtc-codereview.appspot.com/473002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1986 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-04 14:43:54 +00:00
leozwang@webrtc.org
0dc8efe6e6 Fix wrong data type in ReadWavHeader
BUG=409
TEST=media file unit test
Review URL: https://webrtc-codereview.appspot.com/474001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1980 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-04-03 15:11:01 +00:00
henrike@webrtc.org
0ad51862dc Revert 1961 - Clean up TMMBR handling.
Review URL: https://webrtc-codereview.appspot.com/465001

TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/473001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1967 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-30 16:54:13 +00:00
marpan@webrtc.org
3a6080d4c0 FEC Decoding: Insert the received media packets directly into VCM without going
through the recovered packet list. Avoids an issue with very old re-transmitted packets.

Updated the receiver_fec unittest accordingly.
Review URL: https://webrtc-codereview.appspot.com/465003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1966 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-30 16:16:21 +00:00
pwestin@webrtc.org
20f4440c73 Clean up TMMBR handling.
Review URL: https://webrtc-codereview.appspot.com/465001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1961 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-30 11:40:15 +00:00
xians@webrtc.org
010a4e8f0b Fix the converity warnings:
CID 10177: Array compared against 0 (NO_EFFECT)
Comparing an array to null is not useful: "this->_paServerVersion".
Review URL: https://webrtc-codereview.appspot.com/466001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1956 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-29 14:05:04 +00:00
andrew@webrtc.org
952f601405 Fix Linux-release errors and Valgrind errors.
BUG=
TEST=build on Linux release.
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/456008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1949 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-27 18:30:26 +00:00
andrew@webrtc.org
61b1b4b472 Fix neteq-rtpplay Linux build errors.
BUG=
TEST=build on Linux.
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/457007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1948 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-27 17:33:29 +00:00
andrew@webrtc.org
f589dfede4 Merge header-only neteq-rtpplay changes.
TEST=build

Review URL: https://webrtc-codereview.appspot.com/452003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1947 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-27 17:05:44 +00:00
mflodman@webrtc.org
534a435751 Removed RTP Keepalive from RTP module.
Review URL: https://webrtc-codereview.appspot.com/455007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1942 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-27 06:57:41 +00:00
stefan@webrtc.org
38f247d800 Fixes an issue in the FEC decoder where a big jump in sequence numbers may cause new packets to be discarded if there is a wrap around.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/455003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1934 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-23 22:34:14 +00:00
stefan@webrtc.org
af5ffd5bb9 Fixes for coverity warnings.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/461001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1933 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-23 16:01:15 +00:00
tina.legrand@webrtc.org
196e491c46 Adding init decode slave to celt
Minor rearanging of calls. Added a init function to the slave channel of CELT.

Review URL: https://webrtc-codereview.appspot.com/458005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1931 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-23 14:39:46 +00:00
tina.legrand@webrtc.org
c231e4cb03 Fixing bug in re-packing of stereo packets.
BUG=341
TEST=voe_cmd_test, run G.722. First modify /src/modules/audio_coding_main/source/acm_codec_database.cc
@@ -149,7 +149,7 @@ const CodecInst ACMCodecDB::database_[] = {
   {kDynamicPayloadtypes[count_database++], "CELT", 32000, 320, 2, 64000},
 #endif
 #ifdef WEBRTC_CODEC_G722
-  {9, "G722", 16000, 320, 1, 64000},
+  {9, "G722", 16000, 320, 2, 64000},
 #endif
 #ifdef WEBRTC_CODEC_G722_1

Review URL: https://webrtc-codereview.appspot.com/454001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1930 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-23 10:01:11 +00:00
pwestin@webrtc.org
1f569222b2 Clean up coverity warnings.
Review URL: https://webrtc-codereview.appspot.com/456003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1928 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-23 09:04:16 +00:00
phoglund@webrtc.org
4aa57b4150 Extracted a helper library from vie_auto_test.
This CL does not attempt to fix the style issues in the moved tb_ files, at least not yet. In general I've tried to avoid dependencies between the library and vie_auto_test: vie_auto_test depends on the library but not the other way around. I had to make some slight changes to achieve this. I had to remove some ViETest::Log statements in tb_interfaces.cc and I had to move knowledge of where to put output files to the library. I think it ended up being pretty clean in the end but let me know if I missed something. I tried to convert all paths I touched to src-relative paths, so look out if I missed something there.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/450004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1923 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-22 12:56:54 +00:00
stefan@webrtc.org
c8e4886774 Upgrade libvpx to 6b66c01 and enabling temporal denoising.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1921 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-21 16:52:03 +00:00
marpan@webrtc.org
6503ecdc39 Fix to windows test from commit 1914.
Review URL: https://webrtc-codereview.appspot.com/455002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1917 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-21 00:18:13 +00:00
marpan@webrtc.org
3fe3252cb3 Fix to windows build from commit 1914.
Review URL: https://webrtc-codereview.appspot.com/456002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1916 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-20 22:13:24 +00:00
stefan@webrtc.org
e0d6fa4c66 Adding classes for handling multi-frame FEC.
The FEC behavior is unchanged with this commit, we will still be
limited to FEC over one frame for now.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/450006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1915 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-20 22:10:56 +00:00
marpan@webrtc.org
e22d81ce4d Updates to resolution adpatation:
-moved calculation of selected frame size & frame rate to qm_select class.

-various updates to qm_select class (switch to 1/2 from 2 stages of 3/4, 
include native frame rate for going up temporal, favor spatial action for temporal layers,..).

-updates to unittest.
Review URL: https://webrtc-codereview.appspot.com/450008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1914 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-20 18:21:53 +00:00
leozwang@webrtc.org
ac9fd8af09 Change folder name from Android to android
Review URL: https://webrtc-codereview.appspot.com/447012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1912 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-19 21:09:42 +00:00
stefan@webrtc.org
b9c50b68bf Revert commit 1908.
Review URL: https://webrtc-codereview.appspot.com/452009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1909 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-17 00:56:36 +00:00
stefan@webrtc.org
fb5944edf9 Upgrade libvpx to 6b66c01 and enabling temporal denoising.
BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1908 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-17 00:15:13 +00:00
leozwang@webrtc.org
a3736345dd Introduced WEBRTC_ANDROID_PLATFORM_BUILD and make test app build on all platforms
BUG=
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/446012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1907 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-16 21:36:00 +00:00
leozwang@webrtc.org
9a85d8e3dd Remove test apps from Android.mk in APM
BUG=
TEST=build on android and pc platforms
Review URL: https://webrtc-codereview.appspot.com/448005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1905 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-16 18:03:18 +00:00
andrew@webrtc.org
61bf8e33c4 Flush far-end buffers when larger than system delay.
Add a helper function to manage far-end buffer moves.

BUG=issue362
TEST=manually with audioproc

Review URL: https://webrtc-codereview.appspot.com/447007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1899 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-15 19:04:55 +00:00
leozwang@webrtc.org
3053702698 Remove -lasound and -lpulse linking flags
BUG=365
TEST=build on linux
Review URL: https://webrtc-codereview.appspot.com/446007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1898 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-15 18:34:13 +00:00
tina.legrand@webrtc.org
0e0390dc33 Flush NetEQ when receiving payload type switches between mono and stereo.
TEST=voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/448004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1893 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-15 11:23:51 +00:00
andrew@webrtc.org
62283c0ebf Quick fix to avoid non-causal AEC signals on PulseAudio.
BUG=340
TEST=manually with voe_cmd_test

Review URL: https://webrtc-codereview.appspot.com/451007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1884 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-13 19:43:47 +00:00
leozwang@webrtc.org
3a39230fdf Further cleanup WebRtc_Word8 in external video capture
Review URL: https://webrtc-codereview.appspot.com/450003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1881 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-12 21:03:38 +00:00
tina.legrand@webrtc.org
ae1c4547ee Reregister of stereo receiver didn't work.
This CL takes care of the re-registration of codecs, and tests unregistering stereo codecs.

One bug fixed in Celt too.

TEST=audio_coding_module_test: TestStereo.

Review URL: https://webrtc-codereview.appspot.com/436002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1871 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-12 08:41:30 +00:00
leozwang@webrtc.org
f5516240ad Prepare future change of WebRtc_Word8 in udp module
Review URL: https://webrtc-codereview.appspot.com/439007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1870 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-09 22:07:18 +00:00
kjellander@webrtc.org
2050f84b98 audio_device_test_api failing cleaner failure for Linux without audio devices.
BUG=None
TEST=audio_device_test_api on Linux.

Review URL: https://webrtc-codereview.appspot.com/447002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1869 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-09 14:22:27 +00:00
tina.legrand@webrtc.org
0dab9e1523 Revert of r1859
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1866 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-09 10:03:09 +00:00
henrika@webrtc.org
907bc55c19 Removes WebRtc_Word8 dependecy in the AudioDeviceModule.
This CL also modifies the ADM callback interface and introduces void* instead of WebRtc_Word8*
as pointer types for data buffers. This change also affects the VoiceEngine.
Review URL: https://webrtc-codereview.appspot.com/443001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1863 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-09 08:59:19 +00:00
kjellander@webrtc.org
67fdd70e1b Refactoring audio_device_test_api for gtest and execution on all platforms.
All the code that was previously in one single function is now broken up into 44 gtest tests. The creation of the Audio Device is done once (SetUpTestCase) and the audio device is initialized before each test (SetUp) and terminated after each test (TearDown). Doing this, the things that execute are basically the same since the test was structured as different sections separated by these calls:
TEST(audioDevice->Terminate() == 0);
TEST(audioDevice->Init() == 0);

I also cleaned up some unused helper functions and removed old test macros when replacing them by gtest macros.

The parts that are failing for Mac and Windows are excluded using #ifdef. Separate issues are filed for
this code to be fixed.

Formatting is updated to follow the WebRTC style guide.

BUG=None.
TEST=audio_device_test_api in Debug+Release on Linux, Mac and Windows. Test run audio_device_test_func on Linux.

Review URL: https://webrtc-codereview.appspot.com/437002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1861 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-09 08:11:04 +00:00
tina.legrand@webrtc.org
f1befad273 Reregister of stereo receiver didn't work.
This CL takes care of the re-registration of codecs, and tests unregistering stereo codecs.

One bug fixed in Celt too.

TEST=audio_coding_module_test: TestStereo.

Review URL: https://webrtc-codereview.appspot.com/436002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1859 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-08 13:23:18 +00:00
mflodman@webrtc.org
7845d07bf8 VideoCapture now uses pointer constructor of CriticalSectionScoped.
BUG=184
TEST=video_capture_module compiles on all platforms when removing ref ctor of CriticalSectionScoped.

Review URL: https://webrtc-codereview.appspot.com/434001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1855 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-08 08:09:17 +00:00
marpan@webrtc.org
accf607b3e Updates for resolution adaptation.
1) added support for two additional modes: 
    -3/4 spatial down-sampling
    -2/3 frame rate reduction
2) updated unittest and added a few more tests
3) some code refactoring
Review URL: https://webrtc-codereview.appspot.com/429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1854 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-07 17:16:10 +00:00
leozwang@webrtc.org
57da718734 Fix building errors on android
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/441001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1850 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-06 21:28:48 +00:00
leozwang@webrtc.org
77fe431f57 Enable video render test on android
Review URL: https://webrtc-codereview.appspot.com/428006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1849 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-06 20:59:55 +00:00
leozwang@webrtc.org
0975d2158c Cleanup messy data type of unknown_payload_type
BUG=322
TEST=build on all platforms
Review URL: https://webrtc-codereview.appspot.com/430002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1848 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-06 20:59:13 +00:00
andrew@webrtc.org
6f9f817e06 Add an API to offset system delay.
Plumb it through VoiceEngine.

BUG=
TEST=voe_auto_test,audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/428010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1846 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-06 19:03:39 +00:00
mflodman@webrtc.org
0e703f4d0d VideoRender now uses pointer constructor of CriticalSectionScoped.
BUG=184
TEST=video_render_module compiles on all platforms when removing ref ctor of
CriticalSectionScoped.

Review URL: https://webrtc-codereview.appspot.com/427004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1843 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-06 12:02:20 +00:00
leozwang@webrtc.org
db2de5b49f Fix building errors on android
TBR=Tina

BUG=
TEST=build on android
Review URL: https://webrtc-codereview.appspot.com/430001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1840 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-05 19:53:24 +00:00
leozwang@webrtc.org
66487e1629 Enable video test on android
Review URL: https://webrtc-codereview.appspot.com/429006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1839 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-05 19:34:06 +00:00
mflodman@webrtc.org
9ec883e8bd Allow multiple REMB groups and introduce receive channels.
BUG=312
TEST=ViE standard autotest and API test.

Review URL: https://webrtc-codereview.appspot.com/432005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1836 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-05 17:12:41 +00:00
leozwang@webrtc.org
855ced7336 Further cleanup WebRtc_Word8
Review URL: https://webrtc-codereview.appspot.com/426008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1835 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-05 16:07:51 +00:00
mflodman@webrtc.org
fa6bc673b0 Changed default module condition for BW estimate.
Review URL: https://webrtc-codereview.appspot.com/433001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1832 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-05 11:59:55 +00:00
leozwang@webrtc.org
42e362eee5 Fix compilation error on android
Review URL: https://webrtc-codereview.appspot.com/426006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1830 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-02 17:14:09 +00:00
leozwang@webrtc.org
3197d48407 Enable audio device test on android
Review URL: https://webrtc-codereview.appspot.com/428005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1829 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-02 17:12:14 +00:00
marpan@webrtc.org
26762e3e40 Allow for spatial-downsampling without reinitializaing encoder. Change of frame
size will automatically trigger new key frame in codec. This feature is set off
in video engine until we upgrade to a newer version of libvpx.
Review URL: https://webrtc-codereview.appspot.com/427003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1827 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-02 16:48:36 +00:00
leozwang@webrtc.org
fa8c9f7a4f Remove unused variable
Review URL: https://webrtc-codereview.appspot.com/432003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1823 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-02 07:15:03 +00:00
leozwang@webrtc.org
c9a3b81fd2 Further cleanup WebRtc_Word8 in video_capture on mac
BUG=311
TBR=Wu, Mallinath
Review URL: https://webrtc-codereview.appspot.com/431002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1819 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 19:59:52 +00:00
leozwang@webrtc.org
4add6bc603 Fix building errors on window which caused by previous cl
BUG=311
TBR=Wu, Mallinath
Review URL: https://webrtc-codereview.appspot.com/432002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1818 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 19:57:13 +00:00
leozwang@webrtc.org
09e771998c Correct WebRtc_word8 usage in media file module
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/427002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1817 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 18:35:54 +00:00
leozwang@webrtc.org
28f3913ca9 Correct WebRtc_Word8 in adm
Correct WebRtc_Word8 usage in adm

BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=buidl on all platforms

Review URL: https://webrtc-codereview.appspot.com/428001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1814 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 18:01:48 +00:00
leozwang@webrtc.org
0689271d64 nits
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1812 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 16:31:00 +00:00
leozwang@webrtc.org
1745e932cc Correct wrong usage of WebRtc_Word8 in video capture
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1811 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 16:30:40 +00:00
tina.legrand@webrtc.org
1f2cabaecd Crash when deleting Celt.
BUG=issue 6087770
TEST=audio_coding_module_test

Review URL: https://webrtc-codereview.appspot.com/420001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1805 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-03-01 08:46:09 +00:00
kjellander@webrtc.org
132eccbb69 Renamed platform specific code to use GYP conventions.
Restructured GYP files a bit to clean up things.
Removed copying of images to /tmp
Fixed output location of DumpFileName.rtp.

BUG=None
TEST=Tested compiling and running on Mac, Win, Linux.

Review URL: https://webrtc-codereview.appspot.com/406002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1802 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-29 20:55:25 +00:00
leozwang@webrtc.org
07c68b9c9d Correct wrong usage of WebRtc_Word8 in rtp and udp module
BUG=http://code.google.com/p/webrtc/issues/detail?id=311&sort=-id
TEST=build on all platforms

Review URL: https://webrtc-codereview.appspot.com/418001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1798 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-29 16:09:51 +00:00
marpan@webrtc.org
4788bf4256 Fix to warnings on windows.
Review URL: https://webrtc-codereview.appspot.com/415004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1792 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-29 01:11:39 +00:00
marpan@webrtc.org
9d76b4ea54 Updates for resolution adaptation:
1) code cleanup and some updates to selection logic for qm_select.
2) added unit test for the QmResolution class.
3) update codec frame size and reset/update frame rate in media-opt:
4) removed unused motion vector metrics and some related code of content metrics processing.
Review URL: https://webrtc-codereview.appspot.com/405008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1791 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-28 23:39:31 +00:00
andrew@webrtc.org
547c157a49 Temporarily use _Word8 to avoid clang error.
BUG=issue311
TEST=build on clang

Review URL: https://webrtc-codereview.appspot.com/415003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1788 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-28 22:30:30 +00:00
leozwang@webrtc.org
91b359ea9b Change WebRtc_Word8 to char
Review URL: https://webrtc-codereview.appspot.com/407003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1787 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-28 17:26:14 +00:00
stefan@webrtc.org
4ce0ba00de Fix issue 310.
BUG=310
TEST=session_info_unittest.cc

Review URL: https://webrtc-codereview.appspot.com/404004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1782 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-28 12:09:09 +00:00
henrike@webrtc.org
26085e18e0 Coverity fixes for module/media_file.
BUG=Coverity report.
TEST=N/A.

Review URL: https://webrtc-codereview.appspot.com/397003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1780 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 21:50:40 +00:00
leozwang@webrtc.org
ead7d25c1a Revert r1775 which caused building errors.
TBR=pwestin@webrtc.org, mflodman@webrtc.org

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1778 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 19:45:09 +00:00
leozwang@webrtc.org
2559cbf7b7 Change WebRtc_Word8 to char
Review URL: https://webrtc-codereview.appspot.com/405003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1777 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 19:18:25 +00:00
leozwang@webrtc.org
3e9e0f0497 Change WebRtc_Word8 to char
Review URL: https://webrtc-codereview.appspot.com/405004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1776 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 19:17:38 +00:00
leozwang@webrtc.org
adb89f56e0 Change WebRtc_Word8 to char
Review URL: https://webrtc-codereview.appspot.com/405005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1775 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 19:12:19 +00:00
xians@webrtc.org
cf1b6aec30 iReduced the flakiness of the volume tests in linux pulseaudio
Review URL: https://webrtc-codereview.appspot.com/390013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1774 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 17:22:49 +00:00
mflodman@webrtc.org
b4556cd29a Enabling mjpg for Windows.
BUG=306
TEST=ViE loopback call on windows with resolution 960x720
Review URL: https://webrtc-codereview.appspot.com/411003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1770 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 14:02:12 +00:00
stefan@webrtc.org
1bb1da4c30 Enable MFQE if we are recieving temporal layers.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/411002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1769 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 13:52:34 +00:00
mflodman@webrtc.org
f3811194a5 Enable mjpg capture for Linux.
BUG=306
TEST=ViE Loopback test using resolution larger than 640x480.

Review URL: https://webrtc-codereview.appspot.com/411001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1768 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-27 08:10:17 +00:00
mflodman@webrtc.org
8df260023b Prepared for MJPG capture without using MJPG DirectShow filter. MJPG is temporarily disabled and will enabled as soon as MJPG->I420 conversion is available.
Review URL: https://webrtc-codereview.appspot.com/397011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1761 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-24 10:06:30 +00:00
marpan@webrtc.org
946601e408 Change default packetization mode to an equal size mode.
This will produce equal size packets for each frame, which should be somewhat more favorable (less overhead/padding data) for the FEC.
Review URL: https://webrtc-codereview.appspot.com/396013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1754 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-23 18:52:53 +00:00
henrike@webrtc.org
70efc3250d Factory method for the ADM in the interface file.
BUG=N/A
TEST=no

Review URL: https://webrtc-codereview.appspot.com/396017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1753 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-23 17:45:33 +00:00
xians@webrtc.org
6eb0ca2e75 Two problems are fixed:
#1, avoid leaving the lock without entering the lock.
#2, race problems in variables like _playError, _recError, _recWarning, _playWarning.
Review URL: https://webrtc-codereview.appspot.com/400006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1751 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-23 10:39:53 +00:00
mflodman@webrtc.org
4f9e44f5c5 Prepared for MJPG capturing on Linux. MJPG is conversion is not available in libyuv yet, so this CL is only made as preparation.
When this is available in libyuv, I'll remove the ifdef.

BUG=306
TEST=Manual loopback test with a high resolution, verify high FR.

Review URL: https://webrtc-codereview.appspot.com/397008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1748 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-23 09:00:26 +00:00
leozwang@webrtc.org
4ad4c24092 Add android to audio device module
Review URL: https://webrtc-codereview.appspot.com/402001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1745 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-22 16:04:59 +00:00
xians@webrtc.org
539ef94f20 Remove the deprecated kTraceModuleCall trace from audio coding module.
Review URL: https://webrtc-codereview.appspot.com/399002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1741 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-22 08:35:03 +00:00
leozwang@webrtc.org
20e9cf274d Add android to video capture module
Review URL: https://webrtc-codereview.appspot.com/399010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1740 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-22 00:40:45 +00:00
mallinath@webrtc.org
0d757b8610 Fixing coverity issues in capture module.
Review URL: https://webrtc-codereview.appspot.com/399008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1736 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-21 16:47:55 +00:00
niklas.enbom@webrtc.org
7cb0c240cb Trying to free up hellner from review work, since he mainly works in libJingle.
Review URL: https://webrtc-codereview.appspot.com/392020

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1734 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-21 13:58:58 +00:00
xians@webrtc.org
8435e8e3d8 Remove the deprecated kTraceModuleCall trace from audio processing module.
Review URL: https://webrtc-codereview.appspot.com/399003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1733 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-21 10:37:26 +00:00
xians@webrtc.org
20aabbb0be Remove the deprecated kTraceModuleCall trace from audio device module.
Review URL: https://webrtc-codereview.appspot.com/396011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1729 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 09:17:41 +00:00
xians@webrtc.org
9a798d3fca Remove the deprecated kTraceModuleCall trace from video processing module.
Review URL: https://webrtc-codereview.appspot.com/395012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1728 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 09:00:35 +00:00
xians@webrtc.org
843c8c78ff Remove the deprecated kTraceModuleCall trace from video modules.
Review URL: https://webrtc-codereview.appspot.com/391015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1725 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 08:45:02 +00:00
xians@webrtc.org
6bde7a88f1 Remove the deprecated kTraceModuleCall trace from utility module.
Review URL: https://webrtc-codereview.appspot.com/401002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1724 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 08:39:25 +00:00
xians@webrtc.org
57fb09ac18 Remove the deprecated kTraceModuleCall trace from udp transport module.
Review URL: https://webrtc-codereview.appspot.com/395011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1723 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 08:38:21 +00:00
xians@webrtc.org
03039d56e6 Remove the deprecated kTraceModuleCall trace from media file module.
Review URL: https://webrtc-codereview.appspot.com/392016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1722 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 08:37:49 +00:00
xians@webrtc.org
56cfe80c74 Remove the deprecated kTraceModuleCall trace from conference mixer.
Review URL: https://webrtc-codereview.appspot.com/396010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1721 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-20 08:35:37 +00:00
tina.legrand@webrtc.org
145f04f0c4 Changing Celt to use stereo as default.
Review URL: https://webrtc-codereview.appspot.com/397009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1720 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-18 00:32:16 +00:00
marpan@webrtc.org
bd5648f2db Reverting 1718: failed linux video test.
TBR=stefan, andrew, marpan.
Review URL: https://webrtc-codereview.appspot.com/392018

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1719 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 23:16:58 +00:00
marpan@webrtc.org
883e716304 Removed unused motion vector metrics from VideoContentMetrics;
also removed other related unused variables and code. 

Reset frame rate estimate in mediaOpt when frame rate reduction is decided.

Update content_metrics with frame rate and qm_resolution with frame size.
Review URL: https://webrtc-codereview.appspot.com/395007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1718 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 18:35:23 +00:00
mflodman@webrtc.org
4cb060127c Disabled RTPModule VP8 packetizer assert.
BUG=293

Review URL: https://webrtc-codereview.appspot.com/399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1715 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 13:07:01 +00:00
tina.legrand@webrtc.org
79e29e510f Adding option to change bitrate for Celt.
I have updated the code so that Celt rate can be changed to any value between 48 and 128 kbps.
Tests for both mono and stereo are updated.Updated tests for Celt mono.

Review URL: https://webrtc-codereview.appspot.com/391014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1712 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-17 00:38:33 +00:00
mallinath@webrtc.org
ee628358f4 Updating the object-c++ file after change in the API
GetBestMatchedCapability
Review URL: https://webrtc-codereview.appspot.com/396009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1710 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 19:30:37 +00:00
mallinath@webrtc.org
8b4a98d0f4 Change in the interface file for GetBestMatchedCapability method. Updating mac files.
Review URL: https://webrtc-codereview.appspot.com/389013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1709 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 19:00:28 +00:00
mallinath@webrtc.org
12984f0d02 Fixing Coverity issues
Note: This doesn't address Google Code style guidelines issues.
Review URL: https://webrtc-codereview.appspot.com/391011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1707 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 18:18:21 +00:00
mflodman@webrtc.org
f7b6078f6f Allow multiple send channels for REMB. Current implementation splits the remote estimate evenly between all senders.
This CL will be followed by a CL adding support for several REMB groups.

TEST=video_engine_core_unittests

Review URL: https://webrtc-codereview.appspot.com/394002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1705 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 14:50:24 +00:00
stefan@webrtc.org
439be29445 Add APIs for getting receive-side estimated bandwidth and codec target rate.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/391012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1704 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 14:45:37 +00:00
braveyao@webrtc.org
590e5eb283 Convert audio layer to WAV on Vista RTM(without any Service Pack)
Review URL: https://webrtc-codereview.appspot.com/397001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1702 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 03:21:05 +00:00
henrike@webrtc.org
d6d014ff12 Fixes memory leaks introduced in 1698.
Review URL: https://webrtc-codereview.appspot.com/387014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1701 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-16 02:18:09 +00:00
henrike@webrtc.org
f5da4da409 Removes a global non POD instance from the RTP_RTCP module that was introduced in https://code.google.com/p/webrtc/source/detail?r=1076.
Review URL: https://webrtc-codereview.appspot.com/314001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1698 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 23:54:59 +00:00
henrike@webrtc.org
05e0601160 Fixes coverity warnings in the udp_transport module.
BUG=Coverity warnings.
TEST=N/A.

Review URL: https://webrtc-codereview.appspot.com/392012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1696 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 19:43:51 +00:00
henrike@webrtc.org
6b9253eb4f Fixe issues reported by Coverity for modules/utility.
BUG=From Coverity
TEST=N/A

Review URL: https://webrtc-codereview.appspot.com/389011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1695 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 18:48:16 +00:00
henrike@webrtc.org
b38a66aaac Fixes a coverity warning in the mixer module.
Review URL: https://webrtc-codereview.appspot.com/388009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1688 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-15 00:04:27 +00:00
marpan@webrtc.org
79a99de8e4 Reverting 1680: valgrind memory leak reported.
TBR=marpan
Review URL: https://webrtc-codereview.appspot.com/392011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1686 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 22:37:10 +00:00
marpan@webrtc.org
738bcdc4ee Fix to coverity issue 10339.
Review URL: https://webrtc-codereview.appspot.com/391010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1685 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 20:54:57 +00:00
andrew@webrtc.org
737c023e42 Properly disable sse2 source on non-x86.
BUG=
TEST=build on Linux.

Review URL: https://webrtc-codereview.appspot.com/387008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1684 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 19:57:50 +00:00
braveyao@webrtc.org
59d6cec291 Fix the crash at playing 48kHz stereo wav file.
http://code.google.com/p/webrtc/issues/detail?id=208
Review URL: https://webrtc-codereview.appspot.com/396001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1681 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 18:17:16 +00:00
marpan@webrtc.org
4e34dcbd62 Allow for spatial-downsampling without reinitializaing encoder. Change of frame size will automatically trigger new key frame in codec. This feature is set off in vie_encoder until we upgrade to the new libvpx.
Also reset frame rate estimate in mediaOpt when frame rate reduction is decided.
Review URL: https://webrtc-codereview.appspot.com/390006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1680 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 17:26:24 +00:00
mflodman@webrtc.org
d7d46887a6 Update receive only channels with RTT.
vie_autotest_loopback.cc will not be part of the commit, it's only to show the test.

TEST=temporary with attached loopback test.

Review URL: https://webrtc-codereview.appspot.com/390007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1678 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-14 12:49:59 +00:00
pwestin@webrtc.org
c76c096c19 Bugfix issue 273, workaround for compiler issue.
Review URL: https://webrtc-codereview.appspot.com/392005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1675 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-13 09:56:57 +00:00
pwestin@webrtc.org
52fd98d876 Removing encoder reset. Function did not make sence.
Review URL: https://webrtc-codereview.appspot.com/391005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1674 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-13 09:03:53 +00:00
marpan@webrtc.org
567d507707 Fixes a bug when number of media packets in a frame is larger than maximum allowed for the generateFEC.
Review URL: https://webrtc-codereview.appspot.com/391003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1673 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-10 18:56:14 +00:00
mflodman@webrtc.org
8224e19dd9 Fixed incorrect packet loss reported to encoder.
BUG=275

Review URL: https://webrtc-codereview.appspot.com/394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1669 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-10 12:41:57 +00:00
pwestin@webrtc.org
5e954814a8 Clanup handling of key frame requests and FIR.
Review URL: https://webrtc-codereview.appspot.com/387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1667 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-10 12:13:12 +00:00
andrew@webrtc.org
75f1948b0e Restore AECM Coverity fix.
Add a test which would have caught the crash introduced by r1628.

BUG=274
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/388002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1657 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 17:16:18 +00:00
stefan@webrtc.org
4b377414f1 Fix release build errors.
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/394005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1654 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 13:50:57 +00:00
xians@webrtc.org
3dbed8597e This CL makes the playout delay value thread safe.
With the patch, _sndCardPlayDelay is calculated in the DoRenderThread instead of capture thread, an capture thread only gets the _sndCardPlayDelay value.
And _sndCardPlayDelay and _sndCardRecDelay are only changed to be Atomic32 to make them to be accessed by multiple threads.


Test=None
Bug=256
Review URL: https://webrtc-codereview.appspot.com/394001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1653 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 13:50:48 +00:00
stefan@webrtc.org
9c84b0dc9f Fix build errors with GCC.
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/389006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1652 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 13:14:04 +00:00
stefan@webrtc.org
7adab0922d This removes the knowledge of frame completeness from the FEC decoder.
Therefore, with this change a recovered packet is only considered old,
and will be removed, if more than 48 recovered packets are stored.

Packets are immediately reconstructed and sent to the jitter
buffer. Before this CL packets were processed on a frame-by-frame
basis, and all packets belonging to a frame was sent to the
jitter buffer at the same time.

The number of FEC packets is also limited to 48. An FEC packet is
removed if all protected packets have been recovered or if the
FEC packet is considered old.

Lot's of tests added.

Patch-set 2:
- Fixed rtp_fec_unittest.cc to work with the new reconstruction.
- Added reference counting of Packet to be able to keep references to them from FecPacket between different reconstruction runs.
- Rewrote the packet search to use std::set_intersection.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1651 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 12:34:52 +00:00
kjellander@webrtc.org
cf6a295b13 Making video codecs test framework integration test execute in a reproducable fashion.
Fixed reproducable random behavior in packet_manipulator.h.
Test is now fully reproducable (runs on only one core) so much tighter limits are now set for the SSIM/PSNR values for the encoding/decoding (verified on all platforms)

BUG=
TEST=out/Debug/video_codecs_test_framework_integrationtests in Debug+Release on Linux, Mac, Windows and in Linux Valgrind.

Review URL: https://webrtc-codereview.appspot.com/381005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1649 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-09 09:01:51 +00:00
henrike@webrtc.org
d5657c2f69 Refactored files according to google style since http://review.webrtc.org/314001/ is blocked on this and formatting changes should not be done with code changes.
Review URL: https://webrtc-codereview.appspot.com/387005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1648 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 23:41:49 +00:00
andrew@webrtc.org
68da6adafe Remove WebRtc_ types.
Allows us to avoid the "cast to UWord32" Coverity coverup.

BUG=
TEST=test_fec

Review URL: https://webrtc-codereview.appspot.com/379002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1647 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 22:24:14 +00:00
wu@webrtc.org
a8084b07e3 Revert r1628 which causes the crash of voe_auto_test.
With r1628, it's possible the second memcpy got a NULL nearendClean.

TBR=bjornv
Review URL: https://webrtc-codereview.appspot.com/390005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1643 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 17:56:39 +00:00
tina.legrand@webrtc.org
13ac430bef Adding missing timestamp calculation to RTPencode.
Review URL: https://webrtc-codereview.appspot.com/392002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1641 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 13:20:36 +00:00
mflodman@webrtc.org
d2940f71e4 VCM::JB critsect fix.
Review URL: https://webrtc-codereview.appspot.com/390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1639 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 12:42:56 +00:00
stefan@webrtc.org
23307f7c4b Remove frame_list.cc from Andorid.mk.
TBR=mflodman
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/390003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1638 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 10:39:13 +00:00
tina.legrand@webrtc.org
df69775bfa Adding support for full-stereo codec.
This is an experiment, letting Celt represent a full-stereo codec.

Review URL: https://webrtc-codereview.appspot.com/379013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1636 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 10:22:21 +00:00
stefan@webrtc.org
2979461595 Refactored the jitter buffer to use std::list.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/352016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1635 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 08:58:55 +00:00
stefan@webrtc.org
7dfa883954 Disable spatial subsampling for denoiser variance estimation.
With subsampling there are sometimes quite visible trailing
artifacts.

BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/387002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1634 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 08:27:31 +00:00
pwestin@webrtc.org
95392e64ba Bugfix EnableIPV6 issue 255
Review URL: https://webrtc-codereview.appspot.com/378005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1633 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 08:08:37 +00:00
kjellander@webrtc.org
1970b2fcb3 Fixing uninitialized codec settings struct in test.
BUG=
TEST=video_codecs_test_framework_unittests passing in Debug+Release on Linux, Mac and Windows.

Review URL: https://webrtc-codereview.appspot.com/378004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1632 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 07:09:32 +00:00
andrew@webrtc.org
648af7423f Clean up MapSetting().
- Add assert(false) for "impossible" cases.
- Remove tests for invalid enum values.
- Modify MapError() to look the same way.

BUG=
TEST=audioproc_unittest

Review URL: https://webrtc-codereview.appspot.com/386001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1631 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 01:57:29 +00:00
wu@webrtc.org
9143f774d1 Coverity fix for VideoRenderModule including issues 10084, 10226, 10267 and 10340.
Review URL: https://webrtc-codereview.appspot.com/385001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1630 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-08 00:14:25 +00:00
bjornv@webrtc.org
236e842bca Removed memcpy of pointer to itself, triggering Valgrind warning.
BUG=272
Review URL: https://webrtc-codereview.appspot.com/389002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1628 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 17:22:44 +00:00
phoglund@webrtc.org
78088c2f36 Removed warnings on Windows and enabled warnings-as-errors on Windows.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1624 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 14:56:45 +00:00
niklas.enbom@webrtc.org
87885e8409 This CL will look a bit strange. Essentially I've removed sanity checks for > 1 channel and then fixed the bugs that remained. Will add testing in a separate CL.
BUG=
TEST=

Review URL: https://webrtc-codereview.appspot.com/390001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1623 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-02-07 14:48:59 +00:00