pbos@webrtc.org
de1429e9ad
Add thread annotations to Call API.
...
Also constified a lot of pointers and reordered members to make
protected members more grouped together.
R=kjellander@webrtc.org , stefan@webrtc.org
BUG=2770
Review URL: https://webrtc-codereview.appspot.com/15399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 13:00:21 +00:00
kjellander@webrtc.org
7d825e9b2c
Revert "libjingle_unittest now compiles and passes on iOS!"
...
This reverts commit r5986 as it fails compilation on Mac
(non-iOS). The failure was not discovered on the commitbots
since they don't clobber their builds.
BUG=3241
TBR=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5997 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 12:50:47 +00:00
pbos@webrtc.org
c891577e6d
Disable flaky CaptureNtpTimeWithNetworkJitter.
...
TBR=wu@webrtc.org
BUG=3271
Review URL: https://webrtc-codereview.appspot.com/19389005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5996 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:57:32 +00:00
bjornv@webrtc.org
494aa0e93d
AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h
...
These enums are noly used internally in aec_core.c and it makes more sense to put them in aec_core_internal.h
TESTED=trybots
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19379005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5995 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:42:27 +00:00
pbos@webrtc.org
8dfe8ff590
Disable capture test for FrameRate on Windows.
...
Flaky on Windows, has been for a while.
R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=3270
Review URL: https://webrtc-codereview.appspot.com/19389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5994 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 11:27:36 +00:00
henrik.lundin@webrtc.org
e772c71743
Introduce a config struct for AudioCoding module
...
The config struct currently contains the module ID, and the NetEq
config struct, but will be extended in the future. The purpose of this
change is to expose certain NetEq settings to the ACM interface.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5993 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 10:16:57 +00:00
pbos@webrtc.org
f043f79711
Disabling flaky CanReceiveFec.
...
CanReceiveFec is flaky, likely to the test expecting the first FEC
packet to always be decoded and rendered.
R=stefan@webrtc.org
BUG=3269
Review URL: https://webrtc-codereview.appspot.com/19379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5992 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 09:00:50 +00:00
pbos@webrtc.org
69e9950469
Disable flaky RunsRtpRtcpTestWIthoutErrors.
...
BUG=1790
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5991 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:49:07 +00:00
henrik.lundin@webrtc.org
12a34247a4
Fix the NetEq build
...
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5990 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:36:35 +00:00
henrik.lundin@webrtc.org
116ed1d4f0
Include buffer size limits in NetEq config struct
...
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.
The old constants governing the packet buffer limits are deleted.
BUG=3083
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:20:04 +00:00
henrik.lundin@webrtc.org
b08bbf57a6
Add henrik.lundin as owner in AudioCoding module
...
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5988 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-28 08:15:35 +00:00
mallinath@webrtc.org
a0d3067575
Use CreatePeerConnection method which accepts port_allocator.
...
Other method will be removed, in a different CL.
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20369006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-26 00:00:15 +00:00
fischman@webrtc.org
95cd1551f8
libjingle_unittest now compiles and passes on iOS!
...
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && ~/src/ios-deploy/ios-deploy -d -u -v -b ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app
Note that the test's use of signals means that lldb will break in the middle of the suite. To ignore these signals tell lldb:
pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue
BUG=3241
R=noahric@google.com , tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5986 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:59:56 +00:00
andrew@webrtc.org
8f69330310
Replace scoped_array<T> with scoped_ptr<T[]>.
...
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar ...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
buildbot@webrtc.org
658a94595d
(Auto)update libjingle 65619249-> 65622932
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5984 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 22:03:30 +00:00
buildbot@webrtc.org
ff90ed6e96
(Auto)update libjingle 65561104-> 65619249
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5983 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 21:12:10 +00:00
andrew@webrtc.org
2eceb8ef46
Roll third_party/opus 258909:262302
...
Suppresses the nanny "not-optimized" warning.
TBR=minyue
BUG=2864
Review URL: https://webrtc-codereview.appspot.com/14379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5982 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 15:29:33 +00:00
stefan@webrtc.org
0175d76c72
Fix leak in remote bitrate estimator tests introduced in r5980
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5981 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 11:38:57 +00:00
stefan@webrtc.org
4f616a02bd
Support for simulating multiple independent flows in a network.
...
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5980 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 10:59:24 +00:00
asapersson@webrtc.org
46106f2a05
Casting char to int in logs.
...
BUG=3153
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12369006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5979 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 07:02:52 +00:00
buildbot@webrtc.org
2b93402e36
(Auto)update libjingle 65484212-> 65561104
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5978 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 00:18:27 +00:00
jiayl@webrtc.org
cc1ba15fe7
Returns a NULL frame on all platforms if the captured window is closed.
...
Part of the fix for crbug/360181.
On Mac/Linux, it previously continues capturing even if the window is closed.
Now it stops by returning a NULL frame.
On Windows, it used to stop capturing when the window is minimized. Now fixed to match other platforms.
Note: the crbug still needs a chrome side fix to close the notification bar.
This fix only stops the stream (i.e. stream onended event fired).
BUG=crbug/360181
TESTED=manually tested in Chrome
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/12329007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5977 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 23:45:56 +00:00
wu@webrtc.org
cd70119a10
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
...
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 22:10:24 +00:00
wu@webrtc.org
93fd25c20c
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
...
* Cast rtp header extension to int in log in rtp_utility.cc.
BUG=3237
TEST=try bots
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 20:33:08 +00:00
henrik.lundin@webrtc.org
439a4c49f9
Add an output capacity parameter to ACMResampler::Resample10Msec()
...
Also adding a unit tests to make sure that a desired output frequency
of 0 passed to AudioCodingModule::PlayoutData10Ms() is invalid.
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 19:05:33 +00:00
andrew@webrtc.org
103657b484
Add keyboard channel support to AudioBuffer.
...
Also use local aliases for AudioBuffers for brevity.
BUG=2894
R=aluebs@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 18:28:56 +00:00
henrik.lundin@webrtc.org
d57b8149c2
Fix the Android compilation (better structure for NetEq test libs)
...
This change should make the Android targets compile again. The reason
for the failure was a highly dubious structure in the gypi files. With
this fix, the structure is somewhat cleaner. Still room for improvement.
BUG=3254
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5972 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 13:19:04 +00:00
pbos@webrtc.org
5ca6a5387e
Remove TraceCallback use from Call.
...
Non-global logging isn't supported, and having a per-call logging
dispatch seems over-eager and adds more complexity than it's worth. The
current implementation is racy on initialization due to missing atomics
support. Besides, logging support should be separate from use of Call.
R=mflodman@webrtc.org
BUG=3250,3157
Review URL: https://webrtc-codereview.appspot.com/12329006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5971 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 11:35:33 +00:00
pbos@webrtc.org
a5c8d2c9b3
Rename Start/Stop in Video{Send,Receive}Streams.
...
Rename {Start,Stop}{Sending,Receving} to Start/Stop. StartSending
provides no extra information in the context of a VideoSendStream, as
what it does is to send.
R=mflodman@webrtc.org
BUG=3227
Review URL: https://webrtc-codereview.appspot.com/12329005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5970 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 11:13:21 +00:00
henrik.lundin@webrtc.org
0a2277448e
Fixing a bug in ACM2 where the output frame energy was incorrectly set
...
The value of AudioFrame::energy_ must be set to -1 in order to have the
energy calculated later on in the AudioConferenceMixer module. This was
not the case in ACM2, where the value was set to 0 instead. This
resulted in bad audio for multi-party calls (5 or more participants).
Implemented a unit test to verify ACM output frame.
BUG=3255
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5969 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 08:11:39 +00:00
andrew@webrtc.org
f26c9e8369
Use unique filenames in AudioProcessingTests for parallelization.
...
TBR=bjornv
TESTED="gtest-parallel -w 32 --gtest_filter=*AudioProcessingTests*
out/Debug/modules_unittests" passes.
Review URL: https://webrtc-codereview.appspot.com/14369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5968 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 03:46:46 +00:00
buildbot@webrtc.org
3f1aa24078
(Auto)update libjingle 65469804-> 65484212
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5967 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 00:00:12 +00:00
jiayl@webrtc.org
0d915ff603
Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet.
...
R=juberti@webrtc.org , mallinath@webrtc.org
BUG=3244
Review URL: https://webrtc-codereview.appspot.com/12299006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 20:46:29 +00:00
bjornv@webrtc.org
e9d3760d5c
AEC: Adds a reported_delay_enabled_ flag
...
Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.
TESTED=trybots, modules_unittest
R=aluebs@webrtc.org , andrew@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12349005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 13:20:07 +00:00
henrik.lundin@webrtc.org
26e2b687fc
Remove ACM1/ACM2 switching from VoiceEngine tests
...
The option to run VoiceEngine tests with both ACM1 and ACM2 was
introduced while the two versions of AudioCoding module where both
in use. Now, ACM1 is being deprecated, and the tests should use the
defualt one (ACM2).
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5964 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 08:39:41 +00:00
andrew@webrtc.org
db144429b2
Exclude the new AudioProcessingTest from some sanitizer bots.
...
It takes too long.
TBR=kjellander
Review URL: https://webrtc-codereview.appspot.com/12359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5963 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 04:16:24 +00:00
andrew@webrtc.org
46b31b17df
Restore sample_rate_hz() until Chromium is updated to not use it.
...
TBR=bjornv
TESTED=Chromium builds against webrtc head.
BUG=2894
Review URL: https://webrtc-codereview.appspot.com/12349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5962 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 03:33:54 +00:00
buildbot@webrtc.org
504fc89f36
(Auto)update libjingle 65394435-> 65417850
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 03:23:19 +00:00
tkchin@webrtc.org
19b1be159e
Provide GetStats method in RTCPeerConnection
...
BUG=3144
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 21:05:38 +00:00
andrew@webrtc.org
ddbb8a2c24
Support arbitrary input/output rates and downmixing in AudioProcessing.
...
Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org , bjornv@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 21:00:04 +00:00
henrik.lundin@webrtc.org
34fe0153b9
Reland "Stop using ACM factory in VoiceEngine"
...
This change was originally landed as r5954, but had to be reverted in
r5955 due to bots failing. The failures should be fixed in r5956,
so the original change is now relanded.
BUG=2996
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5958 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 19:04:34 +00:00
andrew@webrtc.org
d59359af4d
Remove 44.1 kHz workaround from the iOS AudioDevice.
...
Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.
This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.
BUG=1395
R=fischman@webrtc.org , henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 18:07:49 +00:00
henrik.lundin@webrtc.org
20c71fd1dc
Fix a bug in AcmReceiver::NetworkStatistics
...
One of the variables were not copied between the structs.
BUG=2996
TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5956 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 10:11:21 +00:00
henrik.lundin@webrtc.org
0c108d0b4d
Revert "Stop using ACM factory in VoiceEngine"
...
Some of the bots where breaking.
TBR=henrika@webrtc.org
BUG=2996
Review URL: https://webrtc-codereview.appspot.com/12319004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5955 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 09:44:00 +00:00
henrik.lundin@webrtc.org
139706ec0b
Stop using ACM factory in VoiceEngine
...
The factory injection was introduces in order to facilitate switching
between ACM1 and ACM2. Now, ACM1 is being deprecated, and this switching
mechanism is no longer needed.
BUG=2996
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5954 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:51:21 +00:00
henrik.lundin@webrtc.org
d144bb6812
Let A/V sync test use default AudioCoding module
...
This test used to run with both ACM1 and ACM2, to verify sync with both
versions of the module. ACM1 (and NetEq3) is now being deprecated,
wherefore this test should now use the default one (i.e., ACM2).
BUG=2996
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5953 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:36:33 +00:00
henrik.lundin@webrtc.org
0c1444c748
Create ACM2 instance when calling AudioCodingModule::Create
...
BUG=2996
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5952 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:18:42 +00:00
kjellander@webrtc.org
0a035c8924
Disable tests in common_video_unittests for Dr Memory.
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These tests takes about 1000 seconds to execute under
Dr Memory Full, causing them to be killed if taking more
than 1200 seconds.
* TestScaler.BiLinearScaleTest
* TestScaler.BoxScaleTest
BUG=3247
TBR=zhaoqin@google.com
Review URL: https://webrtc-codereview.appspot.com/12289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5951 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 08:07:03 +00:00
henrik.lundin@webrtc.org
372ae83228
Reland "Make VoiceEngine choose ACM2 by default""
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This cl was originally committed as r5923, but was reverted in r5926
due to a blocking bug (issue 3143). The blocking bug was resolved in
r5936.
BUG=2996
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5950 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 07:21:03 +00:00
bjornv@webrtc.org
5964fe0f86
audio_processing: DestroyHandle() now returns void
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The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.
BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org , aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 06:52:28 +00:00