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Author SHA1 Message Date
tina.legrand@webrtc.org
886aef09a8 Fixing broken tests in voe_auto_test extended 
This CL fixes the problem with voe_auto_test extended-codec test, as well as
extended-file test. First problem was that Opus was not added as a special case, like the other codecs, and the second problem was that the tests were not updated when test files were moved to the resources catalogue.

There are still some tests that fails. Here is a list of all extended tests and their status:

Base: fails - the reason seem to be that external transport has been removed.
CallReport: passes
Codec: passes (with this CL)
DTMF: passes
Encryption: fails or is dissabled?
VoEExternalMedia: passes
File: passes (with this CL)
Hardware: passes
NetEqStats: empty?
Network: passes
RTP_RTCP: fails
VideoSync: fails
VolumeControl: passes

BUG=issue2234
R=andrew@webrtc.org, henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2023004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 10:39:56 +00:00
kjellander@webrtc.org
3555303cb0 Roll chromium_revision 226126:228675 and fix clang warnings 
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.

This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.

TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-15 20:10:17 +00:00
minyue@webrtc.org
cc92e000f3 1. adding request of ACM version in the manual mode of voe_auto_test 
2. adding command line flag for automated mode of voe_auto_test to choose between ACMs

3. adding request of ACM version in voe_cmd_test

R=phoglund@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2281004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-30 08:43:50 +00:00
henrike@webrtc.org
563910bde3 Disables RtpRtcpTest.CanTransmitExtraRtpPacketsWithoutError as it flakily breaks the waterfall. See http://chromegw.corp.google.com/i/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/99/steps/voe_auto_test/logs/stdio the cl triggering it was a no-change (disabled some other broken tests). 
TBR=wu@webrtc.org

BUG=2296

Review URL: https://webrtc-codereview.appspot.com/2098004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4604 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-23 16:16:03 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:38:54 +00:00
pbos@webrtc.org
d65914360a Disable CanTransmitExtraRtpPacketsWithoutError on Windows. 
Flakily crashes on Windows.

BUG=2240
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 09:44:19 +00:00
pbos@webrtc.org
6cd9341801 Hand over loopback packets to a network thread. 
This version of LoopBackTransport hands packets over to a network thread
which will deliver them instead. This allows SendRTP and SendRTCP to
always be able to return, preventing deadlocks in voe_auto_test. The
previous case did not represent actual network usage. Now the send and
receive side can run concurrently with the receiving side. Previously
the sender thread also drove the receiving side, which does not
represent the regular use case where packets are put on a network
socket.

BUG=1568,2081,2178
TEST=Ran VoiceEngine RtpRtcpTest.*, known for deadlocking, 100+ times.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1985005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-09 21:11:57 +00:00
pbos@webrtc.org
676ff1ed89 Ref-counted rewrite of ChannelManager. 
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.

ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.

BUG=2081
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1802004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-07 17:57:36 +00:00
phoglund@webrtc.org
94aca5c7de Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. 
TBR=xians@webrtc.org
BUG=2179

Review URL: https://webrtc-codereview.appspot.com/1955005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 08:20:47 +00:00
phoglund@webrtc.org
bd69d1beaf Disabled SsrcPropagatesCorrectly on Linux. 
BUG=2178
TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1975004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4494 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-06 08:03:16 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
pbos@webrtc.org
a3f30143b7 Default constructor for RtcpAppHandler. 
Whenever this test (RtcpApplicationDefinedPacketsCanBeSentAndReceived) fails
because it's being run on a slower system (such as one running under valgrind),
valgrind reports a lot of undefined-value errors. Initializing the data
makes sure that, while the EXPECT_EQs trigger, they don't cause any errors in
valgrind.

BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1822004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 14:25:45 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
pbos@webrtc.org
5b10d8fb18 Fix some voe_auto_test uninitialised-value errors. 
BUG=
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 15:50:07 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 14:30:48 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
pwestin@webrtc.org
1064cf06b0 Fixed Rtp/Rtcp tests 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1627005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-07 16:03:19 +00:00
andrew@webrtc.org
0a38432ea5 Fix error in mixing test for supported sample rates. 
With the switch to an arbitrary resampler, we now support these strange
rates.

TBR=turaj

Review URL: https://webrtc-codereview.appspot.com/1604004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 22:52:09 +00:00
wu@webrtc.org
fa64a595ad Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. 
This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.

BUG=1828
TEST=unit tests
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1598005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 21:27:57 +00:00
pbos@webrtc.org
956aa7e087 Include files from webrtc/.. paths in voice_engine/ 
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 13:52:32 +00:00
pbos@webrtc.org
8a025e26db Make sure VoiceEngine tests only include one test framework. 
BUG=
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4078 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 11:25:12 +00:00
pbos@webrtc.org
9213521ea9 Remove const for plain data types in voice_engine/ 
BUG=1644
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1463004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 08:31:39 +00:00
phoglund@webrtc.org
258f55efc0 Disabled flaky test. 
BUG=1719
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 12:35:00 +00:00
pwestin@webrtc.org
1de01354e6 Adding playout buffer status to the voe video sync 
Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-11 20:23:35 +00:00
pbos@webrtc.org
6141e13873 WebRtc_Word32 -> int32_t in voice_engine/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1305004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 10:09:10 +00:00
pwestin@webrtc.org
b9e402d99f Remove WEBRTC_*_ENGINE_NETWORK_API use 
Review URL: https://webrtc-codereview.appspot.com/1203009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:51:42 +00:00
henrika@webrtc.org
aa527bbc91 Disabling MixingTests due to race conditions. 
BUG=1580
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/1285005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 15:19:10 +00:00
pwestin@webrtc.org
0c45957e3a Remove UDP transport API from VoE 
Review URL: https://webrtc-codereview.appspot.com/1236004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-03 15:43:57 +00:00
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +00:00
solenberg@webrtc.org
a442d4d983 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. 
Today I had to figure out this code was legacy. Now next person doesn't have to.

BUG=

Review URL: https://webrtc-codereview.appspot.com/1247004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 09:14:36 +00:00
wu@webrtc.org
80fccc29de Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." 
> Removed CPU APIs from VoEHardware. Code is now only used by test applications.
> 
> BUG=8404677
> 
> Review URL: https://webrtc-codereview.appspot.com/1238004

TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 23:38:21 +00:00
henrike@webrtc.org
4c138e8fca Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
BUG=8404677

Review URL: https://webrtc-codereview.appspot.com/1238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 21:23:42 +00:00
pwestin@webrtc.org
e30823911c Move the VoE tests to use external transport instead of the built in udp transport 
Review URL: https://webrtc-codereview.appspot.com/1223006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 16:12:57 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665 
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. 
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-13 17:52:42 +00:00
turaj@webrtc.org
b7edd06530 Remove DTMF detection. Talk team has been in the loop and there is no need for 
DTMF detection at the receiver side.

test=voe_auto_test, VoE extended test DTMF
Review URL: https://webrtc-codereview.appspot.com/1168004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-12 22:27:27 +00:00
turaj@webrtc.org
24045c5a02 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. 
bug=issue1370
test=trybots
Review URL: https://webrtc-codereview.appspot.com/1121007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-05 03:14:22 +00:00
andrew@webrtc.org
6ed8ebcef9 Fix MaxChannels test; 32 -> 100. 
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/1060010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-02 00:05:58 +00:00
andrew@webrtc.org
4a6f62d4dc Remove (in practice) the voice engine channel limit. 
There's really no reason for this limit. I've bumped it to a
practically unreachable ceiling, with a TODO for removing it
entirely.

TBR=henrika
BUG=b/8122300

Review URL: https://webrtc-codereview.appspot.com/1070014

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-01 23:42:44 +00:00
andrew@webrtc.org
08d660f08e Allow for some error in volume testing. 
BUG=616
TESTED=voe_auto_test:VolumeTest.* now passes on a MacBook

Review URL: https://webrtc-codereview.appspot.com/1028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 17:07:02 +00:00
phoglund@webrtc.org
d005468e9b Generalized mechanism for excluding gtests on platforms, disabled broken tests on mac. 
BUG=1268
TEST=vie_auto_test on mac and linux
TBR=mflodman, kjellander

Review URL: https://webrtc-codereview.appspot.com/1027006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-09 16:53:42 +00:00
andrew@webrtc.org
1926d33344 Change Sleep() comment in test fixture. 
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/1023006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-05 03:30:11 +00:00
phoglund@webrtc.org
6f62836ccf Reverting two mixing test patches: seems to introduce a persistent problem for win voe_auto_test (wrapping problem?) 
Revert "Further relax thresholds in mixing test."

This reverts commit 53c7e973a02d65e0b4981129e7ccfc145d955eda.

Revert "Fix implicit conversion error in mixing test."

This reverts commit 68d7e2258082d7d2b9461061e03e2f2d6ae78c4f.

TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1018005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3331 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-03 14:33:00 +00:00
andrew@webrtc.org
201d4b61d1 Fix implicit conversion error in mixing test. 
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1020004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 19:59:53 +00:00
andrew@webrtc.org
b2b628d5cd Further relax thresholds in mixing test. 
TBR=mikhal

Review URL: https://webrtc-codereview.appspot.com/1019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 18:50:13 +00:00
andrew@webrtc.org
00c7c4315b Replace voice engine utility functions with system wrapper variants. 
SLEEP -> SleepMs
GET_TIME_IN_MS -> TickTime::MillisecondTimestamp

These could cause unused function errors on some compilers.

BUG=1228

Review URL: https://webrtc-codereview.appspot.com/1013004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 16:06:39 +00:00
phoglund@webrtc.org
1c75918302 Disabled flaky test. 
From flake in http://webrtc-cb-linux-master.cbf.corp.google.com:8011/builders/LinuxLargeTests/builds/270

Review URL: https://webrtc-codereview.appspot.com/1001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 10:40:05 +00:00
roosa@google.com
b8ba4d8109 Add number of inserted samples to NetEq statistics. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/964030

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 00:06:18 +00:00
roosa@google.com
1b60ceb499 Add GetAudioFrame API to VoiceEngine. 
Allows the caller to pull frames from a channel instead of sending them to the output mixer.

BUG=

Review URL: https://webrtc-codereview.appspot.com/973012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:00:29 +00:00
roosa@google.com
b718619f0a Expose NetEq playout mode off through VoiceEngine. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/971016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 21:59:14 +00:00