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Author SHA1 Message Date
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
minyue@webrtc.org
2a8df7c375 Fixing two bugs in voe_cmd_test. 
I am trying to add a new functionality to voe_cmd_test, and I found two bugs:

1. in r5928, a functionality was removed but the item in the menu was not. Functionalities after it are offset.

r5928: https://code.google.com/p/webrtc/source/detail?r=5928&path=/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

2. in r6736, opus are set to output 48 kHz audio. When mixing Opus output with an audio file, channel.cc may go wrong.

r6736: https://code.google.com/p/webrtc/source/detail?r=6736

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-06 10:05:19 +00:00
minyue@webrtc.org
026859b983 This is related to an earlier CL of enabling Opus 48 kHz. 
https://webrtc-codereview.appspot.com/16619005/

It was reverted due to a build bot error, which this CL is to fix. The problem was that when audio conference mixer receives audio frames all at 48 kHz and mixed them, it uses Audio Processing Module (APM) to do a post-processing. However the APM cannot handle 48 kHz input. The current solution is not to allow the mixer to output 48 kHz.

TEST=locally solved https://webrtc-codereview.appspot.com/16619005/

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6730 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 12:28:28 +00:00
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
aluebs@webrtc.org
9825afc3bd Add ExperimentalNs support in Config 
R=andrew@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-30 17:39:53 +00:00
minyue@webrtc.org
c1a40a7b68 This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. 
This CL is going to be combined with another CL in ACM, which is to be landed.

TEST=passed_try_bots
BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 09:52:06 +00:00
bjornv@webrtc.org
f3e1341da7 VoEVolumeTest: Enabled Linux flaky tests 
Fixed error checks only on Linux to be able to turn on flaky tests. The cause of flaky failures is due to late values in pulse audio.

Related (deleted) CLs:
https://webrtc-codereview.appspot.com/19469007/
https://webrtc-codereview.appspot.com/19469004/

BUG=367
TESTED=trybots, voe_auto_test repeated
R=henrikg@webrtc.org, tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 10:43:42 +00:00
minyue@webrtc.org
2db9f45038 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size 
BUG=webrtc:2925

TEST=passed_all_trybots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 08:33:30 +00:00
solenberg@webrtc.org
57e060251a Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. 
Flakiness was caused by a race condition between two atomic integers shared by two threads. Fixed by counting bad packets (those not containing the expected extension) instead of the good packets.

The CL also eliminates another possible flake by introducing a test fixture which doesn't automatically start sending audio packets when constructed.

BUG=3340,3356
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:27:09 +00:00
henrika@webrtc.org
f383a1b0f2 Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 11:51:45 +00:00
bjornv@webrtc.org
06c1d6f3a1 VoEVolumeTest: Adds error return tests. 
BUG=367
TESTED=trybots, voe_auto_test
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 09:03:33 +00:00
kjellander@webrtc.org
98c76a120d Make vie/voe_auto_test accept non-supported flags without error. 
With the switch recipes on the buildbots and the deprecation of
the custom script at
https://code.google.com/p/webrtc/source/browse/trunk/webrtc/test/buildbot_tests.py
these tests will start failing when Chromium's runtest.py is passing
--brave-new-test-launcher --test-launcher-bot-mode
to the test.
A similar change was made for most of WebRTC's tests (that depends on
the test_support_main target) in
https://webrtc-codereview.appspot.com/2222005

BUG=chromium:346198
TEST=Successfully launched the executables on Linux and Mac using:
out/Release/voe_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --test-launcher-summary-output=/tmp/tmpwhx6Zz
out/Release/vie_auto_test --brave-new-test-launcher --test-launcher-bot-mode --automated --capture_test_ensure_resolution_alignment_in_capture_device=false --test-launcher-summary-output=/tmp/tmpwhx6Zz

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

Review URL: https://webrtc-codereview.appspot.com/13499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 06:01:40 +00:00
bjornv@webrtc.org
8d63d0ee70 Enables VolumeTest.DefaultMicrophoneVolumeIsAtMost255 
Rewritten the test to only check for valid volume when we have actually received a value from the audio device. To check if we have actually received a volume value is out of the scope for this test.

BUG=webrtc:367
TESTED=trybots
R=tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6123 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-13 14:14:56 +00:00
henrika@webrtc.org
6b02eea6ac Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6103 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:24:10 +00:00
henrika@webrtc.org
1cec3957b8 Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 12:19:19 +00:00
henrika@webrtc.org
66021e0fa2 Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 08:53:27 +00:00
henrika@webrtc.org
3b76627afe Removes parts of the webrtc::VoEHardware sub API (relanding) 
Relanding https://webrtc-codereview.appspot.com/18399004/

TBR=niklase

Review URL: https://webrtc-codereview.appspot.com/16489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6092 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:43:00 +00:00
henrika@webrtc.org
3106b706c0 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." 
> Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
> 
> BUG=3206
> R=andrew@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18399004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 11:10:50 +00:00
henrika@webrtc.org
9de3d844ae Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-09 10:55:11 +00:00
henrika@webrtc.org
7f3a041d23 Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test. 
BUG=3289
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 13:59:58 +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
henrika@webrtc.org
66803489f9 Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=henrik.lundin@webrtc.org, juberti@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 10:45:01 +00:00
henrika@webrtc.org
0f7375504a Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=juberti@webrtc.org, niklas.enbom@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 10:38:08 +00:00
aluebs@webrtc.org
f927fd6481 Re-enable AGC tests: 
* AgcConfigTest.HasCorrectDefaultConfiguration
* AgcConfigTest.DealsWithInvalidParameters
* AgcConfigTest.CanGetAndSetAgcStatus
* AgcConfigTest.HasCorrectDefaultRxConfiguration
* AgcConfigTest.DealsWithInvalidRxParameters
* AgcConfigTest.CanGetAndSetRxAgcStatus
* AudioProcessingTest.AgcIsOnByDefault
* AudioProcessingTest.CanEnableAgcWithAllModes
* AudioProcessingTest.RxAgcShouldBeOffByDefault
* AudioProcessingTest.CanTurnOnDigitalRxAcg
* AudioProcessingTest.CannotTurnOnAdaptiveAnalogRxAgc

BUG=webrtc:2784
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5918 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-16 11:58:18 +00:00
fischman@webrtc.org
ca539bbed0 iOS: baby steps to being able to include_tests=1 
- pull iossim in DEPS even when on mac (because bug 2152)
- fix audio_device_test_api.cc's use of bool instead of bool* (!)
- move unused-on-mobile message to non-mobile-only section of
  hardware_before_streaming_test.cc

BUG=3185
R=kjellander@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 20:26:41 +00:00
henrika@webrtc.org
b9309beea4 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 14:12:50 +00:00
andrew@webrtc.org
19018ddb17 Make ACM2 the default in voe_cmd_test. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 20:58:05 +00:00
fischman@webrtc.org
a789f3720a VoiceEngine(iOS & Android): removed NOT_SUPPORTED 
Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
  bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds

BUG=2050,3132
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 00:16:35 +00:00
solenberg@webrtc.org
440fa23553 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. 
BUG=2954
R=mflodman@webrtc.org, stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 19:57:07 +00:00
solenberg@webrtc.org
b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 10:38:25 +00:00
henrika@webrtc.org
800b8dbda6 Removes error printout in voe_cmd_test which was caused by attempts to transmit RTCP packets even if a transport object was not registered. 
BUG=none
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-20 08:07:41 +00:00
wu@webrtc.org
ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. 
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 23:49:08 +00:00
henrika@webrtc.org
b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. 
Reasons for removing:

- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.

BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 08:58:08 +00:00
solenberg@webrtc.org
00844d7bef Remove obsolete voe_unit_test. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 18:50:50 +00:00
solenberg@webrtc.org
a07923339b Remove external encryption API for VoE. 
BUG=
R=henrika@webrtc.org, henrikg@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 11:27:22 +00:00
solenberg@webrtc.org
67e70442b5 Remove unused and not working voe_extended_test. 
BUG=2913
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5533 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-12 09:58:49 +00:00
andrew@webrtc.org
b659e2844d Reduce mixing threshold in test to avoid flakiness. 
Flake observed here:
http://chromegw/i/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D/builds/953/steps/voe_auto_test/logs/stdio

TBR=andresp

Review URL: https://webrtc-codereview.appspot.com/8489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-11 21:52:50 +00:00
andrew@webrtc.org
16c08f03da Restore mixing integration tests. 
These high level tests were disabled over time. Since they depend on
real-time results and the filesystem, they tended to be flaky on the
bots. We now give it a very generous 1 second to start up all channels
before verification and a further relaxed file length check. If we
continue to see problems, I will up the startup delay.

The restored tests would have caught the AGC bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5454

Add a new "real audio" stress test to exercise more code paths. This
would have caught the refactor bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5437

BUG=2164,2844
TESTED=git try. Verified it would have caught the aforementioned bugs
by reintroducing them.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-10 23:04:39 +00:00
pbos@webrtc.org
3e6c41c48f Revert "Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents" 
This reverts commit r5479.

R=henrika@webrtc.org
BUG=2880

Review URL: https://webrtc-codereview.appspot.com/7989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 10:45:14 +00:00
pbos@webrtc.org
064b32acbb Fix locking in LoopBackTransport::StorePacket. 
The critical section in StorePacket was unnamed and only existed in
expression scope. Added GUARDED_BY annotations (which caught the bug),
then fixed it by naming the variable.

BUG=2880
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-04 09:42:02 +00:00
marpan@webrtc.org
f6b8f496ee Disable the test: DtmfTest.ManualSuccessfullySendsIn/OutOfBandTelephoneEvents 
Issue: https://code.google.com/p/webrtc/issues/detail?id=2880

R=andrew@webrtc.org
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-03 21:34:35 +00:00
henrikg@webrtc.org
c693704cc2 Move out typing detection to its own class. 
This will allow an embedder to use it directly.

Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)

R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-30 09:50:46 +00:00
andrew@webrtc.org
7de3bb9df9 Output logs to stderr from voe_cmd_test by default. 
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.

TESTED=logs output to stderr by default and to the usual file when the
flag is specified.

R=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6849005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 22:17:43 +00:00
aluebs@webrtc.org
4371d4650a Temporarily disabling some more audio processing tests. 
R=andrew@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 08:57:22 +00:00
aluebs@webrtc.org
0b72f5863b Add experimental noise suppression dummy API. 
Add this flag to the voe_cmd_test.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-19 15:17:51 +00:00
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