8db5854eb0
Fix potential flakiness in voe_auto_test.
...
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41929004
Cr-Commit-Position: refs/heads/master@{#8362}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8362 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 12:19:42 +00:00
63da1dd972
audio_processing: Now records mic volume level also when using new AGC
...
Previously only mic level calculated by the legacy agc was logged in aecdebug dumps.
Now we log it for any agc.
In addition, it is now possible to turn on and off debug recording in the test tool voe_cmd_test.
BUG=4274
TESTED=verified using voe_cmd_test
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39839004
Cr-Commit-Position: refs/heads/master@{#8274}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8274 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 19:44:46 +00:00
cc64a9cc4f
voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
...
As of r8230 (https://webrtc-codereview.appspot.com/39739004/ ) a new Echo Delay Metric was added calculating the fraction of poor values that may cause the AEC to fail. There are currently two methods for GetDelayMetrics() in webrtc::AutioProcessing and one is deprecated.
This CL updates
- GetEcDelayMetrics()
- voe_auto_test
- talk/media/(fake)webrtcvoiceengine
BUG=N/A
TESTED=locally and trybots
R=pbos@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41749004
Cr-Commit-Position: refs/heads/master@{#8251}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8251 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 12:53:24 +00:00
875c97ed9d
Remove SetNotAlive method from the thread class.
...
Also cleaning up methods with the same name in other classes that are derived from the above method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41759004
Cr-Commit-Position: refs/heads/master@{#8242}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8242 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 11:12:39 +00:00
a671f4b2cb
Fixing a VoE test to set correct rate for iSAC
...
The test was relying on that the code accepted an invalid rate.
Now the test passes a correct rate instead.
COAUTHOR=kwiberg@webrtc.org
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33179004
Cr-Commit-Position: refs/heads/master@{#8217}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8217 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 13:04:47 +00:00
664ccb7d8d
Reland r8125: Modify some tests to never use DTX disable mode
...
DTX disable mode will be removed as a part of the ACM redesign work.
This CL effectively reverts r8129, and relands r8125, but now using
assert instead of DCHECK.
COAUTHOR:kwiberg@webrtc.org
TBR=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37839004
Cr-Commit-Position: refs/heads/master@{#8185}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8185 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 14:49:12 +00:00
456f01441a
Re-allowing RED in voice engine.
...
Path of audio RED packets was blocked in r4692 by accident. It ought be enabled again.
BUG=3619
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8137 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 11:58:42 +00:00
ff108fe508
Revert 8125 "Modify some tests to never use DTX disable mode"
...
Broke compile on the Chromium FYI bots:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3483
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16028
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux/builds/14293
Error:
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
In file included from ../../third_party/webrtc/overrides/webrtc/base/logging.h:35:
../../base/logging.h:367:9:error: 'LOG' macro redefined [-Werror,-Wmacro-redefined]
#define LOG(severity) LAZY_STREAM(LOG_STREAM(severity), LOG_IS_ON(severity))
^
../../third_party/webrtc/system_wrappers/interface/logging.h:123:9: note: previous definition is here
#define LOG(sev) \
^
In file included from ../../third_party/webrtc/voice_engine/channel.cc:13:
In file included from ../../third_party/webrtc/base/checks.h:22:
../../third_party/webrtc/overrides/webrtc/base/logging.h:189:9:error: 'LOG_V' macro redefined [-Werror,-Wmacro-redefined]
#define LOG_V(sev) DIAGNOSTIC_LOG(sev, NONE, 0)
^
../../third_party/webrtc/system_wrappers/interface/logging.h:129:9: note: previous definition is here
#define LOG_V(sev) \
^
2 errors generated.
> Modify some tests to never use DTX disable mode
>
> DTX disable mode will be removed as a part of the ACM redesign work.
>
> COAUTHOR:kwiberg@webrtc.org
>
> R=henrika@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/34769004
TBR=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8129 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 19:02:03 +00:00
043db24767
Modify some tests to never use DTX disable mode
...
DTX disable mode will be removed as a part of the ACM redesign work.
COAUTHOR:kwiberg@webrtc.org
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8125 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 13:30:58 +00:00
4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
ece3890d3a
Report total bitrate for all streams in GetStats.
...
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.
R=stefan@webrtc.org , xians@webrtc.org
BUG=1667
Review URL: https://webrtc-codereview.appspot.com/27179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 11:52:04 +00:00
3cefbc99f4
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
...
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org , pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
4cebd84c79
Reland "Remove DTMF status methods from Voice Engine" r7276
...
This reverts r7277.
TBR=henrika@webrtc.org ,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-01 08:23:21 +00:00
3987f10c11
Revert "Remove DTMF status methods from Voice Engine" r7276
...
This change caused some trouble.
TBR=henrika@webrtc.org ,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 13:15:14 +00:00
bf7b9e0081
Remove DTMF status methods from Voice Engine
...
These methods are not used.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:54:04 +00:00
adee8f9242
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
...
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03
BUG=
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 12:28:06 +00:00
047a46f8b4
Remove Android.mk build files.
...
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.
R=andrew@webrtc.org , glaznev@webrtc.org , henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/15249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +00:00
b96ea2aab5
Remove former team members from OWNERS and WATCHLISTS
...
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@
BUG=
R=henrike@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 06:12:08 +00:00
4521e2d0bd
Adding online bitrate change to voe_cmd_test
...
This is to verify a way of changing the bitrate on-the-fly under current WebRTC implementation.
TEST=changing bit rate for different codecs. sound quality changed when bit rate was set successful. catched error when bit rate is invalid for a running codec.
BUG=
R=andrew@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 12:15:27 +00:00
6aac93bd9c
Adding SetOpusMaxBandwidth in VoE and ACM
...
This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.
TEST = added a test in voe_cmd_test and listened to the result
BUG=
R=henrika@google.com , henrika@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12 08:13:33 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
3106b706c0
Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..."
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> 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
9de3d844ae
Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
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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
7f3a041d23
Removed NetworkTest.CanSwitchToExternalTransport since it tests an unsupported case and we should not maintain such a test.
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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
26e2b687fc
Remove ACM1/ACM2 switching from VoiceEngine tests
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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
66803489f9
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
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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
0f7375504a
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
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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
f927fd6481
Re-enable AGC tests:
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* 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
ca539bbed0
iOS: baby steps to being able to include_tests=1
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- 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
b9309beea4
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
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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
19018ddb17
Make ACM2 the default in voe_cmd_test.
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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
a789f3720a
VoiceEngine(iOS & Android): removed NOT_SUPPORTED
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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
440fa23553
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
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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
b1f5010075
VoE changes to allow forwarding of packets from VoE to ViE BWE.
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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
