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193 Commits

Author SHA1 Message Date
Stefan Holmer
e590416722 Moving the pacer and the pacer thread to ChannelGroup. 
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45549004

Cr-Commit-Position: refs/heads/master@{#8864}
 2015-03-26 10:11:22 +00:00
tommi@webrtc.org
38492c5b6f Re-land 8810 "- Add a SetPriority method to ThreadWr..." 
> Revert 8810 "- Add a SetPriority method to ThreadWrapper"
> Seeing if this is causing roll issues.
> 
> > - Add a SetPriority method to ThreadWrapper
> > - Remove 'priority' from CreateThread and related member variables from implementations
> > - Make supplying a name for threads, non-optional
> > 
> > BUG=
> > R=magjed@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/44729004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/48609004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50459005

Cr-Commit-Position: refs/heads/master@{#8819}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:42:46 +00:00
tommi@webrtc.org
90a1cb4630 Revert 8810 "- Add a SetPriority method to ThreadWrapper" 
Seeing if this is causing roll issues.

> - Add a SetPriority method to ThreadWrapper
> - Remove 'priority' from CreateThread and related member variables from implementations
> - Make supplying a name for threads, non-optional
> 
> BUG=
> R=magjed@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/44729004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48609004

Cr-Commit-Position: refs/heads/master@{#8818}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-22 14:34:46 +00:00
tommi@webrtc.org
b6817d793f - Add a SetPriority method to ThreadWrapper 
- Remove 'priority' from CreateThread and related member variables from implementations
- Make supplying a name for threads, non-optional

BUG=
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44729004

Cr-Commit-Position: refs/heads/master@{#8810}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 15:52:43 +00:00
tommi@webrtc.org
361981faa8 Use scoped_ptr for ThreadWrapper::CreateThread. 
BUG=
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45799004

Cr-Commit-Position: refs/heads/master@{#8794}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-19 14:45:42 +00:00
minyue@webrtc.org
9b2e1144df Supporting Opus DTX in Voice Engine. 
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.

BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43709004

Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 09:38:55 +00:00
pbos@webrtc.org
86639737b8 Remove thread id from ThreadWrapper::Start(). 
Removes ThreadPosix::InitParams and a corresponding wait for an event.
This unblocks ThreadPosix::Start which had to wait for thread scheduling
for an event to trigger on the spawned thread, giving faster Start()
calls.

BUG=4413
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43699004

Cr-Commit-Position: refs/heads/master@{#8709}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 00:07:45 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 13:04:54 +00:00
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +00:00
pkasting@chromium.org
d324546ced Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : 
* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36179004

Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-23 21:29:45 +00:00
minyue@webrtc.org
c0bd7be0df Adding two new stats to VoiceReceiverInfo 
There have been requests of two new stats namely

speech_expand_rate and secondary_decoded_rate.

BUG=3867
R=henrik.lundin@webrtc.org, henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40789004

Cr-Commit-Position: refs/heads/master@{#8415}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-18 15:24:39 +00:00
pkasting@chromium.org
e9facf8bb3 Add range checks in a variety of places where the values will subsequently be 
expected to be 0-127.

BUG=none
TEST=none
R=juberti@webrtc.org
TBR=henrika

Review URL: https://webrtc-codereview.appspot.com/37759004

Cr-Commit-Position: refs/heads/master@{#8399}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8399 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 20:37:35 +00:00
pbos@webrtc.org
d5ce2e63df Remove EventWrapper::Reset(). 
This simplifies the event wrapper which we've recently found issues in.
Also refactoring EndToEndTest.RespectsNetworkState to not depend on it.

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

Review URL: https://webrtc-codereview.appspot.com/41939004

Cr-Commit-Position: refs/heads/master@{#8366}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-13 14:58:38 +00:00
solenberg@webrtc.org
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
bjornv@webrtc.org
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
bjornv@webrtc.org
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
tommi@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
minyue@webrtc.org
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
kjellander@webrtc.org
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
henrik.lundin@webrtc.org
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
pkasting@chromium.org
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
pbos@webrtc.org
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
xians@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
henrik.lundin@webrtc.org
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
minyue@webrtc.org
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
pbos@webrtc.org
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
kjellander@webrtc.org
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
minyue@webrtc.org
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
minyue@webrtc.org
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
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