ff97631e3c
- Add temporary VoEBase::audio_device_module() method.
...
- Remove WVoE::SetAudioDeviceModule() - the ADM is now supplied in ctor.
- Remove WVoE::Init() and WVoE::Terminate().
- Remove MediaEngineInterface::Terminate().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1830213002
Cr-Commit-Position: refs/heads/master@{#12173}
2016-03-31 06:28:56 +00:00
6021fe2b1e
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
...
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1803923003
Cr-Commit-Position: refs/heads/master@{#12003}
2016-03-15 18:41:58 +00:00
e50872be13
Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.
...
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1796183002
Cr-Commit-Position: refs/heads/master@{#11990}
2016-03-14 22:32:53 +00:00
622d8950f5
Remove the VoEDtmf interface.
...
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1723153002
Cr-Commit-Position: refs/heads/master@{#11906}
2016-03-08 12:11:00 +00:00
2515af28e9
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
...
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1493663002
Cr-Commit-Position: refs/heads/master@{#10868}
2015-12-02 14:19:44 +00:00
0ccae13556
Changed FakeVoiceEngine into a MockVoiceEngine.
...
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1402403008 .
Cr-Commit-Position: refs/heads/master@{#10491}
2015-11-03 09:15:59 +00:00
74f0f3551e
Delete a chain of methods in ViE, VoE and ACM
...
The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):
ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay
The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.
This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1421013006
Cr-Commit-Position: refs/heads/master@{#10471}
2015-11-01 19:43:38 +00:00
2d566686a2
Unify Transport and newapi::Transport interfaces.
...
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1369263002
Cr-Commit-Position: refs/heads/master@{#10096}
2015-09-28 16:59:36 +00:00
c14f5ff60f
Improving support for Android Audio Effects in WebRTC.
...
Now also supports AGC and NS effects and adds the possibility
to override default settings.
R=magjed@webrtc.org , pbos@webrtc.org , sophiechang@chromium.org
TBR=perkj
BUG=NONE
Review URL: https://codereview.webrtc.org/1344563002 .
Cr-Commit-Position: refs/heads/master@{#10030}
2015-09-23 12:09:40 +00:00
b04965ccf8
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
...
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
2015-09-09 07:09:49 +00:00
dce40cf804
Update a ton of audio code to use size_t more correctly and in general reduce
...
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org , pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 21:52:45 +00:00
f09e09c7ee
VoE: Remove unused interfaces
...
BUG=4690
I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53519004
Cr-Commit-Position: refs/heads/master@{#9281}
2015-05-26 08:25:00 +00:00
c3f4dbc40b
Remove rtp_rtcp/ dump functionality.
...
Removes RTP dumping from VideoEngine and VoiceEngine.
BUG=1695
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47179004
Cr-Commit-Position: refs/heads/master@{#9234}
2015-05-20 12:10:56 +00:00
2013aeced2
Propagating RTT from send-only channel to receive-only channel.
...
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.
BUG=3978
TEST=chromium with hangout calls
R=henrika@webrtc.org , mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29989004
Cr-Commit-Position: refs/heads/master@{#9186}
2015-05-13 12:14:36 +00:00
300eeb68f5
Remove VideoEngine interfaces.
...
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.
Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).
BUG=1695, 4491
R=mflodman@webrtc.org , solenberg@webrtc.org
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/55379004
Cr-Commit-Position: refs/heads/master@{#9179}
2015-05-12 14:51:08 +00:00
f353dd59b5
VoE: cleanup VoENetwork implementation
...
Changes:
1. Documented return values of VoENetwork methods.
2. In VoENetworkImpl: replaced calls to SetLastError() with LOG_F(). SetLastError() is not used anymore because no one is calling LastError() to check for last error. Also, its usage is being removed in Video Engine and we want to be consistent.
3. In VoENetworkImpl: removed WEBRTC_TRACE() usage.
4. In VoENetworkImpl: replaced some defensive code with assert(). We are now assuming that the caller has called VoEBase::Init() where needed. We are also assuming that it is invalid to pass nullptr where data is expected.
5. Updated unit tests accordingly.
R=henrika@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53369004
Cr-Commit-Position: refs/heads/master@{#9145}
2015-05-06 13:04:29 +00:00
0d266054ac
VoE: apply new style guide on VoE interfaces and their implementations
...
Changes:
1. Ran clang-format on VoE interfaces and their implementations.
2. Replaced virtual with override in derived classes.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49239004
Cr-Commit-Position: refs/heads/master@{#9130}
2015-05-04 12:15:41 +00:00
adf89b7e33
Added SetBitRate function to VoE API to allow changing the audio bitrate.
...
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.
BUG=
R=henrik.lundin@webrtc.org , henrika@webrtc.org , kwiberg@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50789004
Cr-Commit-Position: refs/heads/master@{#9115}
2015-04-29 14:03:45 +00:00
06b08afb67
VoE: VoEBase unit test
...
Changes:
1. Documented return values of VoEBase methods.
2. Added/modified VoEBase unit test.
R=henrika@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51669004
Cr-Commit-Position: refs/heads/master@{#9038}
2015-04-21 09:39:41 +00:00
c317ce5456
VoE: move mock directory 1 level up
...
Changes:
1. Moved directory voice_engine/include/mock to voice_engine/mock (current recommendation).
2. Updated includes where necessary.
Caution:
We need confirmation that these mocks are indeed used only locally.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48089004
Cr-Commit-Position: refs/heads/master@{#9005}
2015-04-15 10:45:09 +00:00
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
474d1eb223
Adds C++/JNI/Java unit test for audio device module on Android.
...
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.
It also:
- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.
BUG=NONE
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40069004
Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 12:40:43 +00:00
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
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
8315d7de85
Remove dual stream functionality in VoiceEngine
...
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.
BUG=3520
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:07:26 +00:00
16825b1a82
Use int64_t more consistently for times, in particular for RTT values.
...
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , holmer@google.com , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
a954c07ee1
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
...
BUG=4034
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-09 16:22:09 +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
64a2f10f4b
Remove Get/SetNetEQPlayoutMode APIs
...
These are not used anymore.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-22 14:30:10 +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
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
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +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
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
...
BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
9f277350f8
Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.
...
BUG=3206
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12299005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6146 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:04:29 +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
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.
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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)
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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.
...
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
8f69330310
Replace scoped_array<T> with scoped_ptr<T[]>.
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scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar ...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
93fd25c20c
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
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* Cast rtp header extension to int in log in rtp_utility.cc.
BUG=3237
TEST=try bots
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 20:33:08 +00:00
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
