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webrtc
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src
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webrtc
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d71b0f63a151c89195b39e1e79afa74509344d01
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voice_engine
/
include
b7b8932
External APM usage downstream dependency support cleanup
by peah
· 8 years ago
76de83e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
bc32410
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
60154fd
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
588f761
Allow an external audio processing module to be used in WebRTC
by peah
· 8 years ago
f4509f8
Removing unnecessary parameters from initializeAndroidGlobals.
by deadbeef
· 8 years ago
191c2da
Remove VoEHardware interface.
by solenberg
· 8 years ago
f5fbd05
Remove VoENetEqStats interface.
by solenberg
· 8 years ago
788c1f7
Remove VoEAudioProcessing interface.
by solenberg
· 8 years ago
07f189f
Remove VoEVolumeControl interface.
by solenberg
· 8 years ago
2633f5d
Remove saturation warning support from TransmitMixer.
by tommi
· 8 years ago
71e8df9
Remove usage of VoEAudioProcessing from WVoE/MC.
by solenberg
· 8 years ago
ac16f1b
Remove VoEVideoSync interface.
by solenberg
· 8 years ago
bda935c
Remove VoEExternalMedia interface.
by solenberg
· 8 years ago
7eb7de8
Remove the unused and untested functions from VoERTP_RTCP.
by solenberg
· 8 years ago
f7c7480
Move AudioDecoder and related stuff to the api/ directory
by kwiberg
· 8 years ago
017ebe5
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
by aleloi
· 8 years ago
e9523f0
Clean up abs-send-time for audio.
by stefan
· 8 years ago
72aebf4
Remove voe::Channel::StopReceive() and associated logic.
by solenberg
· 8 years ago
d208d89
The VoE functionality to apply receive-side processing to VoE channels is unused. I'm removing it so we can avoid instantiating a full APM per channel (and thus also for webrtc::AudioSendStream and webrtc::AudioReceiveStream), and then never use it.
by solenberg
· 8 years ago
a686d5e
Moving/renaming webrtc/common.h.
by solenberg
· 8 years ago
8111a41
VoERTP_RTCP: Remove GetREDStatus and SetREDStatus
by kwiberg
· 8 years ago
bbc45b5
Regression test for issue where Opus DTX status was being forgotten.
by ivoc
· 8 years ago
23ea12e
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
by ivoc
· 9 years ago
822f09e
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
by ivoc
· 9 years ago
1e2f1e5
Move RtcEventLog object from inside VoiceEngine to Call.
by Ivo Creusen
· 9 years ago
1ac1c3f
Moved CreateBuiltinDecoderFactory out to VoEBaseImpl.
by ossu
· 9 years ago
d25cdbb
- Add temporary VoEBase::audio_device_module() method.
by solenberg
· 9 years ago
cf2fc20
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
by solenberg
· 9 years ago
9a75859
Remove unused method OutputMixer::PlayDtmfTone() and infrastructure.
by solenberg
· 9 years ago
39673bb
Remove the VoEDtmf interface.
by solenberg
· 9 years ago
3741297
Removing some unnecessary string manipulation code from VoEBase::GetVersion().
by solenberg
· 9 years ago
4f247a6
Changed FakeVoiceEngine into a MockVoiceEngine.
by Fredrik Solenberg
· 9 years ago
a90d7f5
Delete a chain of methods in ViE, VoE and ACM
by henrik.lundin
· 9 years ago
d5bdda3
Unify Transport and newapi::Transport interfaces.
by pbos
· 9 years ago
e98aa80
Improving support for Android Audio Effects in WebRTC.
by henrika
· 9 years ago
35fd753
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
by ivoc
· 9 years ago
a0ad248
Update a ton of audio code to use size_t more correctly and in general reduce
by Peter Kasting
· 9 years ago
5361f04
VoE: Remove unused interfaces
by Jelena Marusic
· 10 years ago
2450d62
Remove rtp_rtcp/ dump functionality.
by Peter Boström
· 10 years ago
1103af5
Propagating RTT from send-only channel to receive-only channel.
by Minyue
· 10 years ago
26b26b3
Remove VideoEngine interfaces.
by Peter Boström
· 10 years ago
9e50d7e
VoE: cleanup VoENetwork implementation
by Jelena Marusic
· 10 years ago
0705a02
VoE: apply new style guide on VoE interfaces and their implementations
by Jelena Marusic
· 10 years ago
f0b18c6
Added SetBitRate function to VoE API to allow changing the audio bitrate.
by Ivo Creusen
· 10 years ago
7ac069e
VoE: VoEBase unit test
by Jelena Marusic
· 10 years ago
c49056d
VoE: move mock directory 1 level up
by Jelena Marusic
· 10 years ago
0487759
Supporting Opus DTX in Voice Engine.
by minyue@webrtc.org
· 10 years ago
61b3feb
Adds C++/JNI/Java unit test for audio device module on Android.
by henrika@webrtc.org
· 10 years ago
deb9dae
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
by kwiberg@webrtc.org
· 10 years ago
49da516
voice_engine: Updates GetEcDelayMetrics() w.r.t. new metric
by bjornv@webrtc.org
· 10 years ago
c1aed49
Remove dual stream functionality in VoiceEngine
by henrik.lundin@webrtc.org
· 10 years ago
a3166c4
Use int64_t more consistently for times, in particular for RTT values.
by pkasting@chromium.org
· 10 years ago
db5c754
AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer
by henrika@webrtc.org
· 10 years ago
0ab923a
Use size_t more consistently for packet/payload lengths.
by pkasting@chromium.org
· 10 years ago
9735284
Report total bitrate for all streams in GetStats.
by pbos@webrtc.org
· 10 years ago
c10bfea
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
by xians@webrtc.org
· 10 years ago
d29f51c
Reland "Remove DTMF status methods from Voice Engine" r7276
by henrik.lundin@webrtc.org
· 10 years ago
21cb262
Revert "Remove DTMF status methods from Voice Engine" r7276
by henrik.lundin@webrtc.org
· 10 years ago
f70e632
Remove DTMF status methods from Voice Engine
by henrik.lundin@webrtc.org
· 10 years ago
c747067
Remove Get/SetNetEQPlayoutMode APIs
by henrik.lundin@webrtc.org
· 10 years ago
b0aac71
Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate
by minyue@webrtc.org
· 10 years ago
1bfd540
Adding SetOpusMaxBandwidth in VoE and ACM
by minyue@webrtc.org
· 10 years ago
09da1a7
Remove the send-side cname getter APIs from voice and video engine.
by stefan@webrtc.org
· 11 years ago
dd671de
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.
by minyue@webrtc.org
· 11 years ago
22f69bd
Add interface to propagate audio capture timestamp to the renderer.
by wu@webrtc.org
· 11 years ago
c4e54b6
Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
7b2651a
Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
7f5e297
Removes parts of the webrtc::VoEFile sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
d2632a0
Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
12884ba
Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
e639a03
Removes parts of the webrtc::VoEHardware sub API (relanding)
by henrika@webrtc.org
· 11 years ago
b8db407
Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..."
by henrika@webrtc.org
· 11 years ago
a4943ea
Removes parts of the webrtc::VoEHardwareMedia sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
ba47616
Replace scoped_array<T> with scoped_ptr<T[]>.
by andrew@webrtc.org
· 11 years ago
47e54ba
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
by wu@webrtc.org
· 11 years ago
692224a
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
66ccaff
Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
4ff0eda
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
by henrika@webrtc.org
· 11 years ago
2e4c621
(landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds
by henrika@webrtc.org
· 11 years ago
fec6b6e
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 11 years ago
9a82322
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
by wu@webrtc.org
· 11 years ago
4845ee0
Removes VoERTP_RTCP::InsertExtraRTPPacket.
by henrika@webrtc.org
· 11 years ago
a56c5b4
Remove external encryption API for VoE.
by solenberg@webrtc.org
· 11 years ago
87c8b86
Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
by xians@webrtc.org
· 11 years ago
942ba53
Added new capture callback interface to pass the capture callback to a specific voe channel from libjingle webrtcvoiceengine.cc.
by xians@webrtc.org
· 11 years ago
79d6daf
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
by wu@webrtc.org
· 11 years ago
7b72264
Allow opening an AEC dump from an existing file handle.
by henrikg@webrtc.org
· 11 years ago
b43ac9f
Inject config when creating channels to override the existing one.
by turaj@webrtc.org
· 11 years ago
39e22a1
Adds a new voice engine warning for the typing noise off state.
by jiayl@webrtc.org
· 11 years ago
4489c51
This issue is related to https://chromereviews.googleplex.com/9908014/
by minyue@webrtc.org
· 11 years ago
7fc75bb
Update talk to 50918584.
by wu@webrtc.org
· 11 years ago
b3ada15
Ref-counted rewrite of ChannelManager.
by pbos@webrtc.org
· 11 years ago
0ba496b
Revert r4301
by tnakamura@webrtc.org
· 12 years ago
a32d18f
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
by stefan@webrtc.org
· 12 years ago
3b89e10
Proper spacing for end-of-namespace comments.
by pbos@webrtc.org
· 12 years ago
2753b76
Add dummy audio NACK APIs
by niklas.enbom@webrtc.org
· 12 years ago
d557734
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
by turaj@webrtc.org
· 12 years ago
471ae72
Include files from webrtc/.. paths in voice_engine/
by pbos@webrtc.org
· 12 years ago
8510750
Make sure VoiceEngine tests only include one test framework.
by pbos@webrtc.org
· 12 years ago
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