Log - b7e5b2741f9cbd68e7be54e3445d28d266477c92 - fp2-dev/platform/external/chromium_org/third_party/webrtc

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 · 10 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 · 10 years ago
22f69bd Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 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 · 10 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 · 10 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 · 10 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 · 10 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 · 10 years ago
e639a03 Removes parts of the webrtc::VoEHardware sub API (relanding) by henrika@webrtc.org · 10 years ago
b8db407 Revert 6090 "Removes parts of the webrtc::VoEHardwareMedia sub A..." by henrika@webrtc.org · 10 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 · 10 years ago
ba47616 Replace scoped_array<T> with scoped_ptr<T[]>. by andrew@webrtc.org · 10 years ago
47e54ba * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. by wu@webrtc.org · 10 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 · 10 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 · 10 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 · 10 years ago
2e4c621 (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds by henrika@webrtc.org · 10 years ago
fec6b6e VoE changes to allow forwarding of packets from VoE to ViE BWE. by solenberg@webrtc.org · 10 years ago
9a82322 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. by wu@webrtc.org · 10 years ago
4845ee0 Removes VoERTP_RTCP::InsertExtraRTPPacket. by henrika@webrtc.org · 10 years ago
a56c5b4 Remove external encryption API for VoE. by solenberg@webrtc.org · 10 years ago
87c8b86 Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base by xians@webrtc.org · 10 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 · 10 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 · 11 years ago
a32d18f Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
2753b76 Add dummy audio NACK APIs by niklas.enbom@webrtc.org · 11 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 · 11 years ago
471ae72 Include files from webrtc/.. paths in voice_engine/ by pbos@webrtc.org · 11 years ago
8510750 Make sure VoiceEngine tests only include one test framework. by pbos@webrtc.org · 11 years ago
ca7a9a2 Remove const for plain data types in voice_engine/ by pbos@webrtc.org · 11 years ago
f272497 Adding playout buffer status to the voe video sync by pwestin@webrtc.org · 11 years ago
54f03bc WebRtc_Word32 -> int32_t in voice_engine/ by pbos@webrtc.org · 11 years ago
e493218 Remove UDP transport API from VoE by pwestin@webrtc.org · 11 years ago
3b6f728 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
fa2dd22 Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested. by solenberg@webrtc.org · 11 years ago
2ffc8bf Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." by wu@webrtc.org · 11 years ago
365ca40 Removed CPU APIs from VoEHardware. Code is now only used by test applications. by henrike@webrtc.org · 11 years ago
31b4448 Alphabetize include order in fake_voe_external_media.h. by andrew@webrtc.org · 11 years ago
13f66d1 Add some VoE and AudioProcessing mocks. by andrew@webrtc.org · 11 years ago
912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
15a03fd Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
8665399 None of the clients of VoE use SetNetEQBGNMode(), furthermore, NetEq 4 does not provide an API to change the mode of the background noise. by turaj@webrtc.org · 11 years ago
b79627b Expose the capture-side AudioProcessing object and allow it to be injected. by andrew@webrtc.org · 11 years ago
b9e5a3d Make VoiceEngineImpl inherit from VoiceEngine. by tommi@webrtc.org · 11 years ago
ead8a5b Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM. by turaj@webrtc.org · 11 years ago
b9e3afc Add GetAudioFrame API to VoiceEngine. by roosa@google.com · 12 years ago
ca77149 Add API to retreive last received RTP timestamp to VoiceEngine. by roosa@google.com · 12 years ago
7db5290 VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
15e35cc Expose Set and Get Recording/Playout sample rate apis by leozwang@webrtc.org · 12 years ago
c65ae4b Revert 3231 - VoE Changes to enable dual_streaming. by perkj@webrtc.org · 12 years ago
d6f028b VoE Changes to enable dual_streaming. by turaj@webrtc.org · 12 years ago
b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago