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gerrit-public.fairphone.software
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fp2-dev
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platform
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external
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chromium_org
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third_party
/
webrtc
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68a95e15db40860be332f453b6e3ae6b44a327ca
/
common_types.h
514abde
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
bd0a216
Remove usage of webrtc trace in video processing modules.
by asapersson@webrtc.org
· 10 years ago
f50914a
Removing VideoCodecDerived and moving methods inside VideoCodec.
by mallinath@webrtc.org
· 10 years ago
5ca38d1
Add targetBitrate to VideoCodec struct.
by pbos@webrtc.org
· 10 years ago
5f804f8
VoE changes to allow forwarding of packets from VoE to ViE BWE.
by solenberg@webrtc.org
· 10 years ago
40fee00
Adding operator== and != methods for CodecInst and VideoCodec structures.
by mallinath@webrtc.org
· 10 years ago
f39df52
Remove internal codecs from VideoSendStream.
by pbos@webrtc.org
· 10 years ago
2fa9f7e
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
by asapersson@webrtc.org
· 10 years ago
ae50521
Remove external encryption API for VoE.
by solenberg@webrtc.org
· 10 years ago
fa7c4c4
Wire up statistics in video receive stream of new API
by sprang@webrtc.org
· 10 years ago
ef3b24e
Remove empty VideoCodecGeneric struct.
by pbos@webrtc.org
· 11 years ago
ca72300
Wire up statistics in video send stream of new video engine api
by sprang@webrtc.org
· 11 years ago
39139dc
Revert r5294 to re-roll r5293.
by pbos@webrtc.org
· 11 years ago
0af1d21
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
by turaj@webrtc.org
· 11 years ago
ee867fa
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
by solenberg@webrtc.org
· 11 years ago
7f0519e
Update talk to 58174641 together with http://review.webrtc.org/4319005/.
by wu@webrtc.org
· 11 years ago
0e4512b
Callback for send bitrate estimates - new roll
by sprang@webrtc.org
· 11 years ago
3a4fc4b
Update talk to 58127566 together with
by wu@webrtc.org
· 11 years ago
9b3d2bf
Revert 5274 "Update talk to 58113193 together with https://webrt..."
by wu@webrtc.org
· 11 years ago
cde78d6
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
by wu@webrtc.org
· 11 years ago
66e84b0
Revert 5259 "Callback for send bitrate estimates"
by sprang@webrtc.org
· 11 years ago
f1d22d4
Callback for send bitrate estimates
by sprang@webrtc.org
· 11 years ago
cf5c552
Add callbacks for send channel rtp statistics
by sprang@webrtc.org
· 11 years ago
8911937
Add callbacks for send channel rtcp statistics
by sprang@webrtc.org
· 11 years ago
9435a17
Add send frame rate statistics callback
by sprang@webrtc.org
· 11 years ago
e92aec9
Make interface destructor virtual
by sprang@webrtc.org
· 11 years ago
13a4d31
Interface changes to old api, for use by new api transition.
by sprang@webrtc.org
· 11 years ago
4043e7e
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
by sprang@webrtc.org
· 11 years ago
81c4d24
Remove redundant STR_CASE_CMP macro definitions.
by andrew@webrtc.org
· 11 years ago
530f40f
webrtc/common_types.h: Document bitrate fields' units.
by fischman@webrtc.org
· 11 years ago
1e8424f
Replace ExtraCodecOptions with new Config class that supports multiple settings at once.
by andresp@webrtc.org
· 11 years ago
c22830f
Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..."
by pbos@webrtc.org
· 11 years ago
2d6a699
Revert 3933 "Remove traces of deprecated WebRtc_Word types."
by pbos@webrtc.org
· 11 years ago
bbb54b3
Remove traces of deprecated WebRtc_Word types.
by pbos@webrtc.org
· 11 years ago
1efb11c
Adding extra options to interact with external encoder/decoder.
by andresp@webrtc.org
· 11 years ago
1ca9d42
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
90fa4a1
Add min and target bitrate to VideoCodec.
by marpan@webrtc.org
· 11 years ago
a2e9124
Generic video-codec support.
by pbos@webrtc.org
· 11 years ago
d4caede
Remove DTMF detection. Talk team has been in the loop and there is no need for
by turaj@webrtc.org
· 11 years ago
df1cfd1
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
85e2e0e
Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.
by stefan@webrtc.org
· 11 years ago
8d759af
VP8: Making key frame interval a tunnable parameter
by mikhal@webrtc.org
· 11 years ago
d0a9608
Add number of inserted samples to NetEq statistics.
by roosa@google.com
· 12 years ago
e68106f
Expose NetEq playout mode off through VoiceEngine.
by roosa@google.com
· 12 years ago
10e0b28
Add a kTraceTerseInfo level for non-verbose logging.
by andrew@webrtc.org
· 12 years ago
4c8c307
Clean up TraceCallback::Print.
by andrew@webrtc.org
· 12 years ago
75c7043
Add libjingle-style stream-style logging.
by andrew@webrtc.org
· 12 years ago
a7b57da
Move src/ -> webrtc/
by andrew@webrtc.org
· 12 years ago