This is part of rewriting the ConferenceMixer and OutputMixer.
Calls are instead routed through AudioReceiveStream::Start/Stop.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2206223002
Cr-Commit-Position: refs/heads/master@{#13636}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.
BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org
Review URL: https://codereview.webrtc.org/2061193002 .
Cr-Commit-Position: refs/heads/master@{#13210}
Removes the need to use VoEVolume::SetChannelOutputVolumeScaling().
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2062193002
Cr-Commit-Position: refs/heads/master@{#13194}
CreatePeerConnectionFactory does not yet expose the ability to set the
factory from the outside.
Added notry due to android_dbg being broken.
NOTRY=True
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/1991233004
Cr-Commit-Position: refs/heads/master@{#13112}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
VoENetwork is kept for now, but is not really used anylonger.
webrtcvoiceengine is changed to have the same behavior for unsignaled
ssrc as video has, which is reflected by disabling one test case and
this will be discussed and followed up.
BUG=webrtc:5079
TBR=tommi
Review-Url: https://codereview.webrtc.org/1909333002
Cr-Commit-Position: refs/heads/master@{#12555}
Reason for revert:
Breaks GN in chromium.
Original issue's description:
> Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
>
> webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
> depending on voice engine, resulting in a cyclic dependency (which we
> don't detect since we have that check turned off, see webrtc:4243).
>
> BUG=webrtc:4243, webrtc:5589
> R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2
> Cr-Commit-Position: refs/heads/master@{#11766}
TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243, webrtc:5589
Review URL: https://codereview.webrtc.org/1739783002
Cr-Commit-Position: refs/heads/master@{#11769}
This allows other projects to more easily depend on this.
The plan is to move remote_bitrate_estimator and bitrate_controller into this module and reduce the exposed interface to only a simplified version of congestion_controller.h.
No functional changes in this CL.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1718473002 .
Cr-Commit-Position: refs/heads/master@{#11718}
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1702983002
Cr-Commit-Position: refs/heads/master@{#11658}
This adds negotiation of both transport sequence number and transport
feedback. Only offers transport seq num if the
WebRTC-Audio-SendSideBwe finch experiment is enabled.
TBR=mflodman@webrtc.org
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1604563002
Cr-Commit-Position: refs/heads/master@{#11487}
This makes it possible to handle send and receive streams with the same SSRC, which is currently the case in some peer connection tests.
Also moves sending transport feedback to the pacer thread.
BUG=webrtc:5263
Review URL: https://codereview.webrtc.org/1628683002
Cr-Commit-Position: refs/heads/master@{#11443}
Reason for revert:
tommi pointed out that using a refptr for the sink may cause issues. Will reland with a slightly different approach.
Original issue's description:
> Storing raw audio sink for default audio track.
>
> BUG=webrtc:5250
>
> Committed: https://crrev.com/e591f9377f33f3f725a30faecd1bef1a71fa6b99
> Cr-Commit-Position: refs/heads/master@{#11230}
TBR=pthatcher@webrtc.org,solenberg@webrtc.org,pbos@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5250
Review URL: https://codereview.webrtc.org/1588693002
Cr-Commit-Position: refs/heads/master@{#11241}
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
Simplify creation of VoE channels and Call streams in WVoMC.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1454073002
Cr-Commit-Position: refs/heads/master@{#10731}
