This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
Reason for revert:
Broken downstream projects
Original issue's description:
> Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls.
>
> Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
>
> BUG=webrtc:5079
> R=deadbeef@webrtc.org, hbos@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2863123002 .
> Cr-Commit-Position: refs/heads/master@{#18384}
> Committed: e80f4c91d0TBR=hbos@webrtc.org,deadbeef@webrtc.org,holmer@google.com,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2916793003
Cr-Commit-Position: refs/heads/master@{#18386}
Prior to this CL Call::Stats were collected via WebRtcVideoEngine2, but not via WebRtcVoiceEngine, causing these stats to be missing for audio-only calls. Call lives on the peerconnection/session level and should only be collected once independent on how many streams we have.
BUG=webrtc:5079
R=deadbeef@webrtc.org, hbos@webrtc.org
Review-Url: https://codereview.webrtc.org/2863123002 .
Cr-Commit-Position: refs/heads/master@{#18384}
This will later allow calling the "PeerConnectionObserver::OnIceCandidate"
method asynchronously while keeping the object alive.
BUG=webrtc:3721
Review-Url: https://codereview.webrtc.org/2748253003
Cr-Commit-Position: refs/heads/master@{#17380}
for consistency with the WebRTC 1.0 standard as suggested in a TODO.
BUG=None
Review-Url: https://codereview.webrtc.org/2732663004
Cr-Commit-Position: refs/heads/master@{#17077}
Where "TRANSPORT attributes" refers to:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-16
The BUNDLE draft now says that these attributes can
(in fact, MUST) be omitted when m= sections are bundled
(they only need to go in one of the bundled m= sections),
so we should start accepting that SDP.
This CL doesn't fix "a=rtcp-mux", unfortunately. That will be easier
to fix once we've split apart an "RtpTransport" object from
BaseChannel.
BUG=webrtc:6351
Review-Url: https://codereview.webrtc.org/2647593003
Cr-Commit-Position: refs/heads/master@{#16782}
"bundle_transport_name" is no longer relevant here, and
"rtcp_mux_required" is implied by whether or not an RTCP transport is
passed in.
BUG=None
Review-Url: https://codereview.webrtc.org/2689503002
Cr-Commit-Position: refs/heads/master@{#16551}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Previously in the spec, there was a createDtmfSender method on
PeerConnection, but that's been replaced by a "dtmf" attribute
on RtpSender, which allows getting a DTMF sender without having
an audio track.
This also simplifies the code slightly, since tracks are now not
necessary for identification.
BUG=webrtc:4180
Review-Url: https://codereview.webrtc.org/2666853002
Cr-Commit-Position: refs/heads/master@{#16409}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
