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}
Extract the remote addresses from SDP c= line on both session level and
media level. The media level address will overwrite the session level one if
exists.
WebRTC is not using c= and this is used for new SDP parsing API.
BUG=webrtc:7311
Review-Url: https://codereview.webrtc.org/2742903002
Cr-Commit-Position: refs/heads/master@{#17326}
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}
This helps show where ownership is transfered between objects.
Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.
BUG=None
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
This is the naming scheme we've been using for internal interfaces.
Also, this CL will introduce a PacketTransportInterface in the webrtc namespace,
which would get too easily confused with the rtc:: one:
https://codereview.webrtc.org/2675173003/
BUG=None
Review-Url: https://codereview.webrtc.org/2679103006
Cr-Commit-Position: refs/heads/master@{#16539}
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PeerConnection, but that's been replaced by a "dtmf" attribute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}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
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}
