2 Commits

Author SHA1 Message Date
deadbeef
20cb0c1c85 Move DTMF sender to RtpSender (as opposed to WebRtcSession).
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PeerConnection, but that's been replaced by a "dtmf" attribute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}
2017-02-02 04:27:00 +00:00
ossu
7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings.
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}
2017-01-23 12:56:25 +00:00