The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Original-Commit-Position: refs/heads/master@{#15906}
Committed: 67b3bbe639
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15973}
Reason for revert:
Hitting DCHECK in chromium's WebrtcTransportTest.TerminateDataChannel and WebrtcTransportTest.DataStreamLate. Will investigate and reland.
Original issue's description:
> Separating SCTP code from BaseChannel/MediaChannel.
>
> The BaseChannel code is geared around RTP; the presence of media engines,
> send and receive streams, SRTP, SDP directional attribute negotiation, etc.
> It doesn't make sense to use it for SCTP as well. This separation should make
> future work both on BaseChannel and the SCTP code paths easier.
>
> SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
> directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
> doesn't get confused with webrtc::DataChannel any more.
>
> Beyond just moving code around, some consequences of this CL:
> - We'll now stop using the worker thread for SCTP. Packets will be
> processed right on the network thread instead.
> - The SDP directional attribute is ignored, as it's supposed to be.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2564333002
> Cr-Commit-Position: refs/heads/master@{#15906}
> Committed: 67b3bbe639TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2614813003
Cr-Commit-Position: refs/heads/master@{#15908}
The BaseChannel code is geared around RTP; the presence of media engines,
send and receive streams, SRTP, SDP directional attribute negotiation, etc.
It doesn't make sense to use it for SCTP as well. This separation should make
future work both on BaseChannel and the SCTP code paths easier.
SctpDataEngine now becomes SctpTransport, and is used by WebRtcSession
directly. cricket::DataChannel is also renamed, to RtpDataChannel, so it
doesn't get confused with webrtc::DataChannel any more.
Beyond just moving code around, some consequences of this CL:
- We'll now stop using the worker thread for SCTP. Packets will be
processed right on the network thread instead.
- The SDP directional attribute is ignored, as it's supposed to be.
BUG=None
Review-Url: https://codereview.webrtc.org/2564333002
Cr-Commit-Position: refs/heads/master@{#15906}
(This is a re-upload of https://codereview.webrtc.org/2567243003/, the
CQ stopped working there.)
The previously used WebRtcSession::GetTransportStats did a synchronous
invoke per channel (voice, video, data) on the signaling thread to the
network thread - e.g. 3 blocking invokes.
It is replaced by WebRtcSession::GetStats[_s] which can be invoked on
the signaling thread or on any thread if a ChannelNamePairs argument is
present (provided by WebRtcSession::GetChannelNamePairs on the signaling
thread).
With these changes, and changes allowing the getting of certificates
from any thread, the RTCStatsCollector can turn the 3 blocking thread
invokes into 1 non-blocking invoke.
BUG=webrtc:6875, chromium:627816
Review-Url: https://codereview.webrtc.org/2583883002
Cr-Commit-Position: refs/heads/master@{#15672}
New files, classes moved from statscollector_unittest.cc:
+webrtc/api/test/mock_peerconnection.h
for MockPeerConnectionFactory and MockPeerConnection
+webrtc/api/test/mock_webrtcsession.h
for MockWebRtcSession
+webrtc/media/base/test/mock_mediachannel.h
for MockVideoMediaChannel and MockVoiceMediaChannel
The webrtc/media/base/test folder is new.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2238933002
Cr-Commit-Position: refs/heads/master@{#13769}
