This is a partial re-land. The change doesn't make the default Thread ctor protected anymore but it does mark it as deprecated and updates all use of it in WebRTC.
Original issue's description:
Make the default ctor of rtc::Thread, protected.
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2977953002
Cr-Commit-Position: refs/heads/master@{#19031}
Reason for revert:
Break projects.
Original issue's description:
> Make the default ctor of rtc::Thread, protected.
> The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
>
> The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2981623002
> Cr-Commit-Position: refs/heads/master@{#19001}
> Committed: a8a3515997TBR=kthelgason@webrtc.org,tommi@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/2979963002
Cr-Commit-Position: refs/heads/master@{#19003}
The goal is to force use of Thread::Create or Thread::CreateWithSocketServer.
The default constructor constructs a 'default' socket server, which is usually a 'physical' socket server, but not always. Not every instance of Thread actually needs to have network support, so it's better to have this be explicit instead of unknowingly instantiate one.
BUG=none
Review-Url: https://codereview.webrtc.org/2981623002
Cr-Commit-Position: refs/heads/master@{#19001}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
"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}
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}
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}
DtlsTransportChannelWrapper is renamed to be DtlsTransport which inherits from
DtlsTransportInternal. There will be no concept of "channel" in p2p level.
Both P2PTransportChannel and DtlsTransport don't depend on TransportChannel
and TransportChannelImpl any more and they are removed in this CL.
BUG=none
Review-Url: https://codereview.webrtc.org/2606123002
Cr-Commit-Position: refs/heads/master@{#16173}
Reason for revert:
Failed the memory check.
May need to fix the memory leak.
Original issue's description:
> make the DtlsTransportWrapper inherit form DtlsTransportInternal
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2606123002
> Cr-Commit-Position: refs/heads/master@{#16160}
> Committed: 5aed06c8d3TBR=deadbeef@webrtc.org,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/2639203004
Cr-Commit-Position: refs/heads/master@{#16162}
Previously, BaseChannel supported a "no RTCP" mode, which wasn't
being used any more and is being deleted.
Also, "RTCP mux required" previously worked by calling "ActivateRtcpMux"
after construction. Now it works by explicitly passing a
"require_rtcp_mux" parameter into the constructor.
BUG=None
Review-Url: https://codereview.webrtc.org/2622613004
Cr-Commit-Position: refs/heads/master@{#16045}
The BaseChannel can set the transport directly without depending on
TransportController.
When initializing the network of the BaseChannel, the ChannelManager will
create TransportChannels with the TransportController.
When enabling bundling, WebRtcSession will get or create TransportChannels
with the TransportController.
When a TransportChannel of the BaseChannel needs to be destroyed, it will
fire a signal to notify the WebRtcSession.
BUG=none.
Review-Url: https://codereview.webrtc.org/2614263002
Cr-Commit-Position: refs/heads/master@{#16043}
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}
"Crypto required" is a property of the PeerConnection of construction
time; it has nothing to do with SDP. So I'm moving it out of
MediaContentDescription and putting it in the BaseChannel constructor
instead. This is more intuitive, and provides the added assurance that
"secure_required_" can't be flipped from "true" to "false".
BUG=None
Review-Url: https://codereview.webrtc.org/2537343003
Cr-Commit-Position: refs/heads/master@{#15579}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15453}
Reason for revert:
Deletion of transport.h broke downstream builds.
Going to reland with transport.h containing enums/etc.
Original issue's description:
> Refactoring that removes P2PTransport and DtlsTransport classes.
>
> Their base class, Transport, still exists, but it now has a more specific
> role: a helper class that applies TransportDescriptions. And is renamed
> to JsepTransport as a result.
>
> TransportController is now the entity primarily responsible for managing
> TransportChannels. It also starts storing pointers to the DTLS and ICE
> chanels separately, which will make it easier to remove
> TransportChannel/TransportChannelImpl in a subsequent CL.
>
> BUG=None
>
> Committed: https://crrev.com/bd28681d02dee8c185aeb39207e8154f0ad14a37
> Cr-Commit-Position: refs/heads/master@{#15450}
TBR=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/2553043004
Cr-Commit-Position: refs/heads/master@{#15452}
Their base class, Transport, still exists, but it now has a more specific
role: a helper class that applies TransportDescriptions. And is renamed
to JsepTransport as a result.
TransportController is now the entity primarily responsible for managing
TransportChannels. It also starts storing pointers to the DTLS and ICE
chanels separately, which will make it easier to remove
TransportChannel/TransportChannelImpl in a subsequent CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2517883002
Cr-Commit-Position: refs/heads/master@{#15450}
Reason for revert:
This CL probably broke Chromium FYI.
Original issue's description:
> Stop caching supported codecs in WebRtcVideoEngine2
>
> We currently cache the result of GetSupportedCodecs in a member variable
> |video_codecs_| in WebRtcVideoEngine2. This means we need to keep
> |video_codecs_| and the result of GetSupportedCodecs in sync, which is
> error prone. It's simpler to just call GetSupportedCodecs when we need
> it, and we actually end up making fewer calls, so it's faster as well.
> This CL also returns all std::vectors by-value instead of by-ref. Move
> semantic together with in-place filtering of codecs actually end up with
> fewer copies, and it's also simpler to not return references.
>
> BUG=webrtc:6337
>
> Committed: https://crrev.com/9f71ec5a3e3175751f4475b126cfda89767363f2
> Cr-Commit-Position: refs/heads/master@{#15007}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2489173004
Cr-Commit-Position: refs/heads/master@{#15014}
We currently cache the result of GetSupportedCodecs in a member variable
|video_codecs_| in WebRtcVideoEngine2. This means we need to keep
|video_codecs_| and the result of GetSupportedCodecs in sync, which is
error prone. It's simpler to just call GetSupportedCodecs when we need
it, and we actually end up making fewer calls, so it's faster as well.
This CL also returns all std::vectors by-value instead of by-ref. Move
semantic together with in-place filtering of codecs actually end up with
fewer copies, and it's also simpler to not return references.
BUG=webrtc:6337
Review-Url: https://codereview.webrtc.org/2492473002
Cr-Commit-Position: refs/heads/master@{#15007}
Since WebRtcVideoSendStream have reconfigures the send codec to match the incoming captured frames widht and height they have not been used.
Framerate has just been set when parsing sdp to 60fps and not changed elsewhere.
This cl require some upstream projects to change first.
BUG=webrtc:5332
Review-Url: https://codereview.webrtc.org/2408153002
Cr-Commit-Position: refs/heads/master@{#14733}
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.
This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).
BUG=webrtc:6393
Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
With this change, when max-bundle and rtcp-mux are both enabled, we no
longer create and destroy a temporary transport channel when a media
channel gets added. Instead, the media channel uses the correct bundled
transport channel from the start.
This fixes a bug where adding a media type would cause the ICE state to
briefly become Disconnected and then immediately recover. The temporary
channel was created in a non-writable state, which caused the
TransportController to declare the ICE state to be Disconnected (as not
all transport channels were writable). Right after creation, the
temporary channel was then destroyed and the ICE state went back to the
correct one.
BUG=webrtc:5856
Review-Url: https://codereview.webrtc.org/1972493002
Cr-Commit-Position: refs/heads/master@{#12781}
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.
BUG=webrtc:5645
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1903393004 .
Cr-Commit-Position: refs/heads/master@{#12690}
This field only existed as an implementation detail for getting the
codecs sorted, so it doesn't need to be in the public interface.
It cluttered the code and undesirably affected codec comparisons,
causing the video encoder to be reconfigured if a codec's preference
changed but nothing else did.
BUG=webrtc:5690
Review URL: https://codereview.webrtc.org/1845673002
Cr-Commit-Position: refs/heads/master@{#12349}
