This CL is a reland of https://codereview.webrtc.org/2722423003 which got
reverted due to compile errors when rolling into Chromium.
Original CL description:
Improve testing of SRTP external auth code paths.
Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
but developed in WebRTC, which made testing rather complicated. This caused
some trouble in the past (e.g. https://crbug.com/628400#c1)
This CL helps in that the external auth code is now compiled with WebRTC
and the srtpfilter integration gets tested.
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2735613002
Cr-Commit-Position: refs/heads/master@{#17052}
Reason for revert:
Breaks compilation in FYI bots, e.g. here:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/9314
FAILED: obj/third_party/webrtc/pc/rtc_pc/channel.obj
ninja -t msvc -e environment.x86 -- E:\b\c\goma_client/gomacc.exe "E:\b\depot_tools\win_toolchain\vs_files\d3cb0e37bdd120ad0ac4650b674b09e81be45616\VC\bin\amd64_x86/cl.exe" /nologo /showIncludes /FC @obj/third_party/webrtc/pc/rtc_pc/channel.obj.rsp /c ../../third_party/webrtc/pc/channel.cc /Foobj/third_party/webrtc/pc/rtc_pc/channel.obj /Fd"obj/third_party/webrtc/pc/rtc_pc_cc.pdb"
e:\b\c\b\win_builder\src\third_party\webrtc\pc\channel.cc(176): error C2819: type 'cricket::SrtpFilter' does not have an overloaded member 'operator ->'
e:\b\c\b\win_builder\src\third_party\webrtc\pc\srtpfilter.h(45): note: see declaration of 'cricket::SrtpFilter'
e:\b\c\b\win_builder\src\third_party\webrtc\pc\channel.cc(176): note: did you intend to use '.' instead?
e:\b\c\b\win_builder\src\third_party\webrtc\pc\channel.cc(176): error C2232: '->cricket::SrtpFilter::EnableExternalAuth': left operand has 'class' type, use '.'
Original issue's description:
> Improve testing of SRTP external auth code paths.
>
> Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
> but developed in WebRTC, which made testing rather complicated. This caused
> some trouble in the past (e.g. https://crbug.com/628400#c1)
>
> This CL helps in that the external auth code is now compiled with WebRTC
> and the srtpfilter integration gets tested.
>
> BUG=chromium:628400
>
> Review-Url: https://codereview.webrtc.org/2722423003
> Cr-Commit-Position: refs/heads/master@{#17030}
> Committed: ac170d5c21TBR=deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2734643002
Cr-Commit-Position: refs/heads/master@{#17031}
Previously code behind ENABLE_EXTERNAL_AUTH was only compiled with Chromium
but developed in WebRTC, which made testing rather complicated. This caused
some trouble in the past (e.g. https://crbug.com/628400#c1)
This CL helps in that the external auth code is now compiled with WebRTC
and the srtpfilter integration gets tested.
BUG=chromium:628400
Review-Url: https://codereview.webrtc.org/2722423003
Cr-Commit-Position: refs/heads/master@{#17030}
With ENABLE_EXTERNAL_AUTH, external auth will only be used depending
on the selected cipher (allowed for non-GCM, not allowed for GCM).
BUG=webrtc:5222, chromium:628400
Review-Url: https://codereview.webrtc.org/2720663003
Cr-Commit-Position: refs/heads/master@{#16955}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
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}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
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: Failed to setup RTCP mux filter.
Now: rtcpMuxPolicy is 'require', but media description does not
contain 'a=rtcp-mux'.
BUG=webrtc:6966
Review-Url: https://codereview.webrtc.org/2622553003
Cr-Commit-Position: refs/heads/master@{#16062}
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}
Bulk of changes done using
git grep -l 'RTC_DCHECK(false)' | \
xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/'
peerconnection.cc also used RTC_DCHECK(false && "msg") in two places,
which were updated manually.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623313004
Cr-Commit-Position: refs/heads/master@{#16026}
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}
We still DCHECK for RTP, but not RTCP. RTCP packets can be sent before
offer/answer negotiation is complete, due to this bug:
https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
This bug can only occur if the RTCP mux policy is "require", which is
why we started hitting it recently (the default in unit tests was
recently changed to "require").
BUG=webrtc:6776
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2542233002
Cr-Commit-Position: refs/heads/master@{#15369}
Introduce rtc::PacketTransportInterface. Refactor cricket::TransportChannel.
Fix signal slots parameter types in all related code.
BUG=webrtc:6531
Review-Url: https://codereview.webrtc.org/2416023002
Cr-Commit-Position: refs/heads/master@{#14778}
The only real difference between the two is that SetRtcpTransportChannel
had a workaround to prevent a signal from being emitted early.
Basically, in SetTransport, we want to switch the transport channels and
*then* update the state, rather than updating the state after changing
only one transport channel.
But this can be accomplished more easily by simply updating the state in
SetTransport directly.
Review-Url: https://codereview.webrtc.org/2274283004
Cr-Commit-Position: refs/heads/master@{#13945}
There were 3 different meanings for "ReadyToSend", for example, so it
was difficult to understand the meaning at first glance.
Also switching ASSERTs to RTC_DCHECKs.
Review URL: https://codereview.webrtc.org/2269173004 .
Cr-Commit-Position: refs/heads/master@{#13926}
Removing a redundant variable used to track whether or not RTCP mux has
been fully negotiated. It's RtcpMuxFilter's job to do that, and it
already had the state, it just wasn't exposed.
Review-Url: https://codereview.webrtc.org/2260963002
Cr-Commit-Position: refs/heads/master@{#13856}
GCM cipher suites are optional (disabled by default) and can be enabled
through "PeerConnectionFactoryInterface::Options".
If compiled with Chromium (i.e. "ENABLE_EXTERNAL_AUTH" is defined), no
GCM ciphers can be used yet (see https://crbug.com/628400).
BUG=webrtc:5222, 628400
Review-Url: https://codereview.webrtc.org/1528843005
Cr-Commit-Position: refs/heads/master@{#13635}
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.
We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.
I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2192963002 .
Cr-Commit-Position: refs/heads/master@{#13566}
The "should I simulate EWOULDBLOCK?" determination now happens
solely in P2PTransportChannel. This also fixes a bug where the
"last packet id" was set even if no packet was sent.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2099783002 .
Cr-Commit-Position: refs/heads/master@{#13307}
Reason for revert:
It turns out this revert was not necessary because the connection-state mapping for turn-turn connections was not done in connection.
Original issue's description:
> Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ )
>
> Reason for revert:
> ReadyToSendMedia did not consider the new presumed_writable state.
>
> Original issue's description:
> > Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )"
> >
> > This reverts commit 72d41aa6da94dacb8a8464d1abd4ca7d1afffc65.
> >
> > New change made:
> > Do not reset the BWE when the new network route is not ready to send media.
> >
> > BUG=
> > R=pthatcher@webrtc.org, stefan@webrtc.org
> >
TBR=pthatcher@webrtc.org,stefan@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2094863003
Cr-Commit-Position: refs/heads/master@{#13282}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
This means there's only one thread hop to the worker thread.
At the video engine level, SetOptions and SetSource
are combined into one method (all within the same critical section)
which ensures that no frame will be encoded while SetVideoSend
is only partially finished.
BUG=webrtc:5691
Review-Url: https://codereview.webrtc.org/1838413002
Cr-Commit-Position: refs/heads/master@{#13022}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
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}
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1917193008 .
Cr-Commit-Position: refs/heads/master@{#12761}
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 change builds on top of the refactoring in https://codereview.webrtc.org/1841083008/, and enables WebRTC client applications to control the max send bitrate for every audio stream through RtpParameters.
The AudioSendStream now stores the last codec spec, and whenever a global or per-stream bitrate limit changes, the effective limit (smaller of the two) is recomputed and the codec is reconfigured with that bitrate.
TBR=pthatcher
BUG=
Review URL: https://codereview.webrtc.org/1847353004
Cr-Commit-Position: refs/heads/master@{#12290}
so that the call knows which packet ids were sent on the previous candidate pair.
Note that packet_id is actually 16bits, so we can use -1 for values that are not set.
Also moved the tests for candidate pair changes to TestSelectConnectionBeforeNomination.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1842093002 .
Cr-Commit-Position: refs/heads/master@{#12184}
