The former is always defined (by webrtc/base/checks.h) to either 0 or
1, whereas the latter isn't necessarily defined.
NOTRY=true
BUG=webrtc:6451
Review-Url: https://codereview.webrtc.org/2384693002
Cr-Commit-Position: refs/heads/master@{#14474}
The implementation is borrowed from Chromium.
Also change use of raw pointers in VideoSendStreamImpl to use WeakPtr<>
BUG= webrtc:6289
Review-Url: https://codereview.webrtc.org/2367853002
Cr-Commit-Position: refs/heads/master@{#14468}
This means the DTLS handshake can make progress while the SDP answer
containing the fingerprint is still in transit. If the signaling path
if significantly slower than the media path, this can have a moderate
impact on call setup time.
Of course, until the fingerprint is verified no media can be sent. Any
attempted write will result in SR_BLOCK.
This essentially fulfills the requirements of RFC 4572, Section 6.2:
Note that when the offer/answer model is being used, it is possible
for a media connection to outrace the answer back to the offerer.
Thus, if the offerer has offered a 'setup:passive' or 'setup:actpass'
role, it MUST (as specified in RFC 4145 [2]) begin listening for an
incoming connection as soon as it sends its offer. However, it MUST
NOT assume that the data transmitted over the TLS connection is valid
until it has received a matching fingerprint in an SDP answer. If
the fingerprint, once it arrives, does not match the client's
certificate, the server endpoint MUST terminate the media connection
with a bad_certificate error, as stated in the previous paragraph.
BUG=webrtc:6387
Review-Url: https://codereview.webrtc.org/2163683003
Cr-Commit-Position: refs/heads/master@{#14461}
WebRTC polls the list of local IP addresses for both IPv4 and IPv6
every ~2 seconds. It does so by trying to connect() a UDP socket to
an address on the public Internet (without actually sending any
packets).
If the host doesn't have IPv6 (or IPv4) connectivity, it fails with
errno 101 (ENETUNREACH, Linux) or errno 65 (EHOSTUNREACH, Mac).
This is the expected behavior, and we shouldn't be logging these
failures, especially since polling is fairly frequent.
BUG=webrtc:6347
R=deadbeef@webrtc.org, honghaiz@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/2370383002 .
Cr-Commit-Position: refs/heads/master@{#14440}
It's a very general type, and we're about to start needing it in other
places besides AudioCodingModule.
BUG=webrtc:5801
Review-Url: https://codereview.webrtc.org/2380463003
Cr-Commit-Position: refs/heads/master@{#14423}
These are already added in third_party/jsoncpp/BUILD.gn so they
shouldn't be needed.
BUG=webrtc:6412
NOTRY=True
NOTREECHECKS=True
Review-Url: https://codereview.webrtc.org/2368543002
Cr-Commit-Position: refs/heads/master@{#14379}
Reason for revert:
Broke a downstream user of SSLStreamAdapter. Need to add the new interface (returning error code instead of bool) in a backwards compatible way.
Original issue's description:
> Allow the DTLS fingerprint verification to occur after the handshake.
>
> This means the DTLS handshake can make progress while the SDP answer
> containing the fingerprint is still in transit. If the signaling path
> if significantly slower than the media path, this can have a moderate
> impact on call setup time.
>
> Of course, until the fingerprint is verified no media can be sent. Any
> attempted write will result in SR_BLOCK.
>
> This essentially fulfills the requirements of RFC 4572, Section 6.2:
>
> Note that when the offer/answer model is being used, it is possible
> for a media connection to outrace the answer back to the offerer.
> Thus, if the offerer has offered a 'setup:passive' or 'setup:actpass'
> role, it MUST (as specified in RFC 4145 [2]) begin listening for an
> incoming connection as soon as it sends its offer. However, it MUST
> NOT assume that the data transmitted over the TLS connection is valid
> until it has received a matching fingerprint in an SDP answer. If
> the fingerprint, once it arrives, does not match the client's
> certificate, the server endpoint MUST terminate the media connection
> with a bad_certificate error, as stated in the previous paragraph.
>
> BUG=webrtc:6387
> R=mattdr@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/042041bf9585f92e962387c59ca805f1218338f9
> Cr-Commit-Position: refs/heads/master@{#14296}
TBR=pthatcher@webrtc.org,mattdr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6387
Review-Url: https://codereview.webrtc.org/2352863003
Cr-Commit-Position: refs/heads/master@{#14298}
Previously when a Turn port is pruned, if its candidate has been sent to the remote side, the remote side will keep the candidate and use that to create connections.
We now signal the remote side to remove the candidates so that at least no new connection will be created using the removed candidates.
Also updated the virtual socket server to better support our test cases.
1. Allow the virtual socket server to set transit delay for packets sent from a given IP address.
2. Ensure the ordered packet delivery for each socket (Previously the delivery order is enforced on the whole test case, so if a udp packet gets delayed based on its IP address, all TCP packets sent after the UDP packet will be delayed at least until the UDP packet is received).
BUG=webrtc:6380
R=deadbeef@webrtc.org, pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2261523004 .
Cr-Commit-Position: refs/heads/master@{#14297}
This means the DTLS handshake can make progress while the SDP answer
containing the fingerprint is still in transit. If the signaling path
if significantly slower than the media path, this can have a moderate
impact on call setup time.
Of course, until the fingerprint is verified no media can be sent. Any
attempted write will result in SR_BLOCK.
This essentially fulfills the requirements of RFC 4572, Section 6.2:
Note that when the offer/answer model is being used, it is possible
for a media connection to outrace the answer back to the offerer.
Thus, if the offerer has offered a 'setup:passive' or 'setup:actpass'
role, it MUST (as specified in RFC 4145 [2]) begin listening for an
incoming connection as soon as it sends its offer. However, it MUST
NOT assume that the data transmitted over the TLS connection is valid
until it has received a matching fingerprint in an SDP answer. If
the fingerprint, once it arrives, does not match the client's
certificate, the server endpoint MUST terminate the media connection
with a bad_certificate error, as stated in the previous paragraph.
BUG=webrtc:6387
R=mattdr@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2163683003 .
Cr-Commit-Position: refs/heads/master@{#14296}
The two situations are:
1. A thread is in the process of shutting down, so it won't handle any
more messages.
2. A message queue is cleared before it has a chance to process pending
messages.
In both of those cases, we should consider processing done at that
point.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2319303004 .
Cr-Commit-Position: refs/heads/master@{#14245}
Also enforce a minimum inter-frame interval of 1 ms,
fix a bug in the clipping logic, and improve comments.
BUG=webrtc:5740
Review-Url: https://codereview.webrtc.org/2325563002
Cr-Commit-Position: refs/heads/master@{#14206}
Calling VirtualSocketServer::SetSendingBlocked(true) will simulate the
network interface being blocked, and SetSendingBlocked(false) will
simulate it being unblocked, resulting in SignalReadyToSend if
appropriate.
I plan to use this to write tests for upper layers of code that deal
with EWOULDBLOCK/SignalReadyToSend.
Also doing some minor housekeeping in this CL (using RTC_DCHECK,
renaming variables, etc.).
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2284903002 .
Cr-Commit-Position: refs/heads/master@{#14170}
The right thing to do would be to ensure that the multiplication can't
overflow, but that'd be a major change bordering on a rewrite, and
would take too much time. So instead, we just instruct UBSan to look
the other way.
BUG=chromium:634834
Review-Url: https://codereview.webrtc.org/2318083002
Cr-Commit-Position: refs/heads/master@{#14154}
Right now, RuntimeExceptions thrown in an invoke function will kill the handler thread. This CL propagates the exceptions to the calling thread instead, to give the caller a change to handle them.
BUG=webrtc:6327
Review-Url: https://codereview.webrtc.org/2316223002
Cr-Commit-Position: refs/heads/master@{#14132}
Reason for revert:
Chromium build issues have been resolved.
Original issue's description:
> Revert of Remove all reference to carbon api (patchset #2 id:20001 of https://codereview.webrtc.org/2299633002/ )
>
> Reason for revert:
> Breaks chromium build
>
> Original issue's description:
> > Remove all reference to carbon api
> >
> > BUG=webrtc:6282
> >
> > Committed: https://crrev.com/dbd8b6bec4143c940b2f2ca8cd85c25d17327964
> > Cr-Commit-Position: refs/heads/master@{#14080}
>
> TBR=magjed@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6282
>
> Committed: https://crrev.com/b096aa7fd375a980daab3a986596548ca5de2a1c
> Cr-Commit-Position: refs/heads/master@{#14081}
TBR=magjed@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6282
Review-Url: https://codereview.webrtc.org/2321493002
Cr-Commit-Position: refs/heads/master@{#14125}
There is no clear reason to have them in build_overrides, and
webrtc/build seems to be a better place.
Also, delete build_overrides/webrtc.gni
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2309253004
Cr-Commit-Position: refs/heads/master@{#14108}
Reason for revert:
Breaks chromium build
Original issue's description:
> Remove all reference to carbon api
>
> BUG=webrtc:6282
>
> Committed: https://crrev.com/dbd8b6bec4143c940b2f2ca8cd85c25d17327964
> Cr-Commit-Position: refs/heads/master@{#14080}
TBR=magjed@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6282
Review-Url: https://codereview.webrtc.org/2316563002
Cr-Commit-Position: refs/heads/master@{#14081}
In addition to setting or appending from another Buffer, which was
already possible, this allows for e.g. std::vector and rtc::ArrayView
arguments.
Review-Url: https://codereview.webrtc.org/2293983002
Cr-Commit-Position: refs/heads/master@{#14073}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Before this change, the argument still needed those methods, but not
having them resulted in a compilation error. Now, it results in this
constructor being removed from the overload set.
This currently makes no difference, but I'm about to publish a CL that
breaks without this.
Review-Url: https://codereview.webrtc.org/2312473002
Cr-Commit-Position: refs/heads/master@{#14068}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
The sample uses are from when I debugged bug 617124. The change in neteq_network_stats_unittest.cc is a fix for a minor unrelated bug found by the try bots when I tried to land this CL (a test was passing uninitialized packet data to NetEq).
BUG=chromium:617124
Review-Url: https://codereview.webrtc.org/2293893002
Cr-Commit-Position: refs/heads/master@{#14034}
WebRTC no longer has any restriction on what thread frames should be
delivered on. One possible problem with this CL is that NV21->I420
conversion and scaling is done on the thread that delivers frames, which
might cause fps regressions.
R=nisse@webrtc.org, perkj@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/2137503003 .
Cr-Commit-Position: refs/heads/master@{#14021}
