- Place all member function definitions between test fixture
declaration and unit tests.
- Rename GenerateFrame -> PacketizeFrame.
No functional changes are intended by this CL.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2275303003
Cr-Commit-Position: refs/heads/master@{#14488}
Helper class for FlexFEC unit tests to come.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2282473002
Cr-Commit-Position: refs/heads/master@{#14487}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
hbos and hta are already OWNERS of webrtc/stats/ and of rtcstats* files
(per-file rtcstats*=) in webrtc/api/. When the webrtc/api/stats/ folder
was created we forgot to add this OWNERS file (per-file OWNERS does not
apply to subfolders apparently).
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2392633002
Cr-Commit-Position: refs/heads/master@{#14482}
This DCHECK is triggered by org.webrtc.PeerConnectionTest#testTrackRemoval.
BUG=webrtc:6465
Review-Url: https://codereview.webrtc.org/2389703002
Cr-Commit-Position: refs/heads/master@{#14481}
Changing this constant has been empirically proven to solve the issue.
BUG=webrtc:6455,b/31827852
Review-Url: https://codereview.webrtc.org/2382733006
Cr-Commit-Position: refs/heads/master@{#14479}
Main changes:
- Split out general functionality from UlpfecPacketGenerator
into a new class AugmentedPacketGenerator. This will allow
for the addition of a FlexfecPacketGenerator that inherits
from AugmentedPacketGenerator.
- Rename RawRtpPacket to AugmentedPacket. This name is more
reflective of its purpose, i.e., an FEC packet with an
additional WebRtcRTPHeader.
- Return std::unique_ptr's instead of raw pointers.
Minor changes:
- Update names, based on RawRtpPacket -> AugmentedPacket name
change, in FEC unit tests.
- Rename |generator_| to |packet_generator_|, in FEC unit tests.
- Change some int's to size_t's in the packet generator classes.
- Use std::unique_ptr in ProducerFec unittest.
No functionality or performance changes are expected
due to this CL.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2271273004
Cr-Commit-Position: refs/heads/master@{#14477}
- Change some types to size_t.
- Update parameter ordering.
- Run 'git cl format'
- Moved 'using declarations' into unnamed namespaces.
- Removed "::webrtc::" prefix from some using declarations.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2273353002
Cr-Commit-Position: refs/heads/master@{#14475}
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}
Also, change order of definition in fec_test_helper.{h,cc}.
This CL should have no implications on functionality. It is in preparation
for a FlexfecPacketGenerator class, which will be used in the FlexFEC
unit tests.
R=danilchap@webrtc.org
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2276473002
Cr-Commit-Position: refs/heads/master@{#14471}
Add classes that can read and finalize FlexFEC headers.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2269903002
Cr-Commit-Position: refs/heads/master@{#14469}
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}
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.
Original cl description:
Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
The Connection class will now blindly forward SignalReadyToSend, and
P2PTransportChannel will decide whether to forward it further (which
it was already doing).
BUG=webrtc:6448
Review-Url: https://codereview.webrtc.org/2374183005
Cr-Commit-Position: refs/heads/master@{#14462}
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}
This change reduces the number of places where we first fread a I420
frame into a uint8_t buffer, followed by a copy into a frame buffer
object.
BUG=None
Review-Url: https://codereview.webrtc.org/2362683002
Cr-Commit-Position: refs/heads/master@{#14456}
This also makes it possible to drop the RTPFragmentationHeader from
the class VCMEncodedFrame.
BUG=None
Review-Url: https://codereview.webrtc.org/2380933003
Cr-Commit-Position: refs/heads/master@{#14455}
The capture time for a frame (capture_ms) is set later (in ViEEncoder::IncomingCapturedFrame) than the timestamp.
Could potentially cause the RTP timestamp in consecutive RTCP SR to decrease.
Example:
// Frame1 46371: timestamp:2732, capture_ms:46373, rtcp SR ms: 46423 -> estimated current RTP timestamp:2732+(46423-46373)*90 = 7232
// Frame2 46404: timestamp:5702, capture_ms:46412, rtcp SR ms: 46428 -> estimated current RTP timestamp:5702+(46428-46412)*90 = 7142
// Diff: 33 ms: 33 ms, 39 ms, 5 ms
BUG=b/31154867
Review-Url: https://codereview.webrtc.org/2354843003
Cr-Commit-Position: refs/heads/master@{#14454}
Reason for revert:
Downstream code is being fixed.
Original issue's description:
> Revert of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #9 id:160001 of https://codereview.webrtc.org/2262443003/ )
>
> Reason for revert:
> Breaks downstream testcode, still using CapturedFrame.
>
> Original issue's description:
> > Delete VideoFrameFactory, CapturedFrame, and related code.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/66ac50e58c790624d51ede10ae438cbadbca9d2e
> > Cr-Commit-Position: refs/heads/master@{#14315}
>
> TBR=pthatcher@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/66492210e57ee8efce2ad4d45a8781df1fcaa5e3
> Cr-Commit-Position: refs/heads/master@{#14320}
TBR=pthatcher@webrtc.org,perkj@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2370993003
Cr-Commit-Position: refs/heads/master@{#14453}
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2378103005 .
Cr-Commit-Position: refs/heads/master@{#14452}
Reason for revert:
Internal project has been fixed
Original issue's description:
> Revert of Unify the macOS and iOS capturer implementations (patchset #4 id:60001 of https://codereview.webrtc.org/2309253005/ )
>
> Reason for revert:
> Breaks internal project
>
> Original issue's description:
> > Unify the macOS and iOS capturer implementations
> >
> > This removes the QTKit based capturer for mac, and removes the need
> > to link against deprecated system libraries on macOS.
> >
> > BUG=webrtc:3968,webrtc:6275,webrtc:6333
> >
> > Committed: https://crrev.com/242d8bdddd77109781cbb70c59d161be7566ac98
> > Cr-Commit-Position: refs/heads/master@{#14418}
>
> TBR=magjed@webrtc.org,tkchin@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:3968,webrtc:6275,webrtc:6333
>
> Committed: https://crrev.com/eddb7571d81e51a66f4abaf55013c85b4132c837
> Cr-Commit-Position: refs/heads/master@{#14429}
TBR=magjed@webrtc.org,tkchin@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:3968,webrtc:6275,webrtc:6333
Review-Url: https://codereview.webrtc.org/2375273004
Cr-Commit-Position: refs/heads/master@{#14451}
The clang static analyzer seems unable to resolve cpp locks in ObjC code.
As of current time, the clang analyzer has known limitations documented
http://clang.llvm.org/docs/ThreadSafetyAnalysis.html#known-limitations.
From the documentation: "The analysis currently does not do any checking
inside constructors or destructors.
In other words, every constructor and destructor is treated
as if it was annotated with NO_THREAD_SAFETY_ANALYSIS."
This is 'probably' why the analyzer is unable to resolve the lock when
used in ObjC land (the cpp works fine).
The lock can be removed by using atomic property instead.
It's not on performance critical path and we expect updates on just one queue and reads from others. That's why the thread assurance atomic properties bring is enough.
The CL removes rtc_sdk_peerconnection_objc_warnings_config as well as it's no longer needed.
BUG=webrtc:6308
Review-Url: https://codereview.webrtc.org/2372513004
Cr-Commit-Position: refs/heads/master@{#14450}
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
Reason for revert:
It caused some tests in p2ptransportchannel flaky.
Original issue's description:
> Replace RelayPort with TurnPort in p2ptransportchannel tests.
>
> Also remove the relay servers in the tests.
> Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.
>
> BUG=None
> R=deadbeef@webrtc.org
>
> Committed: https://crrev.com/c8d21712dde64c7d613d1ea56c840438505a909f
> Cr-Commit-Position: refs/heads/master@{#14441}
TBR=deadbeef@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/2385563002
Cr-Commit-Position: refs/heads/master@{#14443}
move all logic from that class into RTCPReceiver too,
Simplify and fix style on the way.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2373053002
Cr-Commit-Position: refs/heads/master@{#14442}
Also remove the relay servers in the tests.
Most of the code and the downstream apps are using TurnPort, not RelayPort. Most of the tests in this file are not using RelayPort anyway.
BUG=None
R=deadbeef@webrtc.org
Review URL: https://codereview.webrtc.org/2380923002 .
Cr-Commit-Position: refs/heads/master@{#14441}
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}
