|transport_overhead_per_packet_| and |rtp_overhead_per_packet_| could
be read from and written to on different threads concurrently. This CL
introduces a lock to GUARD these variables.
NOTRY because master.tryserver.webrtc.linux_ubsan_vptr is broken, all
other tests pass.
BUG=webrtc:7231
NOTRY=True
Review-Url: https://codereview.webrtc.org/2710363003
Cr-Commit-Position: refs/heads/master@{#16900}
The class basically implements a timer and can be replaced with a PostDelayedTask call down the line.
BUG=none
Review-Url: https://codereview.webrtc.org/2722613002
Cr-Commit-Position: refs/heads/master@{#16891}
In this CL:
- Add message BweProbeCluster and BweProbeResult to rtc_event_log.proto.
- Add corresponding log functions to RtcEventLog.
- Add optional field |probe_cluster_id| to RtpPacket message and added
an overload function to log with this information.
- Propagate the probe_cluster_id to where RTP packets are logged.
BUG=webrtc:6984
Review-Url: https://codereview.webrtc.org/2666533002
Cr-Commit-Position: refs/heads/master@{#16857}
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/
webrtc::PacedSender::Process <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- this CL end here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
The new constructor introduces two new changes:
* Support specifying thread priority at construction time.
- Moving forward, the SetPriority() method will be removed.
* New thread function type.
- The new type has 'void' as a return type and a polling loop
inside PlatformThread, is not used.
The old function type is still supported until all places have been moved over.
In this CL, the first steps towards deprecating the old mechanism are taken
by moving parts of the code that were simple to move, over to the new callback
type.
BUG=webrtc:7187
Review-Url: https://codereview.webrtc.org/2708723003
Cr-Commit-Position: refs/heads/master@{#16779}
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).
BUG=webrtc:6423
Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
This implementation uses various legacy classes such as EventTimeWrapper,
CriticalSectionWrapper, EventWrapper etc and hasn't been maintained
(or used?) for a long time.
Instead of spending time on testing and updating the class, I think
we should just remove it. For versions of Windows that we support,
following Win7, we use the CoreAudio implementation.
BUG=webrtc:7183
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2700983002 .
Cr-Commit-Position: refs/heads/master@{#16678}
and the method RTPSender::GenerateNewSSRC.
It's now mandatory for higher layers to call SetSSRC, RTPSender
no longer allocates any ssrc by default.
BUG=webrtc:4306,webrtc:6887
Review-Url: https://codereview.webrtc.org/2644303002
Cr-Commit-Position: refs/heads/master@{#16670}
The removed tests are covered by cases in call_perf_tests.
BUG=webrtc:4690
Review-Url: https://codereview.webrtc.org/2672583002
Cr-Commit-Position: refs/heads/master@{#16621}
Note:
* PLR is calculated over all of the known packets.
* RPLR is calculated over all of the known packet *pairs*. That is, only over sets of subsequent packets where the reception status is known for both.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2629883003
Cr-Commit-Position: refs/heads/master@{#16401}
This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder.
BUG=webrtc:6904
Review-Url: https://codereview.webrtc.org/2579613003
Cr-Commit-Position: refs/heads/master@{#16217}
The code previously allowed ipv6 addresses with less than eight sections even without all-zero sections being compacted by a ::.
BUG=webrtc:1028
Review-Url: https://codereview.webrtc.org/2606383003
Cr-Commit-Position: refs/heads/master@{#16108}
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}
This lets the RTP code be unaware of lower layers, and the
SetTransportOverhead method is deleted from RTPSender and RtpRtcp.
Instead, that method is added to CongestionController and
TransportFeedbackAdapter, where it is more appropriate.
BUG=wertc:6847
Review-Url: https://codereview.webrtc.org/2589743002
Cr-Commit-Position: refs/heads/master@{#15995}
Because Voice Engine was the only user.
(We have tried to land this many times before. I'm hoping that this
time all external dependencies on these files will really be gone.)
BUG=none
Review-Url: https://codereview.webrtc.org/2622493002
Cr-Commit-Position: refs/heads/master@{#15978}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
