Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
Enforces previous kProtectionKeyOnLoss as the permanent method which was
the only one used in use. This simplifies SetVideoProtection and
transition over to SetReceiverRobustnessMode.
BUG=webrtc:1596
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1244753002
Cr-Commit-Position: refs/heads/master@{#9641}
Reduces number of running (high-priority) threads, even though the
thread was practically blocked all the time.
Also adding DCHECKs to make sure we're not trying to use certain
sender-only methods on receivers and vice versa.
BUG=webrtc:1675, webrtc:1695
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1222193003
Cr-Commit-Position: refs/heads/master@{#9534}
Registers transport on construction removing the need for ViESender as a
hop and removing a potential deadlock by removing RegisterSendTransport.
BUG=1695, 2999
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57449004
Cr-Commit-Position: refs/heads/master@{#9309}
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
The uint8_t in the log string is interpreted as a char, causing a
character to be logged if the loss is non-zero and terminates the string
with a '\0' in the zero case.
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53449004
Cr-Commit-Position: refs/heads/master@{#9242}
This will make us detect when sources are listed in GYP files that
are no longer present on disk. This check only exists for Windows
but should be enough to keep our GYP files up to date with the file
system.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54409004
Cr-Commit-Position: refs/heads/master@{#9187}
Instead a ViECapturer object is allocated and directly operated on. This
additionally exposes ViESharedData to Call to access the module
ProcessThread, moving towards Call ownership of shared resources.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45339004
Cr-Commit-Position: refs/heads/master@{#9119}
Replaces interface usage with direct calls on ViEEncoder removing a
layer of indirection. Also removing some methods from ViEImageProcess
that were only added for Video{Send,Receive}Stream usage.
BUG=1695
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45319004
Cr-Commit-Position: refs/heads/master@{#9111}
ViECapturer::OnIncomingCapturedFrame can abort if two frames without a
timestamp are delivered at the same time (the same NTP timestamp is
generated). This currently causes captured_frame_ to be in a state
where timestamps (NTP time etc.) are set, but RTP timestamp is not. This
is an issue if the previous frame in captured_frame_ has yet to be
picked up by the capturer thread, as it will pick up captured_frame_ in
a state where the RTP timestamp has not been set.
BUG=chromium:480953
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45299004
Cr-Commit-Position: refs/heads/master@{#9094}
Since RTCP packets are delivered to both senders and receivers that
correspond the receivers currently log that NACKed packets are missing,
since they have no direct connection to the sending side or the RTP
packet history. Also preventing triggering on SR requests and PLI/FIR.
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45249004
Cr-Commit-Position: refs/heads/master@{#9071}
An incoming rtcp report block is inserted to both send and receive channels in Call::DeliverRtcp. The report block may also be accepted by each receive channel (in addition to the send channel) but fails to calculate the rtt (=0).
Remove registration of bandwidth observer for receive channels. Prevents multiple callbacks to the bitrate controller (and with incorrect rtt) for an incoming report block.
BUG=4546
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45209004
Cr-Commit-Position: refs/heads/master@{#9055}
