by creating it on accepted tmmbr/tmmbn rtcp messages
rather on sender/receiver reports.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2702373002
Cr-Commit-Position: refs/heads/master@{#16748}
together with related functions and variables
to stress it is used for Tmmbr only.
This is explicitly pure rename CL with no functional changes.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2707763004
Cr-Commit-Position: refs/heads/master@{#16720}
This a pre-step for improving perfomance of the RTCPReceiver
- rest of the ReceiveInfo is tmmbr related and
can be handled only when tmmbr is explicitly enabled.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2681003003
Cr-Commit-Position: refs/heads/master@{#16667}
Since the only used class is RTCPUtilitiy::NackStats,
rename it to RtcpNackStats and move it into dedicated file.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2680183004
Cr-Commit-Position: refs/heads/master@{#16515}
Rename variables and private functions to follow style,
replace remaining asserts with DCHECKs.
add 'ms' suffix to time variables derived from clock_
add 'ntp' suffix to time variables derived from ntp time.
No functional changes expected.
BUG=None
Review-Url: https://codereview.webrtc.org/2588753002
Cr-Commit-Position: refs/heads/master@{#15706}
Specifically, reject any bitrate allocated for a layer not representable
by the BitrateAllocation struct.
BUG=chromium:671312
Review-Url: https://codereview.webrtc.org/2549233005
Cr-Commit-Position: refs/heads/master@{#15447}
Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
The renaming is to reflect this class is only used for RTCP interaction
and not for other transports.
This Cl will be followed by multiple CLs moving all send-side RTP
functionality to a separate class, rtp module ownership away from
VideoSendStream and use TaskQueue instead of ProcessThread for RTP.
BUG=webrtc:6456
Review-Url: https://codereview.webrtc.org/2390463002
Cr-Commit-Position: refs/heads/master@{#14556}
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}
structs are exactly the same but last one follow naming style.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2368983002
Cr-Commit-Position: refs/heads/master@{#14415}
simplifying cname management.
Remove RTCPUtility::RTCPCnameInformation
since it was last use of the structure.
BUG=webrtc:5565
NOTRY=true
Review-Url: https://codereview.webrtc.org/2354333004
Cr-Commit-Position: refs/heads/master@{#14399}
Use it by pointer instead of by reference.
Renamed PacketInformation members to follow style,
Unused members removed.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2366563002
Cr-Commit-Position: refs/heads/master@{#14375}
App, ExtendedJitterReport and VoipMetric in ExtenedReports are not
used when received (no callbacks, no state change), so removed.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2320703003
Cr-Commit-Position: refs/heads/master@{#14204}
Instead of full RtpRtcpImpl takes interface of all functions it needs from it.
Added single function for parsing packets and sending feedback, moving that
logic from RtpRtcpImpl to RtcpReceiver.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2274573002
Cr-Commit-Position: refs/heads/master@{#13960}
function names style updated,
unused return type removed.
Comment style fixed, redundant comments removed.
pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.
NOTRY=true
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2258523005
Cr-Commit-Position: refs/heads/master@{#13848}
by cleaning RTCPReceiveInfo class
and following cleaning of RTCPReceiver::BoundingSet function.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2254703003
Cr-Commit-Position: refs/heads/master@{#13817}
This CL adds support for an extension on RTP frames to allow the sender
to specify the minimum and maximum playout delay limits.
The receiver makes a best-effort attempt to keep the capture-to-render delay
within this range. This allows different types of application to specify
different end-to-end delay goals. For example gaming can support rendering
of frames as soon as received on receiver to minimize delay. A movie playback
application can specify a minimum playout delay to allow fixed buffering
in presence of network jitter.
There are no tests at this time and most of testing is done with chromium
webrtc prototype.
On chromoting performance tests, this extension helps bring down end-to-end
delay by about 150 ms on small frames.
BUG=webrtc:5895
Review-Url: https://codereview.webrtc.org/2007743003
Cr-Commit-Position: refs/heads/master@{#13059}
indicating the usage of this helper is local.
With local usage critical section become obvisously useless and removed.
BUG=webrtc:5565
R=åsapersson
Review-Url: https://codereview.webrtc.org/1959013003
Cr-Commit-Position: refs/heads/master@{#12881}
Added packet type RpsiItem to destinguish parsed rpsi header and Rpsi body
preventing handling two half-valid (header-only) rpsi packets as one valid,
making test parser calculate rpsi packet once instead of twice.
Added check padding bits doesn't exceed native bit string length.
Marking rpsi received moved after it is validated.
BUG=600977
Review URL: https://codereview.webrtc.org/1880443002
Cr-Commit-Position: refs/heads/master@{#12318}
CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases:
large values consider negative/invalid and result in value of 1.
0 result consider too small and increases to 1.
BUG=590996
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1763823003 .
Cr-Commit-Position: refs/heads/master@{#11928}
Compact NTP representation was designed exactly for that purpose: calculate RTT. No need to map to ms before doing arithmetic on this values.
Because of this change there is no need to keep mapping between compact ntp presentation and milliseconds in the RTCPSender.
BUG=webrtc:5565
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1491843004 .
Cr-Commit-Position: refs/heads/master@{#11710}
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
