- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2067673004 .
Cr-Commit-Position: refs/heads/master@{#13565}
Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too.
It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet.
BUG=webrtc:1600
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1639253007 .
Cr-Commit-Position: refs/heads/master@{#13503}
Reason for revert:
Upstream fixes in place, should be OK now.
Original issue's description:
> Revert of Refactor NACK bitrate allocation (patchset #16 id:300001 of https://codereview.webrtc.org/2061423003/ )
>
> Reason for revert:
> Breaks upstream code.
>
> Original issue's description:
> > Refactor NACK bitrate allocation
> >
> > Nack bitrate allocation should not be done on a per-rtp-module basis,
> > but rather shared bitrate pool per call. This CL moves allocation to the
> > pacer and cleans up a bunch if bitrate stats handling.
> >
> > BUG=
> > R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
> >
> > Committed: 5fc59e810b
>
> TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/e5dd44101eca485f5ad12e5f7ce6f6b0d204116b
> Cr-Commit-Position: refs/heads/master@{#13417}
TBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=
Review-Url: https://codereview.webrtc.org/2146013002
Cr-Commit-Position: refs/heads/master@{#13465}
Reason for revert:
It keeps breaking upstream.
Original issue's description:
> Reland Issue 2061423003: Refactor NACK bitrate allocation
>
> This is a reland of https://codereview.webrtc.org/2061423003/
> Which was reverted in https://codereview.webrtc.org/2131913003/
>
> The reason for the revert was that some upstream code used
> RtpSender::SetTargetBitrate(). I've added that back as a no-op until we
> it's been brought up to date.
>
> TBR=tommi@webrtc.org
>
> Committed: 05ce4ae31fTBR=tommi@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2130423002
Cr-Commit-Position: refs/heads/master@{#13419}
Reason for revert:
Breaks upstream code.
Original issue's description:
> Refactor NACK bitrate allocation
>
> Nack bitrate allocation should not be done on a per-rtp-module basis,
> but rather shared bitrate pool per call. This CL moves allocation to the
> pacer and cleans up a bunch if bitrate stats handling.
>
> BUG=
> R=danilchap@webrtc.org, stefan@webrtc.org, tommi@webrtc.org
>
> Committed: 5fc59e810bTBR=tommi@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2131913003
Cr-Commit-Position: refs/heads/master@{#13417}
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}
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension
The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.
Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.
BUG= webrtc:5895
Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
because in the TMMBRHelp class it is independent of other members.
BUG=webrtc:5565
R=philipel
Review-Url: https://codereview.webrtc.org/1746773002
Cr-Commit-Position: refs/heads/master@{#12669}
- "WebRTC.Video.SendDelayInMs"
Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator).
Add SendDelayStats class for computing delays.
Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer.
Wire up OnSentPacket to SendDelayStats.
BUG=webrtc:5215
Review-Url: https://codereview.webrtc.org/1478253002
Cr-Commit-Position: refs/heads/master@{#12600}
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
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}
Removes per-extension functions in ViEChannel/ViEReceiver and instead
register extensions directly on the RTP module by mapping extension
string to RTP-header-extension type.
BUG=webrtc:5494
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1740133002 .
Cr-Commit-Position: refs/heads/master@{#11786}
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.
Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}
TBR=tommi@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1737013002
Cr-Commit-Position: refs/heads/master@{#11762}
Reason for revert:
Breaks Chromium.
Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58cTBR=tommi@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1736663004
Cr-Commit-Position: refs/heads/master@{#11761}
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1703833002 .
Cr-Commit-Position: refs/heads/master@{#11747}
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}
Also move some stats reporting from vie_channel to send stats proxy
BUG=
Review URL: https://codereview.webrtc.org/1669623004
Cr-Commit-Position: refs/heads/master@{#11688}
TMMBN was capped by configured max bitrate for no apparent reason.
Removing this to not require payload-type reconfiguration on new
video-codec settings. Actual removal of payload-type reconfiguration
will happen in a pending CL.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1702043002 .
Cr-Commit-Position: refs/heads/master@{#11639}
Adds negotiation of rtx codecs for red and vp9. To keep backwards
compatibility with older Chrome versions, this change includes two
hacks:
1. Red packets will be retransmitted over the rtx codec associated with
vp8 if no rtx codec is associated with red. This is how Chrome does
it today and ensures that we still can send red over rtx to older
versions.
2. If rtx packets associated with the media codec (vp8/vp9 etc) are
received and red has been negotiated, we will assume that the sender
incorrectly has packetized red inside the rtx header associated with
media. We will therefore restore it with the red payload type
instead, which ensures that we can still receive rtx associated with
red from old versions.
Offering multiple rtx codecs to older versions should not be a problem
since old versions themselves only try to negotiate rtx for vp8.
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
BUG=webrtc:4024
TEST=Verified by running apprtc and emulating packet loss between Chrome with and without the patch.
Review URL: https://codereview.webrtc.org/1649493004 .
Cr-Commit-Position: refs/heads/master@{#11472}
SetRtpState function was updating only rtp_sender start timestamp.
Now it updates both rtp_sender and rtcp_sender start timestamps.
BUG=webrtc:5433
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1628323003 .
Cr-Commit-Position: refs/heads/master@{#11393}
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1571283002
Cr-Commit-Position: refs/heads/master@{#11336}
This CL makes sure no RTCP SR is sent before there is a valid timestamp
to set in the SR, based on the first sent media packet.
BUG=webrtc:1600
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1506103006 .
Cr-Commit-Position: refs/heads/master@{#10964}
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
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}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
For use when send-side bandwidth estimation is enabled.
Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1290813008
Cr-Commit-Position: refs/heads/master@{#9898}
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}
