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}
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}
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}
The packet size was only used to control how often to output DTMF
packets. However, it likely did not work as intended, since that
interval was only set during initialization. No changes to the packet
size, like what AudioEncoder::Num10MsFramesInNextPacket could
indicate, were picked up. The value was instead taken from an entry in
ACMCodecDB.
Since it was not-fully-functional, its exact value didn't seem to
matter and it was getting in the way of making it possible to supply
an external audio encoder factory, I've decided to remove it
altogether. The DTMF code now uses an interval of 50 ms regardless,
which is a value recommended by the RFC.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2545753002
Cr-Commit-Position: refs/heads/master@{#15380}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
This function has no public use,
removed tests calling it: effect of registering extension is better
tested in AllocatePacket and SendPacket tests.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2530363002
Cr-Commit-Position: refs/heads/master@{#15258}
Prior to this change, FlexFEC packets that were paced would be lost in
the RTPSender, since they were not stored in a packet history. This CL
introduces such a packet history, as well as the needed wireup for
higher layers to be aware that the particular RTPSender is able to
send FlexFEC packets with a particular SSRC.
Updated RTPSender unit test to reflect the fact that paced packets
are now actually sent.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2491293002
Cr-Commit-Position: refs/heads/master@{#15066}
- Change const ptr to const ref in parameter list.
Using nullptr as argument was invalid, so no need to send
pointer instead of reference.
- Change return type to void or bool, where appropriate
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2455963003
Cr-Commit-Position: refs/heads/master@{#14945}
Prior to this change, we signalled that ULPFEC was disabled
through a bool, but that RED was disabled by setting its
payload type to -1. The latter is consistent with how we
disable RED/ULPFEC in the config, so this CL removes the
ULPFEC bool from the {,Set}UlpfecConfig chain of member
functions.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2460533002
Cr-Commit-Position: refs/heads/master@{#14944}
At the same time, change to using int's instead of uint8_t's for the payload type.
This allows us to signal disabled FEC or RED using the sentinel value -1, which
is commonplace in other parts of the code.
These APIs will be deprecated when ULPFEC is deprecated.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2448463003
Cr-Commit-Position: refs/heads/master@{#14942}
Remove functions to enumerate all extensions,
Remove concept of the inactive extension.
Decision if extension should be included into rtp header is done by rtp_sender
GetTotalLengthInBytes now calculates all extension, included or not.
That is used only for calculating how much space to reserve for fec.
Since extension might suddenly be included in the next packet (which still might belong to same fec group), it is safer to calculate all registered extension.
BUG=webrtc:5565, webrtc:1994
Review-Url: https://codereview.webrtc.org/2431253003
Cr-Commit-Position: refs/heads/master@{#14763}
The helpers intended to replace and deprecate BuildRtpHeader when
RtpSenderAudio/RtpSenderVideo will be updated to pass RtpPacket class
instead of raw buffer for sending.
BUG=webrtc:5261
R=sprang@webrtc.org
Review URL: https://codereview.webrtc.org/2303283002 .
Cr-Commit-Position: refs/heads/master@{#14051}
Reason for revert:
Breaks downstream code.
Original issue's description:
> StartTimestamp generated randomly in RtpSender constructor
> instead of not-randomly at SetSendingState(true)
> Renamed to timestamp_offset_ to better match meaning of the variable.
>
> R=asapersson@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> Cr-Commit-Position: refs/heads/master@{#13796}
TBR=asapersson@webrtc.org,terelius@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/2248413002
Cr-Commit-Position: refs/heads/master@{#13798}
The last in-tree call site recently disappeared, so they were unused.
BUG=webrtc:5922
Review-Url: https://codereview.webrtc.org/2066473002
Cr-Commit-Position: refs/heads/master@{#13751}
Reason for revert:
broke browser_tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/4c7f4cd2ef76821edca6d773d733a924b0bedd25
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Original-Commit-Position: refs/heads/master@{#13613}
> Cr-Commit-Position: refs/heads/master@{#13615}
TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2203233002
Cr-Commit-Position: refs/heads/master@{#13616}
Reason for revert:
broke internal tests
Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}
TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691
Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
double check rtp_sender in sending mode when altering sequence_number
adjust test to skip validating timestamp on rtx streams
fix test by waiting for all 3 media streams instead of 3 out 6 media and rtx streams.
BUG=webrtc:4332
Review-Url: https://codereview.webrtc.org/2177523002
Cr-Commit-Position: refs/heads/master@{#13587}
- 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}
This is an issue if the sequence numbers are to be used to compute packet loss statistics since it introduces gaps which are not related to loss.
Also making sure that the header extensions are properly guarded by the send crit sect.
Review-Url: https://codereview.webrtc.org/2190913002
Cr-Commit-Position: refs/heads/master@{#13557}
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}
- "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}
