- 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}
Include CVO in key frame.
Include CVO in delta frame when rotation changes.
Include CVO when it is non zero to support current receiver implementation.
BUG=webrtc:6600
Review-Url: https://codereview.webrtc.org/2452583002
Cr-Commit-Position: refs/heads/master@{#14784}
Rename kFullSendSideEstimator -> kSendSideEstimator and add new class SendSideBweSender (controlled by kSendSideEstimator) that actually uses the send side BWE.
Move the mock to logging/rtc_event_log/mock.
Allow congestion_controller, remote_bitrate_estimator and audio to depend on loggging/rtc_event_log
BUG=webrtc:6526
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2431093003
Cr-Commit-Position: refs/heads/master@{#14772}
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}
Reason for revert:
Downstream build is fixed.
Original issue's description:
> Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
>
> Reason for revert:
> Breaks downstream build.
>
> Original issue's description:
> > Ignore Camera and Flip bits in CVO when parsing video rotation
> >
> > Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
> > set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
> > The Camera and Flip bit is still unimplemented and will just be ignored
> > though.
> >
> > BUG=webrtc:6120
> > R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
> >
> > Committed: f9e1b922ef
>
> TBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6120
>
> Committed: https://crrev.com/97667c7746282704acccd896e26175decee349c0
> Cr-Commit-Position: refs/heads/master@{#14035}
TBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6120
Review-Url: https://codereview.webrtc.org/2320913003
Cr-Commit-Position: refs/heads/master@{#14124}
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 build.
Original issue's description:
> Ignore Camera and Flip bits in CVO when parsing video rotation
>
> Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
> set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
> The Camera and Flip bit is still unimplemented and will just be ignored
> though.
>
> BUG=webrtc:6120
> R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
>
> Committed: f9e1b922efTBR=pthatcher@webrtc.org,danilchap@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6120
Review-Url: https://codereview.webrtc.org/2300323002
Cr-Commit-Position: refs/heads/master@{#14035}
Currently, if WebRTC receives a CVO byte where the Camera or Flip bit is
set, then rotation is incorrectly parsed as 0. This CL fixes that issue.
The Camera and Flip bit is still unimplemented and will just be ignored
though.
BUG=webrtc:6120
R=danilchap@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2280703002 .
Cr-Commit-Position: refs/heads/master@{#14027}
When they are included there will be a mismatch between what the BWE says and
what the encoder is allowed to use, causing us to send more than the network
can handle.
BUG=webrtc:6247
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/2269923003 .
Cr-Commit-Position: refs/heads/master@{#13866}
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}
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}
- 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}
Reason for revert:
Not root cause for perf regression (regression still ongoing).
Original issue's description:
> Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ )
>
> Reason for revert:
> Speculative revert for perf regression.
>
> Original issue's description:
> > Remove VCMQmRobustness.
> >
> > Class contained a lot of not-really-wired-up functionality that ended up
> > being complicated ways of saying return 1; or return false;. This
> > removes this dependency that complicates code readability significantly.
> >
> > BUG=webrtc:5066
> > R=marpan@google.com, marpan@webrtc.org
> > TBR=stefan@webrtc.org
> >
> > Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> > Cr-Commit-Position: refs/heads/master@{#12516}
>
> TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5066, chromium:607838
>
> Committed: https://crrev.com/602316c3cd8556cc78d44f3ea4cd5fc8e70d9417
> Cr-Commit-Position: refs/heads/master@{#12572}
TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838
Review-Url: https://codereview.webrtc.org/1941643002
Cr-Commit-Position: refs/heads/master@{#12583}
Reason for revert:
Speculative revert for perf regression.
Original issue's description:
> Remove VCMQmRobustness.
>
> Class contained a lot of not-really-wired-up functionality that ended up
> being complicated ways of saying return 1; or return false;. This
> removes this dependency that complicates code readability significantly.
>
> BUG=webrtc:5066
> R=marpan@google.com, marpan@webrtc.org
> TBR=stefan@webrtc.org
>
> Committed: https://crrev.com/73894369791cb5eedc8788baf918ec07d11d351d
> Cr-Commit-Position: refs/heads/master@{#12516}
TBR=marpan@webrtc.org,stefan@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5066, chromium:607838
Review-Url: https://codereview.webrtc.org/1935753002
Cr-Commit-Position: refs/heads/master@{#12572}
Class contained a lot of not-really-wired-up functionality that ended up
being complicated ways of saying return 1; or return false;. This
removes this dependency that complicates code readability significantly.
BUG=webrtc:5066
R=marpan@google.com, marpan@webrtc.orgTBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1917083003 .
Cr-Commit-Position: refs/heads/master@{#12516}
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}
