Reason for revert:
Speculative revet for breaking remoting_unittests in fyi bots.
https://build.chromium.org/p/chromium.webrtc.fyi/waterfall?builder=Win7%20Tester
Original issue's description:
> Add a flags field to video timing extension.
>
> The rtp header extension for video timing shuold have an additional
> field for signaling metadata, such as what triggered the extension for
> this particular frame. This will allow separating frames select because
> of outlier sizes from regular frames, for more accurate stats.
>
> This implementation is backwards compatible in that it can read video
> timing extensions without the new flag field, but it always sends with
> it included.
>
> BUG=webrtc:7594
>
> Review-Url: https://codereview.webrtc.org/3000753002
> Cr-Commit-Position: refs/heads/master@{#19353}
> Committed: cf5d485e14TBR=danilchap@webrtc.org,kthelgason@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2995953002
Cr-Commit-Position: refs/heads/master@{#19360}
The rtp header extension for video timing shuold have an additional
field for signaling metadata, such as what triggered the extension for
this particular frame. This will allow separating frames select because
of outlier sizes from regular frames, for more accurate stats.
This implementation is backwards compatible in that it can read video
timing extensions without the new flag field, but it always sends with
it included.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/3000753002
Cr-Commit-Position: refs/heads/master@{#19353}
That way, the debug printout will tell us which of x and y that was false.
BUG=none
Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
Found via supersize query:
size_info.symbols.WhereFullNameMatches(r'\bk[A-Z]').WhereInSection('d')
This moves 90 symbols from .data -> .data.rel.ro (5.50kb)
BUG=chromium:747064
Review-Url: https://codereview.webrtc.org/2986163002
Cr-Commit-Position: refs/heads/master@{#19274}
In preparation of making RTP packet demuxing many-to-one (one SSRC goes to one sink, but one sink may have multiple SSRCs), we need to remove FlexFEC's dependence on being able to register itself with the demuxer. Instead, we register FlexFEC streams with the streams they protect; when those streams get packets, they'll forward them to their associated FlexFEC streams, too.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2974453002
Cr-Commit-Position: refs/heads/master@{#19219}
rtcp_sender accepts nullptr as indication statistics shouldn't be used,
Other uses of NullReceiveStatistcs were already deleted.
BUG=webrtc:8016
Review-Url: https://codereview.webrtc.org/2988143002
Cr-Commit-Position: refs/heads/master@{#19197}
Compose them while creating sr/rr instead of presaving in temporary
member variable
BUG=webrtc:5565, webrtc:8016
Review-Url: https://codereview.webrtc.org/2979413002
Cr-Commit-Position: refs/heads/master@{#19138}
The critical-section's scope can be shrunk (we can hold the lock for a shorter time).
BUG=None
Review-Url: https://codereview.webrtc.org/2984973002
Cr-Commit-Position: refs/heads/master@{#19137}
use (already supported) nullptr as indication for no statistics
BUG=webrtc:8016
Review-Url: https://codereview.webrtc.org/2983363002
Cr-Commit-Position: refs/heads/master@{#19129}
Allows using sizeof() on the class constants and reduces space usage by
a pointer.
Bug: None
Change-Id: Ie919b13094903d50bdadc92b23a5aa5b6cc100ec
Reviewed-on: https://chromium-review.googlesource.com/581878
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19113}
This is the first in a series of CLs to add support for media
identification as part of unified plan SDP.
Bug: webrtc:4050
Change-Id: I0eb5639d240a9a1412c2b047a33d5112e4901f26
Reviewed-on: https://chromium-review.googlesource.com/576374
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19111}
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
Reason for revert:
Fix RtpStreamReceiver to not recover RTX packets with incorrect SSRC.
Original issue's description:
> Revert of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #5 id:120001 of https://codereview.webrtc.org/2893293003/ )
>
> Reason for revert:
> Breaks fuzzer.
>
> Original issue's description:
> > Only compare sequence numbers from the same SSRC in ForwardErrorCorrection.
> >
> > Prior to this CL, the ForwardErrorCorrection state would be reset whenever
> > the difference in sequence numbers of the last recovered media packet
> > and the new packet (media or FEC) was too large. This comparison did not
> > take into account that FlexFEC uses a different SSRC for the FEC packets,
> > meaning that the the state would be reset very frequently when FlexFEC
> > is used. This should not have led to any major problems, except for a
> > decreased decoding efficiency.
> >
> > This CL verifies that whenever we compare sequence numbers in
> > ForwardErrorCorrection, they do indeed belong to the same SSRC.
> >
> > BUG=webrtc:5654
> >
> > Review-Url: https://codereview.webrtc.org/2893293003
> > Cr-Commit-Position: refs/heads/master@{#18399}
> > Committed: 1476a9d789
>
> TBR=stefan@webrtc.org,holmer@google.com
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2919313005
> Cr-Commit-Position: refs/heads/master@{#18446}
> Committed: 92732ecc5cR=stefan@webrtc.org
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2918333002
Cr-Commit-Position: refs/heads/master@{#18827}
Don't disable fec for packets without timing frames extension
even if they are marked as belonging to timing frames.
BUG=webrtc:7894
Review-Url: https://codereview.webrtc.org/2956263002
Cr-Commit-Position: refs/heads/master@{#18826}
Reenable pacer_exit timestamp updates for the timing frames and
exclude timing-frames carrying packets from the FEC.
BUG=webrtc:7859
Review-Url: https://codereview.webrtc.org/2947133002
Cr-Commit-Position: refs/heads/master@{#18702}
Timing information is gathered in EncodedImage,
starting at encoders. Then it's sent using RTP header extension. In the
end, it's gathered at the GenericDecoder. Actual reporting and tests
will be in the next CLs.
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2911193002
Cr-Commit-Position: refs/heads/master@{#18659}
1. To make the files conform to chromium-style guidelines, and stop the compiler from complaing:
1.1. Move constructors out of .h file.
1.2. Move destructors out of .h file.
1.3. Move virtual functions out of .h file.
2. BlockLength() and Create() did not have consistent access modifiers in the various subclasses of RtcpPacket. Change the access level to public throughout.
3. Reorder BlockLength() and Create() where necessary, to reflect the order defined in the parent class (RtcpPacket).
BUG=None
Review-Url: https://codereview.webrtc.org/2937403002
Cr-Commit-Position: refs/heads/master@{#18633}
Specifically, just like SafeMin() and SafeMax() it handles all
combinations of integer and all
combinations of floating-point arguments by picking a
result type that is guaranteed to be able to hold the result.
This CL also replaces a bunch of std::min + std:max call pairs with
calls to SafeClamp()---the ones that could easily be found by grep
because "min" and "max" were on the same line. :-)
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2808513003
Cr-Commit-Position: refs/heads/master@{#18542}
This class is video-specific, and we want to free the name
"RtpStreamReceiver" so it can be reused for a media-independent RTP
receive class.
Also renames related files.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2926253002
Cr-Commit-Position: refs/heads/master@{#18510}
Reason for revert:
Breaks fuzzer.
Original issue's description:
> Only compare sequence numbers from the same SSRC in ForwardErrorCorrection.
>
> Prior to this CL, the ForwardErrorCorrection state would be reset whenever
> the difference in sequence numbers of the last recovered media packet
> and the new packet (media or FEC) was too large. This comparison did not
> take into account that FlexFEC uses a different SSRC for the FEC packets,
> meaning that the the state would be reset very frequently when FlexFEC
> is used. This should not have led to any major problems, except for a
> decreased decoding efficiency.
>
> This CL verifies that whenever we compare sequence numbers in
> ForwardErrorCorrection, they do indeed belong to the same SSRC.
>
> BUG=webrtc:5654
>
> Review-Url: https://codereview.webrtc.org/2893293003
> Cr-Commit-Position: refs/heads/master@{#18399}
> Committed: 1476a9d789TBR=stefan@webrtc.org,holmer@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2919313005
Cr-Commit-Position: refs/heads/master@{#18446}
Reason for revert:
Breaks tests in downstream projects.
Original issue's description:
> Protect new header extension by field trial experiment to allow hardcoding it in SDP
>
> BUG=chrome:718738
>
> Review-Url: https://codereview.webrtc.org/2922683002
> Cr-Commit-Position: refs/heads/master@{#18409}
> Committed: cafa1d6bbeTBR=sprang@webrtc.org,asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chrome:718738
Review-Url: https://codereview.webrtc.org/2922723002
Cr-Commit-Position: refs/heads/master@{#18414}
Move it to include path of the rtp_rtcp module to indicate it is ok to include it outside of the module.
Renamed to match the class it introduce and to reduce confusion with rtp_header_extensions.h
Bug: webrtc:5565
Change-Id: Ic4b4e9f6b75cb9275e23539cd6e88632c1e7c8d2
Reviewed-on: https://chromium-review.googlesource.com/520947
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18402}
Prior to this CL, the ForwardErrorCorrection state would be reset whenever
the difference in sequence numbers of the last recovered media packet
and the new packet (media or FEC) was too large. This comparison did not
take into account that FlexFEC uses a different SSRC for the FEC packets,
meaning that the the state would be reset very frequently when FlexFEC
is used. This should not have led to any major problems, except for a
decreased decoding efficiency.
This CL verifies that whenever we compare sequence numbers in
ForwardErrorCorrection, they do indeed belong to the same SSRC.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2893293003
Cr-Commit-Position: refs/heads/master@{#18399}
Also deletes a couple of includes of call.h, which seem
unnecessary.
BUG=None
Review-Url: https://codereview.webrtc.org/2907403003
Cr-Commit-Position: refs/heads/master@{#18340}
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.
BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc
Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
Before this CL, the RTP state would be re-randomized after a
recreation of VideoSendStream. That might lead to us sending
a non-compliant RTP stream, which is avoided after the
changes in this CL.
BUG=webrtc:5654
TBR=pbos@webrtc.org # Trivial change to fuzzer.
Review-Url: https://codereview.webrtc.org/2912713002
Cr-Commit-Position: refs/heads/master@{#18322}
