All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
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}
The CL in https://codereview.webrtc.org/2918333002/ enabled
FecTest.FlexfecTest and also added a sequence number offset between
the FEC packets and the media packets. This was to simulate that the
sequence numbers were generated from different spaces, i.e., that they
belong to different SSRCs.
The test does not account for sequence number wraparound, which means
that it could fail when the sequence number offset realization was large.
This CL fixes the problem by ensuring that the offset always lies in
[0, 2^15].
This CL also fixes spelling of UlpfecTest.
BUG=webrtc:7912
TESTED=ninja -C out/Debug && third_party/gtest-parallel/gtest-parallel --gtest_filter="*Flexfec*" -r 1000 out/Debug/modules_tests
Review-Url: https://codereview.webrtc.org/2966753002
Cr-Commit-Position: refs/heads/master@{#18863}
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 is to make it possible to override the rtc_task_queue target only.
BUG=none
Review-Url: https://codereview.webrtc.org/2931273002
Cr-Commit-Position: refs/heads/master@{#18534}
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}
Reason for revert:
Fixing and re-landing.
Original issue's description:
> Revert of Enabling `gn check` on webrtc/test (patchset #9 id:160001 of https://codereview.webrtc.org/2911203002/ )
>
> Reason for revert:
> ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
> #include "webrtc/base/checks.h"
> ^-------------------
> The target:
> //webrtc/test:fileutils_unittests
> is including a file from the target:
> //webrtc/base:rtc_base_approved
>
> It's usually best to depend directly on the destination target.
> In some cases, the destination target is considered a subcomponent
> of an intermediate target. In this case, the intermediate target
> should depend publicly on the destination to forward the ability
> to include headers.
>
> Dependency chain (there may also be others):
> //webrtc/test:fileutils_unittests -->
> //webrtc/test:fileutils --[private]-->
> //webrtc/base:rtc_base_approved
>
>
> Original issue's description:
> > Enabling `gn check` on webrtc/test
> >
> > BUG=webrtc:6828
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2911203002
> > Cr-Commit-Position: refs/heads/master@{#18372}
> > Committed: db5bb404b0
>
> TBR=kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6828
>
> Review-Url: https://codereview.webrtc.org/2920763002
> Cr-Commit-Position: refs/heads/master@{#18375}
> Committed: 1a6f143d07TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2918793002
Cr-Commit-Position: refs/heads/master@{#18376}
Reason for revert:
ERROR at //webrtc/test/testsupport/fileutils_unittest.cc:20:11: Can't include this header from here.
#include "webrtc/base/checks.h"
^-------------------
The target:
//webrtc/test:fileutils_unittests
is including a file from the target:
//webrtc/base:rtc_base_approved
It's usually best to depend directly on the destination target.
In some cases, the destination target is considered a subcomponent
of an intermediate target. In this case, the intermediate target
should depend publicly on the destination to forward the ability
to include headers.
Dependency chain (there may also be others):
//webrtc/test:fileutils_unittests -->
//webrtc/test:fileutils --[private]-->
//webrtc/base:rtc_base_approved
Original issue's description:
> Enabling `gn check` on webrtc/test
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2911203002
> Cr-Commit-Position: refs/heads/master@{#18372}
> Committed: db5bb404b0TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2920763002
Cr-Commit-Position: refs/heads/master@{#18375}
Delete unused member |rtp_receiver_|, and simplify a return statement.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2912363002
Cr-Commit-Position: refs/heads/master@{#18354}
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}
Change packetizer interface to explicitly return number of packets
instead of a last flag. Account for extra space needed in the last
packet.
BUG=webrtc:7588,webrtc:7594
Review-Url: https://codereview.webrtc.org/2871173008
Cr-Commit-Position: refs/heads/master@{#18244}
Packet::GetHeader() wasn't copying playout_delay. As result
playout_delay was ignored when flexfec is enabled.
Patch by Rob McCool <rmccool@google.com>
BUG=webrtc:7590
Review-Url: https://codereview.webrtc.org/2899553003
Cr-Commit-Position: refs/heads/master@{#18218}
Since the RtpStreamId and RepairedRtpStreamId extensions can have variable
length, it makes no sense for them to have a constant valueSize field.
The header length calculation in RtpHeaderExtensionMap needed to be changed
for this because it previously worked with the assumption that all header
types have a constant size. Now it's the caller's job to specify the length
of the extensions that it might use.
BUG=webrtc:7433
Review-Url: https://codereview.webrtc.org/2867713003
Cr-Commit-Position: refs/heads/master@{#18179}
