Reason for revert:
Breaks downstream compilation using webrtc/common_audio/real_fourier.h. Let's chat tomorrow on how to coordinate a re-land.
Original issue's description:
> Replace scoped_ptr with unique_ptr in webrtc/common_audio/
>
> BUG=webrtc:5520
>
> Committed: https://crrev.com/79d7a499c0c3e1de8f5ad1138236f0386701053f
> Cr-Commit-Position: refs/heads/master@{#11716}
TBR=henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1726043002
Cr-Commit-Position: refs/heads/master@{#11726}
Removes addition of at least one zero sample in webrtc_perf_tests that
can skew stats differently depending on how often these stats are
updated. Unclear if this skewing is different between now and before.
BUG=chromium:585071, chromium:586216
R=sprang@google.com, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1727583003 .
Cr-Commit-Position: refs/heads/master@{#11720}
This allows other projects to more easily depend on this.
The plan is to move remote_bitrate_estimator and bitrate_controller into this module and reduce the exposed interface to only a simplified version of congestion_controller.h.
No functional changes in this CL.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1718473002 .
Cr-Commit-Position: refs/heads/master@{#11718}
Doesn't utilize the clock or any callbacks out of the receiver but
should still be useful to test input packet parsing.
BUG=webrtc:4771
R=danilchap@webrtc.org
Review URL: https://codereview.webrtc.org/1716143002 .
Cr-Commit-Position: refs/heads/master@{#11717}
Boost low QP threashold to 21, otherwise VGA encoding never
scales up even at 2.5 Mbps.
Also reduce high QP threshold to scale down faster.
BUG=b/26504665
R=jackychen@google.com
Review URL: https://codereview.webrtc.org/1717763003 .
Cr-Commit-Position: refs/heads/master@{#11712}
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}
Moves RtpRtcp module pointers into VideoSendStream and uses them for
simple calls that were only forwarded by ViEChannel.
BUG=webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1693553002 .
Cr-Commit-Position: refs/heads/master@{#11709}
Reduces contention on event_mutex_ while taking gettimeofday(). Impact
highly hypothetical at this point, but less locking is better.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1716563003 .
Cr-Commit-Position: refs/heads/master@{#11706}
avoid clang warnings.
The changes does not change any of the functionality
in the code.
BUG=webrtc:163
Review URL: https://codereview.webrtc.org/1710083006
Cr-Commit-Position: refs/heads/master@{#11705}
This is the only failing test of the currently deployed
at the iOS Simulator bots. Let's disable it so we can promote
the passing tests to the main waterfall and the trybots.
BUG=4755
TBR=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1722523002 .
Cr-Commit-Position: refs/heads/master@{#11703}
Landing https://codereview.webrtc.org/1675923002/ broke some Chromium FYI bots
because the GN build didn't include "sharedexclusivelock.cc" in that scenario.
This CL moves the files from the non-Chromium block into the common sources
list.
BUG=webrtc:5496
Review URL: https://codereview.webrtc.org/1712773003
Cr-Commit-Position: refs/heads/master@{#11699}
Since SSRCs can no longer change on the fly, SSRC code can be made a lot
simpler (and faster). Resulting code has less and shorter locking.
BUG=webrtc:5494
R=danilchap@webrtc.org
Review URL: https://codereview.webrtc.org/1713683003 .
Cr-Commit-Position: refs/heads/master@{#11691}
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}
EncoderStateFeedback is now only connected to one encoder, so remove map
and other complexity to deliver feedback more directly.
BUG=webrtc:5494
R=danilchap@webrtc.org
Review URL: https://codereview.webrtc.org/1706803002 .
Cr-Commit-Position: refs/heads/master@{#11687}
Reason for revert:
Broke chromium.webrtc.fyi bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/9891https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20GN/builds/11416
Fails with
-----
Undefined symbols for architecture x86_64:
"rtc::SharedExclusiveLock::LockShared()", referenced from:
rtc::MessageQueue::DoDestroy() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::socketserver() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::WakeUpSocketServer() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::Quit() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::Get(rtc::Message*, int, bool) in librtc_base.a(messagequeue.o)
rtc::MessageQueue::Post(rtc::MessageHandler*, unsigned int, rtc::MessageData*, bool) in librtc_base.a(messagequeue.o)
rtc::MessageQueue::DoDelayPost(int, unsigned int, rtc::MessageHandler*, unsigned int, rtc::MessageData*) in librtc_base.a(messagequeue.o)
...
"rtc::SharedExclusiveLock::UnlockShared()", referenced from:
rtc::MessageQueue::DoDestroy() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::socketserver() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::WakeUpSocketServer() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::Quit() in librtc_base.a(messagequeue.o)
rtc::MessageQueue::Get(rtc::Message*, int, bool) in librtc_base.a(messagequeue.o)
rtc::MessageQueue::Post(rtc::MessageHandler*, unsigned int, rtc::MessageData*, bool) in librtc_base.a(messagequeue.o)
rtc::MessageQueue::DoDelayPost(int, unsigned int, rtc::MessageHandler*, unsigned int, rtc::MessageData*) in librtc_base.a(messagequeue.o)
...
"rtc::SharedExclusiveLock::SharedExclusiveLock()", referenced from:
rtc::MessageQueue::MessageQueue(rtc::SocketServer*, bool) in librtc_base.a(messagequeue.o)
ld: symbol(s) not found for architecture x86_64
-----
Looks like these are compiling without "webrtc/base/sharedexclusivelock.cc".
Original issue's description:
> Prevent data race in MessageQueue.
>
> The CL prevents a data race in MessageQueue where the variable "ss_" is
> modified without a lock while sometimes read inside a lock.
>
> Also thread annotations have been added to the MessageQueue class.
>
> BUG=webrtc:5496
>
> Committed: https://crrev.com/df88460372e7ce78c871a87774d7e6d82aac6ee3
> Cr-Commit-Position: refs/heads/master@{#11683}
TBR=ivoc@webrtc.org,pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5496
Review URL: https://codereview.webrtc.org/1714463003
Cr-Commit-Position: refs/heads/master@{#11686}
Prevents allocating sequence numbers for packets that go out on the
network even though sending media is disabled.
This race caused a replay of sequence numbers when GetRtpState() on a
stopped stream would not return the last sequence number sent, since the
pacer thread could request and send padding on a later sequence number
before the modules are disconnected from the pacer.
BUG=webrtc:5543
R=stefan@webrtc.org
TEST=Repeating EndToEndTest.RestartingSendStreamPreservesRtpState 1000 times under TSan.
Review URL: https://codereview.webrtc.org/1715703002 .
Cr-Commit-Position: refs/heads/master@{#11685}
The CL prevents a data race in MessageQueue where the variable "ss_" is
modified without a lock while sometimes read inside a lock.
Also thread annotations have been added to the MessageQueue class.
BUG=webrtc:5496
Review URL: https://codereview.webrtc.org/1675923002
Cr-Commit-Position: refs/heads/master@{#11683}
There were two different structures named RtpPacket in webrtc namespace:
RtpPacket defined in fec_test_helper renamed to test::RawRtpPacket
RtpPacket defined in rtp_sender_video and producer_fec removed as unused
BUG=webrtc:5261
R=sprang@google.com, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1710103004 .
Cr-Commit-Position: refs/heads/master@{#11682}
Also added a test for Clear to ensure this invariant holds.
With this change, it is easy to empty a Buffer and reuse its storage. Further down the line, code filling data into a Buffer could be written to just append to it, with the caller determining if the Buffer should first be cleared or not.
There is currently only one use of Buffer::Clear (in AudioEncoderCopyRed::Reset()) and it should benefit from the change, by not requiring a reallocation after Reset.
Review URL: https://codereview.webrtc.org/1707693002
Cr-Commit-Position: refs/heads/master@{#11680}
This CL removes "build/c++11" from the cpplint filters. The same was
changed in "depot_tools" in https://codereview.chromium.org/1573663003/
From the other CL:
-----
The checks are not reliable for Rvalue references, and only are
allowing default/deleted constructors. They are based on the google3
internal rules which do not exactly match our own c++11 rules, and
may diverge more over time.
-----
NOTRY=True
Review URL: https://codereview.webrtc.org/1710293002
Cr-Commit-Position: refs/heads/master@{#11678}
It was hardly used, making the code more complex than needed and caused problems on iOS because it uses system.
BUG=webrtc:5549
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1708353002 .
Cr-Commit-Position: refs/heads/master@{#11677}
