Seconds and fractions parts of the ntp time presented with two values, but used as one.
This helper structure can make that use more clear.
(initially introduced into rtp_rtcp as https://codereview.webrtc.org/1435833003)
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1482593002
Cr-Commit-Position: refs/heads/master@{#10814}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
Enable cpplint and have it use a whitelist that also checks
in subdirectories.
Move the cpplint check so it runs before the pylint check
since that one always run and increases the time to errors
for cpplint.
Fix all cpplint errors in webrtc/video_engine.
BUG=webrtc:5149
TESTED=Fixed issues reported by:
find webrtc/video_engine -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.
R=pbos@chromium.org, phoglund@chromium.orgTBR=pbos@webrtc.org, phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1481723003 .
Cr-Commit-Position: refs/heads/master@{#10808}
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
Due to Chromium moving over to building with a sysroot
image on Linux in
a931efd5dc
we need to disable that until http://crbug.com/561584 is fixed
(libudev.h is missing and is used by talk/media/devices/libudevsymboltable.h).
Change log: 68cf0b8..aa8e58a
Full diff: 68cf0b8..aa8e58a
No dependencies changed.
No update to Clang.
BUG=chromium:561584
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1468313006
Cr-Commit-Position: refs/heads/master@{#10795}
Removes log disabling under Chromium which doesn't compile due to
missing LS_INFO in the override log implementation.
Also removes dependency on webrtc/test/BUILD.gn which doesn't build in
Chromium (due to third_party/gflags not being present). Instead the
no-op implementation of field_trials in system_wrappers is used.
BUG=chromium:561667, webrtc:4771
R=kjellander@webrtc.orgTBR=henrikg@webrtc.org
Review URL: https://codereview.webrtc.org/1473713004 .
Cr-Commit-Position: refs/heads/master@{#10793}
The corresponding set of overrides weren't moved when logging.cc etc.
was moved over. This wasn't noticed because all existing targets before
webrtc fuzzers used to link both rtc_base and rtc_base_approved in
Chromium. Also adding //base:base as a dependency, this used to be
linked in by other targets either way before but generated build errors
when a target solely depends on rtc_base_approved.
BUG=webrtc:4771
R=kjellander@webrtc.orgTBR=henrikg@webrtc.org
Review URL: https://codereview.webrtc.org/1473223005 .
Cr-Commit-Position: refs/heads/master@{#10792}
Reason for revert:
This breaks the Win32 Release [large tests] bot (webrtc_perf_tests times out after 1h23m): https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D
The Mac64 Release [large tests] bot's runtime also increased with +20 minutes.
These bot configs are not a part of the default trybot set, so please run them manually or add this to the CL description:
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal
Original issue's description:
> A unittest that reports the statistics for the duration of an APM stream processing API call.
>
> BUG=webrtc:5099
>
> Committed: https://crrev.com/880896ab0976bbf86a6753d0c900c70e51f421cb
> Cr-Commit-Position: refs/heads/master@{#10786}
TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org,peah@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1473733004
Cr-Commit-Position: refs/heads/master@{#10791}
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
This means that if a TURN server denies permission for an
unreachable address, we'll no longer ping it fruitlessly.
BUG=webrtc:4917
Review URL: https://codereview.webrtc.org/1415313004
Cr-Commit-Position: refs/heads/master@{#10789}
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
This overrides the default (10.6) in Chromium's
build/common.gypi. It's needed since we want ARC and libc++.
TESTED=Ran webrtc/build/gyp_webrtc before this patch and then
grep -r macosx-version-min out/Debug/* | grep 10.6
which gave a lot of output.
Then with this patch applied, there were no output for 10.6 (only 10.7).
R=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1474623002 .
Cr-Commit-Position: refs/heads/master@{#10783}
This is part of the project that makes RTC rendering more
smooth. We've already finished the developement of the
frame selection algorithm in WebMediaPlayerMS, where we
managed a frame pool, and based on the vsync interval, we
actively select the best frame to render in order to
maximize the rendering smoothness.
Thus the frame timeline control in IncomingVideoStream is
no longer needed, because with sophisticated frame
selection algorithm in WebMediaPlayerMS, the time control
in IncomingVideoStream will do nothing but add some extra
delay.
BUG=514873
Review URL: https://codereview.webrtc.org/1419673014
Cr-Commit-Position: refs/heads/master@{#10781}
Remove the headers that were kept to provide non-breaking updates
of downstream code for https://codereview.webrtc.org/1418913006/
and https://codereview.webrtc.org/1417283007/.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
Review URL: https://codereview.webrtc.org/1467173003
Cr-Commit-Position: refs/heads/master@{#10773}
This works aims to:
-More clearly separate the functionalities in the AEC.
-Make the inputs and outputs to functions more clear (currently the state struct is often passed as a parameter to the functions and the functions use members of the state both as inputs and outputs, which reduces the readability of the code and makes it difficult to change/refactor.
What is done in this CL:
-Most of what belongs to the echo subtraction functionality has been moved to a separate function.
-The NonLinearProcessing function has been renamed to EchoSuppressor which I think is more appropriate.
-Part of the code was replaced by a call to the TimeToFrequency function (which was also suggested by an existing todo).
-For consistency, a function FrequencyToTime doing the opposite of TimeToFrequency was added and part of the code was moved to that.
-The ScaleErrorSignal function was changed to no longer have the state as an input parameter. This entailed also changing the corresponding assembly optimized files accordingly.
Testing:
-The changes have been tested for bitexactness on Linux using a fairly extensive test.
-All the unittests pass on linux.
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1455163006
Cr-Commit-Position: refs/heads/master@{#10764}
With this in, the only compilation errors left seems
related to yasm and libjpeg_turbo.
Notice the below example builds x86 builds (not ARM) since if
specifying target_cpu="arm", the gn step fails (separate issue).
BUG=webrtc:5213, webrtc:5195, chromium:459705
TESTED=Passing compilation with:
gn gen --args="target_os=\"ios\"" out/Default
ninja -C out/Default rtc_base audio_device
Review URL: https://codereview.webrtc.org/1471663002
Cr-Commit-Position: refs/heads/master@{#10763}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
This is to make sure that we don't have any assert failure when running with adapter enumeration disabled and we have no IPv6 default local address.
BUG=webrtc:5061
Review URL: https://codereview.webrtc.org/1456663002
Cr-Commit-Position: refs/heads/master@{#10759}
This change allows us to delete AcmReceiver::last_audio_codec_id().
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1467183002
Cr-Commit-Position: refs/heads/master@{#10756}
We don't need it anymore now that we can use std::vector::data().
Review URL: https://codereview.webrtc.org/1470843003
Cr-Commit-Position: refs/heads/master@{#10755}
This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1467163002
Cr-Commit-Position: refs/heads/master@{#10754}
