The callback keeps a reference to an object until the callback goes out of scope.
Review URL: https://codereview.webrtc.org/1487493002
Cr-Commit-Position: refs/heads/master@{#10847}
This should never be using a null decoder, but it looks like it's
crashing out in the field. Adding a CHECK to see if it catches any
interesting stack traces.
Also making the _decoder pointer const to show that it should never be
changing.
BUG=chromium:563299
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1485713002 .
Cr-Commit-Position: refs/heads/master@{#10843}
-Renamed the TimeToFrequency and FrequencyToTime functions.
-Moved the windowing from the TimeToFrequency function.
-Simplified the EchoSubtraction function.
Note that the aec state is still an input to the EchoSubtraction function, and it currently needs to be that in order to support the output of the debug file. The longer-term goal is, however, to order the state into substates. This will simplify the parameter lists to the EchoCancellation function as well as replace the aec state as a parameter
BUG=webrtc:5201
Review URL: https://codereview.webrtc.org/1456123003
Cr-Commit-Position: refs/heads/master@{#10830}
In https://codereview.webrtc.org/1481493004/ some duplicated headers
were left to make it possible to update downstream without breakage.
Now that's done and we can remove these to avoid confusion.
BUG=webrtc:5095
TBR=henrik.lundin@webrtc.org, kwiberg@webrtc.org
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/1477423002
Cr-Commit-Position: refs/heads/master@{#10829}
Multiple decoders were used for the same payload type in this test case,
causing CHECK failures when configuring.
BUG=webrtc:5249
TBR=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1484443003 .
Cr-Commit-Position: refs/heads/master@{#10825}
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1446513002
Cr-Commit-Position: refs/heads/master@{#10823}
Prevents double-initialization of decoders due to resolution changes
between initial database settings and first incoming frame.
BUG=webrtc:5251
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1474193002 .
Cr-Commit-Position: refs/heads/master@{#10822}
Also adds a RTC_CHECK in VideoReceiveStream that verifies that decoders
aren't null, since this will attempt to deregister a codec which would
previously fail with an obscure stack trace not indicating what actually
was wrong.
BUG=webrtc:5249
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1479793002 .
Cr-Commit-Position: refs/heads/master@{#10821}
Also changes presubmit script to not run cpplint on objc dirs.
BUG=
Review URL: https://codereview.webrtc.org/1467173006
Cr-Commit-Position: refs/heads/master@{#10815}
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}
