This will make it possible to remove the build_with_libjingle=1 and key=''
GYP_DEFINES the bots are using (https://codereview.chromium.org/1450313002/).
It will also pave the road for enabling more WebRTC native tests on iOS.
BUG=webrtc:4755,webrtc:3185,webrtc:5165
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
Local compilation with:
GYP_DEFINES='OS=ios target_arch=arm' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm chromium_ios_signing=0' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=ia32' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1457053003 .
Cr-Commit-Position: refs/heads/master@{#10711}
Also clean up some include_dir entries and update the few
references to them with absolute include paths instead.
Finally fixed a few lint errors and invalid header guards.
None of these are used downstream.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1438663003 .
Cr-Commit-Position: refs/heads/master@{#10700}
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_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
R=stefan@webrtc.orgTBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440873005 .
Cr-Commit-Position: refs/heads/master@{#10699}
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
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
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
- Remove myself from OWNERS.
- Add myself to AUTHORS (I signed a CLA).
- Add minyue to audio_conference_mixer which would otherwise be empty.
- Add missing comma in WATCHLISTS.
Review URL: https://codereview.webrtc.org/1458763002
Cr-Commit-Position: refs/heads/master@{#10686}
Removes VP8::Encode trace in favor of VCMGenericEncoder ones and adds
one to InitEncode. Also adds an instant event to ::Encoded since this
can be done on a different thread.
Also adds the corresponding traces to VCMGenericDecoder.
BUG=webrtc:5167
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1412573010
Cr-Commit-Position: refs/heads/master@{#10674}
-Moved the initialize function
-Moved api_format into the shared state
BUG=
Review URL: https://codereview.webrtc.org/1413093002
Cr-Commit-Position: refs/heads/master@{#10668}
Reason for revert:
Potentially breaks a threading test under DrMemory. Rolling back while I investigate.
Original issue's description:
> Several Tick counter improvements.
>
> Move logic into cc file
> Simplify interval calculation
> Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
> Remove double divide on each ::Now() invocation on mac
>
> Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
>
> BUG=
> R=mflodman@webrtc.org, pbos@webrtc.org
>
> Committed: https://crrev.com/4c27e4b62da2047063d88eedfeec3e939fea7843
> Cr-Commit-Position: refs/heads/master@{#10661}
TBR=pbos@webrtc.org,mflodman@webrtc.org,noahric@chromium.org,thaloun@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1450203002
Cr-Commit-Position: refs/heads/master@{#10663}
Move logic into cc file
Simplify interval calculation
Remove unused QUERY_PERFORMANCE_COUNTER windows implementation
Remove double divide on each ::Now() invocation on mac
Move TickTime and TickInterval funcitons to cc file in prep for refactoring.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1415923010 .
Cr-Commit-Position: refs/heads/master@{#10661}
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
The header guards are also identical to avoid mixing them up in the transition.
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
Review URL: https://codereview.webrtc.org/1418913006
Cr-Commit-Position: refs/heads/master@{#10659}
This is to prevent size_t from undefined. This does not happen in current WebRTC since the sources that opus_inst.h gets used have proper definitions. But it would be good to add the definition in itself.
Review URL: https://codereview.webrtc.org/1446093003
Cr-Commit-Position: refs/heads/master@{#10653}
This leaves CodecOwner without a job, so we eliminate it.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1443653004
Cr-Commit-Position: refs/heads/master@{#10650}
to match name given in the RFC5450
private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
to make class usable for parsing packet
Review URL: https://codereview.webrtc.org/1434213004
Cr-Commit-Position: refs/heads/master@{#10636}
Put VideoSender/VideoReceiver flat within the object, not as
scoped_ptrs, giving fewer allocations and looking a bit nicer.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1443613002
Cr-Commit-Position: refs/heads/master@{#10634}
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
This CL attempts to annotate accesses on >16 API levels using as
small scopes as possible. The TargetApi notations mean "yes, I know
I'm accessing a higher API and I take responsibility for gating the
call on Android API level". The Encoder/Decoder classes are annotated
on the whole class, but they're only accessed through JNI; we should
annotate on method level otherwise and preferably on private methods.
This patch also fixes some compiler-level deprecation warnings (i.e.
-Xlint:deprecation), but probably not all of them.
BUG=webrtc:5063
R=henrika@webrtc.org, kjellander@webrtc.org, magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1412673008 .
Cr-Commit-Position: refs/heads/master@{#10624}
This CL does two things:
1) Improves stability in the existing OpenSL ES implementation for devices that
supports OpenSL ES. The cost is a slight increase in latency since the focus here
has been on avoiding audio glitches.
2) Adds a new Java API to exclude usage of OpenSL ES to enable comparisons between
OpenSL ES and Java based audio backends.
BUG=b/22452539
Review URL: https://codereview.webrtc.org/1440623002
Cr-Commit-Position: refs/heads/master@{#10618}
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.
BUG=None
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1405023016
Cr-Commit-Position: refs/heads/master@{#10594}
It can be computed from other members, notably the current encoder's
number of channels.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1423803007
Cr-Commit-Position: refs/heads/master@{#10585}
Receiving RTCP often caused the worker thread to stall for >20 ms
(>100ms observed) due to contention on VideoSender's send_crit_ (used to
protect encoding).
This change removes an unnecessary acquire of send_crit_ and caches
encoder settings in ViEEncoder instead of acquiring them through
::SendCodec() in VCM (which is blocking).
BUG=webrtc:5106
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1433703002 .
Cr-Commit-Position: refs/heads/master@{#10582}
