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}
It do the following:
The SurfaceTexture.updateTexImage() calls are moved from the video renderers into MediaCodecVideoDecoder, and the destructor of the texture frames will signal MediaCodecVideoDecoder that the frame has returned. This CL also removes the SurfaceTexture from the native handle and only exposes the texture matrix instead, because only the video source should access the SurfaceTexture.
It moves the responsibility of calculating the decode time to Java.
Patchset2 Refactor MediaCodecVideoDecoder to drop frames if a texture is not released.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440343002 .
Cr-Commit-Position: refs/heads/master@{#10706}
Reason for revert:
Broke chromium fyi build.
Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}
TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1455233005
Cr-Commit-Position: refs/heads/master@{#10702}
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.
External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1416673006
Cr-Commit-Position: refs/heads/master@{#10701}
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}
Not fully tested yet. Verified in test loopback application
with fake VP9 codec factory.
Assume that encoder generates bitstream in non flexible mode with
one temporal and one spatial layers.
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1451953002 .
Cr-Commit-Position: refs/heads/master@{#10695}
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}
This breaks some client apps in annoying ways, so disable for now.
BUG=webrtc:4776
Review URL: https://codereview.webrtc.org/1461513003
Cr-Commit-Position: refs/heads/master@{#10693}
Since BundleFilter doesn't filter RTCP anymore we can have incoming
RTCPs for audio delivered to video, that delivery will fail when there
are no video receivers causing the log to be spammed.
BUG=webrtc:5223
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1458853002 .
Cr-Commit-Position: refs/heads/master@{#10687}
- 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}
In tcp only call:
Tested with hangout.
Tested with firefox.
To test firefox, goto about:config, search for media.peerconnection.ice.tcp and turn it on.
Existing test case should be suffice to cover this.
R=juberti@google.comTBR=jubert@webrtc.org
BUG=webrtc:3849
Review URL: https://codereview.webrtc.org/1217463004 .
Cr-Commit-Position: refs/heads/master@{#10683}
In preparation for implementing the standardized variant of CHACHA20_POLY1305
(it changed slightly in the standardization process),
TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305 and TLS1_CK_ECDHE_ECDSA_CHACHA20_POLY1305
were renamed to have an _OLD suffix with compatibility unsuffixed #defines
temporarily available.
Update references to include the _OLD suffixed ones. Once we've cycled through
the few consumers of the unsuffixed names (just WebRTC and QUIC), the unsuffixed
names can refer to the to-be-implemented standardized variant and eventually
the draft version will be removed.
(This has no effect on upstream OpenSSL compatibility as OpenSSL never defined
these symbols to begin with. Though probably they will once standardization is
done.)
BUG=none
Review URL: https://codereview.webrtc.org/1412803010
Cr-Commit-Position: refs/heads/master@{#10681}
Do not delete the turn port entry right away when the respective
connection is deleted. The dependency on asyncinvoker has been added
in chromium libjingle-nacl.
BUG=webrtc:5120
Review URL: https://codereview.webrtc.org/1450263002
Cr-Commit-Position: refs/heads/master@{#10679}
