Reason for revert:
ScopedMessageData can't be removed just yet. It broke an internal project.
Original issue's description:
> Delete unused class/template ScopedMessageData.
>
> This appears unused since cl https://codereview.webrtc.org/2564333002
>
> BUG=webrtc:6424
>
> Review-Url: https://codereview.webrtc.org/2652663002
> Cr-Commit-Position: refs/heads/master@{#16229}
> Committed: d83fb921a8TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2654753003
Cr-Commit-Position: refs/heads/master@{#16231}
Reason for revert:
Triggers leak on Linux memcheck (non-default trybot):
### BEGIN MEMORY TOOL REPORT (error hash=#0112A395AF2326BC#)
Command: ../Release/./modules_unittests --isolated-script-test-output=/b/s/w/ioUlJCnu/output.json --isolated-script-test-chartjson-output=/b/s/w/ioUlJCnu/chartjson-output.json --gtest_filter=-CommonFormats/AudioProcessingTest*
Leak_DefinitelyLost
45 bytes in 1 blocks are definitely lost in loss record 118 of 277
operator new[](unsigned long) (m_replacemalloc/vg_replace_malloc.c:363)
webrtc::video_coding::H264SpsPpsTracker::CopyAndFixBitstream(webrtc::VCMPacket*) (/b/s/w/irJgAGsR/out/Release/modules_unittests)
webrtc::video_coding::TestH264SpsPpsTracker_SpsPpsOutOfBand_Test::TestBody() (/b/s/w/irJgAGsR/out/Release/modules_unittests)
Suppression (error hash=#0112A395AF2326BC#):
For more info on using suppressions see http://dev.chromium.org/developers/tree-sheriffs/sheriff-details-chromium/memory-sheriff#TOC-Suppressing-memory-reports
{
<insert_a_suppression_name_here>
Memcheck:Leak
fun:_Zna*
fun:_ZN6webrtc12video_coding17H264SpsPpsTracker19CopyAndFixBitstreamEPNS_9VCMPacketE
fun:_ZN6webrtc12video_coding42TestH264SpsPpsTracker_SpsPpsOutOfBand_Test8TestBodyEv
}
### END MEMORY TOOL REPORT (error hash=#0112A395AF2326BC#)
Original issue's description:
> H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.
>
> - Changed method name to clarify that entire Nalus are expected.
> - Added unit test code.
> - Adjusted InsetSpsPpsNalus() implementation to above requirement.
>
> BUG=webrtc:5948
>
> Review-Url: https://codereview.webrtc.org/2638933002
> Cr-Commit-Position: refs/heads/master@{#16221}
> Committed: f53d7374cfTBR=philipel@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2649113003
Cr-Commit-Position: refs/heads/master@{#16225}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
- Changed method name to clarify that entire Nalus are expected.
- Added unit test code.
- Adjusted InsetSpsPpsNalus() implementation to above requirement.
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2638933002
Cr-Commit-Position: refs/heads/master@{#16221}
webrtcvoiceengine.cc ensured that if the bitrate set for ISAC was 0,
it was changed to -1 so that the codec could manage the bitrate
itself.
webrtcsdp.cc ensured that if the bitrate set for ISAC was 0, it was
explicitly set to default values to avoid the codec's built in bitrate
management.
Eventually, there'll be no codec specific code like this in these
layers. This is one step towards that goal.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2642923003
Cr-Commit-Position: refs/heads/master@{#16220}
First step in moving the content of webrtc/build up one level.
BUG=webrtc:7030
Review-Url: https://codereview.webrtc.org/2653593003
Cr-Commit-Position: refs/heads/master@{#16219}
This CL is to calculate packet loss metrics from TransportFeedback. The outcome of this will be passed down to audio encoder.
BUG=webrtc:6904
Review-Url: https://codereview.webrtc.org/2579613003
Cr-Commit-Position: refs/heads/master@{#16217}
https://codereview.webrtc.org/2514883002/ changed and moved these targets around but did not add public dependencies for the fallbacks, which causes gn gen --check a lot of anger.
NOTRY=true # Only build changes and windows bots are cranky atm.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2651663002
Cr-Commit-Position: refs/heads/master@{#16214}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
This script is similar to header_usage.sh, but it counts usage (number
of files) of each class defined in some header file.
E.g., the following classes appear unused,
AsyncHttpsProxyServerSocket
CompositeMediaEngineWithFakeVoiceEngine
DesktopId
RtpDataCallback
ScopedMessageData
ScreencastEventCatcher
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2649103002
Cr-Commit-Position: refs/heads/master@{#16204}
Reason for revert:
It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places.
Original issue's description:
> Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
>
> Reason for revert:
> This breaks some chromium.webrtc.fyi buildbots with the following error:
>
> ERROR Unresolved dependencies.
> //third_party/webrtc/base:base(//build/toolchain/android:android_arm)
> needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
>
>
> Original issue's description:
> > Creating libwebrtc bundle jar
> >
> > Creates a JAR which includes:
> > - //webrtc/base:base_java
> > - //webrtc/modules/audio_device:audio_device_java
> > - //webrtc/sdk/android:libjingle_peerconnection_java
> > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
> >
> > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2646443002
> > Cr-Commit-Position: refs/heads/master@{#16189}
> > Committed: a62a82b7e7
>
> TBR=kjellander@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2640023010
> Cr-Commit-Position: refs/heads/master@{#16190}
> Committed: 3c9151b953TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646093004
Cr-Commit-Position: refs/heads/master@{#16203}
This is used for fallback from DTLS to SDES encryption, which we probably still
want to support. Setting a certificate puts the DTLS transport in a "DTLS
enabled" mode, so it should be delayed until SDP with "a=fingerprint" is set.
BUG=webrtc:6972
Review-Url: https://codereview.webrtc.org/2641633002
Cr-Commit-Position: refs/heads/master@{#16199}
Pulls in unnecessary GTK dependencies that breaks the chromium GTK3
build. This removes the last of webrtc/media/devices.
BUG=chromium:668446
Review-Url: https://codereview.webrtc.org/2646793008
Cr-Commit-Position: refs/heads/master@{#16198}
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.
Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: eaa826c2eeTBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16196}
The 'src' path referred to in the PATENTS.txt has been renamed
to 'webrtc'.
BUG=webrtc:7021
Review-Url: https://codereview.webrtc.org/2640143004
Cr-Commit-Position: refs/heads/master@{#16195}
These attributes should be parsed in candidate trickling, but when
parsing a full session description, only "a=ice-ufrag"/"a=ice-pwd"
should be used to communicate the ufrag/password.
BUG=chromium:681286
Review-Url: https://codereview.webrtc.org/2639183002
Cr-Commit-Position: refs/heads/master@{#16194}
Initial provisioning was already done in build_ios_libs.sh to support
building the WebRTC framework or static library for iOS (tvOS, watchOS)
with bitcode. Still, the actual build configuration would need to be
modified for each and every part of the build, including 3rd-party libs.
Thus, doing that more universally, at the build/config level, would be
desirable – and actually necessary to provide the intended support.
The patch for enhancing the Chromium build configs with that specific
option was landed in https://codereview.chromium.org/2631573002
NOTRY=True
BUG=webrtc:5085
Review-Url: https://codereview.webrtc.org/2633643003
Cr-Commit-Position: refs/heads/master@{#16193}
The folder webrtc/sdk/ now contains android as well, so we should move the objc DEPS file to the objc folder.
TBR=tommi
BUG=None
Review-Url: https://codereview.webrtc.org/2644733008
Cr-Commit-Position: refs/heads/master@{#16191}
Reason for revert:
This breaks some chromium.webrtc.fyi buildbots with the following error:
ERROR Unresolved dependencies.
//third_party/webrtc/base:base(//build/toolchain/android:android_arm)
needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
Original issue's description:
> Creating libwebrtc bundle jar
>
> Creates a JAR which includes:
> - //webrtc/base:base_java
> - //webrtc/modules/audio_device:audio_device_java
> - //webrtc/sdk/android:libjingle_peerconnection_java
> - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
>
> The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2646443002
> Cr-Commit-Position: refs/heads/master@{#16189}
> Committed: a62a82b7e7TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2640023010
Cr-Commit-Position: refs/heads/master@{#16190}
Creates a JAR which includes:
- //webrtc/base:base_java
- //webrtc/modules/audio_device:audio_device_java
- //webrtc/sdk/android:libjingle_peerconnection_java
- //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646443002
Cr-Commit-Position: refs/heads/master@{#16189}
This CL https://codereview.webrtc.org/2641763003 changed echo return
loss /...enhancement stats from being optional to being undefined
because that was the observed behavior (and a TODO was added to
investigate why).
It turns out that these stats are sometimes available, e.g. if the test
runs for a while like MSAN bot does, so this turned the test flaky.
Example failure:
https://build.chromium.org/p/client.webrtc/builders/Linux%20MSan/builds/8242
This CL reverts that change without reverting the rest of the CL which
other CLs depend on, and updates the TODO.
BUG=chromium:627816
TBR=hta@webrtc.org
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2640743007
Cr-Commit-Position: refs/heads/master@{#16187}
In multi-window mode the non-active activity receives onPause. We
shouldn't stop the camera in this case.
BUG=webrtc:7018
Review-Url: https://codereview.webrtc.org/2648483003
Cr-Commit-Position: refs/heads/master@{#16186}
Moves setting state as stopped to stopInternal. Checks that state is not
stopped in stopInternal.
BUG=webrtc:7015
Review-Url: https://codereview.webrtc.org/2640093003
Cr-Commit-Position: refs/heads/master@{#16182}
