This operation was relatively simple, since no one was doing anything
fishy with this enum. A large number of lines had to be changed
because the enum values now live in their own namespace, but this is
arguably worth it since it is now much clearer what sort of constant
they are.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1424083002
Cr-Commit-Position: refs/heads/master@{#10449}
Negative acknowledgement (NACK) has up to now been implemented in
ACM. But, since NetEq is in charge of the actual packet buffer, it
makes more sense to have the NACK functionlaity in there.
This CL does the following:
- Move nack.{h,cc} and the unit tests from main/acm2 to neteq.
- Move the NACK related code in ACM into NetEq.
- NACK related functions in AcmReceiver are changed to simple
forwarding APIs.
- Remove unused members in AcmReceiver.
- Remove unused API functions in NetEq.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1410073006
Cr-Commit-Position: refs/heads/master@{#10448}
Tested on Linux with the following command lines:
$ ./webrtc/build/gyp_webrtc -Dclang_use_chrome_plugins=1
$ ninja -C out/Release rtc_sound
BUG=webrtc:163
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1425533003
Cr-Commit-Position: refs/heads/master@{#10447}
Used for tests that cannot be run in parallel due to using non-virtual
resources such as filesystems and sockets. Initially moves socket
unittests from rtc_unittest since
PhysicalSocketTest.TestUdpReadyToSendIPv4 is one of the worst flake
offenders.
Future prospect targets are GTEST_DEATH tests that are flaky on Mac in
parallel for instance.
BUG=chromium:445880
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1426643003 .
Cr-Commit-Position: refs/heads/master@{#10446}
This speeds up decode for a 720p workload from 25s to 5s on my machine.
I did some benchmarking and it appears we spend the vast majority of
the time on zxing decoding, so now we will bring all processors on the
the machine to bear.
BUG=webrtc:4986
Review URL: https://codereview.webrtc.org/1428433002
Cr-Commit-Position: refs/heads/master@{#10443}
Plus log first few decoder frames in and out events.
BUG=b/25287910
Review URL: https://codereview.webrtc.org/1423843005
Cr-Commit-Position: refs/heads/master@{#10439}
Following CLs will finish the takeover completely. After that,
RentACodec will also start creating and owning codecs, at which point
its name will start making sense.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1412683006
Cr-Commit-Position: refs/heads/master@{#10432}
Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419193002
Cr-Commit-Position: refs/heads/master@{#10430}
Reason for revert:
Caused compiler warning, breaking Chrome FYI bots.
Original issue's description:
> Adding the ability to change ICE servers through SetConfiguration.
>
> Added a SetIceServers method to PortAllocator. Also added a new
> PeerConnection Initialize method that takes a PortAllocator, in the
> hope that we can get rid of PortAllocatorFactoryInterface, since the
> only substantial thing a factory does is convert the webrtc:: ICE
> servers to cricket:: versions.
>
> Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> Cr-Commit-Position: refs/heads/master@{#10420}
TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1424803004
Cr-Commit-Position: refs/heads/master@{#10421}
Added a SetIceServers method to PortAllocator. Also added a new
PeerConnection Initialize method that takes a PortAllocator, in the
hope that we can get rid of PortAllocatorFactoryInterface, since the
only substantial thing a factory does is convert the webrtc:: ICE
servers to cricket:: versions.
Review URL: https://codereview.webrtc.org/1391013007
Cr-Commit-Position: refs/heads/master@{#10420}
This is necessary in order to support the RTCPeerConnectionState enum in
the future, as well as a correct RTCIceConnectionState (which isn't a
combination ICE and DTLS state).
Review URL: https://codereview.webrtc.org/1414363002
Cr-Commit-Position: refs/heads/master@{#10419}
Reason for revert:
https://codereview.chromium.org/1419253002 is landed to address this linker issue. Keep my fingers crossed.
Original issue's description:
> Revert of Add experiment on weak ping delay during call set up time (patchset #4 id:60001 of https://codereview.webrtc.org/1411883002/ )
>
> Reason for revert:
> This CL breaks Chromium, undefined reference link error to webrtc::field_trial::FindFullName. Adding the dependency system_wrappers to the rtc_p2p target is not enough to fix this.
>
> Looking at field_trial.h (in system_wrappers/interface/, not to be confused with the one in test/) the documentation says "WebRTC clients MUST provide an implementation of: ...FindFullName... Or link with a default one provided in: ...system_wrappers.gyp:field_trial_default).
>
> So maybe just depend on field_trial_default? Not sure what should be done in case there are implications to adding this dependency, I'm reverting this. Sorry :)
>
> Original issue's description:
> > Add experiment on weak ping delay during call set up time
> >
> > BUG=
> > R=pthatcher@webrtc.org
> >
> > Committed: https://crrev.com/3bf69b15f4c0c0ca4ab17c237084684a37bb8279
> > Cr-Commit-Position: refs/heads/master@{#10343}
>
> TBR=pthatcher@webrtc.org,juberti@webrtc.org,guoweis@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
TBR=pthatcher@webrtc.org,juberti@webrtc.org,hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1413603005
Cr-Commit-Position: refs/heads/master@{#10418}
Reason for revert:
Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
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}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1426443007
Cr-Commit-Position: refs/heads/master@{#10417}
The main intended use case is as a function argument, replacing the
harder-to-read and harder-to-use separate pointer and size arguments.
It's easier to read because it's just one argument instead of two, and
with clearly defined semantics; it's easier to use because it has
iterators, and will automatically figure out the size of arrays.
BUG=webrtc:5028
R=andrew@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1408403002 .
Cr-Commit-Position: refs/heads/master@{#10415}
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.
Review URL: https://codereview.webrtc.org/1413713003
Cr-Commit-Position: refs/heads/master@{#10414}
This test was used to verify that VideoEngine handles were handed back
correctly. This is no longer applicable.
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1425673002 .
Cr-Commit-Position: refs/heads/master@{#10412}
This reverts commit f054819e257a4f9cbb7fa82ba51dc2335f4359ec,
2d3747de9b7c3014e106d3766dc07cf5da3e1881 and
7ef0553c85c5b373535d7f6161e9a6d3b5b9a826.
It seems harder than expected to get a GN build for rtc_sound
and we lack sufficient trybot support for the case where
WebRTC is built as part of Chromium.
The Debug builds failed like this:
[6939/7454] SOLINK ./libcontent.so
FAILED: ../../third_party/llvm-build/Release+Asserts/bin/clang++ -shared -Wl,--fatal-warnings -fPIC -Wl,-z,noexecstack -Wl,-z,now -Wl,-z,relro -Wl,-z,defs -B../../third_party/binutils/Linux_x64/Release/bin -fuse-ld=gold -Wl,--icf=all -pthread -m64 -Wl,--export-dynamic -o ./libcontent.so -Wl,-soname=libcontent.so @./libcontent.so.rsp && { readelf -d ./libcontent.so | grep SONAME ; nm -gD -f p ./libcontent.so | cut -f1-2 -d' '; } > ./libcontent.so.tmp && if ! cmp -s ./libcontent.so.tmp ./libcontent.so.TOC; then mv ./libcontent.so.tmp ./libcontent.so.TOC; fi
../../third_party/webrtc/sound/alsasoundsystem.cc:453: error: undefined reference to 'rtc::LateBindingSymbolTable::Load()'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.cc.def:63: error: undefined reference to 'rtc::LateBindingSymbolTable::LateBindingSymbolTable(rtc::LateBindingSymbolTable::TableInfo const*, void**)'
../../third_party/webrtc/base/latebindingsymboltable.cc.def:65: error: undefined reference to 'rtc::LateBindingSymbolTable::~LateBindingSymbolTable()'
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ninja: build stopped: subcommand failed.
BUG=webrtc:4160
TBR=tfarina@chromium.org
Review URL: https://codereview.webrtc.org/1407893005 .
Cr-Commit-Position: refs/heads/master@{#10411}
It can't find //webrtc/base:rtc_base, which is weird, the fix is to use
a relative path.
This should fix the following error:
ERROR at //third_party/webrtc/sound/BUILD.gn:38:5: Can't load input
file.
"//webrtc/base:rtc_base",
^-----------------------
NOTRY=true
BUG=webrtc:4160
TBR=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1419953003
Cr-Commit-Position: refs/heads/master@{#10407}
We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.
As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1415163002
Cr-Commit-Position: refs/heads/master@{#10406}
Tested on Linux with the following command lines:
$ gn gen out-gn/Release --args='is_debug=false target_cpu="x64"
build_with_chromium=false'
$ ninja -C out-gn/Release rtc_sound
BUG=webrtc:4160
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1425583002
Cr-Commit-Position: refs/heads/master@{#10405}
