The json files used to generate the Android.bp files have more flags
than those specified in the BUILD.gn and *.gni configuration files. This
change ignores the ones added by the build toolchain and keeps only
those added by the webrtc authors.
Bug: 269761242
Test: run cuttlefish in x64, build for arm64 and riscv64
Change-Id: I55f2bdea229cf11c21b5780b2639abb6dd7c3268
by making the build file generator script (mostly) architecture independent.
Bug: 269761242
Test: build x64 and arm64 locally
Change-Id: I76ea4bc0ba5e8e5c152b93cb1ad7a385c796adae
-gdwarf-aranges breaks a later linking step on riscv64 with:
ld.lld: error: out/soong/.intermediates/external/webrtc/webrtc_audio_processing/android_vendor.UpsideDownCake_riscv64_static/webrtc_audio_processing.a(audio_processing_impl.o):(.rodata.str1.1): offset is outside the section
Bug: 269343483
Test: lunch aosp_riscv64-userdebug && m libaudiopreprocessing
Change-Id: Iaf96679c3e82229adf958668f9afbdf0d4768c6b
This option is likely not really applicable for Android target builds
which are stripped / have a separate symbols output so just filter it
out. The option breaks on RISC-V due to implicit -mrelax and lack of
toolchain support:
clang++: error: -gsplit-dwarf is unsupported with RISC-V linker relaxation (-mrelax)
Bug: 266468464
Change-Id: Ibdb7bc7e08576c1096148a7a6381554888dfa6b0
For x86, x86_64, arm and arm64
Bug: 261600888
Test: build and run cuttlefish x86, x86_64 and arm64
Change-Id: I3ac4dad1ac9ec83b0e626e64715df450e8809b82
parse
a=msid:<stream_id>
since JSEP stipulates sending this syntax as track identifers
have become meaningless. The track id will be set to a random string.
a=msid:<stream_id> <track_id>
remains supported for backward compability.
BUG=webrtc:14729
Change-Id: I86c073eb97cd613324271125de18a773235fc79d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285783
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38814}
Currently the fixed digital gain is applied after the input volume
controller and before the adaptive digital one. This CL moves its
application after the adaptive digital controller and before the
limiter.
Reasons:
- This change is safe: no production config where both adaptive and
fixed digital controllers are jointly used
- More predictable behavior: when the fixed digital controller is
used after the adaptive digital controller it is easier to describe
the overall behavior - i.e., the fixed digital combined with the
limiter can be used for digital compression
- Allow to remove an unwanted temporal dependency: in a follow-up CL
the input volume controller will use the latest speech level
estimation instead of that from the previously analyzed frame; this
CL makes that change easier.
Bug: webrtc:7494
Change-Id: I2e9869081e0eba1e4f30f11ea93a973ca7fea28c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286340
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38813}
Already implemented for STUN hostname resolution, but TURN port resolves hostnames separately. Reusing the field trial key reserved in bugs.webrtc.org/14334 but with a new parameter so as to not affect ongoing rollouts.
Bug: webrtc:14319, webrtc:14131
Change-Id: Idf771fb2f0de7849f8b701be8ee05a98b8d242f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285981
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#38811}
This cl/ attempts to fix (rather) rare crashes in
OnNetworkConnected_n by loosening the assumptions
that a network handle will keep it's network name.
With this cl/ it is possible that a NetworkHandle
can call OnNetworkConnected_n with one interface name
and then directly afterwards call it with another (
w/o an OnNetworkDisconnected_n inbetween).
This is the only scenario in which I could see the previous
crash occurring.
i.e
OnNetworkConnected(handle, "some-if-name")
OnNetworkConnected(handle, "some-other-name")
- previously this caused crash,
- now this is treated as if there was an OnNetworkDisconnected(handle) in between.
---
Also 1: shamelessly copy TYPE_MOBILE_DUN & TYPE_MOBILE_HIPRI from chromium: 87987f0e76
Also 2: Modify testcase not to use real interface names, so I can ran them on personal test phone w/o the real networks interfering.
Bug: webrtc:13741
Change-Id: I5480d5ce7031c2b5c09b958064076d02b3db1248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285980
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38808}
Differently from the ChromePerfDashboardMetricsExporter, this new flag
doesn't default to storing the output file to NSDocumentDirectory (and
with a default name, for example perftest-output.pb) but instead
just stores the file at the location specified by --webrtc_test_metrics_output_path.
Bug: b/237982523
Change-Id: Ibb504fdbc94ca5179f4b3da5b06d8cea82140140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286280
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38807}
Recently, Chromium added -fsanitize=alignment for `is_ubsan=true`,
and due to this, unaligned memory access was found in several places,
and chromium roll is currently blocked.
Modify unaligned memory access in rtc_base/byte_order.h to use memcpy.
Since libaom and libsrtp perform unaligned memory accesses, add them
to the suppression list.
Also, remove any mention of yasm from the UBSAN suppression list,
as yasm is no longer used.
Bug: chromium:1057551
Change-Id: I4961b66831750f4fa7b6de0b80b2052fe6ef27c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286200
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38806}
Isolates the build targets for the `AdaptiveDigitalGainController`
dependencies that will be moved into `GainController2`.
`AdaptiveDigitalGainController` will be removed because the wrapper
itself adds little - that's the reason why it has no unit tests.
Bug: webrtc:7494
Change-Id: I2ca41f9255c8faefe4b2cb4ec1f8db536e582f39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280482
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38799}
AndroidNetworkMonitor::SetNetworkInfos assumes this method is called
only once, but unittests calls it twice.
One is called by the startMonitoring Java method, and the other is
called by each test.
Because of this, these tests will not succeed if dcheck_always_on is true.
To solve this problem, use OnNetworkConnected_n
instead of SetNetworkInfos in each test.
Bug: None
Change-Id: I027706ad5ccd597a91e3a66f15e181ee22d4aaa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285861
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38798}
This new name emphasizes that the field trial keys are only allowed
within the current scope. We already have test::ScopedFieldTrials that
can be used to ensure that the global field trials string itself is
isolated.
Bug: webrtc:14705
Change-Id: I8b66bbd9c11d97985292c334d2d3496a047074a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284862
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38796}
- use the new naming convention 'input volume'
- fix Yoda-style expressions in the unit tests
- clarify how the gain map values are generated
Bug: webrtc:7494
Change-Id: I4d6ee897a93cdefa6735733b053c57326d01a528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285467
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38795}
This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880.
Reason for revert: Perf regression not affecting open source.
Original change's description:
> Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
>
> This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc.
>
> Reason for revert: Tentative revert due to possible perf regression. b/260123362
>
> Original change's description:
> > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream
> >
> > VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
> > Therefore this cl:
> > - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
> > - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
> > - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.
> >
> > Bug: none
> > Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38698}
>
> Bug: none
> Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38725}
Bug: b/260400659
Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38794}
The current logic has risks if the encoder queue changes, for example when switching from a hardware encoder to a software encoder, or when switching from two different software implementations. Always using a dedicated task queue simplifies the code and is safer.
Bug: chromium:1395308
Change-Id: I0a576ed2f6e892955e0a519567969474d3b99efd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285882
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38793}
