This is currenly tracked in both AcmReceiver and NetEq. Adding this API
enables us to have it in just one place.
Bug: None
Change-Id: Ia537f87f36b0aedf19c00a57bd6cec4425a49df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360743
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42872}
Mark old overload deprecated.
This allows to migrate both calls through AudioDecoderFactory and direct calls to AudioDecpderOpus trait.
Bug: webrtc:356878416
Change-Id: I1502aee5b18aac43a8258e77b770c8e73a056f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359741
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42793}
Finalize change started in https://webrtc-review.googlesource.com/c/src/+/359243
Remove fallback to old interface and unneeded clock member in the config struct.
Bug: None
Change-Id: I4c2b65a09dd1c8a0d44ee76320b095516e2c61fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359561
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42782}
to make it available for creating AudioDecoders
Bug: webrtc:356878416
Change-Id: Ibd24a55df70985dfe02d924da037618f13661032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359241
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42771}
To propagate field trials in addition to clock
Bug: webrtc:356878416
Change-Id: Idefc4848ec4af30c8aed0f93b7fadfc3181bddb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358980
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42761}
To propagate field trials into the NetEq and further towards Audio Decoders
Bug: webrtc:356878416
Change-Id: Ia7cf18451aef70441ca958bf652f492138c6051a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358620
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42739}
Passing Environment would allow to propage field trials with it further to NetEq and AudioDecoders
Bug: webrtc:356878416
Change-Id: Ic68420df3b157ed341146207a2c45cb49e59a931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358501
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42728}
To allow delete old signature of the AudioEncoderOpus::MakeAudioEncoder function and thus guarantee Opus AudioEncoder always has an Environment
Bug: webrtc:343086059
Change-Id: Ib660678aeb5a549dddd1dffa3d8c28b2ec6b9d0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356981
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42690}
This switches from accepting a sample rate and convert to channel
size over to accepting the channel size.
Instead of InitializeIfNeeded:
* Offer a way to explicitly initialize PushResampler via the ctor
(needed for VoiceActivityDetectorWrapper)
* Implicitly check for the right configuration from within Resample().
(All calls to Resample() were preceded by a call to Initialize)
As part of this, refactor VoiceActivityDetectorWrapper (VADW):
* VADW is now initialized in the constructor and more const.
* Remove VADW::Initialize() and instead reconstruct VADW if needed.
Add constants for max sample rate and num channels to audio_util.h
In many cases the numbers for these values are embedded in the code
which has led to some inconsistency.
Bug: chromium:335805780
Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42587}
This is to make sure that the two encoders are "in sync" (the CNG
encoder can be created from an existing speech encoder).
This is a speculative fix for a crash in the CNG encoder where a packet
is unexpectedly emitted from the speech encoder.
Bug: webrtc:42225071
Change-Id: I42571e56e032897f7f083f04d785f6a08ebfb813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355160
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42516}
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default
Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.
Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
The new version of MSan (rolled by [1]) detects the following:
```
==39908==WARNING: MemorySanitizer: use-of-uninitialized-value
#0 0x5591400a52ef in GetPlayoutDelayMs ./../../modules/audio_coding/neteq/decision_logic.cc:466:35
#1 0x5591400a52ef in webrtc::DecisionLogic::ExpectedPacketAvailable(webrtc::NetEqController::NetEqStatus) ./../../modules/audio_coding/neteq/decision_logic.cc:311:36
#2 0x5591400a39e9 in webrtc::DecisionLogic::GetDecision(webrtc::NetEqController::NetEqStatus const&, bool*) ./../../modules/audio_coding/neteq/decision_logic.cc:0:0
#3 0x55913cf590c9 in webrtc::DecisionLogicTest_PreemptiveExpand_Test::TestBody() ./../../modules/audio_coding/neteq/decision_logic_unittest.cc:139:3
#4 0x55913ef28283 in HandleExceptionsInMethodIfSupported<testing::Test, void> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:3
#5 0x55913ef28283 in testing::Test::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2710:5
#6 0x55913ef2ab46 in testing::TestInfo::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:2856:11
#7 0x55913ef2da34 in testing::TestSuite::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:3034:30
#8 0x55913ef621e8 in testing::internal::UnitTestImpl::RunAllTests() ./../../third_party/googletest/src/googletest/src/gtest.cc:5964:44
#9 0x55913ef60f54 in HandleExceptionsInMethodIfSupported<testing::internal::UnitTestImpl, bool> ./../../third_party/googletest/src/googletest/src/gtest.cc:0:0
#10 0x55913ef60f54 in testing::UnitTest::Run() ./../../third_party/googletest/src/googletest/src/gtest.cc:5543:10
#11 0x55913ee1a944 in RUN_ALL_TESTS ./../../third_party/googletest/src/googletest/include/gtest/gtest.h:2334:73
#12 0x55913ee1a944 in webrtc::(anonymous namespace)::TestMainImpl::Run(int, char**) ./../../test/test_main_lib.cc:203:21
#13 0x55913cbd36b8 in main ./../../test/test_main.cc:72:16
#14 0x7fdb18c73082 in __libc_start_main /build/glibc-LcI20x/glibc-2.31/csu/../csu/libc-start.c:308:16
#15 0x55913cb3e1a9 in _start ??:0:0
```
[1] - https://webrtc-review.googlesource.com/c/src/+/353620
Bug: b/344970813
Change-Id: I9b5d7791e68b4c494168ba9f007a3099ae21fed4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353581
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42433}
This is a minor change for places where we use
AudioFrame::kMaxDataSizeSamples sized intermediary buffers. The change
uses `std::array<>` instead of C style arrays which allows for use
of utility templates that incorporate type based buffer size checking.
Also adding `ClearSamples()` method, which complements CopySamples.
Bug: chromium:335805780
Change-Id: I813feb32937e020ceb9ca4b00632dc90907c93fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351681
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42400}
From the new header file:
* MonoView<>: A single channel contiguous buffer of samples.
* InterleavedView<>: Channel samples are interleaved (side-by-side) in
the buffer. A single channel InterleavedView<> is the same thing as a
MonoView<>
* DeinterleavedView<>: Each channel's samples are contiguous within the
buffer. Channels can be enumerated and accessing the
individual channel data is done via MonoView<>.
There are also a few utility functions that offer a unified way to check
the properties regardless of what view type is in use.
Bug: chromium:335805780
Change-Id: I28196f8f4ded4fadc72ee32b62af304c62f4fc47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349300
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42377}
This is a reland of commit 47bfe39ecf
Original change's description:
> Split digest methods from ssl target into digest target
>
> in an attempt to break up the monolithic ssl target.
>
> BUG=None
>
> Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#42249}
Bug: webrtc:339300437
Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42304}
This ensure the absolute capture timestamp from the first audio sample
encoded in the payload is used for the corresponding rtp header.
Bug: webrtc:42226041
Change-Id: Ib8f2e3a5df5c82c5806171bd5b36a26d92fbea72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349265
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42281}
in an attempt to break up the monolithic ssl target.
BUG=None
Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42249}
It is not necessary for the caller to use it and the mute info can be
found on AudioFrame.muted().
Bug: None
Change-Id: I458f1f2e8489c1d8f8a9078b21f889b2540bdab9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42236}
Start introducing ArrayView to AudioFrame and code that flows down
from there. In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
audio buffer. When AudioFrame is not initialized however, data_view()
will return a nullptr whereas the current data() method never returns
nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
the samples per channel and number of channels that's required for
accurately reserving the returned mutable ArrayView.
A notable behavior change is that if the requested number of channels
is larger than supported or the calculated buffer size is too large,
the function will trigger a check.
* Add TODOs for following work.
Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
