We currently have one build target containing everything for audio_device: the interfaces,
the "fine" audio buffer, and the actual implementations for each platform.
Since we are planning to move the Android implementation to the sdk/android folder,
we only want to depend on the interfaces and the "fine" audio buffer, not the other platform
specific implementations. This CL splits the audio_device target into three different targets:
the interfaces, the fine audio buffer, and the platform specific implementations. The default
audio_device target now points to the interfaces instead.
Bug: webrtc:7452
Change-Id: I57e849cc6f4087d950fa02d969ecc682934839cd
Reviewed-on: https://webrtc-review.googlesource.com/61321
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22452}
Why this dep is here is lost to history. Everything works
without it though.
Bug: webrtc:8821
Change-Id: Ie0d763fb8a6508f7177a2f4bc9b7d909b9b02eb6
Reviewed-on: https://webrtc-review.googlesource.com/61962
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22441}
This reverts commit 31a12c557dcd84a31f9c3f2d8858d9646c2a3135.
Reason for revert: Breaks downstream project.
Original change's description:
> Add ability to emulate degraded network in Call via field trial
>
> This is especially useful in Chrome, allowing use to emulate network
> conditions in incoming or outgoing media without the need for platform
> specific tools or hacks. It also doesn't interfere with the rest of the
> network traffic.
>
> Also includes some refactorings.
>
> Bug: webrtc:8910
> Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
> Reviewed-on: https://webrtc-review.googlesource.com/33013
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22418}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I22bda6da01c2ff5abd6f408c5ee9e4fba21294f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8910
Reviewed-on: https://webrtc-review.googlesource.com/61700
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22419}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
Bug: webrtc:8910
Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
Reviewed-on: https://webrtc-review.googlesource.com/33013
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22418}
This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
Bug: webrtc:7671
Change-Id: Ib6e37ce4bc0bae903dd72f49ffdc2ee583d75491
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/61120
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22376}
The number of calls to ComfortNoiseDecoder::Generate() was determined
by the fuzzer input, and was chosen between 0 and 255. This would
sometimes lead to very long runs, with questionable merit. With this
change, the number of call to Generate() is limited to 17 (an
arbitrary small integer).
Bug: chromium:820078
Change-Id: I27b5c7f0b72d53370d002a6b157d4451079a0ba9
Reviewed-on: https://webrtc-review.googlesource.com/60941
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22360}
Previously, the only user of this code was the
VideoProcessorIntegrationTest. We have now changed that
test to directly calculate image quality metrics using libyuv,
similar to how the full stack tests and browser tests work.
Bug: webrtc:8448
Change-Id: Ia7a607d7ddc37741fba76d56aa7297851ffa1c6b
Reviewed-on: https://webrtc-review.googlesource.com/43760
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22341}
This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889
Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}
Bug: webrtc:7671
Change-Id: Iba2e89aee73a73a0372edea26933d6a7ea2e0ec9
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/60600
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22336}
This CL adds two new tests to perf, covering I420 and I420A input to multiplex
codec. In order to have the correct input, it adds I420A case to
SquareGenerator and corresponding PSNR and SSIM calculations.
Bug: webrtc:7671
Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
Reviewed-on: https://webrtc-review.googlesource.com/52180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22330}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. It will be done in 3 steps:
1. Test/fake_audio_device will be moved to production part of WebRTC
source code and renamed to test_audio_device_module. Old header is
replaced with alias to the new one.
2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice.
3. test/fake_audio_device will be removed.
This CL implements 1st step.
Bug: webrtc:8946
Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f
Reviewed-on: https://webrtc-review.googlesource.com/59740
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22325}
This reverts commit 8ea5f9ae5b757aa3a0e6abe46f5c9ef3aaf4b337.
Reason for revert: breaks downstream project
Original change's description:
> Separate test/fake_audio_device on API and implementation.
>
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
>
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
>
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}
TBR=kwiberg@webrtc.org,titovartem@webrtc.org
Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.
Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
We found out that
int16_t x = test::FuzzDataHelper::ReadOrDefaultValue(0)
reads 4 bytes from the fuzzer input instead of 2. That means that
almost half the bits in the input data to audio_processing_fuzzer are
ignored. This change adds template arguments to force reading 2 bytes
when we only need 2.
We also add a small manually generated corpus. During local testing we
let the fuzzer run for a few hours on an empty corpus. Adding the
manually-generated files resulted in an immediate coverage increase by
~3%, and then by another 3% over the next few hours.
The manually generated corpus contains a short segment of speech with
real echo. We suspect that triggering Voice Activity Detection or echo
estimation filter convergence can be difficult for an automatic
fuzzer.
We remove the Level Controller config. We read 20 bytes extra after the
config to guard against future configuration changes.
Bug: webrtc:7820
Change-Id: If60c04f53b27c519c349a40bd13664eef7999368
Reviewed-on: https://webrtc-review.googlesource.com/58744
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22269}
The additional include is needed in order to use EXPECT_NONFATAL_FAILURE()
in unit tests.
Bug: webrtc:8948
Change-Id: If5b9ceb89a3a36480657d094cfabc81c9b0e15b7
Reviewed-on: https://webrtc-review.googlesource.com/58096
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22227}
The length of the fuzzer input can sometimes be really long (more than
1000000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 200000 bytes.
NOTRY=TRUE
Bug: chromium:802149
Change-Id: Ia9d2f080602bba8ff70c5f0575bb9ecfa99c537c
Reviewed-on: https://webrtc-review.googlesource.com/57581
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22183}
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.
NOTRY=TRUE
Bug: chromium:802193
Change-Id: Id32174611fadb480f4e2c6b4f553a2ba0fa5b493
Reviewed-on: https://webrtc-review.googlesource.com/57580
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22182}
The length of the fuzzer input can sometimes be really long (more than
600000 bytes), and this take a very long time to execute. Typically,
the fuzzer times out instead. This change limits the used length of
the fuzzer to 100000 bytes.
NOTRY=TRUE
Bug: chromium:802245
Change-Id: Ibe02b6de932d900408f870d9ba440b7b8e08dc0e
Reviewed-on: https://webrtc-review.googlesource.com/57180
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22181}
The unit tests for AudioSendStream was generating a lot of warnings
about "Uninteresting mock function call" on mocked objects. This is due
to the default gmock implementation being NaggyMock and there was no
NiceMock override.
With this change the mocks are replaced with NiceMock implementations
which do not output warnings for unexpected calls. This makes the error
output from the test runner much easier to visually parse to find the
actual errors in failing tests.
Bug: None
Change-Id: Ic40db78159536ddeaa72a468fc2cb3ec17386d44
Reviewed-on: https://webrtc-review.googlesource.com/56220
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22152}
new is an unsafe construct, while these specific cases were properly
handled it is a code smell and using unique_ptr from the start makes the
code more obviously correct.
Bug: None
Change-Id: I2554cef8d3a8432a3ced1623292fae0adff9421d
Reviewed-on: https://webrtc-review.googlesource.com/56620
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22147}
MockAudioEncoder was calling a mocked Die function on itself in its
destructor. This outputs "Uninteresting mock function call" warning if
the Die call was not expected. This is true even if a NiceMock is used
to suppress the warnings.
The purpose of testing that the destructor is called might be to protect
against memory leaks when audio encoder ownership is transferred using a
raw pointer. However, this case is already covered by msan checks.
Bug: None
Change-Id: I0603c417b4b239027859228e05ebcf83ff5aaf18
Reviewed-on: https://webrtc-review.googlesource.com/56183
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22146}
Since rtp transport controller send owns the congestion controller it
also should own the bitrate configuration logic, this way it can
initialize the send side congestion controller with the bitrate
configuration.
Bug: webrtc:8415
Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553
Reviewed-on: https://webrtc-review.googlesource.com/54303
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22127}
This CL adds the GainCurveApplier (GCA). It owns a
FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve
(IGC). The GCA uses the LE to compute the input signal level, looks up
a gain from IGC and applies it on the signal.
The other IGC and LE submodules were added in previous CLs [1] and
[2].
This CL also turns on AGC2 in the APM fuzzer.
[1] https://webrtc-review.googlesource.com/c/src/+/51920
[2] https://webrtc-review.googlesource.com/c/src/+/52381
Bug: webrtc:7949
Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd
Reviewed-on: https://webrtc-review.googlesource.com/54361
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22103}
Deleting the apparently unused include of api/rtp_headers from
common/video/include/video_frame.h broke the PayloadRouter and
VideoSendStream code under video/. Missing declaration of the
RtpPayloadState struct declared in api/rtp_headers.h. Moving the
declaration of that struct to payload_router.h (outside of the api),
since it's used only internally in video/, and that seemed to be a
more logical place for it.
Bug: webrtc:7504
Change-Id: Ibed8233dfeea8bdf144db5422cdf897da824d6ee
Reviewed-on: https://webrtc-review.googlesource.com/53701
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22080}
When we run webrtc_perf_tests with gtest-parallel, each test is run
individually, and this results in the file with the perf results being
overwritten each time.
To avoid this, we won't use gtest-parallel when running webrtc_perf_tests,
so we will simply run the binary directly.
TBR=phoglund@chromium.org
Bug: chromium:755660
Change-Id: I24db36e512fcf604a3de2adf4d0b4325b2c3d1ae
Reviewed-on: https://webrtc-review.googlesource.com/49340
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21982}
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.
Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
Rtp Packets in webrtc expected to be less that 1500,
i.e. way less that 2^16 bytes for extensions block.
This CL explicitly discards longer extension.
Bug: chromium:809046
Change-Id: Ibed33b51bafc3fd4804ec135f66110c6d2796734
Reviewed-on: https://webrtc-review.googlesource.com/48061
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21910}
When a simulcast stream is enabled or disabled, we want this state
change to be reflected properly in the RtpRtcp modules. Each video send
stream can contain multiple rtp_rtcp_modules pertaining to different
simulcast streams. These modules are currently all turned on/off when
the send stream is started and stopped. This change allows for
individual modules to be turned on/off. This means if a module stops
sending it will send a bye message, so the receiving side will not
expect more frames to be sent when the stream is inactive and the
encoder is no longer encoding/sending images.
Bug: webrtc:8653
Change-Id: Ib6d00240f627b4ff1714646e847026f24c7c3aa4
Reviewed-on: https://webrtc-review.googlesource.com/42841
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21880}
