This solves a circular dep and eliminates a target.
This means we will apply neon copts to some files that weren't before,
but I don't think that is a problem.
Bug: webrtc:6828,webrtc:7042
Change-Id: I3bb656ba5b13d6104b519c2dbf6a4b2814575b87
Reviewed-on: https://webrtc-review.googlesource.com/34183
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21330}
This CL adds windowing of the error signal in echo canceller 3 to
avoid issues with spectral leakage affecting the quality of
the filter estimate.
Bug: webrtc:8661
Change-Id: I3e583f80fe02d7bba387a906bf44fbe7b89a2a6f
Reviewed-on: https://webrtc-review.googlesource.com/34188
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21328}
This CL adds a way to insert a custom render-side pre-processor to
APM. The pre-processor operates in full-band mode before anything
else. Currently the render processing chain is (if everything is
enabled):
Network --> [Pre processing] --> [Band split] -->
[IntelligibilityEnhancer] --> [Echo canceller (read-only)] -->
[Band merge] --> Playout
Since the render pre processor and capture post processor have the
same interface, I renamed webrtc::PostProcessing into
webrtc::CustomProcessing.
The old APM factory method PostProcessing will be deprecated and
dependencies updated as part of webrtc:8665
NOTRY=True
Bug: webrtc:8665
Change-Id: Ia381cbf12e336d6587406a14d77243d931f69a31
Reviewed-on: https://webrtc-review.googlesource.com/29201
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21327}
Currently, when input fps is higher than the configured target fps in
screenshare_layers, we drop frames with the help of a rate tracker using
a one second sliding window. This is not optimal.
For instance, given a 5fps limit and a 30fps capturer, the window will
not be saturated until we have added the first 5 frames. Then we will
drop all frames until the oldest one drops out, at which point we can
immediately fill it's place. This results in quick 5 frame bursts every
second.
In addition to rate tracker, also set a limit on minimum interval
required between input frames, based on target frame rate.
Bug: webrtc:4172
Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57
Reviewed-on: https://webrtc-review.googlesource.com/34380
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21325}
This CL adjusts the filter adaptation behavior to better handle
reverberant environments and environments with poor SNR.
It furthermore updates the unittests to handle the reduced adaptation
speed.
Bug: webrtc:8661
Change-Id: I5f1b5a4a34b333bd6c643ed3727899d0838dbf90
Reviewed-on: https://webrtc-review.googlesource.com/34184
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21323}
Removing enum that was left behind when the metric aec_quality_min was
removed.
Bug: webrtc:8563
Change-Id: I8a8c68659abc6465ef42f002f73bd2607e953ac5
Reviewed-on: https://webrtc-review.googlesource.com/33004
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21321}
It is used only by the test-only http server code.
Bug: webrtc:6424
Change-Id: Id7120ed1ded6773f98472526e8fa282cf0a423e8
Reviewed-on: https://webrtc-review.googlesource.com/33401
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21313}
In balanced adaptation mode, a 1280x720 feed would only ever be reduced in
resolution twice, and would never have its framerate reduced (due to an
interaction with MinFps()).
This change removes the hard limits entirely, instead relying only on
kMinFramerateFps and VideoEncoder::ScalingSettings::min_pixels_per_frame.
Deleted SinkWantsFromOveruseDetector test because it duplicates other tests.
Fixed DoesntAdaptDownPastMinFramerate; it wasn't testing what it claimed to
because it wasn't updating the fake clock correctly, meaning FPS was detected as
0, meaning framerate adaptation was never triggered.
Bug: webrtc:8068, b/38207842
Change-Id: If99d0e74c1334879c1b0c3117eb079f5f2139851
Reviewed-on: https://webrtc-review.googlesource.com/31644
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21312}
The DtlsTransport tests worked by relying on JsepTransport (a helper
class used by higher layers to set everything up in response to SDP).
dtlstransport_unittest has been switched to just calling SetSslRole and
SetRemoteFingerprint directly instead, which were really the only parts
that were necessary.
Some refactoring was also done, and some test coverage was moved to
jseptransport_unittest. jseptransport_unittests has more coverage to
ensure that negotiated parameters are propagated to the DtlsTransport
underneath, which were previously covered by the tests in
dtlstransport_unittest. It also has a test that covers RTP/RTCP not
being multiplexed, which dtlstransport_unittests really doesn't need
to be concerned about.
BUG=NONE
Change-Id: I1d67e9a06486ade39a255555af4052d76191d238
Reviewed-on: https://webrtc-review.googlesource.com/32941
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21309}
This reverts commit 26246cac660a95f439b7d1c593edec2929806d3f.
Reason for revert: Introduces compile failure on MSVC, which is preventing rolls into Chromium.
Sample errors:
[12263/40346] CXX obj/third_party/webrtc/p2p/rtc_p2p/stun.obj
FAILED: obj/third_party/webrtc/p2p/rtc_p2p/stun.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes @obj/third_party/webrtc/p2p/rtc_p2p/stun.obj.rsp /c ../../third_party/webrtc/p2p/base/stun.cc /Foobj/third_party/webrtc/p2p/rtc_p2p/stun.obj /Fd"obj/third_party/webrtc/p2p/rtc_p2p_cc.pdb"
../../third_party/webrtc/p2p/base/stun.cc(1018): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/p2p/base/stun.cc(1018): warning C4267: 'argument': conversion from 'size_t' to 'uint16_t', possible loss of data
Original change's description:
> Add RelayPortFactoryInterface that allows for custom relay (e.g turn) ports
>
> This patch adds a RelayPortFactoryInterface that allows
> for custom relay ports. The factor is added as optional argument
> to BasicPortAlloctor. If none is provided a default implementation
> that mimics existing behavior is created.
>
> The patch also adds 2 stun functions, namely to copy a
> StunAttribute and to remove StunAttribute's from a StunMessage.
>
> Bug: webrtc:8640
> Change-Id: I59bd51f0f5e2f8c187dff9fcf003a24c35ed037f
> Reviewed-on: https://webrtc-review.googlesource.com/32600
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21267}
TBR=jonaso@webrtc.org,pthatcher@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8640
Change-Id: Idf83a1111727d2b5188b9c123f7471be7e99e973
Reviewed-on: https://webrtc-review.googlesource.com/33600
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21304}
TransmitMixer's functionality is moved into the AudioTransportProxy
owned by AudioState. This removes the need for an AudioTransport
implementation in VoEBaseImpl, which means that the proxy is no longer
a proxy, hence AudioTransportProxy is renamed to AudioTransportImpl.
In the short term, AudioState needs to know which AudioDeviceModule is
used, so it is added in AudioState::Config. AudioTransportImpl needs
to know which AudioSendStream:s are currently enabled to send, so
AudioState maintains a map of them, which is reduced into a simple
vector for AudioTransportImpl.
To encode and transmit audio,
AudioSendStream::OnAudioData(std::unique_ptr<AudioFrame> audio_frame)
is introduced, which is used in both the Chromium and standalone use
cases. This removes the need for two different instances of
voe::Channel::ProcessAndEncodeAudio(), so there is now only one,
taking an AudioFrame as argument. Callers need to allocate their own
AudioFrame:s, which is wasteful but not a regression since this was
already happening in the voe::Channel functions.
Most of the logic changed resides in
AudioTransportImpl::RecordedDataIsAvailable(), where two strange
things were found:
1. The clock drift parameter was ineffective since
apm->echo_cancellation()->enable_drift_compensation(false) is
called during initialization.
2. The output parameter 'new_mic_volume' was never set - instead it
was returned as a result, causing the ADM to never update the
analog mic gain
(https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/voe_base_impl.cc?q=voe_base_impl.cc&dr&l=100).
Besides this, tests are updated, and some dead code is removed which
was found in the process.
Bug: webrtc:4690, webrtc:8591
Change-Id: I789d5296bf5efb7299a5ee05a4f3ce6abf9124b2
Reviewed-on: https://webrtc-review.googlesource.com/26681
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21301}
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
The delay_ms stat in AudioprocessStats should be an Optional, because its value is not always computed. This CL changes it to an optional.
Bug: webrtc:8569
Change-Id: I42fd7a86b975c766b685444bf1829511f790da2a
Reviewed-on: https://webrtc-review.googlesource.com/33320
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21293}
The title of Dock window on OSX10.12 is still 'Dock' with layer number 20.
So the removed codes here is reduntant.
Also fix a wrong comment.
Bug: webrtc:8460
Change-Id: I72d4c8f5741a1ccb00aa45897f11e85af8d24e05
Reviewed-on: https://webrtc-review.googlesource.com/33123
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21287}
This reverts commit 59283e4c66d038a00923736685457f4b53f922fe.
Reason for revert: This CL is preventing rolls into Chromium because it fails to compile with MSVC.
Sample error log:
[13258/43857] CXX obj/third_party/webrtc/video/video/send_statistics_proxy.obj
FAILED: obj/third_party/webrtc/video/video/send_statistics_proxy.obj
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes @obj/third_party/webrtc/video/video/send_statistics_proxy.obj.rsp /c ../../third_party/webrtc/video/send_statistics_proxy.cc /Foobj/third_party/webrtc/video/video/send_statistics_proxy.obj /Fd"obj/third_party/webrtc/video/video_cc.pdb"
../../third_party/webrtc/video/send_statistics_proxy.cc(217): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/video/send_statistics_proxy.cc(217): warning C4267: 'initializing': conversion from 'size_t' to 'int', possible loss of data
../../third_party/webrtc/video/send_statistics_proxy.cc(632): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
Original change's description:
> googBandwidthLimitedResolution stat is not always set depending on configuration.
>
> Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
> OnEncodedImage callback.
>
> Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
> on info that is reported to SendStatisticsProxy::OnEncodedImage.
>
> Bug: webrtc:8643
> Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
> Reviewed-on: https://webrtc-review.googlesource.com/31460
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21249}
TBR=brandtr@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8643
Change-Id: Ib9ef55b8894ea72236a5dc1e9a839adecd401afb
Reviewed-on: https://webrtc-review.googlesource.com/33100
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21284}
This reverts commit f711428898cb2f03d4d09738ba751cf0d316c631.
Reason for revert: Breaks downstream projects.
Original change's description:
> Use std::fstream instead of rtc::File to write perf results + rename flag.
>
> Use std::fstream instead of rtc::File to write perf results.
> On Android, when I use rtc::File, the results are not written for some reason.
>
> Also rename the flag to '--chartjson_result_file'.
>
> Bug: webrtc:8566
> Change-Id: I32215e2233e18690c41050dfd35ac77e01d11f35
> Reviewed-on: https://webrtc-review.googlesource.com/32001
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21225}
TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8566
Change-Id: I55611592c3171152cee97e64bff35a0d62cea510
Reviewed-on: https://webrtc-review.googlesource.com/33080
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21283}
Also cleaned up a bit in RtpFrameReferenceFinder.
Bug: chromium:762556
Change-Id: Ib08d2e7ce4b146b359ce9ba823f3aa15776c71bc
Reviewed-on: https://webrtc-review.googlesource.com/32301
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21282}
