This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
Original change's description:
> Put internal video codec factories into separate target
>
> The purpose is to start splitting out the dependencies to the built-in
> SW video codecs, so that clients can decide to not depend on them and
> get a reduction in binary size.
>
> Replaces https://webrtc-review.googlesource.com/c/src/+/29101
>
> Bug: webrtc:7925
> Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> Reviewed-on: https://webrtc-review.googlesource.com/33420
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21381}
Bug: webrtc:7925
Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
Reviewed-on: https://webrtc-review.googlesource.com/35261
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21389}
We currently use raw jobject in our code mixed with sporadic
ScopedLocalRefFrame. This CL moves every jobject into a scoped object,
either local, global, or a parameter. Also, this CL uses the JNI
generation script to generate declaration stubs for the Java->C++
functions so that it no longer becomes possible to mistype them
without getting compilation errors.
TBR=brandt@webrtc.org
Bug: webrtc:8278,webrtc:6969
Change-Id: Ic7bac74a89c11180177d65041086d7db1cdfb516
Reviewed-on: https://webrtc-review.googlesource.com/34655
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21387}
This reverts commit 51698aefd4925f2dfa0310a321f836d433fa9258.
Reason for revert: Breaks builds because badly formatted deps
Original change's description:
> Put internal video codec factories into separate target
>
> The purpose is to start splitting out the dependencies to the built-in
> SW video codecs, so that clients can decide to not depend on them and
> get a reduction in binary size.
>
> Replaces https://webrtc-review.googlesource.com/c/src/+/29101
>
> Bug: webrtc:7925
> Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> Reviewed-on: https://webrtc-review.googlesource.com/33420
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21381}
TBR=magjed@webrtc.org,andersc@webrtc.org
Change-Id: Ib85f77fea756f4beb6a95b45cb132cbdc424ef00
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7925
Reviewed-on: https://webrtc-review.googlesource.com/35260
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21383}
Trivial change. Adding Alex as TBR. Same log exists for playout already.
This change makes is easier to compare logs.
NOTRY=TRUE
TBR=glaznev
Bug: NONE
Change-Id: I5dd23a4435d7816d8c171a0769132ac9d2f7f5aa
Reviewed-on: https://webrtc-review.googlesource.com/34654
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21361}
This reverts commit 28a06b16cc4daa9f380ad45af8acfd11b6057283.
Reason for revert: Causes some unexpected perf changes.
Original change's description:
> Smoother frame dropping when screenshare_layers limits fps
>
> 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}
TBR=ilnik@webrtc.org,sprang@webrtc.org
Change-Id: I7ca5b0c847310dbb11dce28773082eac946c0ba4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:4172
Reviewed-on: https://webrtc-review.googlesource.com/34780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21354}
I realized I could use configs to fix some duplication that I
partially introduced.
Verified APM_DEBUG_DUMP is set appropriately by looking at the
compiler command line.
Bug: webrtc:6828
Change-Id: Ia990e2721546d65639567cd3ab788439e328c5da
Reviewed-on: https://webrtc-review.googlesource.com/34642
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21349}
Changing window type which improves the filter accuracy
at the cost of a slight reduction in convergence time.
Bug: webrtc:8661
Change-Id: Id0e5c66ec179f94471cbca0a2b8d1b94d8146ca6
Reviewed-on: https://webrtc-review.googlesource.com/34501
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21347}
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}
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}
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}
This CL removes the following GN variables: rtc_build_libyuv,
rtc_libyuv_dir (as requested in webrtc:7906).
It also removes some unneeded dependencies on //third_party/libyuv.
WebRTC targets were using public_deps to depend on //third_party/libyuv
and this created a build graph where targets that were depending on
//third_party/libyuv were not declaring the dependency to GN because
they were somehow getting it from another target that was exposing
//third_party/libyuv header files even if it wasn't directly depending
on it.
Bug: webrtc:8605, webrtc:7906
Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb
Reviewed-on: https://webrtc-review.googlesource.com/32201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21275}
With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter
Though hacky, this is very similar to currently used implementation
in the RTCPSender::SendFeedbackPacket
Bug: webrtc:8239
Change-Id: I237b422ae1594dede78cb63daa4aa42b6774d6fe
Reviewed-on: https://webrtc-review.googlesource.com/32680
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21274}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
in particular change bitrate type to int64_t to follow style guide.
With an extra interface it will allow to add both RtpRtcp module
and RtcpTransceiver as feedback sender to PacketRouter
Bug: webrtc:8239
Change-Id: I9ea265686d7cd2d709f0b42e8a983ebe1790a6ba
Reviewed-on: https://webrtc-review.googlesource.com/32302
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21250}
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}
FFMpeg H264 decoder uses YUVJ420 when video_full_range_flag=1 in
bitstream.
Information about color range might be useful for color converter
and renderer. But currently there is no way to extract it from
the wrapper.
Bug: webrtc:8185
Change-Id: Ifd1113f0eee3d7b5906d0cefbc29b4a1061262f6
Reviewed-on: https://webrtc-review.googlesource.com/32000
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21246}
This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
