This work is also done as a preparation for adding stereo support to the
FineAudioBuffer.
Review hints:
Actual changes are in modules/audio_device/fine_audio_buffer.h,cc, the rest is
just adaptations to match these changes.
We do have a forked ADM today, hence, some changes are duplicated.
The changes have been verified on all affected platforms.
Bug: webrtc:6560
Change-Id: I413af41c43809f61455c45ad383fc4b1c65e1fa1
Reviewed-on: https://webrtc-review.googlesource.com/70781
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22938}
Any native call before PeerConnectionFactory.initialize() will fail.
This means creation of JavaAudioDeviceModule will fail if it's created
before PeerConnectionFactory.initialize(). Clients should technically
always call PeerConnectionFactory.initialize() first, but we can help
the situation by deferring creation of the native ADM until it's
actually needed.
Bug: webrtc:7452
Change-Id: I53df2bdb980a8bdc413975f1cea6bcf297b453d5
Reviewed-on: https://webrtc-review.googlesource.com/70763
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22936}
We can then also drop the system_wrappers dependency from the common_video
build target.
Bug: webrtc:6733
Change-Id: I501113d100322d1ebc51b2286970697a24b70a43
Reviewed-on: https://webrtc-review.googlesource.com/70381
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22934}
This makes it easier to refactor the interface in upcoming CLs.
Bug: None
Change-Id: I67d0216e24f087294e95ac96f7278f302bf69832
Reviewed-on: https://webrtc-review.googlesource.com/71041
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22933}
This is a reland of fb82fcc7f9c414dc8ba1ddd314e9524fee54cb80
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This reverts commit 9d8f3850f4c4faad5dc5ab32ab6f2c9c43df7b6c.
Reason for revert: Breaks some trybots: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Release/builds/12793.
Original change's description:
> Adding absl includes and defines to rtc_* templates.
>
> This CL implicitly adds the -I compiler flag and absl macros to WebRTC
> templates. In order to include absl headers using relative paths, WebRTC
> needs to ensure that all its build targets are able to see absl headers.
>
> This can also be done with public_deps, but WebRTC is trying to avoid
> it because it creates problems with other build systems. Given this
> constraint, using rtc_* templates is the most reliable solution.
>
> Please note that rtc_* templates are adding absl includes and defines
> as public_configs, this means that build targets with WebRTC targets
> in their public_deps will propagate these configs following the GN
> guideline.
>
> Bug: webrtc:8821
> Change-Id: I4aa594a524f4bd045bcb3e80d76cc27f06fe01d7
> Reviewed-on: https://webrtc-review.googlesource.com/70367
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22927}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org
Change-Id: Id8e1f881c57553386566eb1970f6b9f8632cab37
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8821
Reviewed-on: https://webrtc-review.googlesource.com/71000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22928}
This CL implicitly adds the -I compiler flag and absl macros to WebRTC
templates. In order to include absl headers using relative paths, WebRTC
needs to ensure that all its build targets are able to see absl headers.
This can also be done with public_deps, but WebRTC is trying to avoid
it because it creates problems with other build systems. Given this
constraint, using rtc_* templates is the most reliable solution.
Please note that rtc_* templates are adding absl includes and defines
as public_configs, this means that build targets with WebRTC targets
in their public_deps will propagate these configs following the GN
guideline.
Bug: webrtc:8821
Change-Id: I4aa594a524f4bd045bcb3e80d76cc27f06fe01d7
Reviewed-on: https://webrtc-review.googlesource.com/70367
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22927}
Change error code for "state mismatch" to "State error",
and also change some parameter errors to "Illegal parameter".
Bug: chromium:819629
Change-Id: I9347d4161344b4ff2bcb58ad82fa6d533cd476fb
Reviewed-on: https://webrtc-review.googlesource.com/69815
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22924}
- protobuf library
- file_wrapper.h
These appear to have been left behind during the AecDump refactoring.
After this CL, APM no longer depends on zlib by default! :)
Bug: webrtc:9139
Change-Id: I12a8df2a17a575515b9c07165825f0879c4e15eb
Reviewed-on: https://webrtc-review.googlesource.com/70762
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22923}
The SequenceBuffer class template implements a linear buffer with a Push
operation that is used to add a fixed size chunk of new samples into the
buffer. Its properties are its size and the size of the chunks that are
pushed. It is used to implement the pitch buffer in the RNN VAD feature
extractor, for which a ring buffer would be a painful choice.
Bug: webrtc:9076
Change-Id: I4767bf06d5a414dbed724a96ea4186ef013a1e30
Reviewed-on: https://webrtc-review.googlesource.com/70204
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22919}
This ensures that packet time effects are simulated properly when using
a demuxer with FakeNetworkPipe. Previously this was only done using a
receiver.
This prepares for a CL with a test depending on this behavior.
Bug: webrtc:9054
Change-Id: I031acc9e18adc2891d3e396352dcd2614211909c
Reviewed-on: https://webrtc-review.googlesource.com/67342
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22918}
This CL adds support for using any externally reported audio buffer
delay to set the initial alignment in AEC3 which is used before the
AEC has been able to detect the delay.
Bug: chromium:834182,webrtc:9163
Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb
Reviewed-on: https://webrtc-review.googlesource.com/70580
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22917}
This CL increases the transparency in AEC3 during regions of low level
echo. What is done is:
-Low-level echoes are smoothly weighted so as to be deemed less
disturbing.
-The time-domain masking effect of the nearend speech is increased for
all frequencies.
-A separate, even more increased, time-domain masking effect is
introduced for lower frequencies.
-The intra-band masking is reduced to reduce the risk of echo leakage.
-The limiting of maximum gain due to filter-bank dynamics is removed
as the usecase for it could no longer be identified.
Bug: webrtc:9159,cromium:833801
Change-Id: I289b92919763124d6c5e5ede19e9a5917877c654
Reviewed-on: https://webrtc-review.googlesource.com/70421
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22915}
This removes posix-only files from the default source set, forcing
them to be explictly included for the desired configurations.
Bug: chromium:812974
Change-Id: Ic4a1561f87395801ca4d3936a95a66fd4270c5ef
Reviewed-on: https://webrtc-review.googlesource.com/70564
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Fabrice de Gans-Riberi <fdegans@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22913}
Signaling per-packet stats invokes GetRemoteAddress and a warning log is
output for each signal in SendTo for UDP sockets. This creates noise in
the debug log.
Bug: webrtc:9103
Change-Id: Ibf4667cf196aef971452f745bef3c02fb1dbf3d4
Reviewed-on: https://webrtc-review.googlesource.com/70364
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22911}
last_target_rate_ and current_target_rate_msg_ both kept track of the
same value. pacer_configured_ was not used at all. This CL removes
both last_target_rate_ and pacer_configured_.
Bug: None
Change-Id: Ieb15c38c7d0c44730bef95492e3e677d505f054e
Reviewed-on: https://webrtc-review.googlesource.com/70184
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22909}
Fixes a mismatch between "useHardware" and "disableBuiltIn" when
creating JavaAudioDeviceModule.
Bug: webrtc:7452
Change-Id: Ia5572822dc4514ff9a06811af1bdbb8362a2c71c
Reviewed-on: https://webrtc-review.googlesource.com/69987
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22908}
This ensures that capture devices that relies on the frame being
released to continue are not blocked by storing the pending frame.
Bug: None
Change-Id: If501bca4ab7bda5e0438d24e98d67df589ad6a6d
Reviewed-on: https://webrtc-review.googlesource.com/70480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22907}
This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
One implication is that encoder is not created until the first
frame arrives, and some of the tests needed updates to emit a
frame or two.
Bug: webrtc:8830
Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
Reviewed-on: https://webrtc-review.googlesource.com/64885
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22905}
Logging the OpenGL shader source code makes it easier to debug problems.
Bug: None
Change-Id: Ie4724b1353511eae3806e98270b04e5daa4c11fc
Reviewed-on: https://webrtc-review.googlesource.com/69322
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22900}
This reverts commit 8628f5bb7c7f5bd0373567095af08cebe8bb7f8d.
Reason for revert: iOS buildbot failing
Original change's description:
> AGC2 RNN VAD: initial build targets
>
> rnn_vad_tool is an executable that reads a wav file of any sample rate
> compatible with 10 ms frames that are resampled and, when the VAD is
> fully landed, will process the resampled frames to compute the VAD
> probability.
>
> To avoid mac, win and ios trybot failures, to_be_removed.h/.cc have
> been added and will be removed as soon as the :lib target includes
> code that leads to a non-empty static lib file on those platforms.
>
> Bug: webrtc:9076
> Change-Id: I810c08acfa1adf2029e3baac2adda3045ae5214a
> Reviewed-on: https://webrtc-review.googlesource.com/70202
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22898}
TBR=alessiob@webrtc.org,aleloi@webrtc.org
Change-Id: Ic6014dde78b0ef371804c52608145ba8acdd9c97
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9076
Reviewed-on: https://webrtc-review.googlesource.com/70144
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22899}
rnn_vad_tool is an executable that reads a wav file of any sample rate
compatible with 10 ms frames that are resampled and, when the VAD is
fully landed, will process the resampled frames to compute the VAD
probability.
To avoid mac, win and ios trybot failures, to_be_removed.h/.cc have
been added and will be removed as soon as the :lib target includes
code that leads to a non-empty static lib file on those platforms.
Bug: webrtc:9076
Change-Id: I810c08acfa1adf2029e3baac2adda3045ae5214a
Reviewed-on: https://webrtc-review.googlesource.com/70202
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22898}
PCMFile.cc uses RTC_DCHECK. include and depend on rtc_base:checks target directly
change usage of value_or by using explicit constructor instead of implicit
Bug: webrtc:9078
Change-Id: I63c596b8a05b387e56df846b15c33a605fbad4e6
Reviewed-on: https://webrtc-review.googlesource.com/69985
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22897}
Commit bbf21a3fd617abb37567a86e65f8ba18b8d64eb2 ("Remove dependencies on
modules:module_api from AudioProcessing") causes the build to fail with
libstdc++ due to several files using memcpy(3) or memset(3) while relying on
string.h being included implicitly by other headers.
Bug: webrtc:9139
Change-Id: Ib73284962f8694d8bed0551968265bfd13cab967
Reviewed-on: https://webrtc-review.googlesource.com/70180
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#22895}
In preparation for moving creation of encoders to VideoStreamEncoder;
then this field is needed for calling
VideoEncoderFactory::CreateVideoEncoder.
Bug: webrtc:8830
Change-Id: Ie107a93596e22a5e7b9a9147bd85a93cf84b15a3
Reviewed-on: https://webrtc-review.googlesource.com/70221
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22894}
I found one additional way a crash could occur: "OnRtpTransportChanged"
being called instead of "OnDtlsTransportChanged", due to a mixup of m=
section types. I could reproduce this by:
1. Applying description with RTP data channel m= section.
2. Applying description with both a rejected RTP data channel m=
section and rejected SCTP m= section.
This is a very strange scenario, but maybe there are other ways to
reproduce that I haven't thought of.
The solution is to combine "OnRtpTransportChanged" and
"OnDtlsTransportChanged", and not do anything with the content type.
This simplifies the code a bit as well.
Bug: chromium:827917
Change-Id: If6818ea0c41573255831534060b30c76a6544e04
Reviewed-on: https://webrtc-review.googlesource.com/70360
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22893}
The existing rule regards a candidate pair as not receiving if it does
not receive any data packet, connectivity check, or connectivity check
response for a timeout period since the last receipt of any packet
above. A backup candidate pair typically sends connectivity checks at a
slow pace to preserve the battery life, and the existing rule however
declares receiving timeout for backup candidate pairs as a side effect.
This is a result of the conflicting value of the receiving timeout
period and the longer default connectivity check interval for backup
candidate pairs.
The new rule regards any candidate pair that has its last connectivity
check acknowledged by a response as receiving.
Bug: webrtc:9145
Change-Id: Ie0171fd83aca3d6a0a465885be32f0854856be7f
Reviewed-on: https://webrtc-review.googlesource.com/69784
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22892}
It seems iOS trybots are the only ones that build "all". This causes
problems when using Abseil because some targets in
//third_party/abseil-cpp fail to build (because they depend on CCTZ).
Bug: webrtc:8821
Change-Id: I017ecb0527a7e3f3c59f41053fa1878d16cbe4e9
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/70140
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22890}
This is a reland of fc43d11717e16dd427ac84fee614e5511e43cefd
Original change's description:
> Add thread checker to PortAllocator and its subclasses and fix a bug
> causing memory contention by threads.
>
> PortAllocator and its subclasses assume all of their methods except the
> constructor must be called on the same thread (the network thread in
> practice). This CL adds a thread checker to PortAllocator and its
> subclasses for thread safety, and fixes bugs of invoking some of their
> methods in PeerConnection on the signaling thread.
>
> Bug: webrtc:9112
> Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
> Reviewed-on: https://webrtc-review.googlesource.com/66945
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22814}
Bug: webrtc:9112
Change-Id: I5c7377f05c0daccbe469e2fdbdfacabc5c222f4c
Reviewed-on: https://webrtc-review.googlesource.com/69422
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22889}
