This CL works around an "Explicit specialization after instantiation
error" when building with clang-cl and is_component_build=true (see
crbug.com/1018579). On top of that it uses "template instantiation
declarations/declarations" in order to avoid to instantiate the
template in clients code.
TBR: hbos@webrtc.org
Bug: webrtc:9419, chromium:1018579
Change-Id: I1b2862de678586afc81e8f7a407947322f8a06c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158795
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29683}
This reverts commit 9dda1b3a484ebeef921e419406402039f3852427.
Reason for revert: The CL is causing downstream issues
Original change's description:
> Correct AEC3 multichannel functionality activation
>
> This CL corrects the AEC3 multichannel activation
> to also work for the case when a factory is used
> for the activation.
>
> Bug: webrtc:10913
> Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29676}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: Ic487f77f5c11485a0f25a2a1d3797d0ec956f913
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158797
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29678}
This CL corrects the AEC3 multichannel activation
to also work for the case when a factory is used
for the activation.
Bug: webrtc:10913
Change-Id: Ic2807d8bcef759261fde14447cff30633ba248dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158794
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29676}
Injecting both a custom NetEqFactory and an AudioDecoderFactory is not
supported, in that case the AudioDecoderFactory should be wrapped inside
the NetEqFactory.
Bug: webrtc:11005
Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29673}
Move the GCM srtp cipher suites below the default SRTP_AES128_CM_SHA1_80 one.
This will not negotiate them by default since they have an impact on packet overhead for audio-only calls.
GCM can still be negotiated if the peer offers it as preferred cipher suite or answers with just that cipher suite.
BUG=chromium:713701
Change-Id: I79bd4ab827e5c7f55f5550d14db3f4217a7eff86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158404
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29672}
This CL also introduces NetEqFactory and NetEqControllerFactory
interfaces, as well as several convenience classes for working with
them: DefaultNetEqFactory, DefaultNetEqControllerFactory and
CustomNetEqFactory.
Bug: webrtc:11005
Change-Id: I1e8fc5154636ac2aad1a856828f80a2a758ad392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156945
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29671}
This is not used and adds a lot of maintenance overhead to
the code since it requires that the transport feedback adapter
communicates directly with audio send stream.
This also means that the packet loss tracker used as input for
this can be removed and a lot of wiring up code overall.
Bug: webrtc:9883
Change-Id: I25689fb622ed89cbb378c27212a159485f5f53be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156502
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29667}
For the automatic content type detection we need to know if the update
rect is trusted or just not available.
Currently we only care if it's not empty, so in case of no update rect
available, full frame resolution was set as a changed region.
This CL makes the update_rect field optional but should be a no-op in the
current code, as absence of update_rect is treated as a full update via
a new getter method |update_rect_or_full_frame()|.
Bug: webrtc:11058
Change-Id: I913545b71ac2fc845861549ac34eb1b630012109
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158673
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29654}
This does not help the attached bugs, but it does allow greater control
over what JSON code is running where. Long-term, the JSON library used
for parsing configuration should likely be a library already present
in Chromium builds, to avoid duplication. And if that happens, then
WebRTC bug 9804 may be passé.
Note that this CL also sorts our poisons alphabetically.
Bug: chromium:895814, webrtc:9804
Change-Id: I70c3efe05a0eba9212895407f73978d8216df920
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158400
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29615}
Chromium tests depend on api/task_queue:task_queue_test but it
cannot be added to the WebRTC component in Chromium (which is not
testonly).
A possible solution is to make api/task_queue:task_queue_test
depend on the WebRTC component which lives in Chromium only
when `build_with_chromium=true`.
Bug: webrtc:9419
Change-Id: I1cbe1fb97f21ef7a201d09d2f0f729104d01ed90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157427
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29578}
Type mismatches will silently fail and skip reading a parameter
in the JSON parsing, except when parsing a size_t from a negative int.
This CL updates the parsing to silently ignore negative values provided
for size_t config parameters, instead of explicitly DCHECKing.
Tested: Ran the fuzzer on the crash test case with + without this fix.
Bug: chromium:1016139
Change-Id: I3899e81e1183aa54b708030efeb6e0006b8cd881
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157894
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29568}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
This CL adds support for multiple channels in the reverb
modelling. As a side effect, it also partly adds multi-channel
supports for the sections of the code.
Beyond adding the multi-channel support, a bug is fixed as part of
this CL. Since the bug fix affects the bitexactness, as a safety
precaution the CL includes the ability to override the bugfix.
Apart from the contributions from the bugfix, the changes have
been verified to be bitexact for a large set of mono recordings.
Bug: webrtc:10913
Change-Id: I1f307b532be85ef4182f8db41384f44d40a25219
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156382
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29456}
This change keeps the original 48 kHz signal and uses it for the
fullband processing given that the following requirements are
fulfilled:
- Input signal is 48 kHz
- Output signal is 48 kHz
- Multiband processing is performed at 32 kHz
- The multiband processing does not modify the original signal
This avoids unnecessary, lossy resampling and band merging.
Bug: b/130016532
Change-Id: I690c26faba07eab0cbff6c0a95a81d89255dd1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155966
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29425}
Just fixing some minor TODOs in my name. Not worth splitting into
separate CLs as the changes are minor.
Bug: webrtc:9883
Change-Id: I05c54b76507a1d51b92cad080ca4e2dfe8546bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155520
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29377}
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total allocatable bitrate.
Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
This CL adds an optional second encoder factory to SimulcastEncoderAdapter,
that can be used to create software fallback adapter per simulcast layer.
It also adds logic to check if the encoder supports simulcast natively, if so
it only allocates a single instance and delegates the simulcast logic to that
encoder instead. This means we will be able to remove EncoderSimulcastProxy.
Bug: webrtc:11000
Change-Id: Ifd5f029cc281ee2cedf9d18efa5e7e460884d6ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155171
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29364}
The header files
api/congestion_control_interface.h
api/data_channel_transport_interface.h
api/datagram_transport_interface.h
api/media_transport_config.h
api/media_transport_interface.h
have been moved into the api/transport/ and api/transport/media
subdirectories.
Bug: webrtc:8733
Change-Id: I98752c4d1306b54559bafa71712b105932c08834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153522
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29357}
This is a reland of 809198edfff416fce8d75b574a43afab5e67b1cd
A fix was made in https://webrtc-review.googlesource.com/c/src/+/154343
which fixed the regression issues caused by the original patch.
Original change's description:
> Fix minor regression caused by a8336d3
>
> VideoEncoder::SetRates was being called unnessesarily when the fields
> appended to RateControlParameters were changed. Since SetRates only
> cares about RateControlParameters, it should have only been called if
> the RateControlParameters themselves were actually changed.
>
> Bug: webrtc:10126
> Change-Id: Ic47d67e642a3043307fec950e5fba970d9f95167
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152829
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#29208}
Bug: webrtc:10126
Change-Id: Iecc3ab6a5cd1193a1fa8e824dcf4f0b8165f9bf8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154359
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29356}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
