This CL changes so that the AGC legacy C code is built as C++.
The CL also
-removes #defines from the header files.
-adds namespaces
-removes unused code.
To simplify the review, the CL is partitioned into different patchsets
where each comprising of one step in the modification of the code
(e.g., patch set 1 performs the renaming of the .c files to .cc).
Bug: webrtc:5298
Change-Id: I362b17bde91142b2f2166acba4f2f888efd50fa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171064
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30847}
This CL renames the shadow filter in AEC3 to have the more accurate name
coarse filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "shadow" with "coarse" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: I28d6041d0d34e85f8f8048d004b44a1a5f07bb07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170981
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30846}
This CL renames the main filter in AEC3 to have the more accurate name
refined filter.
The CL consists of 3 main initial patch sets, designed to simplify
the review:
1) Replaces "main" with "refined" and adds a fall-back functionality
to support the old filter naming.
2) Renames the files according to the new naming.
3) Performs a "git cl format"
Bug: webrtc:8671
Change-Id: Ifd0aab34e291736a2250e0986348404618630b1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170825
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30843}
This CL changes so that the fft4g code is built as C++.
The CL also
-adds namespaces
-removes unused code.
To simplify the review, the CL is partitioned into different patchsets
where each comprising of one step in the modification of the code
(e.g., patch set 1 performs the renaming of the .c files to .cc).
Bug: webrtc:5298
Change-Id: I1dba78cc07c48622888be7b6f49a506795bc22ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171100
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30841}
A follow-up CL will move the rms_level.* files into the new target.
Bug: webrtc:11226
Change-Id: I59579b026346f627c0a2739d25f90c12bffbf248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171102
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30840}
std::vector::reserve has the effect to reserve space in memory but does
not affect the result of size(), which keeps on returning 0. If size is
0, however, data() might either return null or not [1].
This CL fixes the use of reserve() in favour of resize() which
effectively allocates the memory in the vector and updates its size.
This way size() returns a value bigger than 0 and data() returns a valid
pointer.
[1] https://en.cppreference.com/w/cpp/container/vector/data
Fixed: chromium:1059764
Change-Id: Ida3dbe643710c6895f09b9da87b0075b7d7b28df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170470
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Armando Miraglia <armax@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30836}
this is a step towards resolving own todo: making AssembleFrame part of
the VideoRtpDepacketizer interface and replacing codec check with a
call to a virtual function.
RtpVideoStreamReceiver has access to the VideoRtpDepacketizers,
PacketBuffer - hasn't.
Bug: None
Change-Id: I83df09975c092bdb71bab270ced356d79a50683d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168056
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30833}
This CL removes the redundant VAD output from the newly introduced
integer API in AudioProcessing.
Bug: webrtc:5298
Change-Id: Iad2b1b97ada7f4863139655526c110e326c6788a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170824
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30832}
This CL replaces all remaining usage of AudioFrame within APM,
with the exception of the AudioProcessing interface.
The main changes are within the unittests.
Bug: webrtc:5298
Change-Id: I219cdd08f81a8679b28d9dd1359a56837945f3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170362
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30831}
This CL corrects an issue in the storing of the processed capture output
into aecdump recordings for the case when the integer API interface is
used.
Bug: webrtc:11441
Change-Id: I24aad47b5d62e0738d412ec270ad1db3a76aa94f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170823
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30829}
The MockFecController is spitting out lots of warnings, as it is
being called when we don't care about it, in normal tests. Making
it a NiceMock allows it to receive calls without expectation and
not warn.
Bug: None
Change-Id: I1ea219c4665d86917718692dc013ae3ac47222ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30820}
It's better to set this to the right value in the C++ API rather
than the hack in catapult_uploader.py.
Bug: chromium:1029452
Change-Id: Ia942ff22f8422874cd226e6a7fdce20333ac4a50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170632
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30817}
This code was previously only exercised by
video_stream_adapter_unittest.cc and other tests, acting more like
integration tests than unit tests. Now that the VideoStreamAdapter is
in a good state, more extensive test coverage is added.
Testing includes:
- Default restrictions.
- Adapting up or down in "maintain-framerate", "maintain-resolution"
and "balanced", including...
- expecting how frame rate and/or resolution is affected,
- reaching kLimitReached,
- and reaching unrestricted.
- That "disabled" does not adapt.
- When adaptation is not possible, including...
- kInsufficientInput
- kAwaitingPreviousAdaptation
- kIsBitrateConstrained
- PeekNextRestrictions()
- "balanced" + "screenshare" = "maintain-resolution"
- Change degradation preference to/from "balanced" clears restrictions.
- That using invalidated adaptations triggers DCHECKs.
Bug: webrtc:11393
Change-Id: I28e2cf227bc1fd8871ee0d18d9570d4063449160
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170625
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30816}
Track the number of samples that are decoded until a fallback to
software decoder happens.
Bug: chromium:1061376
Change-Id: Ida3ae94034ec83a6d28001cb7be343b8b99b99c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170468
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30814}
This CL moves the implementation of of the AudioFrame
support from the implementation of AudioProcessing
to proxy methods that map the call to the integer
stream interfaces (added in another CL).
The CL also changes the WebRTC code using the AudioFrame
interfaces to instead use the proxy methods.
This CL will be followed by one more CL that removes
the usage of the AudioFrame class from the rest of
APM (apart from the AudioProcessing API).
Bug: webrtc:5298
Change-Id: Iecb72e9fa896ebea3ac30e558489c1bac88f5891
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170110
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30812}
Roll third_party to 67dbe528170eed566f8e8f271b6fa3bed81f9cce to add
third_party/turbine to support its usage in future chromium third_party
rolls.
Bug: None
Change-Id: I07297f5ab45d7d0b29f8c25219513fc6c3f79943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170631
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30811}
To unblock roll of newer build files for libaom that are using
differen build flags to include/exclude libaom
Bug: webrtc:11404
Change-Id: If06b63e0835e65113617efa29f34ba6bb309c16d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170630
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30810}
The problem that has stopped the uploads from working is likely that
json.dump writes -Infinity when encountering float('-inf'), but not
all JSON parsers handle that. Notably, the dashboard JSON library
doesn't when running in compressing mode.
I think the real fix is to land the float->double CL for the histogram
proto - I think we will not get float('inf') values then. Apply
this hack in the meantime.
Also remove allbins and story hacks, they're likely worse than the
defaults anyway.
Bug: chromium:1029452
Tbr: mbonadei@webrtc.org
Change-Id: I98e36307cc144bbe6878ba9d93c0a609cab71418
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170626
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30808}
There was quite a lot of excessive qualification in the test code.
Cleaned up by adding a few using directives.
BUG=None
Change-Id: I03487f457f15f663acf276ce2a4a9cbbc9804101
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170467
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30806}
It was possible before if an input fps dropped due to cpu adaptation
Also, this CL removes occasional test failure (it could've happened if
input framerate got very low)
Bug: webrtc:11432,webrtc:11426
Change-Id: Id1a4df23302f7b8ab6781f1e7cca5112bfcfe9ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170469
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30802}
Removes support for parsing and serializing
a=ssrc:1 mslabel:stream
a=ssrc:1 label:track
which have been superceeded by
a=ssrc:1 msid:stream track
a long time ago.
Bug: webrtc:7110
Change-Id: I3aca47728098b6e7e049b82ed34c59426d411c41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168244
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30801}
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
This CL creates a new stream interface and uses it to replace
most of the usage of AudioFrame in the non-test code.
The CL changes some of the test code as well, as the other
changes required that.
The CL will be followed by 2 more related CLs.
Bug: webrtc:5298
Change-Id: I5cfbe6079f30fc3fbf35b35fd077b6fb49c7def0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170040
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30799}
