This CL reduces the complexity and heap usage of the adaptive filter
in AEC3 by avoiding to compute these for the shadow
filter. In particular it
-Moves to compute the ERL, frequency response and impulse response
on an on-demand basis.
-Stores the ERL, frequency response and impulse response outside
of the adaptive filter.
All the changes have been tested for bitexactness on a sizeable
amount of recordings.
Bug: webrtc:10913
Change-Id: If83c236a6e3f2e489be129b9ebf6143a72f521d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151138
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29081}
All channels are populated by RenderDelayBuffer. but all other
dependent modules are hardcoded to do their regular mono processing
on the first channel.
Bug: webrtc:10913
Tested: Bitexactness on a large set of aecdumps
Change-Id: I11d11aa0ad3da0f244c0ec020d2c9f0f4a735834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151640
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29079}
The VectorBuffer and MatrixBuffer names are too generic for their use case.
Bug: webrtc:10913
Change-Id: Ideecd0d27e07487a85a61dc6474e69733d07dcd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151602
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29076}
All dependent modules are hardcoded to do their regular mono processing on the first channel.
This _almost_ makes RenderBuffer multi-channel: FftData is still only mono.
Bug: webrtc:10913
Change-Id: Id5cc34dbabfe59e1cc72a9675dc7979794e870ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151139
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29074}
This behavior seems to conform to expectations from the rate allocators,
using this signal to chose which layers to enable and then distributing
the remaining bandwidth to the activated layers.
Bug: webrtc:10126
Change-Id: If0e1b27dc672ec2fbb30a5f5ac734e5ed4b42e45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151306
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29065}
This CL propagates the number of render and capture channels into
the echo subtractor and the adaptive filters.
Bug: webrtc:10913
Change-Id: I5ffff24ff64b7cc0f262bf008b34b6dfca1e78f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151300
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29059}
Update the |cumulative_lost_| counter per received packet. The rules
follow from RFC 3550 and are fairly simple: Decrement the counter by
one for every received packet. For every in-order packet, i.e., increasing
|received_seq_max_|, add that change to |cumulative_lost_|.
Net change is zero as long as packets are received in proper sequence.
This way, GetStats() always returns an up-to-date value, independent
of the timing of RTCP report blocks.
For RTCP reports, keep a workaround to never report negative cumulative loss.
Bug: webrtc:10679
Change-Id: I47ff3bf266ff2382f405ec9828d34f7fad7068b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150641
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29058}
These fixes are automatically created by various analysis tools, but have been manually triggered to be applied.
* the 'empty' method should be used to check for emptiness instead of 'size' (3 times)
* using decl 'Return' is unused (4 times)
* using decl '_' is unused (3 times)
* using decl 'DoAll' is unused (2 times)
* using decl 'SetArgPointee' is unused
* using decl 'Dlrr' is unused
* using decl 'IsEmpty' is unused
* redundant get() call on smart pointer
* using decl 'Invoke' is unused (2 times)
* using decl 'SizeIs' is unused (3 times)
* using decl 'make_tuple' is unused
* using decl 'NiceMock' is unused
* using decl 'SaveArg' is unused (2 times)
* using decl 'AtLeast' is unused
* using decl 'ElementsAre' is unused
* using decl 'Gt' is unused
Bug: None
Change-Id: I97658fb0e94620b8319d7c3da29b15e27ec23188
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151133
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29056}
This reverts commit eec5fff4df92b2330e5fec67ff08c7cbb4c4ab8d.
Reason for revert: Some crashes found by the fuzzer
Original change's description:
> Refactor FEC code to use COW buffers
>
> This refactoring helps to reduce unnecessary memcpy calls on the receive
> side.
>
> This CL replaces
> |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class,
> removes |length| field there, and does necessary changes.
>
> This is a reland of these two CLs with fixes:
> https://webrtc-review.googlesource.com/c/src/+/144942
> https://webrtc-review.googlesource.com/c/src/+/144881
>
> Bug: webrtc:10750
> Change-Id: I76f6dee5a57ade59942ea2822ca4737edfe6438b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29035}
TBR=brandtr@webrtc.org,ilnik@webrtc.org,stefan@webrtc.org
Change-Id: Id3d65fb1324b9f1b0446fe217012115ecacf2b40
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10750
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151130
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29043}
This is a reland of a66395e72f9fc86873bf443579ec73c3d78af240
Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
>
> This is a reland of f3a197e55323aee974a932c52dd19fa88e5d4e38
>
> Original change's description:
> > Add core multi-channel pipeline in AEC3
> > This CL adds basic the basic pipeline to support multi-channel
> > processing in AEC3.
> >
> > Apart from that, it removes the 8 kHz processing support in several
> > places of the AEC3 code.
> >
> > Bug: webrtc:10913
> > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29017}
>
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}
Bug: webrtc:10913
Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29042}
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.
Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
This metric is not used anywhere and is not calculated correctly when the delay manager is in relative arrival delay mode.
Bug: webrtc:10333
Change-Id: Iac79ab40b79b17802ad9d626c130e82f761bae26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150786
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29037}
This refactoring helps to reduce unnecessary memcpy calls on the receive
side.
This CL replaces
|uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class,
removes |length| field there, and does necessary changes.
This is a reland of these two CLs with fixes:
https://webrtc-review.googlesource.com/c/src/+/144942https://webrtc-review.googlesource.com/c/src/+/144881
Bug: webrtc:10750
Change-Id: I76f6dee5a57ade59942ea2822ca4737edfe6438b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29035}
This is a reland of f3a197e55323aee974a932c52dd19fa88e5d4e38
Original change's description:
> Add core multi-channel pipeline in AEC3
> This CL adds basic the basic pipeline to support multi-channel
> processing in AEC3.
>
> Apart from that, it removes the 8 kHz processing support in several
> places of the AEC3 code.
>
> Bug: webrtc:10913
> Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29017}
Bug: webrtc:10913
Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29034}
The change is behind a field trial. The intention is to use this
to (heuristically) base the bandwidth estimate only on video packets
even if both audio and video packets have transport sequence numbers.
Bug: webrtc:10932
Change-Id: I6cc5bb9ab6f1a3f25b84ee6ac78e4abb4112032e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150787
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29031}
This reverts commit 7db900e2e78d1644a173a0bc505ad52c61c43f9b.
Reason for revert: Speculative revert
Original change's description:
> Simplify pacer queue
>
> This CL simplifies the pacer queue by removing the now unnecessary
> beginpop/cancelpop/finalizepop methods. Instead there's a const top()
> and a pop() much like an stl queue.
> Old methods using the deprecated pacing code path are cleaned away.
>
> Bug: webrtc:10633
> Change-Id: Ib6da4d46a571bf56415172b790cc9e3f63206a38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150522
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28997}
TBR=sprang@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10633
Change-Id: I38f61afed4f4d542e236bcce3152a3aab52c6e6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29030}
This reverts commit f3a197e55323aee974a932c52dd19fa88e5d4e38.
Reason for revert: Speculative revert, as this may'be broken some build bots
Original change's description:
> Add core multi-channel pipeline in AEC3
> This CL adds basic the basic pipeline to support multi-channel
> processing in AEC3.
>
> Apart from that, it removes the 8 kHz processing support in several
> places of the AEC3 code.
>
> Bug: webrtc:10913
> Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29017}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I877d2993b9ccf024bd1d57bca1513c3e24d0bed3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29022}
This CL adds basic the basic pipeline to support multi-channel
processing in AEC3.
Apart from that, it removes the 8 kHz processing support in several
places of the AEC3 code.
Bug: webrtc:10913
Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29017}
This CL increases the maximum supported sample rate so that all rates
up to 384000 Hz are handled.
The CL also adds tests that verifies that APM works as intended for
different combinations of number of channels and sample rates.
Bug: webrtc:10882
Change-Id: I98738e33ac21413ae00fec10bb43b8796ae2078c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150532
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29014}
Also adding code in preparation of hiding the Module
implementation in PacedSender. The implementation details of
how the PacedSender+ProcessThread interaction works, has now
been moved into PacedSender (and out of RtpTransportControllerSend).
Instead of adding a "GetModuleImplementationForTesting" method
to the PacedSender class (which would have been the lazy way
out), I incorporated MockedProcessThread in the PacedSender tests.
This means more boilerplate code but the Module functionality
can be tested separately from the PacedSender and down the line
I think it would be a good idea to start using a separate thread
in the test, which is how the class under test is really used
in production.
Bug: none
Change-Id: Iec1b7c97cb0b363b331143ca70545e6ebafe2cd4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149176
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29011}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
This fixes a bug in delay manager relative arrival delay mode that caused the effective minimum target level to be 2 packets instead of 1.
Bug: webrtc:10333
Change-Id: I33d32c8da692a3db22179edb923873d307f740fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150785
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29002}
The new target does not depend on libjingle_peerconnection_api, and to
do this, the named "audio" and "video" string literals had to be moved from
media_stream_interface.cc to media_types.cc.
In this cl, the dependency on libjingle_peerconnection_api can be
dropped from a few targets.
No-Presubmit: True
Bug: webrtc:8733
Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28998}
This CL simplifies the pacer queue by removing the now unnecessary
beginpop/cancelpop/finalizepop methods. Instead there's a const top()
and a pop() much like an stl queue.
Old methods using the deprecated pacing code path are cleaned away.
Bug: webrtc:10633
Change-Id: Ib6da4d46a571bf56415172b790cc9e3f63206a38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150522
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28997}
This is a reland of ada8e17125d2124f5bcdc1558182ce95d6311d93
Original change's description:
> Delete mac_utils.h and mac_utils.cc
>
> They defined two functions: ToUtf16 and ToUtf8. The former was unused,
> and the latter is moved to
> modules/desktop_capture/mac/window_list_utils.cc, the only user.
>
> Tbr: sergeyu@chromium.org
> Bug: None
> Change-Id: Ib8a513da42e43ba8d41a2de4c1645b3f48448dc9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148531
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sergey Ulanov <sergeyu@google.com>
> Cr-Commit-Position: refs/heads/master@{#28913}
Tbr: kthelgason@webrtc.org
Bug: None
Change-Id: Icda3a2d6e98cfcaf8618035c248104c234e809ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150109
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28979}
This interface/config field is now unused, let's remove it.
Bug: webrtc:10633
Change-Id: I56ff3d47ba784d973de411ada52ec9485bad9864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150531
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28978}
The packets belonging to a frame were kept in PacketBuffer
until the frame was decoded. This CL clears the dependencies
of an existing RtpFrameObject to PacketBuffer so that we can
free up PacketBuffer as soon as the RtpFrameObject is created.
Bug: none
Change-Id: Ic939be91815519ae1d1c67ada82006417b2d26a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149818
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28977}
Number of received FEC bytes is used for the
WebRTC.Video.FecBitrateReceivedInKbps UMA histogram. Before this cl,
that value is based on a FEC packet counter updated by
ReceiveStatistics::FecPacketReceived. This cl deletes that method, and
instead adds a byte count to the FecPacketCounter struct, which is
maintained by the UlpFecReceiver and used for other FEC-related stats.
Bug: webrtc:10917
Change-Id: I24bd494b6909a2fe109d28e2b71ca8f413d05911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150533
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28976}
Previously the kDontRetransmit value was used to indicate packets that
should not be retransmitted but were put in the RtpPacketHistory anyway
as a temporary storage while waiting for a callback from PacedSender.
Since PacedSender now always owns the delayed packets directly, we can
remove all usage of StorageTye in RtpPacketHistory, and only put
packets there after pacing if RtpPacketToSend::allow_retransmission()
returns true.
Bug: webrtc:10633
Change-Id: I003b76ba43bd87658cc2a39e908fd28ebcd403f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150521
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28974}
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.
Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
This CL removes the old non-paced code path and instead uses a helper
class to just immediately pass the packet through the same code path as
when an actual pacer is used.
Bug: webrtc:10633
Change-Id: Id9a3ee4719829ad07710f5468e5452aa4bc8570b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150530
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28963}
The methods are no longer in use, this CL cleans away references and
updates any tests using them.
Bug: webrtc:10633
Change-Id: I2db301e0a021a2f85a8b9a74e409303baba407da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28956}
This CL corrects the way the audio processing module handles sample rates that
don't allow partitioning the data into evenly sized 10 ms chunks, examples
being 22050 Hz and 11025 Hz.
Bug: webrtc:10882
Change-Id: I35d738f8a0e1debc443fe5d473c0d666a7ba8d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150526
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28953}
This CL removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.
Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
This is a reland of b7b8e30cb44c41f51dbbefb9a9160e6dfe869c5b
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
Bug: webrtc:10863
Change-Id: Ic626b99b099248f0d8a677dc4cfe1505e14ae7cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150330
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28949}
This reverts commit b7b8e30cb44c41f51dbbefb9a9160e6dfe869c5b.
Reason for revert: Broke ApmTest.Process test in internal iOS waterfall
Original change's description:
> Reland Process 8 kHz audio as 16 kHz internally of the audio processing module
>
> This CL relands the code from the CL "Process 8 kHz audio as 16 kHz internally
> of the audio processing module" which by mistake was reverted via a rebase in
> another CL.
>
> The CL changes the behavior of APM for 8 kHz so that it is internally
> processed as 16 kHz.
>
> Bug: webrtc:10863
> Change-Id: I32a57b2d279c2134125667c19b09cfda381a33c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150221
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28944}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: Ia49e07b0c25c49da646917516e317f1d57cc4e84
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150326
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28948}
