Specifically, don't expect the ACM to be able to create encoders; we
have to give it an encoder that we make ourselves.
The new way of creating encoders used a 32 kbit/s bitrate
unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz
and 56 kbit/s for 32 kHz, which is what the old way of creating
encoders has used since forever.
I also had to change some test expectations on Opus, because the new
way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I
believe to be correct), while the old way defaults to 64 kbit/s in
both cases.
Bug: webrtc:8396
Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2
Reviewed-on: https://webrtc-review.googlesource.com/94540
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24375}
This reverts commit b3f5aed4332b25c3fa7e725126efd3d4262c385e.
Reason for revert: speculative revert, possibly breaking downstream projects
Original change's description:
> Remove the flag PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS.
>
> We now always enable any address ports, only using them if they end up
> using interfaces that weren't otherwise accessible. This flag is no
> longer used by downstream projects.
>
> TBR=deadbeef@webrtc.org
>
> Bug: None
> Change-Id: I6e4e93958cbc4300811bafb103f1a2e8732274ed
> Reviewed-on: https://webrtc-review.googlesource.com/85860
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23751}
TBR=qingsi@google.com,qingsi@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: None
Change-Id: I2c3850533cd72c021a42dcb160f73ca54a036a84
Reviewed-on: https://webrtc-review.googlesource.com/95341
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24373}
This reverts commit 625efe6dfe5b5681a2d8d3d5ecaa0c5ac9edd6ee.
Reason for revert: Speculative revert, possibly breaking downstream projects
Original change's description:
> Use also the related address in redundancy detection for candidates from
> the any-address/wildcard ports.
>
> A TURN server can allocate different IPs for different allocation
> requests from the same network interface, and a relayed candidate from a
> wildcard port is not considered duplicate with another relayed candidate
> using the same network interface in the current redundancy detection, if
> their mapped addresses (as the "related address" for relayed candidates)
> are different. Extra candidates would then be surfaced to the
> application unnecessarily.
>
> Bug: webrtc:9469
> Change-Id: I504fde3b70cd727ef6ad4517072dcf37328a8380
> Reviewed-on: https://webrtc-review.googlesource.com/86181
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24108}
TBR=steveanton@webrtc.org,tommi@webrtc.org,qingsi@google.com,qingsi@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Landing with NOTRY because ios64_sim_ios10_dbg bot is broken.
Passing all other bots.
NOTRY=True
Bug: webrtc:9469
Change-Id: Ica382280be5b4470cb99d8cccadcd87d3e7d37bb
Reviewed-on: https://webrtc-review.googlesource.com/95300
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24371}
This reverts commit 84916937b70472715efe5682bc273e91c3a72695.
Reason for revert: breaking downstream projects.
Original change's description:
> Update packetsLost and jitter stats any time a packet is received.
>
> Before this CL, the packetsLost and jitter stats (as returned by
> GetStats, at the API level) were only being updated when an RTCP SR or
> RR is generated. According to the stats spec, "local" stats like this
> should be updated any time a packet is received.
>
> This CL also fixes some minor issues with the calculation of packetsLost
> (and fractionLost):
> * Packets weren't being count as lost if lost over a sequence number
> rollover.
> * Temporary periods of "negative" loss (caused by duplicate or out of
> order packets) weren't being accumulated into the cumulative loss
> counter. Example:
> Period 1: Received packets 1, 2, 4
> Loss over that period: 1 (expected 4 packets, got 3)
> Reported cumulative loss: 1
> Period 2: Received packets 3, 5
> Loss over that period: -1 (expected 1 packet, got 2)
> Reported cumulative loss: 1 (should be 0!)
>
> Landing with NOTRY because Android compile bots are broken for an
> unrelated reason.
> NOTRY=True
>
> Bug: webrtc:8804
> Change-Id: I840ba34de8957b1276f6bdaf93718f805629f5c8
> Reviewed-on: https://webrtc-review.googlesource.com/50020
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23731}
TBR=danilchap@webrtc.org,deadbeef@webrtc.org,ossu@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Landing with NOTRY because ios64_sim_ios10_dbg bot is broken.
Passing all other bots.
NOTRY=True
Bug: webrtc:8804
Change-Id: I07bd6b1206d5a8d211792ad392842f9ed6c505e9
Reviewed-on: https://webrtc-review.googlesource.com/95280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24370}
The PCC congestion control algorithm divides time into consecutive
intervals called monitor intervals. This CL adds a class that is used by
PCC to measure the performance of sending at a certain rate during one
monitor interval.
Bug: webrtc:9434
Change-Id: Ia0447e224067d4ca807bcc6fd8083f9083385b91
Reviewed-on: https://webrtc-review.googlesource.com/87140
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24368}
Some functions were removed from the implementation files when this
flag is unset, but remained in the header files.
Bug: webrtc:7925
Change-Id: I9f8f969fb52f83c05ba02500a62dee78d1bcb0dc
Reviewed-on: https://webrtc-review.googlesource.com/80260
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24366}
This is a part of series of CLs adding PCC (Performance-oriented
Congestion Control).
Bug: webrtc:9434
Change-Id: Idd36d8abea008623ac64b939d0de7ee6001f7f23
Reviewed-on: https://webrtc-review.googlesource.com/86951
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24364}
This CL:
-Adds another config parameter that controls the duration of the initial
state.
-Adds reading of that parameter in audioproc_f from the json settings file.
-Adds missing reading of another parameter in audioproc_f from the json
settings file.
Bug: webrtc:8671
Change-Id: Ie6164c360492de5e6b0ade8838bbabe214560b5e
Reviewed-on: https://webrtc-review.googlesource.com/94621
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24360}
Add FakeVp8Encoder, change FakeEncoder to use BitrateAllocator for simulcast.
Change call_test to use VP8 payload name for simulcast tests.
Bug: none
Change-Id: I5a34c52e66bbd6c05859729ed14ae87ac26b5969
Reviewed-on: https://webrtc-review.googlesource.com/91861
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24359}
This removes the reserved bitrate feature as it is not currently used.
This saves debugging time as there is less code to investigate. This
also makes it easier to compare the code with the task queue based
version which lacks this feature.
Bug: webrtc:9586
Change-Id: I207624ceb8d203c88c3d01bfc753d60523f99fe4
Reviewed-on: https://webrtc-review.googlesource.com/92622
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24357}
Change experimental max bitrate setting from 1.6Mbps to 1.25Mbps in
order to allow a larger fraction of participants to receive this layer.
Add a new field trial to allow setting the number of temporal layers for
the high-quality simulcast stream in screensharing separately from the
temporal layer count for regular video.
Bug: webrtc:9477
Change-Id: I1341b774f870c50710901da24963bd3ede96ffd8
Reviewed-on: https://webrtc-review.googlesource.com/95101
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24356}
- Allow use of flexible mode which was blocked in webrtc:9261 since it
only worked together with old screen sharing. Since webrtc:9244 flexible mode
works with both normal and screen coding modes.
- Add unit test that checks that reference list encoder writes into RTP
payload descriptor and the predefined one match.
Bug: webrtc:9585
Change-Id: I4a1bdc51cbf15e7224cc7c271af8b2e3d46657d1
Reviewed-on: https://webrtc-review.googlesource.com/94778
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24355}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
The was another definition of the --logs flag in full_stack_tests.cc.
The effect was that --logs set an unused bool, and the value of
FLAG_logs in test_main.cc was always false, regardless of actual
command line.
Tbr: sprang@webrtc.org
Bug: None
Change-Id: I073f8025dd897909c7e2b8d7c0ee080cb4b456ca
Reviewed-on: https://webrtc-review.googlesource.com/94900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24347}
Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.
Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}
It didn't account for implicit bitrate allocation, which is used in
some unit tests, when bitrate distribution is done by the encoder
wrapper.
Bug: none
Change-Id: I8fcf28e10f7a6c378580ef917221ad5c8d3869c9
Reviewed-on: https://webrtc-review.googlesource.com/94775
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24343}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I1be59a9277aad8f51765c26e34ab16b63bcaeb42
Reviewed-on: https://webrtc-review.googlesource.com/94774
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24340}
Introduce SimulatedNetworkReceiverInterface and switch DirectTransport
on this interface. Also switch part of related users on
DefaultNetworkSimulationConfig.
This two changes united into single CL to prevent work duplication.
Most changes were done because of stop including fake_network_pipe.h
into direct_transport.h, so splitting this into 2 CLs will require
first fix all imports of fake_network_pipe.h and then replace them
on new API imports again.
Bug: webrtc:9630
Change-Id: I87d4a6ff1bab72d04a9871a40441f4fbe028f4e6
Reviewed-on: https://webrtc-review.googlesource.com/94762
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24336}
Linux has all hw webcams now, but it's tricky to stop invoking this
script just on Linux, so just make it do nothing for now.
It should be safe to turn on video_capture_tests on linux
after this lands.
Bug: webrtc:9292, webrtc:9636
Change-Id: I6e86716b4c7ca43244596f806ff904b7fdf9201a
Reviewed-on: https://webrtc-review.googlesource.com/94769
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24334}
This CL lowers the margins in the AEC3 conservative mode to increase
the transparency when there are audio buffer issues, and during call
startup.
In particular, this CL adjusts the parameters and thresholds to
-Make the requirements for filter divergence more strict, to minimize
the transparency loss during minor filter divergence.
-Decrease the echo power uncertainty used during initial filter
convergence, to increase transparency after audio buffer issues.
-Deactivate the enforcement of conservative suppressor gain after
audio buffer.
Bug: webrtc:9641,chromium:875611
Change-Id: Ie171bb411f17a1e8661c291118debd334f65c74f
Reviewed-on: https://webrtc-review.googlesource.com/94776
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24333}
