The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
Bug: webrtc:9878
Change-Id: Ic912d67455fcef4895566edb8fef62baf62d7cfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156440
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29454}
When one of the sources is a FrameGeneratorCapturer, this implies that
its TaskQueue is stopped. Before this change, the FrameGeneratorCapturer
was destroyed later, by the CallTest destructor, which led to a
use-after-free race on the Clock object passed to the capturer.
Bug: webrtc:11018
Change-Id: I3e53f95a725b6fb53b13e182ecd2caf03ea15bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156170
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29443}
This speed up fuzzing because no more SleepMs in real time.
Bug: chromium:959836, chromium:1009073
Change-Id: Ib00a2ff8d6ca2e0bfc706ee7469e0a9c7fb10758
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156362
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29439}
This can be better used to determine the length of test calls,
rather than using the interval metric.
Bug: webrtc:11017
Change-Id: I69f66fa750b061a7d010d591a718555e2b5b34b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156087
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29413}
Without this, we can end up with negative capture-to-render delays
if the jitter buffer sets the render time to an earlier time.
Bug: webrtc:11017
Change-Id: I590509136f630d025cde6e5e13d4a3ee620267ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156081
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29409}
The new configuration path is via AudioProcessing::ApplyConfig and
AudioProcessing::GetStatistics.
ApmTest.Process passes with unchanged reference files if
audio_processing_impl would initialize the VAD with
VoiceDetection::kLowLikelihood instead of kVeryLowLikelihood.
This was verified by testing this CL with that modification.
Bug: webrtc:9878
Change-Id: I4d08df37a07e5c72feeec02a07d6b9435f917d72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155445
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29395}
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}
Based on similar code in the call level scenario test framework.
Bug: webrtc:10839
Change-Id: I262a890aa2cf905bb81b0f07957c08d0df5f7651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29361}
This started to be detected by a new version of clang and it is blocking
the roll:
../../third_party/webrtc/test/fuzzers/agc_fuzzer.cc:85:29: error: converting the result of '?:' with integer constants to a boolean always evaluates to 'true' [-Werror,-Wtautological-constant-compare]
const bool num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
Bug: chromium:1007367
Change-Id: Ib9a6e4e3c8f109d10845a315dd0782b1498cb54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155166
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29348}
actual task queue implementation for these tests is intentionally unchanged for now.
while at it, change return type of created transports to unique_ptr to note passing ownership.
Bug: webrtc:10933
Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29335}
that allows to use SingleThreadedTaskQueueForTesting as regular TaskQueue.
which allows components that currently depend on SingleThreadedTaskQueueForTesting
to depend on TaskQueueBase interface instead.
Those updates can be done one-by-one and in the end would allow to stop
using SingleThreadedTaskQueueForTesting in favor of other TaskQueue implementations.
Bug: webrtc:10933
Change-Id: I3e642c88c968012588b9d9c09918340f37bbedbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154352
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29307}
This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.
Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
Only the ISAC codec had an non-trivial implementation, for its unused
adaptive mode. This cl deletes that implementation, and the call
from NetEq, and the interface method.
Bug: webrtc:10098
Change-Id: Iaf7667e0ae867fc9d64286dff4c01a8ce0b6e2a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153882
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29279}
The high bitrate smoketest is flaky on some platforms,
this CL reduces the resolution and bitrates to make it less
flaky.
Bug: webrtc:10975
Change-Id: Id271b3c68abfa2011c207e7883cfcb230b1d3e36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153845
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29268}
Verified it no longer times out on the input that spawned the bug.
Bug: chromium:1005853
Change-Id: I5b0ab25aaefdc8b451b4d976b1c3b8f8d38f13e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153840
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29260}
The transport time metric was accidentially changed by the CL
https://webrtc-review.googlesource.com/c/src/+/153660
This CL restore the transport time metric to how it has been
measured before, that is, time from encoder output to decoder input.
Bug: webrtc:10975
Change-Id: I66f022f26976451d28c0374b22849f14f9c02378
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153886
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29259}
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.
The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.
Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
This flag became unused in https://codereview.webrtc.org/2789843002;
it was set, but the setting had no effect.
Bug: webrtc:7135
Change-Id: I012a7c3600bc7a371c7a589695823b30ed5647a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29192}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
Add ability to export internal state of SamplesStatsCounter to be able
then to plot that data.
Bug: webrtc:10138
Change-Id: I5aae5b7dea2989e9f82820933a9ab6f21db17556
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152542
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29173}
