It is a bit more flexible than the current implementation.
Also cleanup ChannelSend::GetRTT since it is not called from the receive stream anymore.
Bug: none
Change-Id: I4403c8b1840012f2287d189be934fd1069de85fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374160
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43728}
When Reconfiguring there's a call to ResetSenderCongestionControlObjects followed by a later call to
RegisterSenderCongestionControlObjects which happen on the worker thread, while enqueuing packets is
happening on a different thread.
If packets are enqueued in between these calls we get a null dereference of the `rtp_packet_pacer_`
This change fixes it by pausing processing of incoming audio in the interim state
Bug: webrtc:358290775
Change-Id: I77cebfb131545ce2a6fdb26105dd999da3f7c443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42815}
Part of a set of CL to allow video to borrow underused audio bitrate.
Bug: webrtc:35055527
Change-Id: Idb504cbbc5794c06b28bdc21b3d860c9da9df175
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358202
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#42733}
To make it available for creating AudioEncoders in follow ups
Bug: webrtc:343086059
Change-Id: I24bb8f7e0494e392210cb1001ea0421030d2766b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352601
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42396}
This pass the absolute capture timestamp at the beginning of the frame
to the rtcp module.
Bug: webrtc:42226041
Change-Id: Iae85a56bfd9d33f7eb9eac3c68961235fe16dc6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350202
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42296}
Calculate the RMS audio level of audio packets being sent before
invoking an encoded frame transform, and pass them with the encode frame
object.
Before this, the audio level was calculated at send time by having rms_levels_ look at all audio samples encoded since the last send. This
is fine without a transform, as this is done synchronously after
encoding, but with an async transform which might take arbitrarily long,
we could end up marking older audio packets with newer audio levels, or
not at all.
This also makes things work correctly if external encoded frames are
injected from elsewhere to be sent, and exposes the AudioLevel on the
TransformableFrame interface.
Bug: chromium:337193823, webrtc:42226202
Change-Id: If55d2c1d30dc03408ca9fb0193d791db44428316
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349263
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#42193}
With this cl
RtpTransportControllerSend::OnAddPacket is instead directly invoked from PacketRouter::SendPacket instead of going via RTP module.
Transport sequence numbers are instead of directly written to header
extension, added to RtpPacketToSendMetaData and written to the extenion
by RTP module.
This is to allow transport sequence numbers without actually sending
them in an extension.
Bug: webrtc:15368
Change-Id: Idd03e02a4257dfc4d0f1898b2803345975d7dad2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344720
Reviewed-by: Erik Språng <sprang@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41974}
To prepare for a new AudioLevel struct to be added to the public WebRTC API, rename the internal RTP extension reader/writer class to AudioLevelExtension. A temporary alias is provided to avoid breaking downstream projects.
Bug: webrtc:15788
Change-Id: Ie231668f25932fd9b539229114128b1d0b949a6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339887
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41787}
Split from
https://webrtc-review.googlesource.com/c/src/+/318283
to reduce CL size. Takes a different and (hopefully) simpler
approach.
BUG=webrtc:15579
Change-Id: I8517ffbeb0f0a76db80e3e367de727fb6976211d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325023
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41073}
Checking in sending classes avoids using global field trial string in favor of the injected one.
In addition to that RateLimiter looks wrong layer for check that field trial:
checking inside RateLimiter class might be surprising if it is used for limiting something else than RTX bitrate.
evaluating field trial for each retransmitting packet might be expensive
Bug: webrtc:15184, webrtc:10335
Change-Id: I87bae3522bbd9692629d4f9b6caa119be03f2bd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322720
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40908}
Combine all parameters into single struct so that it is easier to add and remove optional parameters
Use Timestamp type instad of plain int to represent capture time
Use rtc::ArrayView instead of pointer+size to represent payload
Merge passing audio level into send function.
Bug: webrtc:13757, webrtc:14870
Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40688}
Keep the logic managing whether audio RTP timestamps have the random
start offset added or not inside ChannelSend, so that the
ChannelSendFrameTransformerDelegate doesn't need to worry about it.
Crucially, this means that frames moved between senders by encoded
transforms clients will always use the correct offset for the channel
where we actually get sent.
Also rename TS variables throughout both classes to be explicit over
whether the offset has been added or not.
Bug: chromium:1464847
Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40655}
NetworkLinkRtcpObserver is similar to RtcpBandwidthObserver but pass
time variables using unit types instead of raw integers.
Bug: webrtc:13757
Change-Id: Iaa0bbe0b108620b3a24013c40e7d9004032e904d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305022
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40087}
Delete VoERtcpObserver proxy:
pass BWE related message directly to transport controller
pass ReportBlock directly to ChannelSend, assuming there will be single report block per source ssrc
Bug: None
Change-Id: I8378326bff1dc3c2736960166fc782ee822a9c12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305224
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40081}
ReportBlockData class is better documented and has wider usage.
Bug: webrtc:13757
Change-Id: Ie5f2275f2f0236267172e6dd1ce5c2dfb2193ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304101
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39980}
This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.
The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.
Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}
The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.
Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
This removes the previous approach where we continued to update the timestamp when the capturer is running but the send stream is stopped in favor of a more general approach that also works when the capturer is paused.
Some assumptions for this change to be correct: input sample rate and frame size will be the same before/after the stream is paused.
Bug: webrtc:12397
Change-Id: I3b03741cd6d3285cbc9aee3893800729852e6cfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291526
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39213}
This is to clear any remaining buffers and other state such as the next packet timestamp.
Bug: webrtc:12397
Change-Id: I2ef9a6f7254d82a69a2896ec8aa619ced2694fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291327
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39206}
Remove the default enabled "WebRTC-Audio-FixTimestampStall" field trial which was rolled out 2 years ago without any issues.
Also change the include audio level indication member to be atomic since it is accessed on multiple threads.
Bug: webrtc:14804
Change-Id: I4c5145e1fb03351154162b4293a3bd870e4793cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290886
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39125}
This CL adds functionality to remove packets matching a given SSRC from
the pacer queue, and calls that with any SSRCs used by an RTP module
when that module is removed.
Bug: chromium:1395081
Change-Id: I13c0285ddca600e784ad04a806727a508ede6dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287124
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38880}
This information is now readily available. Let's expose it.
In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.
Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
To mitigate race between ~ChannelSend and task created in ProcessAndEncodeAudio.
as described in the comment next to the task queue member.
Bug: b/228933184
Change-Id: Ia0efd050c76a4539dc2525ef8efc065fab96861c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258983
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36553}
Currently, the RtpRtcp module of AudioSendStream is (de)registered in
the packet router on calls to
(Register|Reset)SenderCongestionControlObjects.
This CL changes that to happen on Start/Stop instead, which allows us
to safely call (Get|Set)RtpState on suspend/resume without the need
for extra locking in the rtp module.
See also https://webrtc-review.googlesource.com/c/src/+/228430
Bug: webrtc:11340
Change-Id: I54243a9ace8a7659924269418468b49b967b9465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228433
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34738}
This change achieves an Idle Wakeup savings of 200 Hz.
ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.
Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
