This is a reland of commit 97ba853295578975a04fc504315cccd465f9f0bd
This cl did not cause the regression in Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks. Real culprit reverted in https://webrtc-review.googlesource.com/c/src/+/290502.
Original change's description:
> Remove use of ReceiveStreamRtpConfig:transport_cc
>
> With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
> I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
>
>
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
>
> Bug: webrtc:14802
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38980}
Bug: webrtc:14802
Change-Id: Ib98e61413161d462da60144942cdb0140e12bc42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38997}
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Bug: webrtc:14802
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38980}
This reverts commit 315b95ca11161bdea715d5316f92828edd41f0d5.
Reason for revert: Breaks internal bots.
Original change's description:
> Enforce stream id uniqueness in RtpSender::set_stream_ids
>
> https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
> has a step saying
> For each stream in streams, add stream.id to
> [[AssociatedMediaStreamIds]] if it's not already there
>
> This applies to addTrack and setStreams and the set of streams in
> addTransceiver.
>
> BUG=webrtc:14769
>
> Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38937}
Bug: webrtc:14769
Change-Id: I6fd22ff0550c0894057fb1dc15f1b95819fa6df2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38940}
https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
has a step saying
For each stream in streams, add stream.id to
[[AssociatedMediaStreamIds]] if it's not already there
This applies to addTrack and setStreams and the set of streams in
addTransceiver.
BUG=webrtc:14769
Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38937}
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.
Test failures seem unrelated, so using No-Try.
No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
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}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
as it is defined in RFC 3550. This avoids implicit casts
between signed and unsigned definitions.
BUG=webrtc:8626
Change-Id: I919b7c38ede1aa8d32f8e31b55660f540e5f5a6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279240
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38522}
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}
RTPHeader is not exported, so the TransformableIncomingAudioFrame can't be mocked in chrome tests, using a getter instead.
Bug: chromium:1247260
Change-Id: I2af4e6a88b3f4772b3bb50ee0ae9d5c80fed3ae4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278785
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38352}
This spills to a few more clasess....
Change-Id: Iea79e3b4ac86b30db6f13da89a47ab7000c5440a
Bug: webrtc:14502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277803
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38334}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
- Propagating `RtpPacketInfo::local_capture_clock_offset`, an
existing field that is related to the abs-capture-timestamp
header extension field `estimated_capture_clock_offset`
- Propagated through `SourceTracker::SourceEntry`
Bug: webrtc:10739, b/246753278
Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38129}
This change increases the number of scenarios where the RTT would be
available to `ChannelReceive`. That's the case since
`ModuleRtpRtcpImpl2::RTT()` falls back on the DLRR-based method when
the report blocks based method is unavailable - i.e., when there is
no audio sender.
Bug: webrtc:10739
Change-Id: Ie2451c739ab5bcfbe7844ee852bb12a97dab2ca4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274581
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38048}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
SetFeedbackParameters no longer recreates the embedded streams for:
- transport cc flag
- rtcp status
Bug: none
Change-Id: If6117a1ae760ca9a02f06bbfa2b46c6e0f448cfc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268281
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37526}
Setting the transport cc flag was only possible post creation for
audio receive streams, while video receive streams need to be recreated.
This CL moves the setter for transport_cc() to where the getter is and
adds boiler plate implementations for the video streams. For audio
streams this splits "SetUseTransportCcAndNackHistory" into two methods,
SetTransportCc and SetNackHistory.
Bug: none
Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37038}
