This reverts commit a22c2a0c58.
Reason for revert: breaks upstream project
Original change's description:
> rtp sender: don't send BYE on deactivating streams
>
> as this breaks RTCP assumptions about SSRCs being no longer
> active as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.6
>
> This should not be sent in reaction to temporarily disabling
> a stream via RTCRtpParameters.active as this does not mean that
> the participant is leaving the session as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
> and does not indicate end of participation as defined in
> https://www.rfc-editor.org/rfc/rfc3550#section-6.1
> which stipulates BYE should be the last packet sent from this SSRC.
>
> BUG=webrtc:11082
>
> Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#38059}
Bug: webrtc:11082
Change-Id: Iaaff0c0d7bb857fe9ce78ebcc716f3c6f1bc5c4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275640
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38097}
Most of the changes are trivial.
Bug: webrtc:14432
Change-Id: I0444527bf57c72c8d65f69754b4a4a1c1d7b2e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38074}
as this breaks RTCP assumptions about SSRCs being no longer
active as defined in
https://www.rfc-editor.org/rfc/rfc3550#section-6.6
This should not be sent in reaction to temporarily disabling
a stream via RTCRtpParameters.active as this does not mean that
the participant is leaving the session as defined in
https://www.rfc-editor.org/rfc/rfc3550#section-6.3.7
and does not indicate end of participation as defined in
https://www.rfc-editor.org/rfc/rfc3550#section-6.1
which stipulates BYE should be the last packet sent from this SSRC.
BUG=webrtc:11082
Change-Id: Ia5144857f85303643146b0759184f0f3f50b66e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273348
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38059}
In some upcoming use cases we might wish to flush pending
retransmissions from the pacer queue. In order to not make those packets
forever inaccessible this CL adds a way to clear their "pending" status
from the packet history.
Bug: webrtc:11340
Change-Id: I9aac44125899a7f1e5a5e5be3390ac07b1e661ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274600
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38037}
Having the minimum value as the default makes more sense than maximum.
Bug: b/232103634
Change-Id: Ia6a97f7a2a47bb74ed3b3316d95a1c6d00e2c16b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274260
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38021}
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}
The data that's used to report the histograms is owned by UlpfecReceiver
and moving the reporting there, simplifies things as configuration
changes happen in RtpVideoStreamReceiver2 (which currently require all
receive streams to be deleted+reconstructed).
Additional updates:
* Consistently using `Clock` for timestamps. Before there was
a mix of Clock and rtc::TimeMillis.
* Update code to use Timestamp and TimeDelta.
Bug: none
Change-Id: I89ca28ec7067a49d6b357315ae733b04e7c5a2e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271027
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37729}
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810
* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats
BUG=webrtc:13756
Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
WebRTC doesn't produces such packet and ignores it when receive.
Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
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}
This reverts commit 791294a647.
Reason for revert: downstream test adjusted
Original change's description:
> Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
>
> This reverts commit a17651f7d8.
>
> Reason for revert: triggers failure in downstream test
>
> Original change's description:
> > Fix overflow due to rounding in AbsoluteSendTime::To24Bits
> >
> > Actual rounding is not important for this time as long it is consistent
> > during the call: only difference between two absolute send time matter
> > Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
> >
> > Bug: None
> > Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37468}
>
> Bug: None
> Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37473}
Bug: None
Change-Id: I99bcc6c6b7c08cd9621bdce336cd5793f78ee657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268190
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37498}
This reverts commit a17651f7d8.
Reason for revert: triggers failure in downstream test
Original change's description:
> Fix overflow due to rounding in AbsoluteSendTime::To24Bits
>
> Actual rounding is not important for this time as long it is consistent
> during the call: only difference between two absolute send time matter
> Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
>
> Bug: None
> Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37468}
Bug: None
Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37473}
Actual rounding is not important for this time as long it is consistent
during the call: only difference between two absolute send time matter
Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
Bug: None
Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37468}
since the fec packets are initialized to 0 there is no need
to special-case the first packet since
A XOR 0
is the identity operator.
BUG=None
Change-Id: I0cb55283ecdca06f8e3a7b5856ec1f9fbbad1ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251522
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37378}
Move warning about missing receive_statistics to AddReceiver to avoid
producing it for rtp send only endpoints.
Remove warning about missing cname as unimportant.
Bug: webrtc:8239
Change-Id: I8a90aa4b378284b9c68f67678b2392b9461c95b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264825
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37093}
BaseTime represents fixed point in time with unknown epoch and thus
make sense to convert to Timestamp type, however Timestamp should always
be positive. however legacy tests expect GetBaseTimeUs to return negative time sometimes.
Bug: webrtc:13757
Change-Id: I3f780a7775fdd1e271402c59384c1298db76f75a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264549
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37076}
to make tests faster and more determenistic.
Bug: webrtc:8239
Change-Id: I18067251a1f1a349fda28bbfbb59bce333bfddca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201737
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36938}
RTCPReceiver::main_ssrc_ and local_media_ssrc() represent the same
value but could get out of sync when `set_media_ssrc()` was called.
Instead of using main_ssrc_, just use the local_media_ssrc() accessor.
Bug: webrtc:11993
Change-Id: I2b034287e6b6025d9b0d2affa391a168896a614b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262663
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36905}
