This is a reland of commit a22c2a0c58
after systems depending on this have been fixed.
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: Iad8b503b3101d1e684a4da2d1547b879e77b85dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293861
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40716}
Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
Use more strict types for the calculation to make it clearer.
Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.
Reland of https://webrtc-review.googlesource.com/c/src/+/315143
Bug: None
Change-Id: I41ce383a2f19d489e4cae0b1bf1f720e0ffdd17a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315460
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40538}
This reverts commit 762f193ca4.
Reason for revert: breaks downstream test
Original change's description:
> Cleanup calculating time between RTCP reports
>
> Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
> Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
> Use more strict types for the calculation to make it clearer.
> Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.
>
> Bug: None
> Change-Id: Ie8c6b9720095cd1cc3f9814b9df16700119337c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315143
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40529}
Bug: None
Change-Id: I8c83013523120a84f236e8efa0d122363e7a228b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315381
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40535}
Move that calculation into dedicated function, move comment why it is calculated the way it is into the same function.
Cleanup that comment - remove parts unused by current code, in particular remove description of code that was deleted a while ago
Use more strict types for the calculation to make it clearer.
Replace DCHECK result can't be zero with a clamp to ensure it can't be zero, because with large bitrates it may.
Bug: None
Change-Id: Ie8c6b9720095cd1cc3f9814b9df16700119337c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315143
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40529}
Instead of using most recent, or most "valuable" packets for padding, use most recent large packet.
The large packet for padding is not culled when acked by the receiver.
The most recent large packet is kept where payload size + 100bytes > currently stored.
Bug: webrtc:15201
Change-Id: I510735b757f99460c477b575061963d2b69016e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306521
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40146}
RtpRtcpInterface::RTT follows discouraged style of using return values,
uses raw integers to represent time delta,
and returns values that no code uses (min, max, average RTT)
added LastRtt function addresses all these stylistic issues.
Bug: webrtc:13757
Change-Id: Iaf947dd1b7139026f2beb991e69634c606c6b608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304520
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40028}
Remove deprecated accessors returning time as raw int
Add setters for all fields to simplify usage of this class in tests
Remove unused min/max RTT fields
Bug: webrtc:13757
Change-Id: Ia8966975c15b9a930f54b4db0fc75f7002dcffe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304461
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40013}
This CL prepares for send packet batching support in later CLs.
Bug: chromium:1439830
Change-Id: I0bbecfa895aa6d4317ef8049b3789272a440d032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304282
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40009}
FieldTrialBasedConfig reads config from the global field trial string
ScopedKeyValueConfig adjust the global field trial string
test::ExplicitKeyValueConfig doesnt touch the global field trial string
Bug: webrtc:11926
Change-Id: I8883634fdc7e1bdb63eec9bf38114a3031103839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299062
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@{#39683}
Replace it with GetSenderReportStats that returns result instead of
filling lots of output parameters
On the way replace pair of uint32_t with dedicated NtpTime type
Bug: None
Change-Id: I5b821b8d000d63ebd8cdc1b9897a86429d97b19b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295560
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39431}
The CSRC concept is really a frame level concept.
Setting it per sender is a quick hack, and should be minimized.
This function doesn't seem to be used anywhere, so removing it
lessens the chance of confusion.
Bug: webrtc:7135
Change-Id: Ia3c27b5984b153e68bc51d93b03f08f7f867adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286426
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#38822}
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}
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}
This reverts commit 2c41cbae37.
Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2.
Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c05.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34897}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34930}
This reverts commit fb0dca6c05.
Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
Original change's description:
> Wire up non-sender RTT for audio, and implement related standardized stats.
>
> The implemented stats are:
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34861}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=hta,hbos,minyue
Bug: webrtc:12951, webrtc:12714
Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34897}
The config flag will be removed once downstream usage is gone.
Bug: webrtc:11340
Change-Id: Iee8816660009211540d9b09bb3cba514455d709b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228431
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34757}
Since https://webrtc-review.googlesource.com/c/src/+/228433 both audio
and video now only call Get/SetRtpState while not registered to the
packet router.
We can thus remove the lock around packet sequencer and just use a
thread checker.
Bug: webrtc:11340
Change-Id: Ie6865cc96c36208700c31a75747ff4dd992ce68d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228435
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34755}
The race can happen when an encoder thread is packetizing a video frame
and is calling RTPSender::AssignSequenceNumber() while the RtpRtcp
module is calling GeneratePadding() and querying
PacketSequencer::CanSendPaddingOnMediaSsrc().
The solution for now is to simply not call
PacketSequencer::CanSendPaddingOnMediaSsrc() from the RtpRtcp module,
as that parameter will be ignored anyway - RTPSender will query that
method internally while holding the send lock.
Once deferred sequencing is implemented, the
can_send_padding_on_media_ssrc parameter can be populated safely since
it is then always called on the pacer thread.
Bug: webrtc:11340, webrtc:12470
Change-Id: I9e90808166453d0e29746df89044e1d3bdffa286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227767
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34655}
This prepares for deferred sequence numbering, and is (sort of)
extracted from
https://webrtc-review.googlesource.com/c/src/+/208584
Bug: webrtc:11340, webrtc:12470
Change-Id: I2f3695309e1591b9f7a1ee98556f4f0758de7f69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227352
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34643}
This change migrates RTCPSender to use webrtc::Timestamp, preparing
for later improvements regarding bugs.webrtc.org/11581.
Fixed: webrtc:12873
Change-Id: I1159701dc373883367d9b2c86823f8fb59904d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222324
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34346}
The class depends on RtcRtcpInterface::Configuration which adds an
unneeded dependency, and inhibits well-manored changes to the
constructor interface.
Fix this so that RTCPSender uses it's own configuration struct which
can be extended in future CLs.
Also add a legacy constructor while downstream dependencies are
updated.
Bug: webrtc:11581
Change-Id: I8d166ab8253b27c08fcbe6aa7c7adde92688b7dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222322
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34343}
The eventual implementation of changing the status will be async so the
return value isn't that useful and was in fact only being used to log
a warning if an error occured.
This change is to facilitate upcoming changes related to media engine.
Bug: webrtc:11993
Change-Id: Ia7f85a9ea18b2648b511fa356918cf32a201461f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215975
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33825}
Add missing members needed to surface `RTCRemoteOutboundRtpStreamStats`
via `ChannelReceive::GetRTCPStatistics()` - i.e., audio streams.
`GetSenderReportStats()` is added to both `ModuleRtpRtcpImpl` and
`ModuleRtpRtcpImpl2` and used by `ChannelReceive::GetRTCPStatistics()`.
Bug: webrtc:12529
Change-Id: Ia8f5dfe2e4cfc43e3ddd28f2f1149f5c00f9269d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211041
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33452}
`LastReceivedNTP()` does not need to be part of the public members of
`ModuleRtpRtcpImpl` and `ModuleRtpRtcpImpl2` since it is used only
once in the same class.
This change is requried by the child CL [1] which adds a public getter
needed to add remote-outbound stats.
[1] https://webrtc-review.googlesource.com/c/src/+/211041
Bug: webrtc:12529
Change-Id: I82cfea5ee795de37fffa3d759ce9f581ca775d55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211043
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33420}
