With this CL, the only usage of RTCStatsMember within WebRTC is the
actual type alias declaration. It's not referenced anywhere anymore.
This allows us to deleting the type alias, but let's do that in a
standalone CL in case it gets reverted.
Bug: webrtc:15164
Change-Id: I766d07abb62b5ddd524859b8ed749394fc439e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41612}
Same method, different name. Unblocks replacing RTCStatsMember<T> with
absl::optional<T>.
Bug: webrtc:15164
Change-Id: I251dd44d3b0f9576b3b68915fe0406d1b3381e5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41573}
This is a reland of commit 84c48ae751
Original change's description:
> [Stats] Move metric names to Attribute, constructed via AttributeInit.
>
> As of this CL, Attribute no longer implements RTCStatsMemberInterface
> and a member no longer owns knowing its own name. The attribute knows
> the name because we pass it down at construction time.
>
> To achieve this, the WEBRTC_RTCSTATS_IMPL() macro is updated to take
> AttributeInits instead of raw member pointers, i.e. (name, ptr) pairs.
>
> By constructing RTCStatsMember<T> without a name parameter, it does the
> same thing as the absl::optional<T> constructor. So RTCStatsMember<T>'s
> days are numbered!
>
> Bug: webrtc:15164
> Change-Id: I560c0134bae1c2d7218426a1576425ecc1b677a7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334203
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41540}
Bug: webrtc:15164
Change-Id: I28f3d588004ff185e5820347ad9513f2f7a6cc66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335020
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41556}
This reverts commit 84c48ae751.
Reason for revert: Breaks downstream project
Original change's description:
> [Stats] Move metric names to Attribute, constructed via AttributeInit.
>
> As of this CL, Attribute no longer implements RTCStatsMemberInterface
> and a member no longer owns knowing its own name. The attribute knows
> the name because we pass it down at construction time.
>
> To achieve this, the WEBRTC_RTCSTATS_IMPL() macro is updated to take
> AttributeInits instead of raw member pointers, i.e. (name, ptr) pairs.
>
> By constructing RTCStatsMember<T> without a name parameter, it does the
> same thing as the absl::optional<T> constructor. So RTCStatsMember<T>'s
> days are numbered!
>
> Bug: webrtc:15164
> Change-Id: I560c0134bae1c2d7218426a1576425ecc1b677a7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334203
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41540}
Bug: webrtc:15164
Change-Id: I9f416838153b26d4560ae98c37fb18a803d0295d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334901
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41550}
As of this CL, Attribute no longer implements RTCStatsMemberInterface
and a member no longer owns knowing its own name. The attribute knows
the name because we pass it down at construction time.
To achieve this, the WEBRTC_RTCSTATS_IMPL() macro is updated to take
AttributeInits instead of raw member pointers, i.e. (name, ptr) pairs.
By constructing RTCStatsMember<T> without a name parameter, it does the
same thing as the absl::optional<T> constructor. So RTCStatsMember<T>'s
days are numbered!
Bug: webrtc:15164
Change-Id: I560c0134bae1c2d7218426a1576425ecc1b677a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334203
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41540}
The plan is to replace Members() with Attributes() instead.
For backwards-compatability during the transition, Attribute implements
RTCStatsMemberInterface but the two classes serve the same purpose
which is to allow iterating all metrics of a stats object.
The reason for moving away from "members" is that we already have a way
to express a variable that maybe has a value: absl::optional<T>. The
only information the member adds is the const char* name(), which we'll
move to Attribute in a future CL.
We don't need to maintain an RTCStatsMemberInterface::Type enum in the
future because absl::variant<T> has absl::holds_alternative<T>.
Step 1: Add Attributes().
Step 2: Migrate to Attributes() and delete Members().
Step 3: Replaces all uses of RTCStatsMember<T> with absl::optional<T>
and delete RTCStatsMember + RTCStatsMemberInterface.
Bug: webrtc:15164
Change-Id: I3fdd5b24214bb5cc340a54a0171df73b516e1803
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41507}
which avoids those stats on datachannel-only or video-only connections.
Note that a receiver always exists, regardless of the transceiver direction.
BUG=None
Change-Id: I1ef33a8446fafe2978ac603e658e67d51d7af904
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330441
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Fredrik Hernqvist <fhernqvist@google.com>
Cr-Commit-Position: refs/heads/main@{#41423}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
which is present if a fec mechanism like FlexFEC is negotiated
spec change:
https://github.com/w3c/webrtc-stats/pull/765
BUG=webrtc:15250
Change-Id: I7d71d49fab0153d734f22831e6684d2acfc647fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314981
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40514}
This extension is documented to carry one bit: Screenshare.
It's been used for carrying simulcast layers and experiment IDs.
This CL removes that usage.
Bug: webrtc:15383
Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40457}
The existing equality check did not always work since content_type
is sometimes overloaded with extra internal information such as simulcast layer index. Fix by using the videocontenttypehelpers::IsScreenshare helper method.
Bug: webrtc:15381
Change-Id: I2fe84e7f036ea2c223e4fa6dd58af1c4c0bcfbdb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312261
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40448}
which means this will not show up in getStats inbound-rtp/outbound-rtp
until the encoder/decoder is known. This has implications in particular
for inbound-rtp where the value is currently "unknown" until video
frames have been received.
This is safe to change as the previous change to gate
decoderImplementation behind getUserMedia access already broke
the assumption that the field is always string.
BUG=webrtc:14906
Change-Id: Ie6040ada3656e80f792c0c32c1b86ad1d6609d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40334}
filtering out the -1 value as it is done for "legacy" stats.
Also change the protocol and don't use "udp" and "tcp" which are misleading since the datachannel protocol is user-supplied.
BUG=webrtc:15071
Change-Id: I45d735fcf30144969630f5b8a91b40f12585bbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300483
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40333}
specified in https://github.com/w3c/webrtc-stats/pull/762
and take FlexFEC into account for receive statistics.
BUG=webrtc:15250
Change-Id: Id85775ab1f29487d5b8bf478da6e22071005901a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40325}
This is a reland of commit ccc87ea3c6
Downstream project has been updated.
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: Ibc7aeb518ed0bd7f1d725f140132c99e5a89bcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308880
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40305}
This reverts commit ccc87ea3c6.
Reason for revert: Breaks downstream project
Original change's description:
> [Stats] Remove enum-like structs in favor of strings.
>
> Due to a limitation of RTCStatsMember<T> not supporting enums, as well
> as the fact that in JavaScript enums are represented as basic strings,
> the stats enums have always been represented by T=std::string.
>
> Now that we have WebIDL-ified[1] all RTCStats dictionaries and enum
> values are simply string-copied (example: [2]) it seems safe to assume
> that "stats enums are just strings" is here to stay.
>
> Therefore there is little value in having C++ structs that look like
> enums so I'm deleting those in favor of std::string operator==()
> comparisons, e.g. `if (rtp_stream.kind == "audio")`. This removes some
> lines of code from our code base.
>
> I mostly want to get rid of these because they were taking up about 20%
> of the rtcstats_objects.h real estate...
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.idl
> [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/renderer/modules/peerconnection/rtc_stats_report.cc;l=667;drc=cf34e84c9df94256abfb1716ba075ed203975755
>
> Bug: webrtc:15245
> Change-Id: Iaf0827d7aecebc1cc02976a61663d5298d684f07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40295}
Bug: webrtc:15245
Change-Id: I05db80ba9f29460239de82cea9d95136e4c708e4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308860
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Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40298}
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530
# Relevant bots already passed
NOTRY=True
Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
The web compat requirement that was the reason for keeping
is now solved in Chromium and its stats bindings.
BUG=webrtc:9674
Change-Id: Ifb722769414b2bcc5f4d36d7dff87a875336e039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303860
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40024}
This reduces dependency on the struct RTCPReportBlock and would allow to
delete it in favor of class ReportBlockData
Bug: None
Change-Id: I93874c4f54cf62af0c16ae26e2231b8fb49f195d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304161
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39985}
* retransmittedBytesReceived
* retransmittedPacketsReceived
added to the specification in
https://github.com/w3c/webrtc-stats/pull/735
BUG=webrtc:15096
Change-Id: I6770e5d8d09ac1c2693c918fd943b0ab257ec7ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295260
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39959}
* The pointer isn't needed for this notification. Arguably using
the internal id is more consistent with the stats code.
* Using the int makes it safer down the line to post the operation
from the network thread to the signaling thread rather than post
an object reference.
Bug: none
Change-Id: I1e9eb31d8386dca3feaa90ee3267ea98eb3e81ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299144
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39696}
Make DataChannelController's AddSctpDataStream and
RemoveSctpDataStream be required to be called on the network thread.
This moves blocking calls within those methods over to the
SctpDataChannel class instead.
For production code there's no functional change in this CL. However, this CL:
1) Introduces an actual dedicated network thread to
DataChannelController and SctpDataChannel tests.
2) Removes two data_channel_transport() checks inside DCC that
were being done on the wrong thread (signaling) and
3) introduces a network calling block to SctpDataChannel, where more
network thread related work needs to be done and can be bundled.
(to be done in follow-up CLs).
Bug: webrtc:11547
Change-Id: I6787ac395e61d4a25ae3a74a123e3357cbb46b54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298052
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39688}
All tests do this already except for RTCStatsCollectorTest.
Bug: none
Change-Id: I318f45a2c79b3d07ca6c92902ebb4f0622ec3200
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297862
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39576}
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.
The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.
BUG=webrtc:14973
Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39497}
This reverts commit 18c869bc36.
Reason for revert: Added a field trial that allows landing the code without affecting performance in prod.
This CL also incorporates subsequent CLs that also had to be reverted.
Original change's description:
> Revert "Use two MediaChannels for 2 directions."
>
> This reverts commit 8981a6fac3.
>
> Reason for revert: Quality regression detected.
>
> Original change's description:
> > Use two MediaChannels for 2 directions.
> >
> > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
> >
> > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
> >
> > Bug: webrtc:13931
> > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39340}
>
> No-Try: true
> Bug: webrtc:13931
> Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39445}
Bug: webrtc:13931
Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39494}
DataChannelController used WeakPtr to clear outstanding references
upon destruction - except for the case of SctpDataChannel where we
had a pointer+flag for the same purpose. This change updates
SctpDataChannel and FakeDataChannelController to use a consistent
approach.
Bug: none
Change-Id: I0248471c241365a2c0de76afbb37302115650194
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39464}
It turns out that there were several sigslot instances across data
channel, pc and stats classes that in practice only served as means
to update two counters in RTCStatsCollector. There's already a
notification path that's suitable.
This also fixes a case where the PC instance sat in the middle
of notifications from datachannels to the datachannel controller.
Bug: webrtc:11943
Change-Id: Ic60b76021584019f82085f6651230fe2fe82d465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39456}
This reverts commit 8981a6fac3.
Reason for revert: Quality regression detected.
Original change's description:
> Use two MediaChannels for 2 directions.
>
> This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
>
> The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
>
> Bug: webrtc:13931
> Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39340}
No-Try: true
Bug: webrtc:13931
Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39445}
This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
Bug: webrtc:13931
Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39340}
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
This is a reland of commit 81aab48878
See diff between Patch Set 1 and latest Patch Set.
The original CL broke this WPT[1] because getStats() with the receiver
as the selector stopped working in the event of unsignalled SSRCs due
to the receiver not knowing what the SSRC was.
This fix is to query media_channel_ for the unsignalled SSRC in the
event that the receiver does not know the SSRC.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html
Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}
Bug: webrtc:14175, webrtc:14811
Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39010}
making it clear what unit is being used.
BUG=webrtc:13756
Change-Id: I6354d35a8e02bb93a905ccf32cb0b294b4813e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289460
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39008}
This reverts commit 81aab48878.
Reason for revert: external/wpt/webrtc/simulcast/setParameters-active.https.html is failing with this change
Original change's description:
> Remove 'trackId' dependency in stats selector algorithm.
>
> In preparation for the deletion of deprecated 'track' stats, the
> stats selector algorithm needs to be rewritten not to use 'trackId'.
>
> This is achieved by finding RTP stats by their SSRC, as obtained via
> getParameters(). This unfortunately adds a block-invoke (in the sender
> case the block-invoke happens inside GetParametersInternal and in the
> receiver case the block-invoke is explicit at the calling place), but
> it can't be helped and it's just once per getStats() call and only if
> the selector argument is used.
>
> Bug: webrtc:14175
> Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38981}
Bug: webrtc:14175
Change-Id: Id1cbe892250fe88bd6db0b47269bcefa346709b4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290502
Commit-Queue: Christoffer Jansson <jansson@google.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#38993}
The tests should say something about RTPs too, because what they say
about track will soon be irrelevant. A drive-by fix is to GetStatsOfType
as RTCOutboundRTPStreamStats instead of RTPStreamStats, this is because
of a limitation with using T::kType for runtime type information...
GetStatsOfType<T> does not work for Ts higher up in the stats hierarchy,
so it would always have returned an empty list even if
RTCOutboundRTPStreamStats were present. (Room for improvement here, but
that's a different issue.)
Bug: webrtc:14175
Change-Id: I0235bc0b66c52081859ee621c58249a6b6e98738
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288583
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38984}
In preparation for the deletion of deprecated 'track' stats, the
stats selector algorithm needs to be rewritten not to use 'trackId'.
This is achieved by finding RTP stats by their SSRC, as obtained via
getParameters(). This unfortunately adds a block-invoke (in the sender
case the block-invoke happens inside GetParametersInternal and in the
receiver case the block-invoke is explicit at the calling place), but
it can't be helped and it's just once per getStats() call and only if
the selector argument is used.
Bug: webrtc:14175
Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38981}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
As the exposure of power efficient stats to JavaScript are limited as
to reduce the fingerprinting surface to getStats, a new RTCStatsMember
derivation, RTCLimitedStatsMember, was added in this change. This sets
the exposure criteria of the stat on the type, which keeps the size of
the RTCStatsMember class the same and allows for extension in the future
for new types of stat restrictions.
Bug: webrtc:14483
Change-Id: Ib0303050a112441ba2416fd5f004dd8be26b47ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279021
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38576}
These have all been moved to "inbound-rtp" and now that upstream
projects have migrated we can delete the old location.
Unblocks https://crbug.com/webrtc/14175
Bug: webrtc:14521, webrtc:14524
Change-Id: Ia2bfa399d62304cc0ead0e65c340dfad20acc530
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281183
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38532}
These metrics were not only non-standard, but residing in the
non-standard "track" stats object that we want to delete. As per
https://github.com/w3c/webrtc-stats/issues/695#issuecomment-1259611462
these metrics are no longer needed because we already have
inbound-rtp.totalInterFrameDelay/totalSquaredInterFrameDelay which is
basically the same thing.
// mac_rel infra failures are unrelated
NOTRY=True
Bug: webrtc:14522
Change-Id: I565da42514a93f15532ba8357dd006547a5296ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38509}
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}
