Whether a metric is to be exposed to JavaScript or not is a blink
implementation detail that the WebRTC repository does not need to be
concerned with.
This CL removes unused code and paves the way for the possibility of
making the one and only RTCStatsMember class be absl::optional<>-based
in the future.
Bug: webrtc:15162
Change-Id: I578715f48b8fcc3534b72b4c700fd6567f8d553e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304722
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40139}
NonStandardGroupId is no longer wired up to Chrome, but if we did want
to only expose certain metrics if a field trial was enabled then the
right place to do that would be in blink, where WebIDL lives.
This was only used prior to the WebRtcStatsReportIdl launch and
experiments haven't been active in several years so its dead code.
Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4514754
Bug: webrtc:15162
Change-Id: Ia41a4d21d7e5f029ddb121183fbd69ae7f98fac4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304720
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40132}
Step 1: Make reading RTCStatsMember look the same as reading
absl::optional (this CL).
Step 2: Migrate uses of "is_defined()" to "has_value()".
Step 3: Delete "is_defined()".
Step 4: Make RTCStatsMember+Interface an implementation detail of
RTCStats::Members(), only used for abstract iteration ("for
each metric"). Lazy instantiate it upon Members().
Step 5: Replace RTCStatsMember with absl::optional for use in RTCStats
dictionaries (rtcstats_objects.h/cc).
Bug: webrtc:15164
Change-Id: I5a2c9fe56707e3c7d89e8ea62fb37171ae806a7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304840
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40048}
in favor of the Timestamp constructor and method.
The constructor is most likely not used outside libWebRTC,
the call to
.timestamp_us()
can be replaced with
.timestamp().us()
BUG=webrtc:14813
Change-Id: Id166b4f85b2425ecec1c7ebb81406f82ff9d95c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290727
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39066}
Adds a new StatExposureCriteria for non-standard stats. This removes the
virtual call to is_standardized() which can simply use the
StatExposureCriteria.
Bug: webrtc:14546
Change-Id: If4174019ff8cc6559ab0dc9a04e0f8a6631b9842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279045
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39057}
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}
which is a more modern way to represent something that either has a value or is not set
BUG=webrtc:14544
Change-Id: I0a06b30b1c7f802247eb1f60e69271594b94a6f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278421
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@{#38443}
Recent WebRTC stats spec changes have added restrictions on what stats
are available to JavaScript. This is done to reduce that fingerprinting
surface of WebRTC getStats. For example, stats exposing hardware
capabilities have requirements that must be met by the browser. See [1]
for more details.
This CL adds the types and the enumerations. Stats with these
restrictions should not be added until Chromium has implemented
filtering based on the stat type.
[1] https://w3c.github.io/webrtc-stats/#limiting-exposure-of-hardware-capabilities
Bug: webrtc:14546
Change-Id: I6dae5d4921c7a2bc828a4fc8f7d68e0c59f3be82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279043
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38381}
Polymorphic comparison operators doesn't work in C++20.
(-Wambiguous-reversed-operator)
Fix this issue by using the non-virtual interface pattern.
Bug: chromium:1284275
Change-Id: I79e2bbcd3ae2f3b089183146f7e7c775c493e3f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38210}
apart from the certificate stats which need to update the
reference to the previous certificate stats in the chain.
BUG=None
Change-Id: I27f58084b849fd9afe236e5b57139bedb8eb1811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274175
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38026}
Starting from [1], explicit template declaration/definition is in use
for this template so there is no need to RTC_EXPORT its declaration.
Doing so leads to this error on clang-cl:
../../third_party/webrtc\api/stats/rtc_stats.h(372,1): error: explicit instantiation declaration should not be 'dllexport' [-Werror,-Wdllexport-explicit-instantiation-decl]
WEBRTC_DECLARE_RTCSTATSMEMBER(bool);
^
../../third_party/webrtc\api/stats/rtc_stats.h(369,3): note: expanded from macro 'WEBRTC_DECLARE_RTCSTATSMEMBER'
extern template class RTC_EXPORT_TEMPLATE_DECLARE(RTC_EXPORT) \
^
../../third_party/webrtc\api/stats/rtc_stats.h(287,7): note: attribute is here
class RTC_EXPORT RTCStatsMember : public RTCStatsMemberInterface {
^
../..\third_party/webrtc/rtc_base/system/rtc_export.h(24,31): note: expanded from macro 'RTC_EXPORT'
Full log: https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/430931
[1] - https://webrtc-review.googlesource.com/c/src/+/158795
Bug: webrtc:9419
Change-Id: I9f0893ae26b45049f186e19f862a1d138a320a24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158891
Reviewed-by: Nico Weber <thakis@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29703}
This CL works around an "Explicit specialization after instantiation
error" when building with clang-cl and is_component_build=true (see
crbug.com/1018579). On top of that it uses "template instantiation
declarations/declarations" in order to avoid to instantiate the
template in clients code.
TBR: hbos@webrtc.org
Bug: webrtc:9419, chromium:1018579
Change-Id: I1b2862de678586afc81e8f7a407947322f8a06c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158795
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29683}
This implements RTCAudioSourceStats and RTCVideoSourceStats, both
inheriting from abstract dictionary RTCMediaSourceStats:
https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
All members are implemented except for the total "frames" counter:
- trackIdentifier
- kind
- width
- height
- framesPerSecond
This means to make googFrameWidthInput, googFrameHeightInput and
googFrameRateInput obsolete.
Implemented using the same code path as the goog stats, there are
some minor bugs that should be fixed in the future, but not this CL:
1. We create media-source objects on a per-track attachment basis.
If the same track is attached multiple times this results in
multiple media-source objects, but the spec says it should be on a
per-source basis.
2. framesPerSecond is only calculated after connecting (when we have a
sender with SSRC), but if collected on a per-source basis the source
should be able to tell us the FPS whether or not we are sending it.
Bug: webrtc:10453
Change-Id: I23705a79f15075dca2536275934af1904a7f0d39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28028}
