759f161182
Use explicit template decl/def for RTCNonStandardStatsMember.
...
This should fix the error raised by lld-link while testing the WebRTC
component build (see [1]).
[1] - https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/436729
Bug: webrtc:9419
Change-Id: Ia239f544432c55ae69e2da474b8284143c94a003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159697
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29786}
2019-11-13 11:12:40 +00:00
f47c2ab889
RTC_EXPORT webrtc::RTCNonStandardStatsMember<T>.
...
Since webrtc::RTCNonStandardStatsMember<T> extends some explicit
template initializations, its symbols need to be exported as well.
Example of an lld-link error:
https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/435974
Bug: webrtc:9419
Change-Id: I48bd484b821b17a214e0180e78ba34c38e247947
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159693
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29780}
2019-11-12 21:19:11 +00:00
62a19d0798
RTC_EXPORT RTCStatsMember::StaticType and VideoFrame::Builder.
...
Bug: webrtc:9419
Change-Id: I093a00926af5f5169e14d2680a65c8cbda6cbfba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159542
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29760}
2019-11-11 19:40:25 +00:00
54875d05f1
Add missing RTC_EXPORT for the Chromium Windows build.
...
After fixing the issue with crbug.com/1018579, lld-link complained
that some symbols need to be exported, see [1].
[1] - https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/432025
Bug: webrtc:9419
Change-Id: I9107a9c76361f4c66463a9af2e81a3991ae14df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159007
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29721}
2019-11-07 09:32:24 +00:00
1e6aa1f8bf
Remove RTC_EXPORT from RTCStatsMember<T> template declaration.
...
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}
2019-11-06 10:26:45 +00:00
054f18513e
Use template instantiation declaration/definition for RTCStatsMember<T>.
...
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}
2019-11-04 16:24:37 +00:00
5cb7807a36
Implement crypto stats on DTLS transport
...
Bug: chromium:1018077
Change-Id: I585d4064f39e5f9d268b408ebf6ae13a056c778a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158403
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29628}
2019-10-28 11:30:23 +00:00
fcf79cca7b
Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.
...
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
Partial implementation: currently only populated when a/v sync is enabled.
Bug: webrtc:7065
Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29581}
2019-10-23 07:46:39 +00:00
ac0a4cbbd8
Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
...
This is a reland of fbde32e596f06893d6dda13eb7d29f4c251cf08b
The chromium problem should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/1862437
Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
>
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Niels Moller <nisse@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29462}
Tbr: kwiberg@webrtc.org
Bug: webrtc:10525
Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045
Reviewed-by: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29485}
2019-10-15 10:43:59 +00:00
ef0627fb50
Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received"
...
This reverts commit fbde32e596f06893d6dda13eb7d29f4c251cf08b.
Reason for revert: It seems to break WebRTC FYI tests in Chromium.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4763
Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
>
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
> Reviewed-by: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Harald Alvestrand <hta@webrtc.org >
> Commit-Queue: Niels Moller <nisse@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#29462}
TBR=kwiberg@webrtc.org ,hbos@webrtc.org ,nisse@webrtc.org ,hta@webrtc.org
Change-Id: I6a983ea4d5ff38e49f096a8ff5cd9b426768f955
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29478}
2019-10-15 08:55:06 +00:00
fbde32e596
Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
...
Changes the standard GetStats, legacy GetStats unchanged.
Bug: webrtc:10525
Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29462}
2019-10-14 13:07:13 +00:00
cc62b16658
Add qualityLimitationResolutionChanges stat
...
Implements the stat qualityLimitationResolutionChanges [1].
[1] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
Bug: webrtc:10935
Change-Id: I391f2be5958a96b442e32c40ab7043752f3f71dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150882
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Evan Shrubsole <eshr@google.com >
Cr-Commit-Position: refs/heads/master@{#29113}
2019-09-09 15:22:57 +00:00
149dc72dfa
Add support for RTCTransportStats.selectedCandidatePairChanges
...
This patch adds accounting and reporting needed for
newly added RTCTransportStats.selectedCandidatePairChanges,
https://w3c.github.io/webrtc-stats/#dom-rtctransportstats-selectedcandidatepairchanges
a) P2PTransportChannel counts everytime selected_connection_
is modified and reports this counter in the GetStats()-call.
b) RTCStatsCollector puts the counter into the standardized
stats object.
Bug: webrtc:10900
Change-Id: Ibaeca18706b8edcbcb44b0c6f2754854bcb545ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149830
Reviewed-by: Qingsi Wang <qingsi@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28987}
2019-08-28 13:22:08 +00:00
6b430867b8
Reland "[GetStats] Expose video codec implementation in standardized metrics."
...
This is a reland of 2b9fa09fa3e3379fd8e76490c394f25670352ef2.
It got reverted because I forgot to whitelist the new metrics in chromium,
which has now been done:
https://chromium-review.googlesource.com/c/chromium/src/+/1760209
Relanding requires no changes to the CL.
Original change's description:
> [GetStats] Expose video codec implementation in standardized metrics.
>
> Spec issue: https://github.com/w3c/webrtc-stats/issues/445
> Spec PR: https://github.com/w3c/webrtc-stats/pull/473
>
> Now that the spec's RTCCodecStats.implementation has moved to
> RTCOutboundRtpStreamStats.encoderImplementation and
> RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
> using the same string that the legacy getStats() API used.
>
> Bug: webrtc:10890
> Change-Id: Ic43ce44735453626791959df3061ee253356015a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
> Commit-Queue: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#28877}
TBR=ilnik@webrtc.org
Bug: webrtc:10890
Change-Id: Ib874b608856c2795b1ca08f6af43c61dd859ea21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149800
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28887}
2019-08-19 09:09:18 +00:00
df625f46c0
Revert "[GetStats] Expose video codec implementation in standardized metrics."
...
This reverts commit 2b9fa09fa3e3379fd8e76490c394f25670352ef2.
Reason for revert: speculative revert since it seems to break Chrome FYI bots. See https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4206
Original change's description:
> [GetStats] Expose video codec implementation in standardized metrics.
>
> Spec issue: https://github.com/w3c/webrtc-stats/issues/445
> Spec PR: https://github.com/w3c/webrtc-stats/pull/473
>
> Now that the spec's RTCCodecStats.implementation has moved to
> RTCOutboundRtpStreamStats.encoderImplementation and
> RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
> using the same string that the legacy getStats() API used.
>
> Bug: webrtc:10890
> Change-Id: Ic43ce44735453626791959df3061ee253356015a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
> Commit-Queue: Henrik Boström <hbos@webrtc.org >
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#28877}
TBR=ilnik@webrtc.org ,hbos@webrtc.org
Change-Id: Ia0b7f9806564cf28881c50d6371b8141a22e3431
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10890
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149175
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28879}
2019-08-16 15:29:28 +00:00
2b9fa09fa3
[GetStats] Expose video codec implementation in standardized metrics.
...
Spec issue: https://github.com/w3c/webrtc-stats/issues/445
Spec PR: https://github.com/w3c/webrtc-stats/pull/473
Now that the spec's RTCCodecStats.implementation has moved to
RTCOutboundRtpStreamStats.encoderImplementation and
RTCInboundRtpStreamStats.decoderImplementation, this CL implements them
using the same string that the legacy getStats() API used.
Bug: webrtc:10890
Change-Id: Ic43ce44735453626791959df3061ee253356015a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149168
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28877}
2019-08-16 14:10:46 +00:00
a4d873786f
Format almost everything.
...
This CL was generated by running
git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format
Most of these changes are clang-format grouping and reordering includes
differently.
Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
d2c336f892
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
...
Implements RTCAudioSourceStats members:
- audioLevel
- totalAudioEnergy
- totalSamplesDuration
In this CL description these are collectively referred to as the audio
levels.
The audio levels are removed from sending "track" stats (in Chrome,
these are now reported as undefined instead of 0).
Background:
For sending tracks, audio levels were always reported as 0 in Chrome
(https://crbug.com/736403 ), while audio levels were correctly reported
for receiving tracks. This problem affected the standard getStats() but
not the legacy getStats(), blocking some people from migrating. This
was likely not a problem in native third_party/webrtc code because the
delivery of audio frames from device to send-stream uses a different
code path outside of chromium.
A recent PR (https://github.com/w3c/webrtc-stats/pull/451 ) moved the
send-side audio levels to the RTCAudioSourceStats, while keeping the
receive-side audio levels on the "track" stats. This allows an
implementation to report the audio levels even if samples are not sent
onto the network (such as if an ICE connection has not been established
yet), reflecting some of the current implementation.
Changes:
1. Audio levels are added to RTCAudioSourceStats. Send-side audio
"track" stats are left undefined. Receive-side audio "track" stats
are not changed in this CL and continue to work.
2. Audio level computation is moved from the AudioState and
AudioTransportImpl to the AudioSendStream. This is because a) the
AudioTransportImpl::RecordedDataIsAvailable() code path is not
exercised in chromium, and b) audio levels should, per-spec, not be
calculated on a per-call basis, for which the AudioState is defined.
3. The audio level computation is now performed in
AudioSendStream::SendAudioData(), a code path used by both native
and chromium code.
4. Comments are added to document behavior of existing code, such as
AudioLevel and AudioSendStream::SendAudioData().
Note:
In this CL, just like before this CL, audio level is only calculated
after an AudioSendStream has been created. This means that before an
O/A negotiation, audio levels are unavailable.
According to spec, if we have an audio source, we should have audio
levels. An immediate solution to this would have been to calculate the
audio level at pc/rtp_sender.cc. The problem is that the
LocalAudioSinkAdapter::OnData() code path, while exercised in chromium,
is not exercised in native code. The issue of calculating audio levels
on a per-source bases rather than on a per-send stream basis is left to
https://crbug.com/webrtc/10771 , an existing "media-source" bug.
This CL can be verified manually in Chrome at:
https://codepen.io/anon/pen/vqRGyq
Bug: chromium:736403, webrtc:10771
Change-Id: I8036cd9984f3b187c3177470a8c0d6670a201a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143789
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28480}
2019-07-04 08:13:45 +00:00
bfd343b9be
Add totalDecodeTime to RTCInboundRTPStreamStats
...
Pull request to WebRTC stats specification:
https://github.com/w3c/webrtc-stats/pull/450
Bug: webrtc:10775
Change-Id: Id032cb324724329fee284ebc84595b9c39208ab8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144035
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28440}
2019-07-02 08:28:06 +00:00
2efae7786e
Add RTCStats for keyFramesEncoded, keyFramesDecoded.
...
This implements the correspondingly named JavaScript fields defined in
https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict * and
https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict *.
Bug: webrtc:7066
Change-Id: I431045bca80bf5faf27132c54f59c1f723c92952
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143683
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28404}
2019-06-27 14:59:47 +00:00
3472b9ae22
Delete RTCInboundRTPStreamStats::fraction_lost
...
And delete corresponding plumbing via the internal stats attribute
MediaReceiverInfo::fraction_lost. The latter attribute is not deleted
yet, since downstream projects have to be updated first.
Bug: webrtc:10744
Change-Id: Id5401aeee7e5637a406ddf2fa33fbfe336abec9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143178
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28385}
2019-06-26 11:43:23 +00:00
ce33b6a4cf
Implement QualityLimitationReasonTracker and expose "reason".
...
This CL implements the logic behind qualityLimitationReason[1] and
qualityLimitationDurations[2]
This CL also exposes qualityLimitationReason in the standard getStats()
API, but does not expose qualityLimitationDurations because that is
blocked on supporting the "record<>" type in RTCStatsMember[3].
[1] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason
[2] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
[3] https://crbug.com/webrtc/10685
TBR=stefan@webrtc.org
Bug: webrtc:10451, webrtc:10686
Change-Id: Ifff0be4ddd64eaec23d59c02af99fdbb1feb3841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138825
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28090}
2019-05-28 16:23:55 +00:00
883eefc59e
Implement RTCRemoteInboundRtpStreamStats for both audio and video.
...
This implements the essentials of RTCRemoteInboundRtpStreamStats. This
includes:
- ssrc
- transportId
- codecId
- packetsLost
- jitter
- localId
- roundTripTime
https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict *
The following members are not implemented because they require more
work...
- From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded,
packetsRepaired, burstPacketsLost, burstPacketsDiscarded,
burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate,
gapLossRate and gapDiscardRate.
- From RTCRemoteInboundRtpStreamStats: fractionLost.
Bug: webrtc:10455, webrtc:10456
Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28073}
2019-05-27 12:45:22 +00:00
646fda0212
Implement RTCMediaSourceStats and friends in standard getStats().
...
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}
2019-05-22 16:03:41 +00:00
23aff9b737
Implement RTCOutboundRtpStreamStats.totalEncodedBytesTarget.
...
This is a standardized metric:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
We estimate the target frame size in bytes from the current encoder
target bitrate and encoder framerate.
We would expect that the average bytes produced by the encoder would
over time match the average target, which is calculated by polling
getStats() twice and dividing the delta totalEncodedBytesTarget with
the delta framesEncoded. This is meant to make googTargetEncBitrate
obsolete.
Bug: webrtc:10446
Change-Id: Ib10ce236476a2f965582d5c536f419952926d4e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137200
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28022}
2019-05-22 10:59:39 +00:00
9fe1834d5d
Implement RTCOutboundRtpStreamStats.totalPacketSendDelay for video.
...
This is a standardized metric. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
It is meant to replace the legacy googBucketDelay. The average
packet delay over any interval can be calculated as the delta
totalPacketSendDelay divided by the delta packetsSent between two
calls to getStats().
Bug: webrtc:10506
Change-Id: I3d6c6d66e5a06937d0ea8d182a82cd255084ad19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137044
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27979}
2019-05-17 18:53:20 +00:00
8d8ffdbcca
Expose new audio stats on the API
...
Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887 . This CL
adds these to the GetStats API.
Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27839}
2019-05-03 10:10:15 +00:00
44125faba5
Reland "Piping audio interruption metrics to API layer"
...
The metrics are now added as RTCNonStandardStatsMember objects in
RTCMediaStreamTrackStats. Unit tests are updated.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/134303 ,
with fixes.
TBR=kwiberg@webrtc.org
Bug: webrtc:10549
Change-Id: I29dcc6fbfc69156715664e71acfa054c1b2d9038
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134500
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27806}
2019-04-29 15:39:50 +00:00
cf96e0f87d
Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.
...
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
These are already existed in StreamDataCounters. This CL takes care of
the plumbing of these values to the standard stats collector.
TBR=solenberg@webrtc.org
Bug: webrtc:10447
Change-Id: I27d6c3ee3ab627d306303e6ee67e586ddf31cc81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132012
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27663}
2019-04-17 13:04:50 +00:00
01738c63aa
Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.
...
This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.
Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27628}
2019-04-15 16:06:01 +00:00
2e06926c95
Implement RTC[In/Out]boundRtpStreamStats.contentType.
...
Spec: https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
This already exists as a goog-stat. This CL only plumbs the value to the
new stats collector.
Note: There is currently no distinction between the extension being
missing and it being present but the value being "unspecified". Until
https://crbug.com/webrtc/10529 is fixed, this metric is only exposed if
SCREENSHARE was present.
Bug: webrtc:10452
Change-Id: Ic8723f4d0efb43ab72a560e954676facd3b90659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131946
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27520}
2019-04-09 13:02:03 +00:00
f71362f0cf
Wire up RTCOutboundRtpStreamStats.totalEncodeTime.
...
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/130517 that calculated
this metric.
This CL is purely plumbing, exposing
VideoSendStream::total_encode_time_ms in standard getStats() as
RTCOutboundRtpStreamStats.totalEncodeTime (in seconds):
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime
Bug: webrtc:10448
Change-Id: I715f1ef937e441169dee55b5e8d4fbf98811c5f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131940
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27501}
2019-04-09 07:34:38 +00:00
aa023e2553
Add test-only group id.
...
Bug: chromium:943076
Change-Id: Ife855fe59fe5e358bc94e4bb0da704ee2647dbd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129900
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27356}
2019-03-29 15:53:16 +00:00
2293622f02
Add group_ids to RTCStatsMemberInterface.
...
Bug: chromium:943076
Change-Id: I1bd40cbbd121dbc3c961650926d9a28fe90f6883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128769
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27235}
2019-03-22 11:45:51 +00:00
758d946106
Add origin trial ids to non-standard stats members.
...
Bug: chromium:943076
Change-Id: I2d8211d3acd844cf602ed1c7de08bb7441263950
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128420
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#27186}
2019-03-19 16:53:47 +00:00
232b3fda92
Expose relative packet arrival delay metric in stats API.
...
The metric is non-standard and documented in: https://github.com/henbos/webrtc-provisional-stats/pull/14
Bug: webrtc:10333
Change-Id: Ie5b4bbad5b1e2c9104742931529bab8f48f51f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125861
Reviewed-by: Minyue Li <minyue@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26999}
2019-03-06 16:35:16 +00:00
22f9925b3e
webrtc: Remove semicolons.
...
Bug: chromium:926235
Change-Id: I66c10ab3df38adf87152d1f18cc8162afedca7e4
Reviewed-on: https://webrtc-review.googlesource.com/c/123560
Commit-Queue: Tommi <tommi@webrtc.org >
Reviewed-by: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26780}
2019-02-20 16:02:59 +00:00
0237106559
Expose video freeze metrics in GetStats.
...
This adds the following non-standardized metrics to video receiver
stats:
- freezeCount
- pauseCount
- totalFreezesDuration
- totalPausesDuration
- totalFramesDuration
- sumOfSquaredFrameDurations
For description of these metrics see
https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict *
Bug: webrtc:10145
Change-Id: I4c76d5651102e73b1592ffed561e6224f2badeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/114840
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26523}
2019-02-04 09:58:08 +00:00
6c70e16a4a
Fix RTCStatsReport::ConstIterator move constructor.
...
This problem has been originally discovered by clang-tidy while trying
to apply performance-move-const-arg [1] on [2].
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-move-const-arg.html
[2] - https://webrtc-review.googlesource.com/c/src/+/120350
Bug: webrtc:10252
Change-Id: I2b0a116f78ca8096a6cf2bc23e2f4b8f372ca04f
Reviewed-on: https://webrtc-review.googlesource.com/c/120815
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26512}
2019-02-01 14:12:43 +00:00
f380284035
(7) Rename files to snake_case: remove forwarding headers
...
Bug: webrtc:10159
Change-Id: I2ba899e0283b953538c7941c8790213e36d7c70e
Reviewed-on: https://webrtc-review.googlesource.com/c/118561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26417}
2019-01-26 00:33:46 +00:00
d970807e0c
Remove rtc_base/scoped_ref_ptr.h.
...
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o .
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26414}
2019-01-25 20:29:58 +00:00
0acffb5b36
Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats().
...
NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.
Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Chen Xing <chxg@google.com >
Cr-Commit-Position: refs/heads/master@{#26276}
2019-01-16 11:44:10 +00:00
aec15aa810
(5) Rename files to snake_case: install forwarding headers
...
Mechanically generated with this command:
tools_webrtc/do-renames.sh install all-renames.txt && git cl format
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ic8e99f71f2da62e5c99863c6d24a8cfe311466cd
Reviewed-on: https://webrtc-review.googlesource.com/c/115682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26227}
2019-01-11 17:13:36 +00:00
10542f21c8
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
...
Mechanically generated by running this command:
tools_webrtc/do-renames.sh update all-renames.txt && git cl format
Then manually updating:
tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
1c05765831
(3) Rename files to snake_case: move the files
...
Mechanically generated with this command:
tools_webrtc/do-rename.sh move all-renames.txt
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
ffa6ce4714
Add integration test for new GetStats() with many tracks
...
Bug: None
Change-Id: Ia134b4563edbb40e5082592666aad8ad70f1f1a4
Reviewed-on: https://webrtc-review.googlesource.com/c/112186
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25859}
2018-11-30 18:09:31 +00:00
3e70781361
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
...
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280 .
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25830}
2018-11-28 18:25:07 +00:00
352ce5c419
Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
...
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI
Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25802}
2018-11-27 15:10:09 +00:00
8af8896596
Expose jitter buffer flushes metric in new getStats api.
...
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#
Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Ruslan Burakov <kuddai@google.com >
Cr-Commit-Position: refs/heads/master@{#25768}
2018-11-23 11:41:43 +00:00
276827cbdb
Export symbols needed by the Chromium component build (part 3).
...
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).
Bug: webrtc:9419
Change-Id: I4d4e2ae52ee01de68147fd0f2cfe4c92d600ad94
Reviewed-on: https://webrtc-review.googlesource.com/c/106343
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#25207}
2018-10-16 12:57:04 +00:00
