The integration test sets up a loopback call, verifies media is flowing,
and then asserts which metrics should be available.
One of the things it asserted was that audioLevel is positive. This
could flake in rare circumstances because audioLevel requires a certain
number of samples to have been received before it is updated or else it
would have its default value zero.
This test is a broad asserting things about 150+ metrics; it's not worth
adding a dependency on the "implementation detail" about how long you
have to wait before this specific metric is non-zero. The fix for the
flake is to only require the metric to have been set, but zero is also
an acceptable value.
We don't lose much test coverage; we're still asserting that other
audio metrics originating from the same class have positive values.
Bug: webrtc:10962
Change-Id: I5def9193da7150492d89ea62031858bac5c41646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152821
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29179}
silentConcealedSamples, insertedSamplesForDeceleration and
removedSamplesForAcceleration were implemented in M76, but we forgot to
add them to the WEBRTC_RTCSTATS_IMPL list, meaning the "iterate all
members" method, RTCStats::Members(), did not contain these metrics.
As a consequence, Chrome did not pick up these members for exposure to
JavaScript.
Also fix the test coverage in rtc_stats_integrationtest.cc where code
paths that did not apply to audio track stats were not explicitly
asserting that they must be undefined in those cases.
Bug: chromium:996146, webrtc:10903
Change-Id: I00e7ddee600818ee4d561b88e005391830adcf3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149816
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28925}
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}
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}
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}
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}
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}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
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}
