Adds two new stats to RTCMediaStreamTrackStats:
* totalSamplesReceived is the total number of samples received on
the audio channel and includes real and synthetic samples.
* concealedSamples is the total number of synthetic samples
received on the audio channel used to conceal packet loss.
Bug: webrtc:8076
Change-Id: I36e9828525ec341490cf3310a972b56a8b443667
Reviewed-on: https://chromium-review.googlesource.com/615902
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19506}
NetEq network statistics contains discard rate but has not been used and even not been implemented until recently.
According to w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded,
this statistics needs to be replaced with an accumulative stats. Such work will be carried out separately.
Meanwhile, we need to add a rate to reflect rate of discarded redundant packets. See webrtc:8025.
In this CL, we replace the existing discard rate with secondary discarded rate, so as to
1. fulfill the requests on webrtc:8025
2. get ready to implement an accumulative statistics for discarded packets.
BUG: webrtc:7903,webrtc:8025
Change-Id: Idbf143a105db76ca15f0af54848e1448f2a810ec
Reviewed-on: https://chromium-review.googlesource.com/582863
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19495}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
After a Merge operation, the statistics for number of samples
generated using Expand must be corrected, and the correction can in
fact be negative. However, a bug was introduced in
https://codereview.webrtc.org/1230503003 which uses a size_t to
represent the correction, which leads to wrap-around of the negative
value. This is not a problem in itself, since this value is added to
another size_t, with the effect that the desired subtraction happens
anyway.
The actual problem arises if the statistics are polled/reset before a
subtraction happens -- that is, between an Expand and a Merge
operation. This will lead to an actual wrap-around of the stats value,
and large expand_rate (16384) is reported.
BUG=webrtc:7554
Review-Url: https://codereview.webrtc.org/2859483005
Cr-Commit-Position: refs/heads/master@{#18029}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
With this change, the aggregates for packet waiting times are
calculated in NetEq's StatisticsCalculator insead of in
AcmReceiver. This simplifies things somewhat, and avoids having to
copy the raw data on polling.
R=ivoc@webrtc.org, minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1296633002 .
Cr-Commit-Position: refs/heads/master@{#9778}
This CL adds calculation and logging of average excess buffer delay
and number of delayed packet outage events per minute.
The first is the average of time spent in the packet buffer for all
packets. The average is calculated for intervals of one minute, and
the result is logged to the UMA stat
WebRTC.Audio.AverageExcessBufferDelayMs.
The second is a counter of delayed packet outage events that is
restarted every minute, and the result is logged to the UMA stat
WebRTC.Audio.DelayedPacketOutageEventsPerMinute. For a description of
delayed packet outages, see previous CL implementing a duration log
for these events.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1287333005 .
Cr-Commit-Position: refs/heads/master@{#9731}
Measures the duration of each packet loss concealment (a.k.a. expand)
event that is not followed by a merge operation.
Having decoded and played packet m−1, the next expected packet is
m. If packet m arrives after some time of packet loss concealment, we
have a delayed packet outage event. However, if instead packet n>m
arrives, we have a lost packet outage event. In NetEq, the two outage
types results in different operations. Both types start with expand
operations to generate audio to play while the buffer is empty. When a
lost packet outage happens, the expand operation(s) are followed by
one merge operation. For delayed packet outages, merge is not done,
and the expand operations are immediately followed by normal
operations.
This change also includes unit tests for the new statistics.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1290113002 .
Cr-Commit-Position: refs/heads/master@{#9725}
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
