Ultimately, IDs should be random according to spec[1], so we shouldn't
rely on the ID to convey easily readable information. By making the IDs
shorter we reduce the overhead of string copies and make report dumps a
little bit smaller.
Drive-by: Add "DEPRECATED_" prefic to the RTCMediaStreamStats ID.
[1] https://w3c.github.io/webrtc-pc/#dom-rtcstats-id
# Examples of IDs before and after this CL #
RTCDataChannel_3
-> D3
RTCPeerConnection
-> P
RTCTransport_0_1
-> T01
RTCCodec_RTCTransport_0_1_100_minptime=10;useinbandfec=1
-> CIT01_100_minptime=10;useinbandfec=1
RTCInboundRTPAudioStream_6666
-> IA6666
RTCAudioSource_1
-> SA1
RTCOutboundRTPAudioStream_2943129392
-> OA2943129392
RTCRemoteInboundRtpAudioStream_3541280085
-> RIA3541280085
RTCIceCandidate_6cWRqicY
-> I6cWRqicY
RTCIceCandidatePair_6cWRqicY_haEcM2xD
-> CP6cWRqicY_haEcM2xD
RTCCertificate_FD1:BC:58:90:DF:E8:40:58:8D:04:91:44:93:4E:6C:52:9E:F0:14:98:AA:67:7B:8B:C8:30:C8:31:D0:84:1B:BF
-> CFD1:BC:58:90:DF:E8:40:58:8D:04:91:44:93:4E:6C:52:9E:F0:14:98:AA:67:7B:8B:C8:30:C8:31:D0:84:1B:BF
DEPRECATED_RTCMediaStreamTrack_receiver_3
-> DEPRECATED_TI3
RTCMediaStream_45a6e766-5d1a-40f9-a55c-ea8fdefcde49
-> DEPRECATED_S45a6e766-5d1a-40f9-a55c-ea8fdefcde49
Bug: webrtc:14416, webrtc:14419
Change-Id: I11f0a8b8354203fea1df1093d8864a6d47ee71e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273709
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37992}
The code incorrectly assumed that codecs exist on a per-mid/transceiver
basis, but codec payload types are unique on a per-transport basis and
in practise most applications use BUNDLE (single transport for the
entire PC).
This CL makes the codecs per-transport instead of per-transceiver. We
still need to iterate transceivers because codecs are exposed on a
per-transceiver basis and as shown in
https://jsfiddle.net/henbos/7kqxgnr8/ it is possible for FMTP lines to
be different on different m= sections despite BUNDLE.
Manual testing shows that this CL brings down the number of "codec"
stats in Google Meet 50p from 872 objects to 43 objects.
Bug: webrtc:14414
Change-Id: Ic854b31bd595799554b99fff22cbd48264ebd141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273707
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37989}
There's no way to add a deprecation warning unique to using
RTCMediaStreamTrackStats, but we could signal to users that it is
deprecated by adding "DEPRECATED_" to its ID.
This could break apps with hardcoded assumptions about what the stats
IDs are, but apps doing this are using the API incorrectly anyway, so
if anyone is affected by this change that would be a good time to
remove any dependency on this (see https://crbug.com/webrtc/10656
regading the fact that IDs should be unpredictable).
Bug: webrtc:14175
Change-Id: I6242c4efc08e9570420c00af5aaf491b1af819f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269004
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37595}
increasing precision since summing up rounded values leads to
a rounding error, in particular for small frames which take very
little time to decode.
BUG=webrtc:12526,webrtc:13756
Change-Id: I647c702808856a002c746ed9f115aa9bcaddc1f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262810
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37249}
This should allow standard stats users not to have to rely on the
obsolete "track" stats.
Bug: webrtc:14174
Change-Id: I24e5e1478ee47c73c12fcdecf7314f41fcc76bc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37246}
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
Since the lifetime of an SctpDataChannel is not strictly controlled
by its controller, the controller might go away before the channel
does. This CL guards against this.
Bug: webrtc:13931
Change-Id: I07046fe896d1a66bf89287429beb0587382a13a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36852}
This better reflects the ownership passing of AddTrack, and is more
consistent for RemoveTrack.
Bug: webrtc:13980
Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36603}
This cl/ extends the RTCIceCandidateStats object with
network_adapter_type and vpn, so that it maps the underlying
WebRTC objects completly.
Bug: webrtc:13773
Change-Id: I5cf79972c60ca6bf2a127dc96fa90811263ba6fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36110}
This cl/ changes so that the RTCTransportStats bytes/packets
sent/recevied is computed in P2PTransportChannel. Previously
they were computed by aggregating over the Connections, but that
does not work when Connections are created and destroyed.
Bug: webrtc:13769
Change-Id: Ia97dfae70b5aced897d4813ec007ba61bc032f87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36103}
Also apply IWYU to all .cc files in pc/, and correct BUILD file to match.
Note: Some files came out wrong when iwyu was applied. These are not included.
Bug: none
Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36064}
This makes SetChannel() consistently make 2 invokes instead of a
multiple of senders+receivers (previous minimum was 4 but could be
larger).
* Stop() doesn't hop to the worker thread.
* SetMediaChannel(), an already-required step on the worker thread for
senders and *sometimes* for receivers[1], is now consistently required
for both. This simplifies transceiver teardown and enables the next
bullet.
* Transceiver stops all senders and receivers in one go rather than
ping ponging between threads.
[1] When not required, it was done implicitly inside of Stop().
See changes in `RtpTransceiver::SetChannel`
Bug: webrtc:13540
Change-Id: Ied61636c8ef09d782bf519524fff2a31e15219a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249797
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36057}
This is a straightforward replacement; simplifications due to the ability
to inline functions in the lambdas are for a later CL.
Bug: webrtc:11943
Change-Id: I7274cedde507b954f1d8aa8bc560861102eeb264
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35936}
Move deletion of channel objects over to the RtpTransceiver instead
of having it done by SdpOfferAnswer.
The deletion is now also done via PostTask rather than Invoke.
Bug: webrtc:11992, webrtc:13540
Change-Id: I5aff14956d5e572ca8816bbfef8739bb609b4484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248170
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35798}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This makes relay candidates identifiable in scenarios
where the TURN server is behind another entity and the peer
sees a different ip address:
client -> turn -> relay address -> third party relay + address -> peer
In those cases, the relay candidate will become peer-reflexive
since the peer sends the third party relay's address in the xor-mapped
address and it is currently not easily possible to determine this is a
relay candidate anymore.
BUG=webrtc:13392
Change-Id: I6787339d0abdc735f8a43f636a676cccd8cadcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35346}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
