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}
The flexibility offered by the GN `rtc_jsoncpp_root` should be enough
to wire a different version of jsoncpp.
Bug: b/281848049
Change-Id: I8af39fec2406e27c3af7b7ec1949a4762dce762f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304861
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40045}
With intent to fully replace RtcpBandwidthObserver interface
and half of the TransportFeedbackObserver interface
RtcpBandwidthObserver interfaces passed bitrate and time variables as
raw ints, NetworkLinkRtcpObserver uses more expressive types.
Bug: webrtc:13757, webrtc:8239
Change-Id: I0a8c8de626fbe0c190a0a1a9f6733d863494401c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304700
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40043}
OpenSSL removed ability to generate C code:
a18cf8fc63
CL rewrites generation script to use pure Python asn1crypto library.
The changes in generated code leading to huge diff in generated file:
- Certificate array names are based on certificate fingerprints instead
of semi-human readable names, which were not referenced externally;
- Order of arrays in generated file matches the order of certificates
as they are appeared in source pem file. Previously re-ordering happen
due to writing temporary files on disk;
Bug: webrtc:11710
Change-Id: Ie7a97b3658f6ccb397f0fd0c21d341934a2cc12e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304642
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40039}
If the caller calls RegisterObserver() on the network thread while the
state is not kOpen but there are queued received data, those received
data will be immediately delivered to the observer before the state is
transitioned to kOpen, which may break the observer's assertions and
cause problems.
The problem turns out to be that, when SctpDataChannel::RegisterObserver
calls DeliverQueuedReceivedData(), the data will be passed to the
observer without checking the |state_| first, meanwhile
SctpDataChannel::UpdateState does effectively check the state and
null-check |observer_| before delivering the received data. This CL
fixes this by simply making DeliverQueuedReceivedData() also check
`state_ == kOpen`. In case the state transitions to kOpen after
RegisterObserver() is called, the first DeliverQueuedReceivedData()
call will be no-op, while the second DeliverQueuedReceivedData() call
will do the work.
Bug: chromium:1442696
Change-Id: If25ce6a038d704939b1a8ae73d7ced110448b050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304687
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40036}
The VCMReceiveStatisticsCallback interface is both implemented (by ReceiveStatisticsProxy) and called (by VideoStreamBufferController) in `video/`, so there's no reason it should be declared in `modules/video_coding`. I also took the opportunity to update the name.
No functional changes are intended by this change, but following CLs will make some changes.
Bug: webrtc:15085
Change-Id: Ib8da30ca56675e4f638d0b9778c329b9c1138acf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304662
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40034}
This is especially needed for datachannels that get created in
response to an OPEN message and RegisterObserver() is called from
within the OnDataChannel callback. More details in the associated bug.
Bug: webrtc:15165
Change-Id: I833db6c3c503623d482808dc5a02f03b9821a5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304721
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40032}
which avoids an infinitely growing SDP if the remote end rejects
the datachannel section. This will reactivate the m-line even if
all datachannels are closed.
BUG=chromium:1442604
Change-Id: If60f93b406271163df692d96102baab701923602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40029}
RtpRtcpInterface::RTT follows discouraged style of using return values,
uses raw integers to represent time delta,
and returns values that no code uses (min, max, average RTT)
added LastRtt function addresses all these stylistic issues.
Bug: webrtc:13757
Change-Id: Iaf947dd1b7139026f2beb991e69634c606c6b608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304520
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40028}
The field trial has been enabled-by-default for several years, I
suspect it was needed during its development but there doesn't seem to
be any reason to maintain it going forward.
Its very existence blocks our long term objective to have our APIs
behave according to the W3C standards and any apps still depending on
it, if there are any, should make sure to use the APIs correctly
instead. I assume they already do any any references to this is us
forgetting to clean things up.
Bug: webrtc:15161
Change-Id: I4a6a44a15219d2e045f3d8d857b5197a064f049c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304660
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40025}
The web compat requirement that was the reason for keeping
is now solved in Chromium and its stats bindings.
BUG=webrtc:9674
Change-Id: Ifb722769414b2bcc5f4d36d7dff87a875336e039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303860
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40024}
These tests were failing when building WebRTC against OpenSSL instead of
BoringSSL. The reason is that OpenSSLStreamAdapter::SSLVerifyCallback in
the BoringSSL mode returns the full cert_chain by calling
SSL_get0_peer_certificates. This API does not exist in OpenSSL, instead
only a single certificate is fetched via X509_STORE_CTX_get0_cert.
ifdef out the parts of the test that assert on cert[1] and cert[2].
An alternative but more involved way to fix these tests could be to use
X509_STORE_CTX_get1_chain to fetch the full chain on the OpenSSL path.
Bug: webrtc:15153
Change-Id: I1ede6a3c5a63d4afd2de849f5e44fcd67592aa3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40022}
Same information is already passed using ReporBlockData class
Bug: webrtc:8239
Change-Id: Iaae0edd34941c45527414a20923b5238e7a822fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304641
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40021}
As pointed out in issue webrtc:15146 this Mac/iOS specific variable,
makes the SequenceChecker behave incorrectly on those platforms.
The variable was introduced in a CL that merged the previous checker
classes, ThreadChecker and SequencedTaskChecker, but curiously neither
one of them had such a variable. So I'm not exactly sure what problem
was being solved. Hence I'm wondering if we actually need it.
Reference: https://webrtc-review.googlesource.com/c/src/+/129721
Bug: webrtc:15146
Change-Id: Ia7a9eb17b993c4f8a1e8204c658bf0b3dbdaa1e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304401
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40019}
This CL removes a PostTask in response to packet receipt reception.
This is made possible due to PacketRouter lock removal in
https://webrtc-review.googlesource.com/c/src/+/300964.
Depending on how transport code is organized, this may lead to
possibility of packet receipts arriving in
RtpTransportControllerSend which may re-enter the PacingController's
ProcessPackets method, leading to out-of-order packet sends. Fix
this by detecting re-entry and avoiding a second ProcessPackets call
in the TaskQueuePacedSender.
Bug: chromium:1373439
Change-Id: I24928f2d28a240d0860fe7e4a114cedf1f13d2bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304580
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40017}
Remove deprecated accessors returning time as raw int
Add setters for all fields to simplify usage of this class in tests
Remove unused min/max RTT fields
Bug: webrtc:13757
Change-Id: Ia8966975c15b9a930f54b4db0fc75f7002dcffe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304461
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40013}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
This CL prepares for send packet batching support in later CLs.
Bug: chromium:1439830
Change-Id: I0bbecfa895aa6d4317ef8049b3789272a440d032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304282
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40009}
There is a lot of different delays tracked by this class. Renaming this
member clarifies what delay it actually tracks.
Bug: webrtc:15085
Change-Id: I8b038ecf84ca262efdc9f69c0f9a2a9eaad81d37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304402
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40007}
rtcp::ReportBlock class is designed for serialization while
ReportBlockData designed for passing report block information across
multiple components.
This slightly reduce how RtcpTransceiver and RtcpReceiver interact with other webrtc components
Bug: webrtc:8239
Change-Id: I582e3d7b32dc6357954b29a1def37e2e72116a74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304285
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40006}
guarding the change in receive stats behind the
WebRTC-Stats-RtxReceiveStats
field trial which acts as a killswitch.
BUG=webrtc:15096
Change-Id: I475a2ce4fe4bddd454aa6477f8818384696c007b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304160
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@{#40005}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: I580e8412d379931bfdf9517e0a8be25c19e0cd32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304100
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40004}
