The tests should say something about RTPs too, because what they say
about track will soon be irrelevant. A drive-by fix is to GetStatsOfType
as RTCOutboundRTPStreamStats instead of RTPStreamStats, this is because
of a limitation with using T::kType for runtime type information...
GetStatsOfType<T> does not work for Ts higher up in the stats hierarchy,
so it would always have returned an empty list even if
RTCOutboundRTPStreamStats were present. (Room for improvement here, but
that's a different issue.)
Bug: webrtc:14175
Change-Id: I0235bc0b66c52081859ee621c58249a6b6e98738
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288583
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38984}
This unblocks the deletion of this deprecated stats object.
Bug: webrtc:14175
Change-Id: I850c028fc9556a36191909afa3d635a7e6b65b69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288582
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38983}
In preparation for the deletion of deprecated 'track' stats, the
stats selector algorithm needs to be rewritten not to use 'trackId'.
This is achieved by finding RTP stats by their SSRC, as obtained via
getParameters(). This unfortunately adds a block-invoke (in the sender
case the block-invoke happens inside GetParametersInternal and in the
receiver case the block-invoke is explicit at the calling place), but
it can't be helped and it's just once per getStats() call and only if
the selector argument is used.
Bug: webrtc:14175
Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38981}
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Bug: webrtc:14802
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38980}
An RTP dump may or may not include the payload of the recorded RTP packets. When the payload is not present packets should still be created with their original packet length.
Bug: webrtc:14801
Change-Id: Ice74cb5f7d370aaefac5f370445ffd3f2fc5924c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289920
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38979}
rtc::CopyOnWriteBuffer::SetSize extends buffer with uninitialized memory by design.
It is up to the user of the rtc::CopyOnWriteBuffer to ensure it is initialized.
Bug: chromium:1404299
Change-Id: I41f3f91bf20ff440984d78ed81e01f5db36ff509
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38972}
rtc::CopyOnWriteBuffer::SetSize extends buffer with uninitialized memory by design.
It is up to the user of the rtc::CopyOnWriteBuffer to ensure it is initialized.
Bug: chromium:1403397
Change-Id: Ic0111a84bda32379770ddb1c7d24bee10d96b7a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289041
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38959}
preparing to put them at session level when max-bundle is set.
Drive-by: move m= serialization to helper.
BUG=None
Change-Id: I04d918ee8eb70c0cc40baf8ebc12054c6b3a2a15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288820
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38950}
which needs to be taken into account separately if the
primary SSRC has been acknowledged but the RTX SSRC has
not.
If nothing has been acknowledged, mid+rid are sent on the primary SSRC and mid+rrid are sent on the RTX SSRC.
If the primary SSRC has been acknowledged, no extensions are sent on the primary SSRC and mid+rrid are sent on the RTX SSRC.
If both the primary SSRC and the RTX SSRC have been ack'd, no extensions are sent on either primary or RTX SSRC.
BUG=webrtc:13896
Change-Id: Ice251fae23a881ee9c9edc71b5d5c45a32ac76d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38949}
Instead of getting header extension mapping from a receiver object, get the mapping from the received packet.
The purpose is to be able to remove extension information from webrtc/call/receive_stream.h.
Header extensions are negotiated per mid, not per receive stream.
The goal is to reduce the number of places where packets are parsed and demuxed.
Bug: webrtc:7135, webrtc:14795
Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38944}
UnitBase own function to divide with rounding overflows when dividend is close to the max int64_t
Bug: b/262999013
Change-Id: I5b0c3f4408165a0f03690cab80bd098e506fc984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38943}
https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
has a step saying
For each stream in streams, add stream.id to
[[AssociatedMediaStreamIds]] if it's not already there
This applies to addTrack and setStreams and the set of streams in
addTransceiver.
Tests that default to the stream id as sync group add
"-sync" as a postfix
BUG=webrtc:14769
Change-Id: I806d2fd87a98d50e54709755541f3f1efff1d8ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288701
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38942}
This reverts commit 315b95ca11161bdea715d5316f92828edd41f0d5.
Reason for revert: Breaks internal bots.
Original change's description:
> Enforce stream id uniqueness in RtpSender::set_stream_ids
>
> https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
> has a step saying
> For each stream in streams, add stream.id to
> [[AssociatedMediaStreamIds]] if it's not already there
>
> This applies to addTrack and setStreams and the set of streams in
> addTransceiver.
>
> BUG=webrtc:14769
>
> Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38937}
Bug: webrtc:14769
Change-Id: I6fd22ff0550c0894057fb1dc15f1b95819fa6df2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38940}
The STUN message about mismatched passwords is not interesting.
Only emit it in VERBOSE mode when DCHECK is on.
Bug: webrtc:14578
Change-Id: Ie83080d88be6da24e7f2f79d7eb279087f84c2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288740
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38938}
https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
has a step saying
For each stream in streams, add stream.id to
[[AssociatedMediaStreamIds]] if it's not already there
This applies to addTrack and setStreams and the set of streams in
addTransceiver.
BUG=webrtc:14769
Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38937}
This CL removes a couple more opportunities for client code
to interact directly with the MediaChannel implementation classes.
No-try because of infra failure.
No-Try: true
Bug: webrtc:13931
Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38935}
