Commit Graph

1655 Commits

Author SHA1 Message Date
1859dc04fd Revert "Refactoring DataContentDescription class"
This reverts commit 8a9193c217d818fea77b9540bd4ca7ebad53db76.

Reason for revert: Breaks downstreams

Original change's description:
> Refactoring DataContentDescription class
> 
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::DataContentDescription (used for RTP data) and
> cricket::SctpDataContentDescription (used for SCTP only).
> 
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
> 
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> 
> Bug: webrtc:10358
> Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27651}

TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org

Change-Id: I3b8a68cd481c41ce30eeb5ffbc5da735a9659019
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133360
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27652}
2019-04-16 20:56:06 +00:00
8a9193c217 Refactoring DataContentDescription class
This CL splits the cricket::DataContentDescription class into
two classes: cricket::DataContentDescription (used for RTP data) and
cricket::SctpDataContentDescription (used for SCTP only).

SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".

Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#

Bug: webrtc:10358
Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27651}
2019-04-16 20:34:34 +00:00
70c2db1aa0 Reland "Make negotiationneeded processing in PeerConnection spec compliant."
The new processing applies only in Unified Plan mode.
Plan B retains the old-style processing.

This is a reland of 1fa06041bcd8a0119e557d16e7b54a9110c5ad03

Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}

Bug: chromium:740501
Change-Id: I048ae81b2b00086f6d669e94eecf426f0db0ec08
TBR: steveanton@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133162
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27640}
2019-04-16 12:04:33 +00:00
6cf61f53ad Delete unneeded includes of async_invoker.h
Bug: None
Change-Id: I3753592f8eb53eb2b31cf645b80c446bd2251404
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133027
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27636}
2019-04-16 07:28:06 +00:00
884adca3a0 Log details when RtpDemuxer fails to deliver a packet
Bug: None
Change-Id: Ie9dc5c3c545073d2e43b464d2585cb945eb346fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131360
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27634}
2019-04-16 00:47:53 +00:00
6df49d8983 Fixing issue with creating StreamParams when track id is not signaled.
Current logic requires a stream id and track ids when creating a stream
that does not have SSRCs signaled.
This change removes the requirement for stream ids. The requirement for
track id is softer, as one should be generated when it is not present.

Bug: webrtc:10551
Change-Id: Ibc0cc181c6b18efa8394b6c0e4820e3a13da70c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133080
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27633}
2019-04-16 00:34:04 +00:00
01738c63aa Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp.
This collects this metric for both audio and video streams.
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp

This is a follow-up to https://webrtc-review.googlesource.com/c/src/+/130479
which calculated this metric. This CL is purely plumbing from
"StreamDataCounters::last_packet_received_timestamp_ms" to
RTCInboundRtpStreamStats.


Bug: webrtc:10449
Change-Id: I757ad19b5b8e84553da5edd4a75efa3e1fe30b56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131397
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27628}
2019-04-15 16:06:01 +00:00
0006a625b1 Remove HKDF implementation from WebRTC.
We no longer have a need for a HKDF implementation in WebRTC. To keep
code quality high it makes sense to delete this dead code path.

Bug: webrtc:9600
Change-Id: Ibe6ee9150acd9dbf59452372242d857c5ffa65c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132802
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27604}
2019-04-13 00:53:44 +00:00
668a42b84f Revert "Make negotiationneeded processing in PeerConnection spec compliant."
This reverts commit 1fa06041bcd8a0119e557d16e7b54a9110c5ad03.

Reason for revert: Likely cause for breaking downstream projects

Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
> 
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
> 
> 
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}

TBR=steveanton@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,hbos@webrtc.org,guidou@webrtc.org

Change-Id: Iad7b7d4e37227fa6a76ff830160ca3da9dbe4719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132761
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27599}
2019-04-12 16:14:07 +00:00
1fa06041bc Make negotiationneeded processing in PeerConnection spec compliant.
This CL fixes the problem of misfired negotiationneeded notifications due
to the lack of a NegotiationNeeded slot and the proper procedure to
update it.


Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27594}
2019-04-12 13:58:33 +00:00
114871b1ba Expose SSL ciphersuite on the webrtc::DtlsTransport interface
This allows the ciphersuite to be accessed from Blink code, which is
useful if we want to emit deprecation messages.

Bug: none
Change-Id: Idcf1f5402948406e4cf7c6e54b67a622a1a403ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132004
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27581}
2019-04-12 08:29:59 +00:00
4d7160e41d Code cleanup: Make JsepSessionDescription::Initialize take std::unique_ptr
Bug: none
Change-Id: I5e03ff6a6f16bd2e67fa4830e218b510b702d1d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132321
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27577}
2019-04-12 05:42:46 +00:00
7061e51b48 Expose DtlsTransport::remote_ssl_certificates
Bug: chromium:907849
Change-Id: If990d541099edb9a327230e1d78a03b406269885
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131951
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27558}
2019-04-11 09:47:33 +00:00
48cce4d9e8 Parse "max-message-size" parameter from SCTP SDP description
Bug: chromium:943975
Change-Id: I559093cfa3686f2a388b872774df8f0737963281
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132224
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27555}
2019-04-11 08:44:44 +00:00
0810a7c25a Add base class NetworkPredictor and NetworkPredictorFactory and wire up.
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.

Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
2019-04-10 12:38:58 +00:00
f4770401dc Reland "Adding a restriction for legal RID values."
This is a reland of 07f3279a730980583403b78c3762c5d246d1d9be

Original change's description:
> Adding a restriction for legal RID values.
>
> According to the spec, RID values should be constrained to only
> alpha-numeric values. This was not enforced in our implementation to
> allow for more flexibility.
> It has been brought to our attention that some values that we currently
> consider legal (such as the '~', '=' ';' characters) might cause confusion
> with the simulcast syntax that uses these characters to indicate other
> meanings.
> What's worse, is that some characters, when used in RIDs (such as
> \u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
> users which might also be a health hazard.
> This change resolves these issues by restricting RIDs to alpha-numeric.
>
> Bug: webrtc:10491
> Change-Id: I16e262c87525d0289764beacd098e1525a355463
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27499}

TBR=steveanton@webrtc.org

Bug: webrtc:10491
Change-Id: I856581306a9258480ee9184f12b55c2a23dd8636
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131983
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27530}
2019-04-09 21:19:31 +00:00
6a489f22c7 Fully qualify googletest symbols.
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}
2019-04-09 17:18:20 +00:00
a857698d54 Revert "Adding a restriction for legal RID values."
This reverts commit 07f3279a730980583403b78c3762c5d246d1d9be.

Reason for revert: Suspect of producing consistent failure in some Chrome trybots, blocking rolls.

Failed test:
external/wpt/webrtc/RTCPeerConnection-addTransceiver.https.html


First failure:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/64597

Original change's description:
> Adding a restriction for legal RID values.
> 
> According to the spec, RID values should be constrained to only
> alpha-numeric values. This was not enforced in our implementation to
> allow for more flexibility.
> It has been brought to our attention that some values that we currently
> consider legal (such as the '~', '=' ';' characters) might cause confusion
> with the simulcast syntax that uses these characters to indicate other
> meanings.
> What's worse, is that some characters, when used in RIDs (such as
> \u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
> users which might also be a health hazard.
> This change resolves these issues by restricting RIDs to alpha-numeric.
> 
> Bug: webrtc:10491
> Change-Id: I16e262c87525d0289764beacd098e1525a355463
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27499}

TBR=steveanton@webrtc.org,amithi@webrtc.org

Change-Id: I89f9d8a8d3fa82de8a7d429f11ad7cc30812ba7c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132244
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27525}
2019-04-09 17:12:33 +00:00
2e06926c95 Implement RTC[In/Out]boundRtpStreamStats.contentType.
Spec: https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype

This already exists as a goog-stat. This CL only plumbs the value to the
new stats collector.

Note: There is currently no distinction between the extension being
missing and it being present but the value being "unspecified". Until
https://crbug.com/webrtc/10529 is fixed, this metric is only exposed if
SCREENSHARE was present.

Bug: webrtc:10452
Change-Id: Ic8723f4d0efb43ab72a560e954676facd3b90659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27520}
2019-04-09 13:02:03 +00:00
88718fdce7 Fix comment - ice transport is cleared on the networking thread.
Bug: chromium:907849
Change-Id: I15afe28d3fb2a175a6e2e7491dc33d36f9f9cd32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132008
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27509}
2019-04-09 09:12:10 +00:00
f3736ed3d8 Datachannel: Use absl::optional for maxRetransmits and maxRetransmitTime.
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.

Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.

Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
2019-04-09 08:32:43 +00:00
f73f7d684c Add thread safety annotations for some more PeerConnection members (part 13)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: Ib0814a02bd277005c8f4c1848421b70f847b5549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131339
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27505}
2019-04-09 08:16:20 +00:00
f71362f0cf Wire up RTCOutboundRtpStreamStats.totalEncodeTime.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/130517 that calculated
this metric.

This CL is purely plumbing, exposing
VideoSendStream::total_encode_time_ms in standard getStats() as
RTCOutboundRtpStreamStats.totalEncodeTime (in seconds):
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime

Bug: webrtc:10448
Change-Id: I715f1ef937e441169dee55b5e8d4fbf98811c5f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131940
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27501}
2019-04-09 07:34:38 +00:00
07f3279a73 Adding a restriction for legal RID values.
According to the spec, RID values should be constrained to only
alpha-numeric values. This was not enforced in our implementation to
allow for more flexibility.
It has been brought to our attention that some values that we currently
consider legal (such as the '~', '=' ';' characters) might cause confusion
with the simulcast syntax that uses these characters to indicate other
meanings.
What's worse, is that some characters, when used in RIDs (such as
\u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
users which might also be a health hazard.
This change resolves these issues by restricting RIDs to alpha-numeric.

Bug: webrtc:10491
Change-Id: I16e262c87525d0289764beacd098e1525a355463
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27499}
2019-04-08 22:41:24 +00:00
c01367db40 Deprecating ThreadChecker specific interface.
All changes outside thread_checker.h are by:
s/CalledOnValidThread/IsCurrent/
s/DetachFromThread/Detach/

Bug: webrtc:9883
Change-Id: Idbb1086bff0817db58e770116acf4c9d60fae8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27494}
2019-04-08 16:58:07 +00:00
1e2d436cbf Change PlayoutLatency setLatency zero-threshold value.
This is needed to match behaviour described in this spec:
https://github.com/henbos/webrtc-timing/pull/2

Bug: webrtc:10287
Change-Id: Idce9af2ec63705dfbfb500b7dbbf755ed3eab571
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131336
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27480}
2019-04-08 10:27:06 +00:00
4bac79ece2 Add SetJitterBufferMinimumDelay method to RtpReceiverInterface.
This change is required to allow modification of Jitter Buffer delay
in javascript via Origin Trial Experiment.
Link to experiment description:
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/Tgm4qiNepJc

Bug: webrtc:10287
Change-Id: I4f21380aad5982a4a60c55683b5173ce72ce0392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131144
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27444}
2019-04-04 09:00:16 +00:00
739506e45e Add thread safety annotations for some more PeerConnection members (part 12)
Plus all the annotations that were necessary to make things compile
again. I also had to send copies of some values owned by the signal
thread to the network thread, instead of letting the latter read them
itself.

Bug: webrtc:9987
Change-Id: Ic4b38696245584bab44956e60ac63753146e3ff4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131020
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27437}
2019-04-03 10:28:54 +00:00
78a5e96001 Reland "Add thread guards to JsepTransport"
This reverts commit caedb5db82b2bc8273910f4a0d1afb1d0e2994f3.

Reason for revert: Fixed issue (allow SetNeedsIceRestart from off-thread).

Original change's description:
> Revert "Add thread guards to JsepTransport"
>
> This reverts commit 7e1db52c93c57a180073906eda6a58919a9fd537.
>
> Reason for revert: Breaks downstream.
>
> Original change's description:
> > Add thread guards to JsepTransport
> >
> > This ensures that JsepTransport's methods are either only accessed on the thread
> > that creates it, or using methods that are marked for off-thread use
> > (using a lock to prevent simultaneous access).
> >
> > The intent is to document the existing contract, and to make it easy to find the
> > actions needed to convert the class to a pure single-threaded class.
> >
> > Bug: webrtc:10300
> > Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27427}
>
> TBR=steveanton@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
>
> Change-Id: I30c65d2161de9376ccd1172e2b261f2280fb1d75
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10300
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130519
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27429}

Change-Id: Ic32bfc04d96e657fc67c3d3999f77969e55ed994
Bug: webrtc:10300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130962
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27434}
2019-04-03 09:41:38 +00:00
d9bf720c20 Add thread safety annotations for some more PeerConnection members (part 11)
After reviewer feedback, this CL was reduced to just adding scary
comments on two variables.

Bug: webrtc:9987
Change-Id: Id1e251ffd02e4ca8050235bd9f3971b5363f0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27431}
2019-04-03 09:05:41 +00:00
caedb5db82 Revert "Add thread guards to JsepTransport"
This reverts commit 7e1db52c93c57a180073906eda6a58919a9fd537.

Reason for revert: Breaks downstream.

Original change's description:
> Add thread guards to JsepTransport
> 
> This ensures that JsepTransport's methods are either only accessed on the thread
> that creates it, or using methods that are marked for off-thread use
> (using a lock to prevent simultaneous access).
> 
> The intent is to document the existing contract, and to make it easy to find the
> actions needed to convert the class to a pure single-threaded class.
> 
> Bug: webrtc:10300
> Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27427}

TBR=steveanton@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org

Change-Id: I30c65d2161de9376ccd1172e2b261f2280fb1d75
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130519
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27429}
2019-04-03 07:15:38 +00:00
7e1db52c93 Add thread guards to JsepTransport
This ensures that JsepTransport's methods are either only accessed on the thread
that creates it, or using methods that are marked for off-thread use
(using a lock to prevent simultaneous access).

The intent is to document the existing contract, and to make it easy to find the
actions needed to convert the class to a pure single-threaded class.

Bug: webrtc:10300
Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27427}
2019-04-02 21:49:36 +00:00
9435c61021 Expose TaskQueueFactory for webrtc::Call in peer connection api
making a step for GlobalTaskQueueFactory to be optional way
to provide TaskQueueFactory

Bug: webrtc:10284
Change-Id: Ife838b3691c256820973118bc5b3cb372dea09cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130488
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27423}
2019-04-02 18:46:54 +00:00
7a651c6e58 Add thread safety annotations for some more PeerConnection members (part 10)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I2b08c7db10dda7b18ad4ba036125f2a56ebf80a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130478
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27419}
2019-04-02 17:40:37 +00:00
2cc368fd7a Add thread safety annotations for some more PeerConnection members (part 9)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: Ie958f4d86319e86527567ca1273a0595ccceee17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130490
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27411}
2019-04-02 10:48:16 +00:00
66e7679fb8 Export symbols needed by the Chromium component build (part 8).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 10:13:36 +00:00
5c4d2ee615 RTCDataChannel: Ignore "id" when "negotiated" is false
This updates behavior to be aligned with the WebRTC spec.

Bug: chromium:948055
Change-Id: Id3bbf05b3df084c9b7f7d12598c09187679d60fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130493
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27394}
2019-04-01 14:06:29 +00:00
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
1f928d3316 Close data channels when ID assignment fails.
This prevents crashes due to unassigned IDs.

Bug: chromium:945256
Change-Id: I63f3a17cc7dff07dab58a6bc59fe3606b23e8e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129902
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27349}
2019-03-28 17:34:07 +00:00
e46f5db8bf Add missing using declarations for names in testing namespace.
This code was unnecessarily depending on ADL
(https://abseil.io/tips/49).

Bug: None
Change-Id: I4f130fbd46bf3c7cc3b4313c9c85f1ac9dc64cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129764
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27340}
2019-03-28 13:20:00 +00:00
efe4c92d54 Use RtpSender/RtpReceiver track ID for legacy GetStats
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.

This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
   the unsigneled RtpReceiver track ID for both Plan B and Unified
   Plan.
2) Removes a couple methods on PeerConnection that were only used by
   the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
   the code easier to understand.

Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
2019-03-27 18:14:00 +00:00
b9bb3718e3 Revert "Don't crash when a datachannel can't get an ID"
This reverts commit 77c442ca1946924b0acfc9c0ba469ef6a2c3178f.

Reason for revert: Breakage on internal build tests, random breakage elsewhere. Timing issues suspected.

Original change's description:
> Don't crash when a datachannel can't get an ID
> 
> This is exercised by WPT test RTCDataChannel-id.
> 
> Bug: chromium:945256
> Change-Id: I53781dc874134f8c68a49c201848377b93b8858f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128871
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27300}

TBR=hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org

Change-Id: Ib7eaf8880e8ce21226b84b3b2283be93acb8dc8b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:945256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129766
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27302}
2019-03-27 07:23:41 +00:00
77c442ca19 Don't crash when a datachannel can't get an ID
This is exercised by WPT test RTCDataChannel-id.

Bug: chromium:945256
Change-Id: I53781dc874134f8c68a49c201848377b93b8858f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128871
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27300}
2019-03-26 22:06:24 +00:00
07122bc87e Use TaskQueueForTest instead or TaskQueue in unittests
To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue

Bug: webrtc:10284
Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:42:49 +00:00
a58e169269 Add thread safety annotations for some more PeerConnection members (part 8)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I452c17f52302fb28d37d9b570ef3b7ab3d023f77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129443
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27289}
2019-03-26 13:17:19 +00:00
ac025898e1 Fix misunderstanding: OnTransportChanged is called on network thread
Earlier CLs assumed that the object pointed to by call_ had to be
accessed on the worker thread. While this is generally the case,
Call::MediaTransportChange is explicitly thread safe, so
PeerConnection::OnTransportChanged doesn't have to run on the worker
thread for that reason.

Which is fortunate, because it actually runs on the network thread.
The RTC_RUN_ON(worker_thread()) annotation on the method declaration
was ineffective because this method is being called via a base class
pointer; replacing it with a call to
RTC_DCHECK_RUN_ON(worker_thread()) in the function body immediately
triggered assertions in the unit tests.

Bug: webrtc:9987
Change-Id: I08cf558a74f4ca2b2eff8ef4810ebbd1287a9726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129442
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27287}
2019-03-26 12:51:34 +00:00
12ba3adcaf Move unique_ptr into task instead of using a raw pointer
The raw pointer would have leaked if the task was ever destroyed
without being run.

Bug: webrtc:9987
Change-Id: Iddeb1adf0f836b8fec3056eab89bce7b9f034ca7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128865
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27284}
2019-03-26 11:28:25 +00:00
6cab5c8718 Add thread safety annotations for some more PeerConnection members (part 5)
Plus all the annotations that were necessary to make things compile
again.

We needed a special twist for call_. The value it points to is owned
by the worker thread, but the signal thread needs to read the pointer.
We could have made the pointer const, except that we explicitly reset
it in the destructor (in an invoke to the worker thread).

Bug: webrtc:9987
Change-Id: I31f024547f4be0e50967133b0d452c80ae38d7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128863
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27278}
2019-03-26 09:35:20 +00:00
0cd95081c0 Delete last traces of RtpTransportAdapter
Bug: None
Change-Id: I6bfac26ebd924c83f2f8a3adae983feb7d5bf00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128770
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27258}
2019-03-25 08:52:45 +00:00
85a4a93e77 Add thread safety annotations for some more PeerConnection members (part 4)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: Ia421e4dc0e1bbc81c3976cc7530d44de934d33bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128882
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27257}
2019-03-23 06:54:31 +00:00