Commit graph

29 commits

Author SHA1 Message Date
Niels Möller
16e27a1dc5 Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
Original cl: https://webrtc-review.googlesource.com/95721

Bug: webrtc:9239
Change-Id: I7eac85839182bbcecd0d9bd71ae26f6a1c516df4
Reviewed-on: https://webrtc-review.googlesource.com/96401
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24529}
2018-09-03 09:00:01 +00:00
Niels Moller
ec4a060a55 Revert "Delete leftover includes and declarations for MediaConstraintsInterface"
This reverts commit a1e4ae2371.

Reason for revert: Breakage in downstream code still using constraints.

Original change's description:
> Delete leftover includes and declarations for MediaConstraintsInterface
> 
> Bug: webrtc:9239
> Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
> Reviewed-on: https://webrtc-review.googlesource.com/95721
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24442}

TBR=kwiberg@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: Idbef4c57a0d3b82e94a431c5407a86c9fcd4be41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9239
Reviewed-on: https://webrtc-review.googlesource.com/96160
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24444}
2018-08-27 11:26:42 +00:00
Niels Möller
a1e4ae2371 Delete leftover includes and declarations for MediaConstraintsInterface
Bug: webrtc:9239
Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
Reviewed-on: https://webrtc-review.googlesource.com/95721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24442}
2018-08-27 10:41:57 +00:00
Niels Möller
f06f923ef0 Delete almost all use of MediaConstraintsInterface in the PeerConnection API
Bug: webrtc:9239
Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b
Reviewed-on: https://webrtc-review.googlesource.com/74420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24396}
2018-08-23 07:14:37 +00:00
Jeroen de Borst
4f6d233dcc Added explicit EOR to sctp messages and coalesce messages on the receiving side.
TBR=pthatcher@webrtc.org

Bug: webrtc:7774
Change-Id: I41d1cd98d1e7b2ad479177eb2e328a5e2c704824
Reviewed-on: https://webrtc-review.googlesource.com/88900
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24031}
2018-07-19 01:26:59 +00:00
Mirko Bonadei
db90556972 Re-enable skipped test.
TBR=kwiberg@webrtc.org

Bug: webrtc:9442
Change-Id: I1cde15deac8202ce90c31578efd32f6cc4aabfca
Reviewed-on: https://webrtc-review.googlesource.com/88569
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23978}
2018-07-16 07:31:07 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Mirko Bonadei
e12c1fe8d9 Removing warning suppression flags from pc/.
Bug: webrtc:9251
Change-Id: Ic12126fc03309448fe71a17e6b65343949496f4f
Reviewed-on: https://webrtc-review.googlesource.com/86820
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23838}
2018-07-04 10:35:27 +00:00
Mirko Bonadei
776199a55a Enable PeerConnectionEndToEndTest.CallWithLegacySdp on ASan.
Bug: None
Change-Id: I9f695bd0a13b0130f4d773803e010b69020c2ac1
Reviewed-on: https://webrtc-review.googlesource.com/86131
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23777}
2018-06-29 05:00:46 +00:00
Mirko Bonadei
82d171c824 Skip PeerConnectionEndToEndTest.CallWithCustomCodec on Win ASan builds.
Bug: None
Change-Id: Iaee0bdee03e23aae916a641c6230e14ae229c6df
Reviewed-on: https://webrtc-review.googlesource.com/86130
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23776}
2018-06-29 04:57:36 +00:00
Mirko Bonadei
d5b8ee1e17 Re-enable PeerConnectionEndToEndTest.Call on TSan.
Bug: webrtc:4719
Change-Id: Ic24c0921892a45bd28cd91f8ce6bdd9593ef1d59
Reviewed-on: https://webrtc-review.googlesource.com/85281
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23729}
2018-06-25 15:36:02 +00:00
Danil Chapovalov
66cadcc6b9 Replace rtc::Optional with absl::optional in pc
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'pc'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 20:55:07 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Niels Möller
2d02e085de Delete deprecated CreateAudioSource method, with constraints.
Bug: webrtc:9239
Change-Id: I5025b7fd103247e0426ceabedc1216a4f0f0ab34
Reviewed-on: https://webrtc-review.googlesource.com/76560
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23501}
2018-06-04 08:19:30 +00:00
Taylor Brandstetter
cdd05f0cc1 Implement proper SCTP data channel closing procedure.
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!

https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7

However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.

This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.

This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.

TBR=magjed@webrtc.org

Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
2018-05-31 21:01:53 +00:00
Kári Tristan Helgason
983042b84a Re-enable test on mac now that issue is fixed.
Bug: webrtc:5231
Change-Id: I92bc4838c7c17f512caca994232916b9144e37c4
Reviewed-on: https://webrtc-review.googlesource.com/68522
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22853}
2018-04-13 11:29:28 +00:00
Karl Wiberg
5bdc82a60f Test for audio codec pair ID assignment
We want to ensure that encoders and decoders actually get IDs with the
desired properties.

Bug: webrtc:8941
Change-Id: Ie64b67c2e9cb67171725d27f92e954afd1b77834
Reviewed-on: https://webrtc-review.googlesource.com/63300
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22547}
2018-03-22 00:11:23 +00:00
Karl Wiberg
d6fbf2a4b1 Tests: Pass codec ID argument to audio codecs
Bug: webrtc:8941
Change-Id: Ia6d51dcbf7d69b38f3615e01d3f7031b8f5c31d0
Reviewed-on: https://webrtc-review.googlesource.com/58092
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22383}
2018-03-12 13:25:29 +00:00
Karl Wiberg
17668ec4a5 Audio codec implementations: Take optional codec ID argument
None of the built-in codecs do anything with the ID, but callers will
soon require them to accept it.

Bug: webrtc:8941
Change-Id: I0eb77db82d72c7d34cff639fecb67c1e6ec421bf
Reviewed-on: https://webrtc-review.googlesource.com/58089
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22259}
2018-03-01 19:28:38 +00:00
Steve Anton
36da6ff582 Parameterize PeerConnection interface tests for Unified Plan
Bug: webrtc:8765
Change-Id: I550164bc8c6cf133f7b72a22d86bd4a704a8c1d3
Reviewed-on: https://webrtc-review.googlesource.com/47242
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22065}
2018-02-17 00:07:39 +00:00
Steve Anton
191c39f307 Parameterize PeerConnection end to end tests for Unified Plan
Bug: webrtc:8765
Change-Id: If4b797be7876a7680e99c698631c29b412f7a455
Reviewed-on: https://webrtc-review.googlesource.com/41540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21755}
2018-01-25 04:03:18 +00:00
Oskar Sundbom
63e232a2cf Optional: Use nullopt and implicit construction in /pc/peerconnectionendtoend_unittest.cc
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=hbos@webrtc.org

Bug: None
Change-Id: I2fc923e788b744fc167f02f9fecd19bea512af81
Reviewed-on: https://webrtc-review.googlesource.com/23613
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20834}
2017-11-22 14:21:57 +00:00
Mirko Bonadei
c61ce0d0cd Fixing some clang-tidy findings.
Bug: None
Change-Id: I949c1ff35284ce79c99e8f76148f63b8bba965a9
Reviewed-on: https://webrtc-review.googlesource.com/24041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20818}
2017-11-21 16:43:07 +00:00
Steve Anton
36b29d1df3 Enable cpplint in pc/
Enable cpplint check in the PRESUBMIT for pc/ and fix all existing
warnings.

Bug: webrtc:5583
Change-Id: If39994692ab6f6f3c83c74f23850f02fdfe810e8
Reviewed-on: https://webrtc-review.googlesource.com/16540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20482}
2017-10-30 18:08:29 +00:00
Steve Anton
c4faa9c4e1 Remove QUIC transport/data channel
Originally, the idea was to implement QUIC data channels as a
PeerConnection API. Now, the effort has shifted to implementing it as a
part of ORTC which will live in Chromium. Since this code has not been
maintained and is not currently being used, remove it to reduce
maintenance overhead while a copy will be retained in the Git history.

Bug: webrtc:8385
Change-Id: I2719c007a0de0118b67d41a425f900b66c52f65a
Reviewed-on: https://webrtc-review.googlesource.com/14100
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20412}
2017-10-24 16:14:18 +00:00
Karl Wiberg
c5bb00b6bf PeerConnection end-to-end test with a non-builtin codec
BUG=webrtc:8159

Change-Id: I27f3fc8a6204115c480d70238225b0565c41bd81
Reviewed-on: https://webrtc-review.googlesource.com/6283
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20245}
2017-10-11 11:48:54 +00:00
Patrik Höglund
563934e726 Clean up dependencies of peerconnection_unittest.
There were a number of unused includes and undeclared
dependencies. I removed the includes that were causing
problems and added dependencies for the includes that
turned out to be needed.

Bug: webrtc:7239,webrtc:6828
Change-Id: I5b57f9b8411d969e96eaa46fb49101b7b7c32284
Reviewed-on: https://webrtc-review.googlesource.com/1185
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19858}
2017-09-15 12:51:00 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/pc/peerconnectionendtoend_unittest.cc (Browse further)