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}
The last mention was in a unit test, where speex was used to name an
arbitrary codec. The name "foo" is now used instead.
Bug: webrtc:4844
Change-Id: Ia1ede8512b894e6c16c0c168a50dc4d62d6911ad
Reviewed-on: https://webrtc-review.googlesource.com/74781
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23160}
This profile will now not be used unless the application explicitly
sets the flag in CryptoOptions to true. As a result, an 80-bit
authentication tag will be used instead of a 32-bit one. See bug for
more details.
Bug: webrtc:7670
Change-Id: I7c0a118fd7b1e7aac23b9eb8717099f055de0441
Reviewed-on: https://webrtc-review.googlesource.com/66600
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22757}
This change replaces the use of sync_label from StreamParams with
the new stream_labels() and set_stream_labels() getter and setter.
Bug: webrtc:7932
Change-Id: Ibd6d38f7d4efed37ac07963e6fbe377c93a28fd6
Reviewed-on: https://webrtc-review.googlesource.com/58540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22257}
This reverts commit 6780c51b23.
Reason for revert:
More details in crbug.com/810292
Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
>
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
>
> R=deadbeef@webrtc.org
>
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org
Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
from native apps if really necessary.
R=deadbeef@webrtc.org
Bug: webrtc:7670
Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
Reviewed-on: https://webrtc-review.googlesource.com/41420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21952}
Previously, if a media section is rejected by the answerer, the TransportInfo
of that section will not be added to the answer but when answer SDP is
deserialized by the offerer, the rejected TransportInfo will be added.
This CL fixes this inconsistency by adding the TransportInfo of all the m=
sections including the rejected ones.
Bug: webrtc:8818
Change-Id: I3b163245979c4ac4df31db39262e499f1c4901c4
Reviewed-on: https://webrtc-review.googlesource.com/46380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21815}
This changes all internal code to use the media_description() helper
for ContentInfo along with the as_audio, as_video, and as_data casting
methods introduced in a previous CL. Reduces the total number of
pointer static_casts in pc/ from 351 to 122.
Bug: webrtc:8620
Change-Id: I996f49b55f1501c758a9e5223e30539a9f8d4eac
Reviewed-on: https://webrtc-review.googlesource.com/35921
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21419}
This makes the following changes:
- Replaces ContentDescription with its only subclass,
MediaContentDescription
- Adds helpers to cast a MediaContentDescription to its
audio, video, and data subclasses.
- Changes ContentInfo.type to a new enum, MediaProtocolType.
Bug: webrtc:8620
Change-Id: I5eb0811cb16a51b0b9d73ecc4fe8edc7037f1aed
Reviewed-on: https://webrtc-review.googlesource.com/35100
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21401}
Replaces cricket::RtpTransceiverDirection with
webrtc::RtpTransceiverDirection, which is part of the public API.
Bug: webrtc:8558
Change-Id: Ibfc9373e25187e98fb969e7ac937a1371c8fa4c7
Reviewed-on: https://webrtc-review.googlesource.com/24129
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20899}
Previously when creating an answer, we used the codecs from the remote list
to get correctly mapped codec id, but the side effect is that we also used
the remote codec parameters which is wrong.
With this change, we use the local codecs with local parameters and the
|NegotiateCodecs| method will take care of the codec id mapping.
Bug: webrtc:8550
Change-Id: Ib7b2fe5f6ed3a72c58cf5df8bf2c1d00476c141c
Reviewed-on: https://webrtc-review.googlesource.com/25285
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20843}
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}
The first argument of FindMatchingCodec is supposed to be the list that
contains the codec to be found, specifically to handle RTX codecs that
point to other codecs. But this wasn't being done everywhere, and wasn't
noticed because *most of the time* it just results in adding the RTX
codec in a different location, which isn't an issue.
But, it's still not standards-compliant. And it sometimes is an issue
when talking to older endpoints.
Adding a regression test, and DCHECK in FindMatchingCodec to ensure this
doesn't happen by accident again.
Bug: webrtc:8332
Change-Id: I5def056b245c6d00a49a59d429f1dee303fb7cef
Reviewed-on: https://webrtc-review.googlesource.com/6240
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20130}
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}
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/mediasession_unittest.cc (Browse further)
