Move base64.h to the proper location and put redirect header into the
old place to be able to switch downstream users on new location.
Bug: webrtc:8366
Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291
Reviewed-on: https://webrtc-review.googlesource.com/88223
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24069}
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}
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 previously used "memset(ptr, 0, size)" can get optimized away by compilers
if "ptr" is not used afterwards.
A new class "ZeroOnFreeBuffer" is introduced that can hold sensitive data and
that automatically clears underlying memory when it's no longer used.
Bug: webrtc:8806, webrtc:8897, webrtc:8905
Change-Id: Iedddddf80790f9af0addaab3346ec5bff102917d
Reviewed-on: https://webrtc-review.googlesource.com/41941
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22334}
JsepTransportController process the entire SDP and handle the RTCP-mux,
SRTP setup, BUNDLE related logic internally. This will replace the current
TransportController.
JsepTransport2 is used by the JsepTransportController which processes the
transport part of SDP and owns the DtlsTransport created internally.
JsepTransport2 will replace JsepTransport and be renamed eventually.
Bug: webrtc:8587
Change-Id: Ib02dfa52fe9b7a5b8b132afcc8e4363eb8bd9cf4
Reviewed-on: https://webrtc-review.googlesource.com/48841
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22164}
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.
TBR=pthatcher@webrtc.org
Bug: None
Change-Id: If41c462dc3ddff664d0b70d249d760e2ca4c8ab3
Reviewed-on: https://webrtc-review.googlesource.com/23576
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20820}
This is the same thing we're doing for usrsctp. Before this CL, the
first SrtpSession to call SetKey would initialize libsrtp, and
ChannelManager's destructor would deinitialize it. This works for an
application that only uses one instance of ChannelManager simultaneously
(or one instance of PeerConnectionFactory), but not one that uses
multiple.
Now, libsrtp is effectively reference-counted, with the first
SrtpSession to take a reference initializing it, and the last to remove
its reference deinitializing it.
This issue was discovered recently due to a change that resulted in
using srtp_update. Without using that method, the issue went unnoticed;
maybe srtp_protect/srtp_unprotect don't require initialization?
Bug: webrtc:8388
Change-Id: If1329360f0b469e454810e62e9b5acfbd4cba100
Reviewed-on: https://webrtc-review.googlesource.com/9000
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20262}
The SrtpTransport takes the SRTP responsibilities from the BaseChannel
and SrtpFilter. SrtpTransport is now responsible for setting the crypto
keys, protecting and unprotecting the packets. SrtpTransport doesn't
know if the keys are from SDES or DTLS handshake.
BaseChannel is now only responsible setting the offer/answer for SDES
or extracting the key from DtlsTransport and configuring the
SrtpTransport.
SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
BUG=webrtc:7013
Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e
Reviewed-on: https://webrtc-review.googlesource.com/2560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19977}
Reason for revert:
This seems to be causing some video freezes. See https://bugs.chromium.org/p/webrtc/issues/detail?id=8251
Original issue's description:
> Completed the functionalities of SrtpTransport.
>
> The SrtpTransport takes the SRTP responsibilities from the BaseChannel
> and SrtpFilter. SrtpTransport is now responsible for setting the crypto
> keys, protecting and unprotecting the packets. SrtpTransport doesn't know
> if the keys are from SDES or DTLS handshake.
>
> BaseChannel is now only responsible setting the offer/answer for SDES
> or extracting the key from DtlsTransport and configuring the
> SrtpTransport.
>
> SrtpFilter is used by BaseChannel as a helper for SDES negotiation.
>
> BUG=webrtc:7013
>
> Review-Url: https://codereview.webrtc.org/2997983002
> Cr-Commit-Position: refs/heads/master@{#19636}
> Committed: e683c6871fTBR=deadbeef@webrtc.org,pthatcher@google.com,zhihuang@webrtc.org
Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/3018513002
Cr-Commit-Position: refs/heads/master@{#19895}
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}
