Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328.
Reason for revert: Breaks downstream project
Original change's description:
> Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
>
> Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
> that only handles SRTP configuration to a more generic structure that can be
> used and extended for all per peer connection CryptoOptions that can be on a
> given PeerConnection.
>
> Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
> accessed as crypto_options.srtp.whatever_option_name. This is more inline with
> other structures we have in WebRTC such as VideoConfig. As additional features
> are added over time this will allow the structure to remain compartmentalized
> and concerned components can only request a subset of the overall configuration
> structure e.g:
>
> void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
>
> In addition to this it made little sense for sslstreamadapter.h to hold all
> Srtp related configuration options. The header has become loo large and takes on
> too many responsibilities and spilting this up will lead to more maintainable
> code going forward.
>
> This will be used in a future CL to enable configuration options for the newly
> supported Frame Crypto.
>
> Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
> Bug: webrtc:9681
> Reviewed-on: https://webrtc-review.googlesource.com/c/105180
> Reviewed-by: Emad Omara <emadomara@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25130}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
Bug: webrtc:9681
Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff
Reviewed-on: https://webrtc-review.googlesource.com/c/105541
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25133}diff --git a/api/peerconnectioninterface.h b/api/peerconnectioninterface.h
index 141b2c9..3d8a9c1 100644
--- a/api/peerconnectioninterface.h
+++ b/api/peerconnectioninterface.h
@@ -77,7 +77,6 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/call/callfactoryinterface.h"
-#include "api/crypto/cryptooptions.h"
#include "api/datachannelinterface.h"
#include "api/fec_controller.h"
#include "api/jsep.h"
@@ -1181,7 +1180,7 @@
public:
class Options {
public:
- Options() {}
+ Options() : crypto_options(rtc::CryptoOptions::NoGcm()) {}
// If set to true, created PeerConnections won't enforce any SRTP
// requirement, allowing unsecured media. Should only be used for
@@ -1210,7 +1209,7 @@
rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
// Sets crypto related options, e.g. enabled cipher suites.
- CryptoOptions crypto_options = CryptoOptions::NoGcm();
+ rtc::CryptoOptions crypto_options;
};
// Set the options to be used for subsequently created PeerConnections.