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Remove SignalSSLHandshakeError signal from SSLStreamAdapter.

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Also removing has_slots depdency from OpenSSLStreamAdapter and moving
it to the  OpenSSLStreamAdapter subclass where it's still needed.

Bug: webrtc:11943
Change-Id: Ibcae5ea1efff146d78b32bb0eca63d7f44ed08c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318885
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40702}
This commit is contained in:
Tommi 2023-09-05 13:16:47 +02:00 committed by WebRTC LUCI CQ
parent 7b6faa1243
commit 48df56e9ac
3 changed files with 4 additions and 10 deletions
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4
rtc_base/openssl_stream_adapter.cc
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@ -942,10 +942,6 @@ int OpenSSLStreamAdapter::ContinueSSL() {
<< ERR_GET_REASON(err_code); << ERR_GET_REASON(err_code);
if (handshake_error_) { if (handshake_error_) {
handshake_error_(ssl_handshake_err); handshake_error_(ssl_handshake_err);
} else {
// TODO(bugs.webrtc.org/11943): SignalSSLHandshakeError usage has been
// deprecated. Remove once external usage has been cleaned up.
SignalSSLHandshakeError(ssl_handshake_err);
} }
return (ssl_error != 0) ? ssl_error : -1; return (ssl_error != 0) ? ssl_error : -1;
} }

4
rtc_base/openssl_stream_adapter.h
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@ -34,6 +34,7 @@
#include "rtc_base/stream.h" #include "rtc_base/stream.h"
#include "rtc_base/system/rtc_export.h" #include "rtc_base/system/rtc_export.h"
#include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace rtc { namespace rtc {
@ -71,7 +72,8 @@ class SSLCertChain;
// configuration is restored. // configuration is restored.
RTC_EXPORT void SetAllowLegacyTLSProtocols(const absl::optional<bool>& allow); RTC_EXPORT void SetAllowLegacyTLSProtocols(const absl::optional<bool>& allow);
class OpenSSLStreamAdapter final : public SSLStreamAdapter { class OpenSSLStreamAdapter final : public SSLStreamAdapter,
public sigslot::has_slots<> {
public: public:
OpenSSLStreamAdapter( OpenSSLStreamAdapter(
std::unique_ptr<StreamInterface> stream, std::unique_ptr<StreamInterface> stream,

6
rtc_base/ssl_stream_adapter.h
View File

@ -24,7 +24,6 @@
#include "rtc_base/ssl_certificate.h" #include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_identity.h" #include "rtc_base/ssl_identity.h"
#include "rtc_base/stream.h" #include "rtc_base/stream.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace rtc { namespace rtc {
@ -113,7 +112,7 @@ enum { SSE_MSG_TRUNC = 0xff0001 };
// Used to send back UMA histogram value. Logged when Dtls handshake fails. // Used to send back UMA histogram value. Logged when Dtls handshake fails.
enum class SSLHandshakeError { UNKNOWN, INCOMPATIBLE_CIPHERSUITE, MAX_VALUE }; enum class SSLHandshakeError { UNKNOWN, INCOMPATIBLE_CIPHERSUITE, MAX_VALUE };
class SSLStreamAdapter : public StreamInterface, public sigslot::has_slots<> { class SSLStreamAdapter : public StreamInterface {
public: public:
// Instantiate an SSLStreamAdapter wrapping the given stream, // Instantiate an SSLStreamAdapter wrapping the given stream,
// (using the selected implementation for the platform). // (using the selected implementation for the platform).
@ -263,9 +262,6 @@ class SSLStreamAdapter : public StreamInterface, public sigslot::has_slots<> {
// authentication. // authentication.
bool GetClientAuthEnabled() const { return client_auth_enabled_; } bool GetClientAuthEnabled() const { return client_auth_enabled_; }
// TODO(bugs.webrtc.org/11943): Remove after updating downstream code.
sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError;
private: private:
// If true (default), the client is required to provide a certificate during // If true (default), the client is required to provide a certificate during
// handshake. If no certificate is given, handshake fails. This applies to // handshake. If no certificate is given, handshake fails. This applies to