Make ECDSA default for RTCPeerConnection

webrtc/api/objctests/RTCConfigurationTest.mm

 /*
  *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #import <Foundation/Foundation.h>
 
 #include <vector>
 
 #include "webrtc/base/gunit.h"
 
 #import "webrtc/api/objc/RTCConfiguration.h"
 #import "webrtc/api/objc/RTCConfiguration+Private.h"
 #import "webrtc/api/objc/RTCIceServer.h"
 #import "webrtc/base/objc/NSString+StdString.h"
 
 @interface RTCConfigurationTest : NSObject
 - (void)testConversionToNativeConfiguration;
-- (void)testInitFromNativeConfiguration;
 @end
 
 @implementation RTCConfigurationTest
 
 - (void)testConversionToNativeConfiguration {
   NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
   RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
 
-  RTCConfiguration *config =
-      [[RTCConfiguration alloc] initWithIceServers:@[ server ]
-                                iceTransportPolicy:RTCIceTransportPolicyRelay
-                                      bundlePolicy:RTCBundlePolicyMaxBundle
-                                     rtcpMuxPolicy:RTCRtcpMuxPolicyNegotiate
-                                tcpCandidatePolicy:RTCTcpCandidatePolicyDisabled
-                       audioJitterBufferMaxPackets:60
-                     iceConnectionReceivingTimeout:1
-                iceBackupCandidatePairPingInterval:2];
+  RTCConfiguration *config = [[RTCConfiguration alloc] init];
+  config.iceServers = @[ server ];
+  config.iceTransportPolicy = RTCIceTransportPolicyRelay;
+  config.bundlePolicy = RTCBundlePolicyMaxBundle;
+  config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
+  config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
+  const int maxPackets = 60;
+  const int timeout = 1;
+  const int interval = 2;
+  config.audioJitterBufferMaxPackets = maxPackets;
+  config.iceConnectionReceivingTimeout = timeout;
+  config.iceBackupCandidatePairPingInterval = interval;
 
   webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig =
       config.nativeConfiguration;
   EXPECT_EQ(1u, nativeConfig.servers.size());
   webrtc::PeerConnectionInterface::IceServer nativeServer =
       nativeConfig.servers.front();
   EXPECT_EQ(1u, nativeServer.urls.size());
   EXPECT_EQ("stun:stun1.example.net", nativeServer.urls.front());
 
   EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig.type);
   EXPECT_EQ(webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle,
             nativeConfig.bundle_policy);
   EXPECT_EQ(webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate,
             nativeConfig.rtcp_mux_policy);
   EXPECT_EQ(webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled,
             nativeConfig.tcp_candidate_policy);
-  EXPECT_EQ(60, nativeConfig.audio_jitter_buffer_max_packets);
-  EXPECT_EQ(1, nativeConfig.ice_connection_receiving_timeout);
-  EXPECT_EQ(2, nativeConfig.ice_backup_candidate_pair_ping_interval);
-}
-
-- (void)testInitFromNativeConfiguration {
-  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
-
-  webrtc::PeerConnectionInterface::IceServer nativeServer;
-  nativeServer.username = "username";
-  nativeServer.password = "password";
-  nativeServer.urls.push_back("stun:stun.example.net");
-  webrtc::PeerConnectionInterface::IceServers servers { nativeServer };
-
-  nativeConfig.servers = servers;
-  nativeConfig.type = webrtc::PeerConnectionInterface::kNoHost;
-  nativeConfig.bundle_policy =
-      webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
-  nativeConfig.rtcp_mux_policy =
-      webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
-  nativeConfig.tcp_candidate_policy =
-      webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
-  nativeConfig.audio_jitter_buffer_max_packets = 40;
-  nativeConfig.ice_connection_receiving_timeout =
-      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined;
-  nativeConfig.ice_backup_candidate_pair_ping_interval =
-      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined;
-
-  RTCConfiguration *config =
-      [[RTCConfiguration alloc] initWithNativeConfiguration:nativeConfig];
-
-  EXPECT_EQ(1u, config.iceServers.count);
-  RTCIceServer *server = config.iceServers.firstObject;
-  EXPECT_EQ(1u, server.urlStrings.count);
-  EXPECT_TRUE([@"stun:stun.example.net" isEqualToString:
-      server.urlStrings.firstObject]);
-
-  EXPECT_EQ(RTCIceTransportPolicyNoHost, config.iceTransportPolicy);
-  EXPECT_EQ(RTCBundlePolicyMaxCompat, config.bundlePolicy);
-  EXPECT_EQ(RTCRtcpMuxPolicyRequire, config.rtcpMuxPolicy);
-  EXPECT_EQ(RTCTcpCandidatePolicyEnabled, config.tcpCandidatePolicy);
-  EXPECT_EQ(40, config.audioJitterBufferMaxPackets);
-  EXPECT_EQ(-1, config.iceConnectionReceivingTimeout);
-  EXPECT_EQ(-1, config.iceBackupCandidatePairPingInterval);
+  EXPECT_EQ(maxPackets, nativeConfig.audio_jitter_buffer_max_packets);
+  EXPECT_EQ(timeout, nativeConfig.ice_connection_receiving_timeout);
+  EXPECT_EQ(interval, nativeConfig.ice_backup_candidate_pair_ping_interval);
 }
 
 @end
 
 TEST(RTCConfigurationTest, NativeConfigurationConversionTest) {
   @autoreleasepool {
     RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
     [test testConversionToNativeConfiguration];
   }
 }
 
-TEST(RTCConfigurationTest, InitFromConfigurationTest) {
-  @autoreleasepool {
-    RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
-    [test testInitFromNativeConfiguration];
-  }
-}