Update API for Objective-C RTCConfiguration.

webrtc/api/objctests/RTCConfigurationTest.mm

Index: webrtc/api/objctests/RTCConfigurationTest.mm
diff --git a/webrtc/api/objctests/RTCConfigurationTest.mm b/webrtc/api/objctests/RTCConfigurationTest.mm
new file mode 100644
index 0000000000000000000000000000000000000000..565cf24d2d61f11056ec7d77b2dc4d082ca6c143
--- /dev/null
+++ b/webrtc/api/objctests/RTCConfigurationTest.mm
@@ -0,0 +1,118 @@
+/*
+ *  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];
+
+  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig =
+      config.nativeConfiguration;
+  EXPECT_EQ((size_t)1u, nativeConfig.servers.size());
+  webrtc::PeerConnectionInterface::IceServer nativeServer =
+      nativeConfig.servers.front();
+  EXPECT_EQ((size_t)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((size_t)1, config.iceServers.count);
+  RTCIceServer *server = config.iceServers.firstObject;
+  EXPECT_EQ((size_t)1, 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);
+}
+
+@end
+
+TEST(RTCConfigurationTest, NativeConfigurationConversionTest) {
+  @autoreleasepool {
+    RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
+    [test testConversionToNativeConfiguration];
+  }
+}
+
+TEST(RTCConfigurationTest, InitFromConfigurationTest) {
+  @autoreleasepool {
+    RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
+    [test testInitFromNativeConfiguration];
+  }
+}