[iOS] Added an initialization method to RTCConfiguration that takes a

native configuration.
Added a getConfiguration getter method to RTCPeerConnection to return
the RTCConfiguration.

BUG=webrtc:7431

Review-Url: https://codereview.webrtc.org/2790833002
Cr-Commit-Position: refs/heads/master@{#17517}

webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm

@@ -21,6 +21,7 @@
 
 @interface RTCConfigurationTest : NSObject
 - (void)testConversionToNativeConfiguration;
+- (void)testNativeConversionToConfiguration;
 @end
 
 @implementation RTCConfigurationTest
@@ -74,12 +75,60 @@
   EXPECT_EQ(true, nativeConfig->prune_turn_ports);
 }
 
+- (void)testNativeConversionToConfiguration {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
+
+  RTCConfiguration *config = [[RTCConfiguration alloc] init];
+  config.iceServers = @[ server ];
+  config.iceTransportPolicy = RTCIceTransportPolicyRelay;
+  config.bundlePolicy = RTCBundlePolicyMaxBundle;
+  config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
+  config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
+  config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
+  const int maxPackets = 60;
+  const int timeout = 1;
+  const int interval = 2;
+  config.audioJitterBufferMaxPackets = maxPackets;
+  config.audioJitterBufferFastAccelerate = YES;
+  config.iceConnectionReceivingTimeout = timeout;
+  config.iceBackupCandidatePairPingInterval = interval;
+  config.continualGatheringPolicy =
+      RTCContinualGatheringPolicyGatherContinually;
+  config.shouldPruneTurnPorts = YES;
+
+  webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig =
+      [config createNativeConfiguration];
+  RTCConfiguration *newConfig = [[RTCConfiguration alloc] initWithNativeConfiguration:nativeConfig];
+  EXPECT_EQ([config.iceServers count], newConfig.iceServers.count);
+  RTCIceServer *newServer = newConfig.iceServers[0];
+  RTCIceServer *origServer = config.iceServers[0];
+  EXPECT_EQ(origServer.urlStrings.count, server.urlStrings.count);
+  std::string origUrl = origServer.urlStrings.firstObject.UTF8String;
+  std::string url = newServer.urlStrings.firstObject.UTF8String;
+  EXPECT_EQ(origUrl, url);
+
+  EXPECT_EQ(config.iceTransportPolicy, newConfig.iceTransportPolicy);
+  EXPECT_EQ(config.bundlePolicy, newConfig.bundlePolicy);
+  EXPECT_EQ(config.rtcpMuxPolicy, newConfig.rtcpMuxPolicy);
+  EXPECT_EQ(config.tcpCandidatePolicy, newConfig.tcpCandidatePolicy);
+  EXPECT_EQ(config.candidateNetworkPolicy, newConfig.candidateNetworkPolicy);
+  EXPECT_EQ(config.audioJitterBufferMaxPackets, newConfig.audioJitterBufferMaxPackets);
+  EXPECT_EQ(config.audioJitterBufferFastAccelerate, newConfig.audioJitterBufferFastAccelerate);
+  EXPECT_EQ(config.iceConnectionReceivingTimeout, newConfig.iceConnectionReceivingTimeout);
+  EXPECT_EQ(config.iceBackupCandidatePairPingInterval,
+            newConfig.iceBackupCandidatePairPingInterval);
+  EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
+  EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
+}
+
 @end
 
 TEST(RTCConfigurationTest, NativeConfigurationConversionTest) {
   @autoreleasepool {
     RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
     [test testConversionToNativeConfiguration];
+    [test testNativeConversionToConfiguration];
   }
 }
 

webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm

@@ -0,0 +1,91 @@
+/*
+ *  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 "NSString+StdString.h"
+#import "RTCConfiguration+Private.h"
+#import "WebRTC/RTCConfiguration.h"
+#import "WebRTC/RTCPeerConnection.h"
+#import "WebRTC/RTCPeerConnectionFactory.h"
+#import "WebRTC/RTCIceServer.h"
+#import "WebRTC/RTCMediaConstraints.h"
+
+@interface RTCPeerConnectionTest : NSObject
+- (void)testConfigurationGetter;
+@end
+
+@implementation RTCPeerConnectionTest
+
+- (void)testConfigurationGetter {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
+
+  RTCConfiguration *config = [[RTCConfiguration alloc] init];
+  config.iceServers = @[ server ];
+  config.iceTransportPolicy = RTCIceTransportPolicyRelay;
+  config.bundlePolicy = RTCBundlePolicyMaxBundle;
+  config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
+  config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
+  config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
+  const int maxPackets = 60;
+  const int timeout = 1;
+  const int interval = 2;
+  config.audioJitterBufferMaxPackets = maxPackets;
+  config.audioJitterBufferFastAccelerate = YES;
+  config.iceConnectionReceivingTimeout = timeout;
+  config.iceBackupCandidatePairPingInterval = interval;
+  config.continualGatheringPolicy =
+      RTCContinualGatheringPolicyGatherContinually;
+  config.shouldPruneTurnPorts = YES;
+
+  RTCMediaConstraints *contraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{}
+      optionalConstraints:nil];
+  RTCPeerConnectionFactory *factory = [[RTCPeerConnectionFactory alloc] init];
+  RTCPeerConnection *peerConnection = [factory peerConnectionWithConfiguration:config
+      constraints:contraints delegate:nil];
+
+  RTCConfiguration *newConfig = peerConnection.configuration;
+
+  EXPECT_EQ([config.iceServers count], [newConfig.iceServers count]);
+  RTCIceServer *newServer = newConfig.iceServers[0];
+  RTCIceServer *origServer = config.iceServers[0];
+  std::string origUrl = origServer.urlStrings.firstObject.UTF8String;
+  std::string url = newServer.urlStrings.firstObject.UTF8String;
+  EXPECT_EQ(origUrl, url);
+
+  EXPECT_EQ(config.iceTransportPolicy, newConfig.iceTransportPolicy);
+  EXPECT_EQ(config.bundlePolicy, newConfig.bundlePolicy);
+  EXPECT_EQ(config.rtcpMuxPolicy, newConfig.rtcpMuxPolicy);
+  EXPECT_EQ(config.tcpCandidatePolicy, newConfig.tcpCandidatePolicy);
+  EXPECT_EQ(config.candidateNetworkPolicy, newConfig.candidateNetworkPolicy);
+  EXPECT_EQ(config.audioJitterBufferMaxPackets, newConfig.audioJitterBufferMaxPackets);
+  EXPECT_EQ(config.audioJitterBufferFastAccelerate, newConfig.audioJitterBufferFastAccelerate);
+  EXPECT_EQ(config.iceConnectionReceivingTimeout, newConfig.iceConnectionReceivingTimeout);
+  EXPECT_EQ(config.iceBackupCandidatePairPingInterval,
+            newConfig.iceBackupCandidatePairPingInterval);
+  EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
+  EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
+}
+
+@end
+
+TEST(RTCPeerConnectionTest, ConfigurationGetterTest) {
+  @autoreleasepool {
+    RTCPeerConnectionTest *test = [[RTCPeerConnectionTest alloc] init];
+    [test testConfigurationGetter];
+  }
+}
+
+