[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/BUILD.gn

@@ -268,6 +268,7 @@ if (is_ios || is_mac) {
         "objc/Framework/UnitTests/RTCIceCandidateTest.mm",
         "objc/Framework/UnitTests/RTCIceServerTest.mm",
         "objc/Framework/UnitTests/RTCMediaConstraintsTest.mm",
+        "objc/Framework/UnitTests/RTCPeerConnectionTest.mm",
         "objc/Framework/UnitTests/RTCSessionDescriptionTest.mm",
         "objc/Framework/UnitTests/avformatmappertests.mm",
       ]

webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h

@@ -65,6 +65,9 @@ NS_ASSUME_NONNULL_BEGIN
 - (webrtc::PeerConnectionInterface::RTCConfiguration *)
     createNativeConfiguration;
 
+- (instancetype)initWithNativeConfiguration:
+    (const webrtc::PeerConnectionInterface::RTCConfiguration *)config NS_DESIGNATED_INITIALIZER;
+
 @end
 
 NS_ASSUME_NONNULL_END

webrtc/sdk/objc/Framework/Classes/RTCConfiguration.mm

@@ -40,38 +40,49 @@
 @synthesize iceCheckMinInterval = _iceCheckMinInterval;
 
 - (instancetype)init {
+  // Copy defaults.
+  webrtc::PeerConnectionInterface::RTCConfiguration config(
+    webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive);
+  return [self initWithNativeConfiguration:&config];
+}
+
+- (instancetype)initWithNativeConfiguration:
+    (const webrtc::PeerConnectionInterface::RTCConfiguration *)config {
+  NSParameterAssert(config);
   if (self = [super init]) {
-    _iceServers = [NSMutableArray array];
+    NSMutableArray *iceServers = [NSMutableArray array];
-    // Copy defaults.
+    for (const webrtc::PeerConnectionInterface::IceServer& server : config->servers) {
-    webrtc::PeerConnectionInterface::RTCConfiguration config(
+      RTCIceServer *iceServer = [[RTCIceServer alloc] initWithNativeServer:server];
-        webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive);
+      [iceServers addObject:iceServer];
+    }
+    _iceServers = iceServers;
     _iceTransportPolicy =
-        [[self class] transportPolicyForTransportsType:config.type];
+        [[self class] transportPolicyForTransportsType:config->type];
     _bundlePolicy =
-        [[self class] bundlePolicyForNativePolicy:config.bundle_policy];
+        [[self class] bundlePolicyForNativePolicy:config->bundle_policy];
     _rtcpMuxPolicy =
-        [[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
+        [[self class] rtcpMuxPolicyForNativePolicy:config->rtcp_mux_policy];
     _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
-        config.tcp_candidate_policy];
+        config->tcp_candidate_policy];
     _candidateNetworkPolicy = [[self class]
-        candidateNetworkPolicyForNativePolicy:config.candidate_network_policy];
+        candidateNetworkPolicyForNativePolicy:config->candidate_network_policy];
     webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
-        config.continual_gathering_policy;
+    config->continual_gathering_policy;
     _continualGatheringPolicy =
         [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
-    _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
+    _audioJitterBufferMaxPackets = config->audio_jitter_buffer_max_packets;
-    _audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
+    _audioJitterBufferFastAccelerate = config->audio_jitter_buffer_fast_accelerate;
-    _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
+    _iceConnectionReceivingTimeout = config->ice_connection_receiving_timeout;
     _iceBackupCandidatePairPingInterval =
-        config.ice_backup_candidate_pair_ping_interval;
+        config->ice_backup_candidate_pair_ping_interval;
     _keyType = RTCEncryptionKeyTypeECDSA;
-    _iceCandidatePoolSize = config.ice_candidate_pool_size;
+    _iceCandidatePoolSize = config->ice_candidate_pool_size;
-    _shouldPruneTurnPorts = config.prune_turn_ports;
+    _shouldPruneTurnPorts = config->prune_turn_ports;
     _shouldPresumeWritableWhenFullyRelayed =
-        config.presume_writable_when_fully_relayed;
+        config->presume_writable_when_fully_relayed;
-    if (config.ice_check_min_interval) {
+    if (config->ice_check_min_interval) {
       _iceCheckMinInterval =
-          [NSNumber numberWithInt:*config.ice_check_min_interval];
+          [NSNumber numberWithInt:*config->ice_check_min_interval];
     }
   }
   return self;

webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm

@@ -288,6 +288,12 @@ void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
   return _peerConnection->SetConfiguration(*config);
 }
 
+- (RTCConfiguration *)configuration {
+  webrtc::PeerConnectionInterface::RTCConfiguration config =
+    _peerConnection->GetConfiguration();
+  return [[RTCConfiguration alloc] initWithNativeConfiguration:&config];
+}
+
 - (void)close {
   _peerConnection->Close();
 }

webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h

@@ -109,7 +109,7 @@ RTC_EXPORT
  */
 @property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
 
-- (instancetype)init NS_DESIGNATED_INITIALIZER;
+- (instancetype)init;
 
 @end
 

webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h

@@ -123,6 +123,7 @@ RTC_EXPORT
 @property(nonatomic, readonly) RTCSignalingState signalingState;
 @property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
 @property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
+@property(nonatomic, readonly, copy) RTCConfiguration *configuration;
 
 /** Gets all RTCRtpSenders associated with this peer connection.
  *  Note: reading this property returns different instances of RTCRtpSender.

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];
+  }
+}
+
+