[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}
diff --git a/webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h b/webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h
index f20cdc5..05a5034 100644
--- a/webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h
+++ b/webrtc/sdk/objc/Framework/Classes/RTCConfiguration+Private.h
@@ -65,6 +65,9 @@
- (webrtc::PeerConnectionInterface::RTCConfiguration *)
createNativeConfiguration;
+- (instancetype)initWithNativeConfiguration:
+ (const webrtc::PeerConnectionInterface::RTCConfiguration *)config NS_DESIGNATED_INITIALIZER;
+
@end
NS_ASSUME_NONNULL_END
diff --git a/webrtc/sdk/objc/Framework/Classes/RTCConfiguration.mm b/webrtc/sdk/objc/Framework/Classes/RTCConfiguration.mm
index ae37fc1..54d8eac 100644
--- a/webrtc/sdk/objc/Framework/Classes/RTCConfiguration.mm
+++ b/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];
- // Copy defaults.
- webrtc::PeerConnectionInterface::RTCConfiguration config(
- webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive);
+ NSMutableArray *iceServers = [NSMutableArray array];
+ for (const webrtc::PeerConnectionInterface::IceServer& server : config->servers) {
+ RTCIceServer *iceServer = [[RTCIceServer alloc] initWithNativeServer:server];
+ [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;
- _audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
- _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
+ _audioJitterBufferMaxPackets = config->audio_jitter_buffer_max_packets;
+ _audioJitterBufferFastAccelerate = config->audio_jitter_buffer_fast_accelerate;
+ _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;
- _shouldPruneTurnPorts = config.prune_turn_ports;
+ _iceCandidatePoolSize = config->ice_candidate_pool_size;
+ _shouldPruneTurnPorts = config->prune_turn_ports;
_shouldPresumeWritableWhenFullyRelayed =
- config.presume_writable_when_fully_relayed;
- if (config.ice_check_min_interval) {
+ config->presume_writable_when_fully_relayed;
+ if (config->ice_check_min_interval) {
_iceCheckMinInterval =
- [NSNumber numberWithInt:*config.ice_check_min_interval];
+ [NSNumber numberWithInt:*config->ice_check_min_interval];
}
}
return self;
diff --git a/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm b/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm
index de7608c..7c8eb4d 100644
--- a/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm
+++ b/webrtc/sdk/objc/Framework/Classes/RTCPeerConnection.mm
@@ -288,6 +288,12 @@
return _peerConnection->SetConfiguration(*config);
}
+- (RTCConfiguration *)configuration {
+ webrtc::PeerConnectionInterface::RTCConfiguration config =
+ _peerConnection->GetConfiguration();
+ return [[RTCConfiguration alloc] initWithNativeConfiguration:&config];
+}
+
- (void)close {
_peerConnection->Close();
}
diff --git a/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
index c516240..862a01d 100644
--- a/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
+++ b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
@@ -109,7 +109,7 @@
*/
@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
-- (instancetype)init NS_DESIGNATED_INITIALIZER;
+- (instancetype)init;
@end
diff --git a/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
index 4438eb2..13617b5 100644
--- a/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
+++ b/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
@@ -123,6 +123,7 @@
@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.
diff --git a/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm b/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm
index 79ffd17..fee2a49 100644
--- a/webrtc/sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm
+++ b/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];
}
}
diff --git a/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm b/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm
new file mode 100644
index 0000000..bd2db30
--- /dev/null
+++ b/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];
+ }
+}
+
+