92 lines
3.6 KiB
Plaintext
92 lines
3.6 KiB
Plaintext
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/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#import <Foundation/Foundation.h>
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#include <vector>
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#include "webrtc/base/gunit.h"
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#import "NSString+StdString.h"
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#import "RTCConfiguration+Private.h"
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#import "WebRTC/RTCConfiguration.h"
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#import "WebRTC/RTCPeerConnection.h"
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#import "WebRTC/RTCPeerConnectionFactory.h"
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#import "WebRTC/RTCIceServer.h"
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#import "WebRTC/RTCMediaConstraints.h"
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@interface RTCPeerConnectionTest : NSObject
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- (void)testConfigurationGetter;
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@end
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@implementation RTCPeerConnectionTest
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- (void)testConfigurationGetter {
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NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
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RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
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RTCConfiguration *config = [[RTCConfiguration alloc] init];
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config.iceServers = @[ server ];
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config.iceTransportPolicy = RTCIceTransportPolicyRelay;
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config.bundlePolicy = RTCBundlePolicyMaxBundle;
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config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
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config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
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config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
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const int maxPackets = 60;
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const int timeout = 1;
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const int interval = 2;
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config.audioJitterBufferMaxPackets = maxPackets;
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config.audioJitterBufferFastAccelerate = YES;
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config.iceConnectionReceivingTimeout = timeout;
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config.iceBackupCandidatePairPingInterval = interval;
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config.continualGatheringPolicy =
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RTCContinualGatheringPolicyGatherContinually;
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config.shouldPruneTurnPorts = YES;
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RTCMediaConstraints *contraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{}
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optionalConstraints:nil];
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RTCPeerConnectionFactory *factory = [[RTCPeerConnectionFactory alloc] init];
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RTCPeerConnection *peerConnection = [factory peerConnectionWithConfiguration:config
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constraints:contraints delegate:nil];
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RTCConfiguration *newConfig = peerConnection.configuration;
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EXPECT_EQ([config.iceServers count], [newConfig.iceServers count]);
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RTCIceServer *newServer = newConfig.iceServers[0];
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RTCIceServer *origServer = config.iceServers[0];
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std::string origUrl = origServer.urlStrings.firstObject.UTF8String;
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std::string url = newServer.urlStrings.firstObject.UTF8String;
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EXPECT_EQ(origUrl, url);
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EXPECT_EQ(config.iceTransportPolicy, newConfig.iceTransportPolicy);
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EXPECT_EQ(config.bundlePolicy, newConfig.bundlePolicy);
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EXPECT_EQ(config.rtcpMuxPolicy, newConfig.rtcpMuxPolicy);
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EXPECT_EQ(config.tcpCandidatePolicy, newConfig.tcpCandidatePolicy);
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EXPECT_EQ(config.candidateNetworkPolicy, newConfig.candidateNetworkPolicy);
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EXPECT_EQ(config.audioJitterBufferMaxPackets, newConfig.audioJitterBufferMaxPackets);
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EXPECT_EQ(config.audioJitterBufferFastAccelerate, newConfig.audioJitterBufferFastAccelerate);
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EXPECT_EQ(config.iceConnectionReceivingTimeout, newConfig.iceConnectionReceivingTimeout);
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EXPECT_EQ(config.iceBackupCandidatePairPingInterval,
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newConfig.iceBackupCandidatePairPingInterval);
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EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
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EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
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}
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@end
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TEST(RTCPeerConnectionTest, ConfigurationGetterTest) {
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@autoreleasepool {
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RTCPeerConnectionTest *test = [[RTCPeerConnectionTest alloc] init];
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[test testConfigurationGetter];
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}
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}
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