blob: 63617cbe5a9e26cf3174e9b32777f2d895daebc8 [file] [log] [blame]
/*
* 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 "rtc_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 = 1500;
const int interval = 2000;
config.audioJitterBufferMaxPackets = maxPackets;
config.audioJitterBufferFastAccelerate = YES;
config.iceConnectionReceivingTimeout = timeout;
config.iceBackupCandidatePairPingInterval = interval;
config.continualGatheringPolicy =
RTCContinualGatheringPolicyGatherContinually;
config.shouldPruneTurnPorts = YES;
config.activeResetSrtpParams = YES;
RTCMediaConstraints *contraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{}
optionalConstraints:nil];
RTCPeerConnectionFactory *factory = [[RTCPeerConnectionFactory alloc] init];
RTCConfiguration *newConfig;
@autoreleasepool {
RTCPeerConnection *peerConnection =
[factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
newConfig = peerConnection.configuration;
EXPECT_TRUE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:100000]
currentBitrateBps:[NSNumber numberWithInt:5000000]
maxBitrateBps:[NSNumber numberWithInt:500000000]]);
EXPECT_FALSE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:2]
currentBitrateBps:[NSNumber numberWithInt:1]
maxBitrateBps:nil]);
}
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);
EXPECT_EQ(config.activeResetSrtpParams, newConfig.activeResetSrtpParams);
}
@end
TEST(RTCPeerConnectionTest, ConfigurationGetterTest) {
@autoreleasepool {
RTCPeerConnectionTest *test = [[RTCPeerConnectionTest alloc] init];
[test testConfigurationGetter];
}
}