mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
7bca8ca4e2
This CL separates the files under sdk/objc into logical directories, replacing the previous file layout under Framework/. A long term goal is to have some system set up to generate the files under sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter term the goal is to abstract out shared concepts from these classes in order to make them as uniform as possible. The separation into base/, components/, and helpers/ are to differentiate between the base layer's common protocols, various utilities and the actual platform specific components. The old directory layout that resembled a framework's internal layout is not necessary, since it is generated by the framework target when building it. Bug: webrtc:9627 Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f Reviewed-on: https://webrtc-review.googlesource.com/94142 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24493}
101 lines
4.3 KiB
Text
101 lines
4.3 KiB
Text
/*
|
|
* 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 "api/peerconnection/RTCConfiguration+Private.h"
|
|
#import "api/peerconnection/RTCConfiguration.h"
|
|
#import "api/peerconnection/RTCIceServer.h"
|
|
#import "api/peerconnection/RTCMediaConstraints.h"
|
|
#import "api/peerconnection/RTCPeerConnection.h"
|
|
#import "api/peerconnection/RTCPeerConnectionFactory.h"
|
|
#import "helpers/NSString+StdString.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];
|
|
}
|
|
}
|