mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00

In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
144 lines
6.4 KiB
Text
144 lines
6.4 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 "NSString+StdString.h"
|
|
#import "RTCConfiguration+Private.h"
|
|
#import "WebRTC/RTCConfiguration.h"
|
|
#import "WebRTC/RTCIceServer.h"
|
|
#import "WebRTC/RTCIntervalRange.h"
|
|
|
|
@interface RTCConfigurationTest : NSObject
|
|
- (void)testConversionToNativeConfiguration;
|
|
- (void)testNativeConversionToConfiguration;
|
|
@end
|
|
|
|
@implementation RTCConfigurationTest
|
|
|
|
- (void)testConversionToNativeConfiguration {
|
|
NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
|
|
RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
|
|
RTCIntervalRange *range = [[RTCIntervalRange alloc] initWithMin:0 max:100];
|
|
|
|
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;
|
|
config.iceRegatherIntervalRange = range;
|
|
|
|
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
|
|
nativeConfig([config createNativeConfiguration]);
|
|
EXPECT_TRUE(nativeConfig.get());
|
|
EXPECT_EQ(1u, nativeConfig->servers.size());
|
|
webrtc::PeerConnectionInterface::IceServer nativeServer =
|
|
nativeConfig->servers.front();
|
|
EXPECT_EQ(1u, nativeServer.urls.size());
|
|
EXPECT_EQ("stun:stun1.example.net", nativeServer.urls.front());
|
|
|
|
EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig->type);
|
|
EXPECT_EQ(webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle,
|
|
nativeConfig->bundle_policy);
|
|
EXPECT_EQ(webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate,
|
|
nativeConfig->rtcp_mux_policy);
|
|
EXPECT_EQ(webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled,
|
|
nativeConfig->tcp_candidate_policy);
|
|
EXPECT_EQ(webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost,
|
|
nativeConfig->candidate_network_policy);
|
|
EXPECT_EQ(maxPackets, nativeConfig->audio_jitter_buffer_max_packets);
|
|
EXPECT_EQ(true, nativeConfig->audio_jitter_buffer_fast_accelerate);
|
|
EXPECT_EQ(timeout, nativeConfig->ice_connection_receiving_timeout);
|
|
EXPECT_EQ(interval, nativeConfig->ice_backup_candidate_pair_ping_interval);
|
|
EXPECT_EQ(webrtc::PeerConnectionInterface::GATHER_CONTINUALLY,
|
|
nativeConfig->continual_gathering_policy);
|
|
EXPECT_EQ(true, nativeConfig->prune_turn_ports);
|
|
EXPECT_EQ(range.min, nativeConfig->ice_regather_interval_range->min());
|
|
EXPECT_EQ(range.max, nativeConfig->ice_regather_interval_range->max());
|
|
}
|
|
|
|
- (void)testNativeConversionToConfiguration {
|
|
NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
|
|
RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
|
|
RTCIntervalRange *range = [[RTCIntervalRange alloc] initWithMin:0 max:100];
|
|
|
|
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;
|
|
config.iceRegatherIntervalRange = range;
|
|
|
|
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);
|
|
EXPECT_EQ(config.iceRegatherIntervalRange.min, newConfig.iceRegatherIntervalRange.min);
|
|
EXPECT_EQ(config.iceRegatherIntervalRange.max, newConfig.iceRegatherIntervalRange.max);
|
|
}
|
|
|
|
@end
|
|
|
|
TEST(RTCConfigurationTest, NativeConfigurationConversionTest) {
|
|
@autoreleasepool {
|
|
RTCConfigurationTest *test = [[RTCConfigurationTest alloc] init];
|
|
[test testConversionToNativeConfiguration];
|
|
[test testNativeConversionToConfiguration];
|
|
}
|
|
}
|
|
|