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This reverts commit 181ea6e414
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Reason for revert: Breaks downstream project. Kári will help to land it next week.
Original change's description:
> Add a prefix for objc category.
>
> According to the Google Objective-C style [1], category names should
> start with an appropriate prefix. WebRTC has some category definitions
> for system interfaces, but it doesn't use prefixes.
>
> $ otool -ov WebRTC.framework/WebRTC | grep -E "^[0-9a-z]{16} 0x[0-9a-z]+ __OBJC_._CATEGORY" | grep -v "_RTC"
> 0000000002160840 0x217c3c0 __OBJC_$_CATEGORY_UIDevice_$_H264Profile
> 0000000002160850 0x21808b8 __OBJC_$_CATEGORY_AVCaptureSession_$_DevicePosition
> 0000000002160858 0x2180968 __OBJC_$_CATEGORY_NSString_$_StdString
> 0000000002160860 0x21809c8 __OBJC_$_CATEGORY_NSString_$_AbslStringView
>
> To avoid conflicts, prefix the names and methods of those categories.
> Also remove sdk/objc/Framework/Classes/Common/NSString+StdString.h as
> it is not used by any other files.
>
> [1] https://google.github.io/styleguide/objcguide.html#category-naming
>
> Bug: webrtc:13884
> Change-Id: I2cf2742af198ab4e0bfb15c0476d72971e50ceee
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262341
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36880}
Bug: webrtc:13884
Change-Id: I85257088e4a3a62e01ff925ab5e77af83b078ef3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262420
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36885}
204 lines
9.5 KiB
Text
204 lines
9.5 KiB
Text
/*
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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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#import <XCTest/XCTest.h>
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#include <memory>
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#include <vector>
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#include "rtc_base/gunit.h"
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#import "api/peerconnection/RTCConfiguration+Private.h"
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#import "api/peerconnection/RTCConfiguration.h"
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#import "api/peerconnection/RTCCryptoOptions.h"
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#import "api/peerconnection/RTCIceCandidate.h"
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#import "api/peerconnection/RTCIceServer.h"
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#import "api/peerconnection/RTCMediaConstraints.h"
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#import "api/peerconnection/RTCPeerConnection.h"
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#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"
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#import "api/peerconnection/RTCPeerConnectionFactory.h"
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#import "api/peerconnection/RTCSessionDescription.h"
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#import "helpers/NSString+StdString.h"
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@interface RTCPeerConnectionTests : XCTestCase
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@end
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@implementation RTCPeerConnectionTests
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- (void)testConfigurationGetter {
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NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
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RTC_OBJC_TYPE(RTCIceServer) *server =
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[[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
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RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
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config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
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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 = 1500;
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const int interval = 2000;
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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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config.activeResetSrtpParams = YES;
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config.cryptoOptions =
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[[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES
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srtpEnableAes128Sha1_32CryptoCipher:YES
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srtpEnableEncryptedRtpHeaderExtensions:NO
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sframeRequireFrameEncryption:NO];
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RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
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[[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
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optionalConstraints:nil];
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RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
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[[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
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RTC_OBJC_TYPE(RTCConfiguration) * newConfig;
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@autoreleasepool {
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RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
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[factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
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newConfig = peerConnection.configuration;
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EXPECT_TRUE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:100000]
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currentBitrateBps:[NSNumber numberWithInt:5000000]
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maxBitrateBps:[NSNumber numberWithInt:500000000]]);
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EXPECT_FALSE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:2]
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currentBitrateBps:[NSNumber numberWithInt:1]
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maxBitrateBps:nil]);
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}
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EXPECT_EQ([config.iceServers count], [newConfig.iceServers count]);
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RTC_OBJC_TYPE(RTCIceServer) *newServer = newConfig.iceServers[0];
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RTC_OBJC_TYPE(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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EXPECT_EQ(config.activeResetSrtpParams, newConfig.activeResetSrtpParams);
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EXPECT_EQ(config.cryptoOptions.srtpEnableGcmCryptoSuites,
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newConfig.cryptoOptions.srtpEnableGcmCryptoSuites);
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EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher,
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newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher);
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EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions,
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newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
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EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
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newConfig.cryptoOptions.sframeRequireFrameEncryption);
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}
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- (void)testWithDependencies {
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NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
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RTC_OBJC_TYPE(RTCIceServer) *server =
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[[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
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RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
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config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
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config.iceServers = @[ server ];
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RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
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[[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
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optionalConstraints:nil];
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RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
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[[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
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std::unique_ptr<webrtc::PeerConnectionDependencies> pc_dependencies =
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std::make_unique<webrtc::PeerConnectionDependencies>(nullptr);
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@autoreleasepool {
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RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
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[factory peerConnectionWithDependencies:config
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constraints:contraints
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dependencies:std::move(pc_dependencies)
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delegate:nil];
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ASSERT_NE(peerConnection, nil);
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}
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}
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- (void)testWithInvalidSDP {
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RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
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[[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
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RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
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config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
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RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
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[[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
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optionalConstraints:nil];
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RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
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[factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
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dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
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[peerConnection setRemoteDescription:[[RTC_OBJC_TYPE(RTCSessionDescription) alloc]
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initWithType:RTCSdpTypeOffer
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sdp:@"invalid"]
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completionHandler:^(NSError *error) {
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ASSERT_NE(error, nil);
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if (error != nil) {
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dispatch_semaphore_signal(negotiatedSem);
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}
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}];
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NSTimeInterval timeout = 5;
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ASSERT_EQ(
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0,
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dispatch_semaphore_wait(negotiatedSem,
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dispatch_time(DISPATCH_TIME_NOW, (int64_t)(timeout * NSEC_PER_SEC))));
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[peerConnection close];
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}
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- (void)testWithInvalidIceCandidate {
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RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
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[[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
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RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
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config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
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RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
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[[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
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optionalConstraints:nil];
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RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
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[factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
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dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
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[peerConnection addIceCandidate:[[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithSdp:@"invalid"
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sdpMLineIndex:-1
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sdpMid:nil]
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completionHandler:^(NSError *error) {
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ASSERT_NE(error, nil);
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if (error != nil) {
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dispatch_semaphore_signal(negotiatedSem);
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}
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}];
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NSTimeInterval timeout = 5;
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ASSERT_EQ(
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0,
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dispatch_semaphore_wait(negotiatedSem,
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dispatch_time(DISPATCH_TIME_NOW, (int64_t)(timeout * NSEC_PER_SEC))));
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[peerConnection close];
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}
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@end
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