From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../sdk/objc/unittests/RTCConfigurationTest.mm | 162 +++++++++++++++++++++ 1 file changed, 162 insertions(+) create mode 100644 third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm (limited to 'third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm') diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm new file mode 100644 index 0000000000..18cc97191e --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm @@ -0,0 +1,162 @@ +/* + * 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 +#import + +#include + +#include "rtc_base/gunit.h" + +#import "api/peerconnection/RTCConfiguration+Private.h" +#import "api/peerconnection/RTCConfiguration.h" +#import "api/peerconnection/RTCIceServer.h" +#import "helpers/NSString+StdString.h" + +@interface RTCConfigurationTest : XCTestCase +@end + +@implementation RTCConfigurationTest + +- (void)testConversionToNativeConfiguration { + NSArray *urlStrings = @[ @"stun:stun1.example.net" ]; + RTC_OBJC_TYPE(RTCIceServer) *server = + [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings]; + + RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(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.cryptoOptions = + [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES + srtpEnableAes128Sha1_32CryptoCipher:YES + srtpEnableEncryptedRtpHeaderExtensions:YES + sframeRequireFrameEncryption:YES]; + config.rtcpAudioReportIntervalMs = 2500; + config.rtcpVideoReportIntervalMs = 3750; + + std::unique_ptr + 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(true, nativeConfig->crypto_options->srtp.enable_gcm_crypto_suites); + EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher); + EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_encrypted_rtp_header_extensions); + EXPECT_EQ(true, nativeConfig->crypto_options->sframe.require_frame_encryption); + EXPECT_EQ(2500, nativeConfig->audio_rtcp_report_interval_ms()); + EXPECT_EQ(3750, nativeConfig->video_rtcp_report_interval_ms()); +} + +- (void)testNativeConversionToConfiguration { + NSArray *urlStrings = @[ @"stun:stun1.example.net" ]; + RTC_OBJC_TYPE(RTCIceServer) *server = + [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings]; + + RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(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.cryptoOptions = + [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES + srtpEnableAes128Sha1_32CryptoCipher:NO + srtpEnableEncryptedRtpHeaderExtensions:NO + sframeRequireFrameEncryption:NO]; + config.rtcpAudioReportIntervalMs = 1500; + config.rtcpVideoReportIntervalMs = 2150; + + webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig = + [config createNativeConfiguration]; + RTC_OBJC_TYPE(RTCConfiguration) *newConfig = + [[RTC_OBJC_TYPE(RTCConfiguration) alloc] initWithNativeConfiguration:*nativeConfig]; + EXPECT_EQ([config.iceServers count], newConfig.iceServers.count); + RTC_OBJC_TYPE(RTCIceServer) *newServer = newConfig.iceServers[0]; + RTC_OBJC_TYPE(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.cryptoOptions.srtpEnableGcmCryptoSuites, + newConfig.cryptoOptions.srtpEnableGcmCryptoSuites); + EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher, + newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher); + EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions, + newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions); + EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption, + newConfig.cryptoOptions.sframeRequireFrameEncryption); + EXPECT_EQ(config.rtcpAudioReportIntervalMs, newConfig.rtcpAudioReportIntervalMs); + EXPECT_EQ(config.rtcpVideoReportIntervalMs, newConfig.rtcpVideoReportIntervalMs); +} + +- (void)testDefaultValues { + RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init]; + EXPECT_EQ(config.cryptoOptions, nil); +} + +@end -- cgit v1.2.3