1 files changed, 162 insertions, 0 deletions
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 <Foundation/Foundation.h>
+#import <XCTest/XCTest.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 "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<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(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