Obj-C SDK Cleanup
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}diff --git a/sdk/objc/unittests/RTCConfigurationTest.mm b/sdk/objc/unittests/RTCConfigurationTest.mm
new file mode 100644
index 0000000..1ef718a
--- /dev/null
+++ b/sdk/objc/unittests/RTCConfigurationTest.mm
@@ -0,0 +1,143 @@
+/*
+ * 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/RTCIntervalRange.h"
+#import "helpers/NSString+StdString.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];
+ }
+}