Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

sdk/objc/Framework/UnitTests/RTCConfigurationTest.mm

@@ -0,0 +1,144 @@
+/*
+ *  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 "webrtc/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];
+  }
+}
+