rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm

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2016-06-21 20:13:05 +00:00
/*
* libjingle
* Copyright 2015 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#import "talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h"
#import "talk/app/webrtc/objc/RTCEnumConverter.h"
#import "talk/app/webrtc/objc/RTCICEServer+Internal.h"
#import "talk/app/webrtc/objc/public/RTCLogging.h"
#include <memory>
#include "webrtc/base/rtccertificategenerator.h"
@implementation RTCConfiguration
@synthesize iceTransportsType = _iceTransportsType;
@synthesize iceServers = _iceServers;
@synthesize bundlePolicy = _bundlePolicy;
@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
@synthesize iceBackupCandidatePairPingInterval = _iceBackupCandidatePairPingInterval;
@synthesize keyType = _keyType;
- (instancetype)init {
if (self = [super init]) {
// Copy defaults.
webrtc::PeerConnectionInterface::RTCConfiguration config;
_iceTransportsType = [RTCEnumConverter iceTransportsTypeForNativeEnum:config.type];
_bundlePolicy = [RTCEnumConverter bundlePolicyForNativeEnum:config.bundle_policy];
_rtcpMuxPolicy = [RTCEnumConverter rtcpMuxPolicyForNativeEnum:config.rtcp_mux_policy];
_tcpCandidatePolicy =
[RTCEnumConverter tcpCandidatePolicyForNativeEnum:config.tcp_candidate_policy];
_audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
_iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
_iceBackupCandidatePairPingInterval = config.ice_backup_candidate_pair_ping_interval;
_keyType = kRTCEncryptionKeyTypeECDSA;
}
return self;
}
- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
bundlePolicy:(RTCBundlePolicy)bundlePolicy
rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval {
if (self = [super init]) {
_iceTransportsType = iceTransportsType;
_bundlePolicy = bundlePolicy;
_rtcpMuxPolicy = rtcpMuxPolicy;
_tcpCandidatePolicy = tcpCandidatePolicy;
_audioJitterBufferMaxPackets = audioJitterBufferMaxPackets;
_iceConnectionReceivingTimeout = iceConnectionReceivingTimeout;
_iceBackupCandidatePairPingInterval = iceBackupCandidatePairPingInterval;
}
return self;
}
#pragma mark - Private
- (webrtc::PeerConnectionInterface::RTCConfiguration *)
createNativeConfiguration {
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
nativeConfig(new webrtc::PeerConnectionInterface::RTCConfiguration());
nativeConfig->type =
[RTCEnumConverter nativeEnumForIceTransportsType:_iceTransportsType];
for (RTCICEServer *iceServer : _iceServers) {
nativeConfig->servers.push_back(iceServer.iceServer);
}
nativeConfig->bundle_policy =
[RTCEnumConverter nativeEnumForBundlePolicy:_bundlePolicy];
nativeConfig->rtcp_mux_policy =
[RTCEnumConverter nativeEnumForRtcpMuxPolicy:_rtcpMuxPolicy];
nativeConfig->tcp_candidate_policy =
[RTCEnumConverter nativeEnumForTcpCandidatePolicy:_tcpCandidatePolicy];
nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
nativeConfig->ice_connection_receiving_timeout =
_iceConnectionReceivingTimeout;
nativeConfig->ice_backup_candidate_pair_ping_interval =
_iceBackupCandidatePairPingInterval;
rtc::KeyType keyType =
[[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
if (keyType != rtc::KT_DEFAULT) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate =
rtc::RTCCertificateGenerator::GenerateCertificate(
rtc::KeyParams(keyType), rtc::Optional<uint64_t>());
if (!certificate) {
RTCLogError(@"Failed to generate certificate.");
return nullptr;
}
nativeConfig->certificates.push_back(certificate);
}
return nativeConfig.release();
}
+ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:
(RTCEncryptionKeyType)keyType {
switch (keyType) {
case kRTCEncryptionKeyTypeRSA:
return rtc::KT_RSA;
case kRTCEncryptionKeyTypeECDSA:
return rtc::KT_ECDSA;
}
}
@end