bc7dd7e023
BUG=4658 R=jiayl@webrtc.org Review URL: https://webrtc-codereview.appspot.com/56419004 Cr-Commit-Position: refs/heads/master@{#9335}
88 lines
3.9 KiB
Plaintext
88 lines
3.9 KiB
Plaintext
/*
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* libjingle
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* Copyright 2015 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#import "talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h"
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#import "talk/app/webrtc/objc/RTCEnumConverter.h"
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#import "talk/app/webrtc/objc/RTCICEServer+Internal.h"
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@implementation RTCConfiguration
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@synthesize iceTransportsType = _iceTransportsType;
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@synthesize iceServers = _iceServers;
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@synthesize bundlePolicy = _bundlePolicy;
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@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
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@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
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@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
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- (instancetype)init {
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if (self = [super init]) {
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// Copy defaults.
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webrtc::PeerConnectionInterface::RTCConfiguration config;
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_iceTransportsType = [RTCEnumConverter iceTransportsTypeForNativeEnum:config.type];
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_bundlePolicy = [RTCEnumConverter bundlePolicyForNativeEnum:config.bundle_policy];
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_rtcpMuxPolicy = [RTCEnumConverter rtcpMuxPolicyForNativeEnum:config.rtcp_mux_policy];
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_tcpCandidatePolicy =
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[RTCEnumConverter tcpCandidatePolicyForNativeEnum:config.tcp_candidate_policy];
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_audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
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}
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return self;
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}
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- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
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bundlePolicy:(RTCBundlePolicy)bundlePolicy
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rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
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tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
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audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets {
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if (self = [super init]) {
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_iceTransportsType = iceTransportsType;
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_bundlePolicy = bundlePolicy;
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_rtcpMuxPolicy = rtcpMuxPolicy;
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_tcpCandidatePolicy = tcpCandidatePolicy;
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_audioJitterBufferMaxPackets = audioJitterBufferMaxPackets;
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}
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return self;
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}
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#pragma mark - Private
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- (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfiguration {
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webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
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nativeConfig.type = [RTCEnumConverter nativeEnumForIceTransportsType:_iceTransportsType];
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for (RTCICEServer *iceServer : _iceServers) {
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nativeConfig.servers.push_back(iceServer.iceServer);
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}
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nativeConfig.bundle_policy = [RTCEnumConverter nativeEnumForBundlePolicy:_bundlePolicy];
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nativeConfig.rtcp_mux_policy = [RTCEnumConverter nativeEnumForRtcpMuxPolicy:_rtcpMuxPolicy];
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nativeConfig.tcp_candidate_policy =
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[RTCEnumConverter nativeEnumForTcpCandidatePolicy:_tcpCandidatePolicy];
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nativeConfig.audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
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return nativeConfig;
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}
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@end
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