Revert of Implement RTCConfiguration.iceCandidatePoolSize. (patchset #7 id:120001 of https://codereview.webrtc.org/1956453003/ )
Reason for revert:
Breaks remoting_unittests. They defined their own operator== which conflicts with this one.
I'll remove the operator== in a roll CL. But until it's approved, I'm reverting this so the FYI bots will pass.
Original issue's description:
> Implement RTCConfiguration.iceCandidatePoolSize.
>
> It works by creating pooled PortAllocatorSessions which can be picked up
> by a P2PTransportChannel when needed (after a local description is set).
>
> This can optimize candidate gathering time when there is some time between
> creating a PeerConnection and setting a local description.
>
> R=pthatcher@webrtc.org
>
> Committed: 48e9d05f51
TBR=pthatcher@webrtc.org,honghaiz@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/1972043004
Cr-Commit-Position: refs/heads/master@{#12709}
This commit is contained in:
@ -270,30 +270,42 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
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static const int kAudioJitterBufferMaxPackets = 50;
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// TODO(pthatcher): Rename this ice_transport_type, but update
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// Chromium at the same time.
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IceTransportsType type = kAll;
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IceTransportsType type;
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// TODO(pthatcher): Rename this ice_servers, but update Chromium
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// at the same time.
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IceServers servers;
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BundlePolicy bundle_policy = kBundlePolicyBalanced;
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RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyNegotiate;
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TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
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int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
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bool audio_jitter_buffer_fast_accelerate = false;
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int ice_connection_receiving_timeout = kUndefined; // ms
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int ice_backup_candidate_pair_ping_interval = kUndefined; // ms
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ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
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BundlePolicy bundle_policy;
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RtcpMuxPolicy rtcp_mux_policy;
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TcpCandidatePolicy tcp_candidate_policy;
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int audio_jitter_buffer_max_packets;
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bool audio_jitter_buffer_fast_accelerate;
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int ice_connection_receiving_timeout; // ms
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int ice_backup_candidate_pair_ping_interval; // ms
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ContinualGatheringPolicy continual_gathering_policy;
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std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
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bool prioritize_most_likely_ice_candidate_pairs = false;
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bool prioritize_most_likely_ice_candidate_pairs;
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struct cricket::MediaConfig media_config;
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// Flags corresponding to values set by constraint flags.
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// rtc::Optional flags can be "missing", in which case the webrtc
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// default applies.
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bool disable_ipv6 = false;
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bool enable_rtp_data_channel = false;
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bool disable_ipv6;
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bool enable_rtp_data_channel;
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rtc::Optional<int> screencast_min_bitrate;
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rtc::Optional<bool> combined_audio_video_bwe;
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rtc::Optional<bool> enable_dtls_srtp;
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int ice_candidate_pool_size = 0;
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RTCConfiguration()
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: type(kAll),
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bundle_policy(kBundlePolicyBalanced),
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rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
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tcp_candidate_policy(kTcpCandidatePolicyEnabled),
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audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
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audio_jitter_buffer_fast_accelerate(false),
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ice_connection_receiving_timeout(kUndefined),
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ice_backup_candidate_pair_ping_interval(kUndefined),
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continual_gathering_policy(GATHER_ONCE),
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prioritize_most_likely_ice_candidate_pairs(false),
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disable_ipv6(false),
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enable_rtp_data_channel(false) {}
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};
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struct RTCOfferAnswerOptions {
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