Set the IceConnectionReceivingTimeout as a RTCConfiguration parameter.
BUG= 4901
Review URL: https://codereview.webrtc.org/1315503003
Cr-Commit-Position: refs/heads/master@{#9832}
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index ee81f7c..a3af0e7 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -233,6 +233,9 @@
// TODO(hbos): Change into class with private data and public getters.
struct RTCConfiguration {
+ static const int kUndefined = -1;
+ // Default maximum number of packets in the audio jitter buffer.
+ static const int kAudioJitterBufferMaxPackets = 50;
// TODO(pthatcher): Rename this ice_transport_type, but update
// Chromium at the same time.
IceTransportsType type;
@@ -247,6 +250,7 @@
TcpCandidatePolicy tcp_candidate_policy;
int audio_jitter_buffer_max_packets;
bool audio_jitter_buffer_fast_accelerate;
+ int ice_connection_receiving_timeout;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
RTCConfiguration()
@@ -255,8 +259,9 @@
bundle_policy(kBundlePolicyBalanced),
rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
tcp_candidate_policy(kTcpCandidatePolicyEnabled),
- audio_jitter_buffer_max_packets(50),
- audio_jitter_buffer_fast_accelerate(false) {}
+ audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
+ audio_jitter_buffer_fast_accelerate(false),
+ ice_connection_receiving_timeout(kUndefined) {}
};
struct RTCOfferAnswerOptions {
@@ -358,8 +363,6 @@
// The |observer| callback will be called when done.
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) = 0;
- // Sets the ICE connection receiving timeout value in milliseconds.
- virtual void SetIceConnectionReceivingTimeout(int timeout_ms) {}
// Restarts or updates the ICE Agent process of gathering local candidates
// and pinging remote candidates.
virtual bool UpdateIce(const IceServers& configuration,
