// Copyright 2017 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
module blink.mojom;
import "mojo/public/mojom/base/values.mojom";
// This enum is a typemap of base::PowerObserver::DeviceThermalState.
enum DeviceThermalState {
kUnknown,
kNominal,
kFair,
kSerious,
kCritical,
};
// This constant mirrors base::PowerThermalObserver::kSpeedLimitMax
const int32 kSpeedLimitMax = 100;
struct PeerConnectionInfo {
// ID of the peer connection. Unique only within the renderer process.
int32 lid;
// Serialized version of RTCConfiguration.
string rtc_configuration;
// The URL of the blink::WebLocalFrame within which this peer connection
// lives. Used for debugging purposes (displayed by WebRTC-Internals).
// May be null for eg pages with opaque URLs.
// TODO(1274640): Use a mojo URL instead of a string.
string? url;
};
// This interface collects data about each peer connection and sends it to the
// browser process.
interface PeerConnectionManager {
// Called when the browser process reports a suspend event from the OS.
OnSuspend();
// Called when the browser process reports a new thermal state from the OS.
OnThermalStateChange(DeviceThermalState thermal_state);
// Called when the browser process reports a new CPU speed limit change from
// the OS. Values are normally between 0 and 100.
OnSpeedLimitChange(int32 speed_limit);
// Enable WebRtc event log output.
StartEventLog(int32 peer_connection_local_id, int32 output_period_ms);
// Disable WebRtc event log output.
StopEventLog(int32 peer_connection_local_id);
// Requests standard PeerConnection stats for webrtc-internals.
GetStandardStats();
// Requests a state dump for webrtc-internals.
GetCurrentState();
};
// This interface allows forwarding PeerConnection events to WebRTCInternals in
// the browser process.
interface PeerConnectionTrackerHost {
// PeerConnectionTrackerHost forwards these events to WebRTCInternals.
// TODO(vm.arjun): Migrate rest of the messages,
// https://bugs.chromium.org/p/chromium/issues/detail?id=792801
// Notifies WebRTCInternals about the addition of the peer connection
// whose data is specified in |info|.
AddPeerConnection(PeerConnectionInfo info);
// Notifies WebRTCInternals about the removal of the peer connection
// identified with local id |lid|.
RemovePeerConnection(int32 lid);
// This method is called to update PeerConnection.
// |lid| is the ID of the peer connection
// |type| is the update type. ex: "createOffer"
// |value| is the detail of the update.
// ex: "options: {offerToReceiveAudio: 0}"
UpdatePeerConnection(int32 lid, string type, string value);
// This method is called to inform that a peer connection's session
// description has been assigned an ID.
OnPeerConnectionSessionIdSet(int32 lid, string session_id);
// Lets WebRTCInternals know that a getUserMedia() call has been made.
// |request_id| is an id assigned to the getusermedia call and its
// callback/error
// |audio| is true if audio stream is requested
// |video| is true if video stream is requested
// |audio_constraints| is constraints for audio like audio source device id
// |video_constraints| is constraints for video like audio source device id
// The constraint strings are for display only and should not be parsed
// by the browser for security reasons.
GetUserMedia(int32 request_id, bool audio, bool video,
string audio_constraints, string video_constraints);
// Lets WebRTCInternals know that a getUserMedia() call has resolved.
// |request_id| is an id assigned to the getusermedia call and its
// callback/error
// |stream_id| is information about the stream such as the stream id.
// |audio_track_info| describes the streams audio track (if any).
// |video_track_info| describes the streams video track (if any).
// The stream_id, audio_track_info and video_track_info strings are
// for display only and should not be parsed by the browser for
// security reasons.
GetUserMediaSuccess(int32 request_id, string stream_id,
string audio_track_info, string video_track_info);
// Lets WebRTCInternals know that a getUserMedia() call has failed.
// |request_id| is an id assigned to the getusermedia call and its
// callback/error
// |error| the (DOM) error.
// |error_message| the error message.
// The error and error message strings are for display only and should not
// be parsed by the browser for security reasons.
GetUserMediaFailure(int32 request_id, string error, string error_message);
// Lets WebRTCInternals know that a getUserMedia() call has been made.
// |request_id| is an id assigned to the getusermedia call and its
// callback/error
// |audio| is true if audio stream is requested
// |video| is true if video stream is requested
// |audio_constraints| is constraints for audio like audio source device id
// |video_constraints| is constraints for video like audio source device id
// The constraint strings are for display only and should not be parsed
// by the browser for security reasons.
GetDisplayMedia(int32 request_id, bool audio, bool video,
string audio_constraints, string video_constraints);
// Lets WebRTCInternals know that a getDisplayMedia() call has resolved.
// |request_id| is an id assigned to the getusermedia call and its
// callback/error
// |stream_id| is information about the stream such as the stream id.
// |audio_track_info| describes the streams audio track (if any).
// |video_track_info| describes the streams video track (if any).
// The stream_id, audio_track_info and video_track_info strings are
// for display only and should not be parsed by the browser for
// security reasons.
GetDisplayMediaSuccess(int32 request_id, string stream_id,
string audio_track_info, string video_track_info);
// Lets WebRTCInternals know that a getDisplayMedia() call has failed.
// |request_id| is an id assigned to the getusermedia call and its
// callback/error
// |error| the (DOM) error.
// |error_message| the error message.
// The error and error message strings are for display only and should not
// be parsed by the browser for security reasons.
GetDisplayMediaFailure(int32 request_id, string error, string error_message);
// Logs WebRtc log for the connection id.
WebRtcEventLogWrite(int32 lid, array<uint8> output);
// Adds standard PeerConnection stats for webrtc-internals.
// |lid| is the renderer local id,
// |value| is the list of stats reports.
AddStandardStats(int32 lid, mojo_base.mojom.ListValue value);
// Adds legacy PeerConnection stats for webrtc-internals.
// |lid| is the renderer local id,
// |value| is the list of stats reports.
AddLegacyStats(int32 lid, mojo_base.mojom.ListValue value);
};
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