[wip] now display user's camera

next tricky thing is connecting users

priv/static/index.html

@@ -17,6 +17,7 @@
 
         <ul>
             <li><a href="/echo.html">Echo</a></li>
+            <li><a href="/webrtc.html">WebRTC</a></li>
             <li><a href="/wfc.html">WFC</a></li>
         </ul>
     </div>

priv/static/js/dist/libfewd.js.map

@@ -1 +1 @@
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priv/static/js/dist/webrtc.d.ts

@@ -0,0 +1,22 @@
+/**
+ * webrtc page script
+ *
+ * Author: Peter Harpending <peterharpending@qpq.swiss>
+ * Date: 2026-02-04
+ * Copyright: Copyright (c) 2026 QPQ AG
+ *
+ * Reference: https://git.qpq.swiss/QPQ-AG/research-megadoc/src/commit/c7c4592d4b21ad120145ef63334471a1a7ec1e60/paste/2026-02/grok-webrtc.html
+ *
+ * @module
+ */
+declare function main(): Promise<void>;
+/**
+ * Try to get the user's camera
+ *
+ * tries to get screenshare if user camera not available for whatever reason
+ *
+ * chimps out if
+ * - not on https
+ * - user says no
+ */
+declare function getUserCamera(): Promise<MediaStream>;

priv/static/js/dist/webrtc.js

@@ -0,0 +1,41 @@
+"use strict";
+/**
+ * webrtc page script
+ *
+ * Author: Peter Harpending <peterharpending@qpq.swiss>
+ * Date: 2026-02-04
+ * Copyright: Copyright (c) 2026 QPQ AG
+ *
+ * Reference: https://git.qpq.swiss/QPQ-AG/research-megadoc/src/commit/c7c4592d4b21ad120145ef63334471a1a7ec1e60/paste/2026-02/grok-webrtc.html
+ *
+ * @module
+ */
+main();
+async function main() {
+    console.log('peepy');
+    let video = await getUserCamera();
+    console.log('poopy');
+    console.log('poopu');
+    let moi = document.getElementById('me');
+    moi.srcObject = video;
+    console.log('poopa');
+}
+/**
+ * Try to get the user's camera
+ *
+ * tries to get screenshare if user camera not available for whatever reason
+ *
+ * chimps out if
+ * - not on https
+ * - user says no
+ */
+async function getUserCamera() {
+    if (!(navigator.mediaDevices)) {
+        console.error('navigator.mediaDevices is null; user not on https or something');
+    }
+    let availableDevices = await navigator.mediaDevices.enumerateDevices();
+    console.log(availableDevices);
+    return await navigator.mediaDevices.getUserMedia({ video: true, audio: false });
+    // return await navigator.mediaDevices.getDisplayMedia();
+}
+//# sourceMappingURL=webrtc.js.map

priv/static/js/dist/webrtc.js.map

@@ -0,0 +1 @@
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priv/static/js/dist/wfc.js.map

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priv/static/js/ts/webrtc.ts

@@ -0,0 +1,54 @@
+/**
+ * webrtc page script
+ *
+ * Author: Peter Harpending <peterharpending@qpq.swiss>
+ * Date: 2026-02-04
+ * Copyright: Copyright (c) 2026 QPQ AG
+ *
+ * Reference: https://git.qpq.swiss/QPQ-AG/research-megadoc/src/commit/c7c4592d4b21ad120145ef63334471a1a7ec1e60/paste/2026-02/grok-webrtc.html
+ *
+ * @module
+ */
+
+main();
+
+async function
+main
+    ()
+    : Promise<void>
+{
+    console.log('peepy');
+    let video : MediaStream      = await getUserCamera();
+    console.log('poopy');
+    console.log('poopu');
+    let moi = document.getElementById('me') as HTMLVideoElement;
+    moi.srcObject = video;
+    console.log('poopa');
+}
+
+
+/**
+ * Try to get the user's camera
+ *
+ * tries to get screenshare if user camera not available for whatever reason
+ *
+ * chimps out if
+ * - not on https
+ * - user says no
+ */
+async function
+getUserCamera
+    ()
+    : Promise<MediaStream>
+{
+    if (!(navigator.mediaDevices)) {
+        console.error('navigator.mediaDevices is null; user not on https or something');
+    }
+
+    let availableDevices: Array<MediaDeviceInfo> =
+        await navigator.mediaDevices.enumerateDevices();
+    console.log(availableDevices);
+
+    return await navigator.mediaDevices.getUserMedia({video: true, audio: false});
+    // return await navigator.mediaDevices.getDisplayMedia();
+}

priv/static/webrtc.html

@@ -0,0 +1,27 @@
+<!DOCTYPE html>
+<html lang="en">
+<head>
+    <meta charset="utf-8">
+    <title>FEWD: voice chat demo</title>
+    <link rel="stylesheet" href="/css/default.css">
+</head>
+<body>
+    <div id="titlebar">
+        <div class="content">
+            <a href="/" class="tb-home">Home</a>
+        </div>
+    </div>
+
+    <div class="content">
+        <h1 class="content-title">FEWD: webrtc demo</h1>
+
+        <div class="content-body">
+            <textarea disabled id="wfc-output"></textarea>
+
+            <video id="me" autoplay muted></video>
+        </div>
+    </div>
+
+    <script src="/js/dist/webrtc.js"></script>
+</body>
+</html>

scratch/grok-webrtc.html

@@ -0,0 +1,166 @@
+<!DOCTYPE html>
+<html lang="en">
+<head>
+  <meta charset="UTF-8" />
+  <title>Super Simple WebRTC – Manual Signaling</title>
+  <style>
+    body { font-family: sans-serif; margin: 20px; }
+    textarea { width: 100%; height: 90px; font-family: monospace; }
+    button { margin: 6px 0; padding: 8px 16px; }
+    video { max-width: 320px; border: 1px solid #444; margin: 8px; background: #000; }
+  </style>
+</head>
+<body>
+
+<h2>WebRTC 1:1 test (copy-paste signaling)</h2>
+
+<div>
+  <button id="btnCreateOffer">1. Create Offer (Peer A)</button>
+  <button id="btnSetRemoteOffer">2. Paste offer → Set remote description</button><br>
+  <textarea id="offerSdp" placeholder="Offer SDP will appear here (Peer A) or paste offer here (Peer B)"></textarea>
+</div>
+
+<div style="margin-top: 24px;">
+  <button id="btnCreateAnswer">3. Create Answer (Peer B)</button>
+  <button id="btnSetRemoteAnswer">4. Paste answer → Set remote description</button><br>
+  <textarea id="answerSdp" placeholder="Answer SDP will appear here (Peer B) or paste answer here (Peer A)"></textarea>
+</div>
+
+<div style="margin-top: 20px;">
+  <button id="btnAddIce">Add ICE candidate manually (if needed)</button><br>
+  <textarea id="iceInput" placeholder="Paste remote ICE candidate here"></textarea>
+  <div id="iceLog"></div>
+</div>
+
+<div style="margin-top: 20px;">
+  <video id="localVideo" autoplay playsinline muted></video>
+  <video id="remoteVideo" autoplay playsinline></video>
+</div>
+
+<pre id="status">Status: ready</pre>
+
+<script>
+// ────────────────────────────────────────────────
+const $ = s => document.querySelector(s);
+const status = $('pre#status');
+const offerArea  = $('#offerSdp');
+const answerArea = $('#answerSdp');
+const iceInput   = $('#iceInput');
+const iceLog     = $('#iceLog');
+
+let pc = null;
+let localStream = null;
+
+// ────────────────────────────────────────────────
+async function initPeerConnection() {
+  if (pc) pc.close();
+
+  pc = new RTCPeerConnection({
+    iceServers: [
+      { urls: "stun:stun.l.google.com:19302" },
+      { urls: "stun:stun.stunprotocol.org" }
+    ]
+  });
+
+  pc.onicecandidate = e => {
+    if (e.candidate) {
+      console.log("New ICE candidate:", e.candidate);
+      iceLog.innerHTML += "<div style='color:#c33'>→ " + e.candidate.candidate + "</div>";
+    }
+  };
+
+  pc.oniceconnectionstatechange = () => {
+    status.textContent = "ICE state: " + pc.iceConnectionState;
+  };
+
+  pc.ontrack = e => {
+    console.log("ontrack", e);
+    const remoteVideo = $('#remoteVideo');
+    remoteVideo.srcObject = e.streams[0];
+    status.textContent = "ICE state: " + pc.iceConnectionState + " (receiving media!)";
+  };
+
+  // Add local stream if we have camera/mic
+  if (localStream) {
+    localStream.getTracks().forEach(track => {
+      pc.addTrack(track, localStream);
+    });
+  }
+}
+
+// ────────────────────────────────────────────────
+async function startLocalCamera() {
+  try {
+    localStream = await navigator.mediaDevices.getUserMedia({
+      video: true,
+      audio: false    // change to true if you want audio too
+    });
+    $('#localVideo').srcObject = localStream;
+  } catch (err) {
+    console.error("Camera access failed", err);
+    status.textContent = "Camera access failed: " + err.message;
+  }
+}
+
+// ────────────────────────────────────────────────
+$('#btnCreateOffer').onclick = async () => {
+  await initPeerConnection();
+  await startLocalCamera();
+
+  const offer = await pc.createOffer();
+  await pc.setLocalDescription(offer);
+
+  offerArea.value = JSON.stringify(pc.localDescription, null, 2);
+  status.textContent = "Offer created – copy and send to peer B";
+};
+
+// ────────────────────────────────────────────────
+$('#btnCreateAnswer').onclick = async () => {
+  await initPeerConnection();
+  await startLocalCamera();
+
+  const offer = JSON.parse(offerArea.value);
+  await pc.setRemoteDescription(offer);
+
+  const answer = await pc.createAnswer();
+  await pc.setLocalDescription(answer);
+
+  answerArea.value = JSON.stringify(pc.localDescription, null, 2);
+  status.textContent = "Answer created – copy and send to peer A";
+};
+
+// ────────────────────────────────────────────────
+$('#btnSetRemoteOffer').onclick = async () => {
+  if (!pc) await initPeerConnection();
+
+  const remote = JSON.parse(offerArea.value);
+  await pc.setRemoteDescription(remote);
+  status.textContent = "Remote offer set";
+};
+
+// ────────────────────────────────────────────────
+$('#btnSetRemoteAnswer').onclick = async () => {
+  const remote = JSON.parse(answerArea.value);
+  await pc.setRemoteDescription(remote);
+  status.textContent = "Remote answer set – waiting for ICE/media";
+};
+
+// ────────────────────────────────────────────────
+$('#btnAddIce').onclick = async () => {
+  if (!iceInput.value.trim()) return;
+  try {
+    const candidate = new RTCIceCandidate(JSON.parse(iceInput.value));
+    await pc.addIceCandidate(candidate);
+    iceLog.innerHTML += "<div style='color:#3a3'>← added ICE candidate</div>";
+    iceInput.value = "";
+  } catch (err) {
+    console.error("Bad ICE candidate", err);
+  }
+};
+
+// Optional: auto-start camera when page loads
+// startLocalCamera();
+
+</script>
+</body>
+</html>

scratch/webrtc01.ts

@@ -0,0 +1,127 @@
+/**
+ * webrtc page script
+ *
+ * Author: Peter Harpending <peterharpending@qpq.swiss>
+ * Date: 2026-02-04
+ * Copyright: Copyright (c) 2026 QPQ AG
+ *
+ * Reference: https://git.qpq.swiss/QPQ-AG/research-megadoc/src/commit/c7c4592d4b21ad120145ef63334471a1a7ec1e60/paste/2026-02/grok-webrtc.html
+ *
+ * @module
+ */
+
+type state =
+    {whoami: string,
+     roster: Array<string>};
+
+let st: state =
+    {whoami: "",
+     roster: []};
+
+main();
+
+async function
+main
+    ()
+    : Promise<void>
+{
+    // start websocket immediately
+    let ws: WebSocket;
+    let init = document.getElementById('init') as HTMLDivElement;
+    let init_join = document.getElementById('init-join') as HTMLButtonElement;
+
+    // handle button click
+    init_join.addEventListener('click',
+        function() {
+            init.hidden = true;
+            document.getElementById('run')!.hidden = false;
+            ws = new WebSocket('/ws/webrtc');
+            ws.onopen    = function(e) { console.log('ws open'); };
+            ws.onclose   = function(e) { console.warn('ws closed'); };
+            ws.onerror   = function(e) { console.error('ws error:', e); };
+            ws.onmessage =
+                function(e) {
+                    // console.log('ws message:', e.data);
+                    let message = JSON.parse(e.data) as ws_msg;
+                    handle_ws_msg(message);
+                };
+        }
+    );
+}
+
+type ws_msg = ["whoami", string]
+            | ["roster", Array<string>];
+
+function
+handle_ws_msg
+    (message: ws_msg)
+    : void
+{
+    console.log('message from server:', message);
+    switch(message[0]) {
+        case "whoami":
+            st.whoami = message[1];
+            break;
+        case "roster":
+            st.roster = (message[1] as Array<string>);
+            break;
+        default:
+            console.warn("unknown message", message);
+    }
+
+    render_state();
+}
+
+
+
+function
+ws_send_json
+    (ws : WebSocket,
+     x  : any)
+    : void
+{
+    let s: string = JSON.stringify(x, undefined, 4);
+    console.log('sending:\n', s);
+
+    ws.send(s);
+}
+
+
+
+function
+render_state
+    ()
+    : void
+{
+    console.log('st', st);
+    // whoami
+    document.getElementById('whoami')!.innerText = st.whoami;
+    //
+    let roster_ul = document.getElementById('roster-ul') as HTMLUListElement;
+    let newChildren : Array<HTMLElement> = [];
+    for (let nick of st.roster) {
+        if (!(nick === st.whoami)) {
+            let li = nickkk(nick);
+            newChildren.push(li);
+        }
+    }
+    roster_ul.replaceChildren(...newChildren);
+}
+
+function
+nickkk
+    (nick: string)
+    : HTMLElement
+{
+    let li = document.createElement('li');
+
+    li.innerText += nick;
+    li.innerText += ' ';
+
+    let call_a = document.createElement('button');
+    call_a.innerText = 'call';
+
+    li.appendChild(call_a);
+
+    return li;
+}

src/fd_httpd.erl

@@ -23,6 +23,12 @@ start_link() ->
 
 init([]) ->
     RestartStrategy = {one_for_one, 1, 60},
+    WebRTC    = {fd_httpd_webrtc,
+                 {fd_httpd_webrtc, start_link, []},
+                 permanent,
+                 5000,
+                 worker,
+                 [fd_httpd_webrtc]},
     FileCache = {fd_httpd_sfc,
                  {fd_httpd_sfc, start_link, []},
                  permanent,
@@ -35,5 +41,6 @@ init([]) ->
                  5000,
                  supervisor,
                  [fd_httpd_clients]},
-    Children  = [FileCache, Clients],
+    Children  = [WebRTC, FileCache, Clients],
+    %Children  = [FileCache, Clients],
     {ok, {RestartStrategy, Children}}.

src/fd_httpd_client.erl

@@ -135,9 +135,19 @@ loop(Parent, Debug, State = #s{socket = Socket, next = Next0}) ->
             Received = <<Next0/binary, Message/binary>>,
             case qhl:parse(Socket, Received) of
                 {ok, Req, Next1} ->
-                    Next2 = handle_request(Socket, Req, Next1),
+                    %% FIXME: unfuck received logic here
+                    %% handle_request should eventually call back into
+                    %% loop/3 or close the socket
+                    %%
+                    %% also loop/3 should be loop/1, because we need to pass
+                    %% the state down the callchains
+                    Next2 =
+                        case Next1 of
+                            none -> <<>>;
+                            Bin  -> Bin
+                        end,
                     NewState = State#s{next = Next2},
-                    loop(Parent, Debug, NewState);
+                    handle_request(Req, Parent, Debug, NewState);
                 Error ->
                     %% should trigger bad request
                     tell(error, "~p QHL parse error: ~tp", [?LINE, Error]),
@@ -210,40 +220,44 @@ system_replace_state(StateFun, State) ->
 
 %%% http request handling
 
--spec handle_request(Sock, Request, Received) -> NewReceived
-    when Sock        :: gen_tcp:socket(),
-         Request     :: request(),
-         Received    :: binary(),
-         NewReceived :: binary().
+-spec handle_request(Request, Parent, Debug, State) -> no_return()
+    when Request :: request(),
+         Parent  :: pid(),
+         Debug   :: [sys:dbg_opt()],
+         State   :: state().
 
-handle_request(Sock, R = #request{method = M, path = P}, Received) when M =/= undefined, P =/= undefined ->
-    tell("~tp ~tp ~ts", [self(), M, P]),
-    route(Sock, M, P, R, Received).
+handle_request(Req = #request{method = Method, path = Path}, Parent, Debug, State)
+        when Method =/= undefined,
+             Path   =/= undefined ->
+    tell("~tp ~tp ~ts", [self(), Method, Path]),
+    route(Method, Path, Req, Parent, Debug, State).
 
 
-
--spec route(Sock, Method, Route, Request, Received) -> NewReceived
-    when Sock        :: gen_tcp:socket(),
-         Method      :: get | post,
-         Route       :: binary(),
-         Request     :: request(),
-         Received    :: binary(),
-         NewReceived :: binary().
-
-route(Sock, get, Route, Request, Received) ->
+% hardcode routes first
+route(get, Route, Request, P, D, S = #s{socket = Sock}) ->
     case Route of
-         <<"/ws/echo">>   -> ws_echo(Sock, Request)                 , Received;
-         <<"/">>          -> route_static(Sock, <<"/index.html">>)  , Received;
-         _                -> route_static(Sock, Route)              , Received
+         <<"/ws/echo">>   -> ws_echo(Sock, Request);
+         <<"/ws/webrtc">> ->
+            ws_webrtc(Request, S);
+         <<"/">> ->
+            route_static(Sock, <<"/index.html">>),
+            loop(P, D, S);
+         _ ->
+            route_static(Sock, Route),
+            loop(P, D, S)
     end;
-route(Sock, post, Route, Request, Received) ->
+route(post, Route, Request, P, D, S = #s{socket = Sock}) ->
     case Route of
-         <<"/wfcin">>       -> wfcin(Sock, Request) , Received;
-         _                  -> fd_httpd_utils:http_err(Sock, 404)  , Received
+         <<"/wfcin">> ->
+            wfcin(Sock, Request),
+            loop(P, D, S);
+         _ ->
+            fd_httpd_utils:http_err(Sock, 404),
+            loop(P, D, S)
     end;
-route(Sock, _, _, _, Received) ->
+route(_, _, _, P, D, S = #s{socket = Sock}) ->
     fd_httpd_utils:http_err(Sock, 404),
-    Received.
+    loop(P, D, S).
 
 
 
@@ -279,6 +293,69 @@ respond_static(Sock, not_found) ->
     fd_httpd_utils:http_err(Sock, 404).
 
 
+%% ------------------------------
+%% webrtc
+%% ------------------------------
+
+ws_webrtc(Request, State = #s{socket = Sock}) ->
+    try
+        case qhl_ws:handshake(Request) of
+            {ok, Response} ->
+                fd_httpd_utils:respond(Sock, Response),
+                % wait for username
+                ok = fd_httpd_webrtc:join(),
+                ws_webrtc_loop(State);
+            Error ->
+                tell("ws_webrtc: error: ~tp", [Error]),
+                fd_httpd_utils:http_err(Sock, 400),
+                exit(bad_request)
+        end
+    catch
+        exit:Y:Z ->
+            tell("~tp ws_webrtc: dying peacefully: ~tp:~tp", [self(), Y, Z]),
+            gen_tcp:close(Sock);
+        error:Y:Z ->
+            tell(error, "~tp ws_webrtc: ERROR ~tp:~tp", [self(), Y, Z]),
+            fd_httpd_utils:http_err(Sock, 500),
+            error(internal_server_error)
+    end.
+
+
+-spec ws_webrtc_loop(state()) -> no_return().
+
+% we have no tcp bytes waiting to be parsed
+ws_webrtc_loop(State = #s{socket = Socket, next = <<>>}) ->
+    ok = inet:setopts(Socket, [{active, once}]),
+    receive
+        {tcp, Socket, Message} ->
+            NewState = State#s{next = Message},
+            ws_webrtc_loop(NewState);
+        {webrtc, Message} ->
+            ok = ws_webrtc_relay(Socket, s2c, Message),
+            ws_webrtc_loop(State);
+        {tcp_closed, Socket} ->
+            ok = tell("~p Socket closed, retiring.~n", [self()]),
+            exit(normal)
+    end;
+% we have TCP bytes sitting and waiting to be parsed
+ws_webrtc_loop(State = #s{socket = Sock, next = Recv}) ->
+    {ok, Message, NewRecv} = qhl_ws:recv_json(Sock, Recv, 5000),
+    ok = ws_webrtc_relay(Sock, c2s, Message),
+    ws_webrtc_loop(State#s{next = NewRecv}).
+
+
+% relay
+% message is a zj:value()
+ws_webrtc_relay(Socket, Direction, ZJValue) ->
+    tell("~p ws_webrtc_relay ~p: ~p", [self(), Direction, ZJValue]),
+    case Direction of
+        c2s -> fd_httpd_webrtc:hi(ZJValue);
+        s2c -> qhl_ws:send_json(Socket, ZJValue)
+    end,
+    ok.
+
+
+
 %% ------------------------------
 %% echo
 %% ------------------------------

src/fd_httpd_webrtc.erl

@@ -0,0 +1,165 @@
+% @doc webrtc peer pool
+-module(fd_httpd_webrtc).
+
+-behavior(gen_server).
+
+%% api (caller context)
+-export([
+    join/0, hi/1
+]).
+
+% startup/gen_server callbacks
+-export([
+    % caller context
+    start_link/0,
+    % process context
+    init/1, handle_call/3, handle_cast/2, handle_info/2,
+    code_change/3, terminate/2
+]).
+
+-include("$zx_include/zx_logger.hrl").
+
+
+-record(u, {pid :: pid()}).
+-type user() :: #u{}.
+
+-record(s, {roster = [] :: [user()]}).
+-type state() :: #s{}.
+
+
+%%-----------------------------------------------------------------------------
+%% caller context
+%%-----------------------------------------------------------------------------
+
+
+start_link() ->
+    gen_server:start_link({local, ?MODULE}, ?MODULE, none, []).
+
+
+
+-spec join() -> ok | {error, any()}.
+
+join() ->
+    gen_server:call(?MODULE, join).
+
+
+
+-spec hi(Message) -> ok
+    when Message :: zj:value().
+
+hi(Message) ->
+    gen_server:cast(?MODULE, {hi, self(), Message}).
+
+
+
+%%-----------------------------------------------------------------------------
+%% process context
+%%-----------------------------------------------------------------------------
+
+%% gen_server callbacks
+
+init(none) ->
+    tell("starting fd_httpd_webrtc", []),
+    InitState = #s{},
+    {ok, InitState}.
+
+
+handle_call(join, _From = {PID, _Tag}, State) ->
+    {Reply, NewState} = do_join(PID, State),
+    {reply, Reply, NewState};
+handle_call(Unexpected, From, State) ->
+    tell("~tp: unexpected call from ~tp: ~tp", [?MODULE, Unexpected, From]),
+    {noreply, State}.
+
+
+handle_cast(Unexpected, State) ->
+    tell("~tp: unexpected cast: ~tp", [?MODULE, Unexpected]),
+    {noreply, State}.
+
+
+handle_info({'DOWN', _Ref, process, PID, _Reason}, State) ->
+    NewState = do_down(PID, State),
+    {noreply, NewState};
+handle_info(Unexpected, State) ->
+    tell("~tp: unexpected info: ~tp", [?MODULE, Unexpected]),
+    {noreply, State}.
+
+
+code_change(_, State, _) ->
+    {ok, State}.
+
+terminate(_, _) ->
+    ok.
+
+
+%%---------------------
+%% doers
+%%---------------------
+
+-spec do_join(PID, State) -> {Reply, NewState}
+    when PID      :: pid(),
+         State    :: state(),
+         Reply    :: ok
+                   | {error, any()},
+         NewState :: state().
+% @private join a user to a pool
+
+do_join(PID, State = #s{roster = Users}) ->
+    % see if pid is already there
+    case lists:keymember(PID, #u.pid, Users) of
+        true -> {error, already_joined};
+        false -> do_join2(PID, State)
+    end.
+
+do_join2(PID, State = #s{roster = Users}) ->
+    monitor(process, PID),
+    NewUser = #u{pid = PID},
+    NewRoster = usort([NewUser | Users]),
+    NewState  = State#s{roster = NewRoster},
+    ok = whisper(whoami, PID),
+    ok = gossip(roster, NewState),
+    {ok, NewState}.
+
+usort(Users) ->
+    lists:keysort(#u.pid, Users).
+
+
+-spec do_down(PID, State) -> NewState
+    when PID      :: pid(),
+         State    :: state(),
+         NewState :: state().
+% @private handle a user dying
+
+do_down(PID, State = #s{roster = Users}) ->
+    % remove from roster
+    NewUsers = lists:keydelete(PID, #u.pid, Users),
+    NewState = State#s{roster = NewUsers},
+    % broadcast username
+    ok = gossip(roster, NewState),
+    NewState.
+
+
+% @doc send a message to a specific person
+whisper(whoami, PID) ->
+    PID ! {webrtc, ["whoami", pidstr(PID)]},
+    ok.
+
+
+% @doc send a message to everyone
+gossip(roster, State = #s{roster = Us}) ->
+    Pids = [pidstr(PID) || #u{pid = PID} <- Us],
+    gossip_json(["roster", Pids], State).
+
+pidstr(PID) ->
+    unicode:characters_to_list(io_lib:format("~tp", [PID])).
+
+
+-spec gossip_json(zj:value(), state()) -> ok.
+% @private gossip a message to everyone
+
+gossip_json(Message, #s{roster = Users}) ->
+    GossipTo =
+        fun(#u{pid = PID}) ->
+            PID ! {webrtc, Message}
+        end,
+    lists:foreach(GossipTo, Users).

src/qhl_ws.erl

@@ -16,7 +16,9 @@
     %% porcelain
     handshake/1,
     recv/3, recv/4,
-    send/2
+    recv_strict/3, recv_json/3,
+    send/2,
+    send_dwim/2, send_json/2
 ]).
 
 -include("http.hrl").
@@ -259,6 +261,54 @@ response_token(ChallengeToken) when is_binary(ChallengeToken) ->
     base64:encode(Sha1).
 
 
+-spec recv_json(Socket, Received, TimeoutMS) -> Result
+    when Socket    :: gen_tcp:socket(),
+         Received  :: binary(),
+         TimeoutMS :: non_neg_integer(),
+         Result    :: {ok, zj:value(), Remainder}
+                    | {error, Reason},
+         Remainder :: binary(),
+         Reason    :: any().
+% @doc
+% asserts response is text
+
+recv_json(Sock, Recv, TimeoutMS) ->
+    case recv_strict(Sock, Recv, TimeoutMS) of
+        {ok, {Type, Payload}, NewRecv} when Type =:= text orelse
+                                            Type =:= binary ->
+            case zj:decode(Payload) of
+                {ok, Value} ->
+                    io:format("~p value: ~p~n", [self(), Value]),
+                    {ok, Value, NewRecv};
+                Error       -> {error, {zj_decode, Error}}
+            end;
+        Error ->
+            Error
+    end.
+
+
+-spec recv_strict(Socket, Received, TimeoutMS) -> Result
+    when Socket    :: gen_tcp:socket(),
+         Received  :: binary(),
+         TimeoutMS :: non_neg_integer(),
+         Result    :: {ok, Message, Remainder}
+                    | {error, Reason},
+         Message   :: ws_msg(),
+         Remainder :: binary(),
+         Reason    :: any().
+% @doc
+% Almost equivalent to recv/3, but asserts resulting frames are empty
+
+recv_strict(Sock, Recv, TimeoutMS) ->
+    case recv(Sock, Recv, TimeoutMS) of
+        {ok, Message, [], Remainder} ->
+            {ok, Message, Remainder};
+        Illegal = {ok, _, _NonEmptyFrames, _} ->
+            {error, {bad_frame_stack, Illegal}};
+        Error ->
+            Error
+    end.
+
 
 -spec recv(Socket, Received, TimeoutMS) -> Result
     when Socket    :: gen_tcp:socket(),
@@ -604,10 +654,15 @@ recv_frame(Frame = #frame{payload_length = Len, payload = none}, Sock, Received,
 
 %% factoring this out into a function to reduce repetition
 recv_frame_await(Frame, Sock, Received, Timeout) ->
+    io:format("~p called: recv_frame_await(~p, ~p, ~p, ~p)~n",
+              [self(), Frame, Sock, Received, Timeout]),
     case inet:setopts(Sock, [{active, once}]) of
         ok ->
             receive
-                {tcp, Sock, Bin}          -> recv_frame(Frame, Sock, <<Received/bits, Bin/binary>>, Timeout);
+                {tcp, Sock, Bin}          ->
+                    io:format("~p calling: recv_frame(~p, ~p, ~p, ~p)~n",
+                        [self(),Frame, Sock, <<Received/bits, Bin/binary>>, Timeout]),
+                    recv_frame(Frame, Sock, <<Received/bits, Bin/binary>>, Timeout);
                 {tcp_closed, Sock}        -> {error, tcp_closed};
                 {tcp_error, Sock, Reason} -> {error, {tcp_error, Reason}}
             after Timeout ->
@@ -618,6 +673,34 @@ recv_frame_await(Frame, Sock, Received, Timeout) ->
     end.
 
 
+send_json(Socket, X) ->
+    send_dwim(Socket, {json, X}).
+
+
+-spec send_dwim(Socket, Message) -> Result
+    when Socket  :: gen_tcp:socket(),
+         Message :: string()
+                  | binary()
+                  | {json, zj:value()}
+                  | ws_msg(),
+         Result  :: ok
+                  | {error, Reason},
+         Reason  :: any().
+% @doc
+% equivalent to send/2 but assumes iolists/strings are meant to be `text`, and
+% naked binaries are meant to be `binary`
+%
+% lists are assumed to be unicode iolists and are converted to strings via
+% unicode:characters_to_list
+%
+% json is encoded as text and sent as such
+% @end
+
+send_dwim(Socket, X) when is_list(X)   -> send(Socket, {text, X});
+send_dwim(Socket, X) when is_binary(X) -> send(Socket, {binary, X});
+send_dwim(Socket, {json, X})           -> send(Socket, {text, zj:encode(X)});
+send_dwim(Socket, {Type, Payload})     -> send(Socket, {Type, Payload}).
+
 
 -spec send(Socket, Message) -> Result
     when Socket  :: gen_tcp:socket(),