webrtc-unidirectional-h264.c 21.4 KB
Newer Older
Jan Schmidt's avatar
Jan Schmidt committed
1
2
3
4
5
#include <locale.h>
#include <glib.h>
#include <gst/gst.h>
#include <gst/sdp/sdp.h>

6
7
8
9
#ifdef G_OS_UNIX
#include <glib-unix.h>
#endif

Jan Schmidt's avatar
Jan Schmidt committed
10
11
12
13
14
15
16
17
#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>

#include <libsoup/soup.h>
#include <json-glib/json-glib.h>
#include <string.h>

#define RTP_PAYLOAD_TYPE "96"
18
#define RTP_AUDIO_PAYLOAD_TYPE "97"
Jan Schmidt's avatar
Jan Schmidt committed
19
20
21
#define SOUP_HTTP_PORT 57778
#define STUN_SERVER "stun.l.google.com:19302"

22
23
24
25
26
27
#ifdef G_OS_WIN32
#define VIDEO_SRC "mfvideosrc"
#else
#define VIDEO_SRC "v4l2src"
#endif

28
29
30
31
gchar *video_priority = NULL;
gchar *audio_priority = NULL;


Jan Schmidt's avatar
Jan Schmidt committed
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
typedef struct _ReceiverEntry ReceiverEntry;

ReceiverEntry *create_receiver_entry (SoupWebsocketConnection * connection);
void destroy_receiver_entry (gpointer receiver_entry_ptr);

void on_offer_created_cb (GstPromise * promise, gpointer user_data);
void on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data);
void on_ice_candidate_cb (GstElement * webrtcbin, guint mline_index,
    gchar * candidate, gpointer user_data);

void soup_websocket_message_cb (SoupWebsocketConnection * connection,
    SoupWebsocketDataType data_type, GBytes * message, gpointer user_data);
void soup_websocket_closed_cb (SoupWebsocketConnection * connection,
    gpointer user_data);

void soup_http_handler (SoupServer * soup_server, SoupMessage * message,
    const char *path, GHashTable * query, SoupClientContext * client_context,
    gpointer user_data);
void soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
    SoupWebsocketConnection * connection, const char *path,
    SoupClientContext * client_context, gpointer user_data);

static gchar *get_string_from_json_object (JsonObject * object);

struct _ReceiverEntry
{
  SoupWebsocketConnection *connection;

  GstElement *pipeline;
  GstElement *webrtcbin;
};

const gchar *html_source = " \n \
<html> \n \
  <head> \n \
    <script type=\"text/javascript\" src=\"https://webrtc.github.io/adapter/adapter-latest.js\"></script> \n \
    <script type=\"text/javascript\"> \n \
      var html5VideoElement; \n \
      var websocketConnection; \n \
      var webrtcPeerConnection; \n \
      var webrtcConfiguration; \n \
      var reportError; \n \
 \n \
 \n \
      function onLocalDescription(desc) { \n \
        console.log(\"Local description: \" + JSON.stringify(desc)); \n \
        webrtcPeerConnection.setLocalDescription(desc).then(function() { \n \
          websocketConnection.send(JSON.stringify({ type: \"sdp\", \"data\": webrtcPeerConnection.localDescription })); \n \
        }).catch(reportError); \n \
      } \n \
 \n \
 \n \
      function onIncomingSDP(sdp) { \n \
        console.log(\"Incoming SDP: \" + JSON.stringify(sdp)); \n \
        webrtcPeerConnection.setRemoteDescription(sdp).catch(reportError); \n \
        webrtcPeerConnection.createAnswer().then(onLocalDescription).catch(reportError); \n \
      } \n \
 \n \
 \n \
      function onIncomingICE(ice) { \n \
        var candidate = new RTCIceCandidate(ice); \n \
        console.log(\"Incoming ICE: \" + JSON.stringify(ice)); \n \
        webrtcPeerConnection.addIceCandidate(candidate).catch(reportError); \n \
      } \n \
 \n \
 \n \
      function onAddRemoteStream(event) { \n \
        html5VideoElement.srcObject = event.streams[0]; \n \
      } \n \
 \n \
 \n \
      function onIceCandidate(event) { \n \
        if (event.candidate == null) \n \
          return; \n \
 \n \
        console.log(\"Sending ICE candidate out: \" + JSON.stringify(event.candidate)); \n \
        websocketConnection.send(JSON.stringify({ \"type\": \"ice\", \"data\": event.candidate })); \n \
      } \n \
 \n \
 \n \
      function onServerMessage(event) { \n \
        var msg; \n \
 \n \
        try { \n \
          msg = JSON.parse(event.data); \n \
        } catch (e) { \n \
          return; \n \
        } \n \
 \n \
        if (!webrtcPeerConnection) { \n \
          webrtcPeerConnection = new RTCPeerConnection(webrtcConfiguration); \n \
          webrtcPeerConnection.ontrack = onAddRemoteStream; \n \
          webrtcPeerConnection.onicecandidate = onIceCandidate; \n \
        } \n \
 \n \
        switch (msg.type) { \n \
          case \"sdp\": onIncomingSDP(msg.data); break; \n \
          case \"ice\": onIncomingICE(msg.data); break; \n \
          default: break; \n \
        } \n \
      } \n \
 \n \
 \n \
      function playStream(videoElement, hostname, port, path, configuration, reportErrorCB) { \n \
        var l = window.location;\n \
        var wsHost = (hostname != undefined) ? hostname : l.hostname; \n \
        var wsPort = (port != undefined) ? port : l.port; \n \
        var wsPath = (path != undefined) ? path : \"ws\"; \n \
        if (wsPort) \n\
          wsPort = \":\" + wsPort; \n\
        var wsUrl = \"ws://\" + wsHost + wsPort + \"/\" + wsPath; \n \
 \n \
        html5VideoElement = videoElement; \n \
        webrtcConfiguration = configuration; \n \
        reportError = (reportErrorCB != undefined) ? reportErrorCB : function(text) {}; \n \
 \n \
        websocketConnection = new WebSocket(wsUrl); \n \
        websocketConnection.addEventListener(\"message\", onServerMessage); \n \
      } \n \
 \n \
      window.onload = function() { \n \
        var vidstream = document.getElementById(\"stream\"); \n \
        var config = { 'iceServers': [{ 'urls': 'stun:" STUN_SERVER "' }] }; \n\
        playStream(vidstream, null, null, null, config, function (errmsg) { console.error(errmsg); }); \n \
      }; \n \
 \n \
    </script> \n \
  </head> \n \
 \n \
  <body> \n \
    <div> \n \
      <video id=\"stream\" autoplay playsinline>Your browser does not support video</video> \n \
    </div> \n \
  </body> \n \
</html> \n \
";

169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
static gboolean
bus_watch_cb (GstBus * bus, GstMessage * message, gpointer user_data)
{
  switch (GST_MESSAGE_TYPE (message)) {
    case GST_MESSAGE_ERROR:
    {
      GError *error = NULL;
      gchar *debug = NULL;

      gst_message_parse_error (message, &error, &debug);
      g_error ("Error on bus: %s (debug: %s)", error->message, debug);
      g_error_free (error);
      g_free (debug);
      break;
    }
    case GST_MESSAGE_WARNING:
    {
      GError *error = NULL;
      gchar *debug = NULL;

      gst_message_parse_warning (message, &error, &debug);
      g_warning ("Warning on bus: %s (debug: %s)", error->message, debug);
      g_error_free (error);
      g_free (debug);
      break;
    }
    default:
      break;
  }

  return G_SOURCE_CONTINUE;
}

202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
static GstWebRTCPriorityType
_priority_from_string (const gchar * s)
{
  GEnumClass *klass =
      (GEnumClass *) g_type_class_ref (GST_TYPE_WEBRTC_PRIORITY_TYPE);
  GEnumValue *en;

  g_return_val_if_fail (klass, 0);
  if (!(en = g_enum_get_value_by_name (klass, s)))
    en = g_enum_get_value_by_nick (klass, s);
  g_type_class_unref (klass);

  if (en)
    return en->value;

  return 0;
}

Jan Schmidt's avatar
Jan Schmidt committed
220
221
222
223
224
225
226
ReceiverEntry *
create_receiver_entry (SoupWebsocketConnection * connection)
{
  GError *error;
  ReceiverEntry *receiver_entry;
  GstWebRTCRTPTransceiver *trans;
  GArray *transceivers;
227
  GstBus *bus;
Jan Schmidt's avatar
Jan Schmidt committed
228
229
230
231
232
233
234
235
236
237

  receiver_entry = g_slice_alloc0 (sizeof (ReceiverEntry));
  receiver_entry->connection = connection;

  g_object_ref (G_OBJECT (connection));

  g_signal_connect (G_OBJECT (connection), "message",
      G_CALLBACK (soup_websocket_message_cb), (gpointer) receiver_entry);

  error = NULL;
Costa Shulyupin's avatar
Costa Shulyupin committed
238
239
240
  receiver_entry->pipeline =
      gst_parse_launch ("webrtcbin name=webrtcbin stun-server=stun://"
      STUN_SERVER " "
241
242
      VIDEO_SRC
      " ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=15/1 ! videoconvert ! queue max-size-buffers=1 ! x264enc bitrate=600 speed-preset=ultrafast tune=zerolatency key-int-max=15 ! video/x-h264,profile=constrained-baseline ! queue max-size-time=100000000 ! h264parse ! "
243
      "rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
Jan Schmidt's avatar
Jan Schmidt committed
244
      "application/x-rtp,media=video,encoding-name=H264,payload="
245
      RTP_PAYLOAD_TYPE " ! webrtcbin. "
246
      "autoaudiosrc is-live=1 ! queue max-size-buffers=1 leaky=downstream ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
247
      RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error);
Jan Schmidt's avatar
Jan Schmidt committed
248
249
250
251
252
253
254
255
256
257
  if (error != NULL) {
    g_error ("Could not create WebRTC pipeline: %s\n", error->message);
    g_error_free (error);
    goto cleanup;
  }

  receiver_entry->webrtcbin =
      gst_bin_get_by_name (GST_BIN (receiver_entry->pipeline), "webrtcbin");
  g_assert (receiver_entry->webrtcbin != NULL);

Costa Shulyupin's avatar
Costa Shulyupin committed
258
259
  g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers",
      &transceivers);
260
  g_assert (transceivers != NULL && transceivers->len > 1);
Jan Schmidt's avatar
Jan Schmidt committed
261
262
  trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);
  trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
  if (video_priority) {
    GstWebRTCPriorityType priority;

    priority = _priority_from_string (video_priority);
    if (priority)
      gst_webrtc_rtp_sender_set_priority (trans->sender, priority);
  }
  trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 1);
  trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
  if (audio_priority) {
    GstWebRTCPriorityType priority;

    priority = _priority_from_string (audio_priority);
    if (priority)
      gst_webrtc_rtp_sender_set_priority (trans->sender, priority);
  }
Jan Schmidt's avatar
Jan Schmidt committed
279
  g_array_unref (transceivers);
Jan Schmidt's avatar
Jan Schmidt committed
280
281
282
283
284
285
286

  g_signal_connect (receiver_entry->webrtcbin, "on-negotiation-needed",
      G_CALLBACK (on_negotiation_needed_cb), (gpointer) receiver_entry);

  g_signal_connect (receiver_entry->webrtcbin, "on-ice-candidate",
      G_CALLBACK (on_ice_candidate_cb), (gpointer) receiver_entry);

287
288
289
290
291
292
293
  bus = gst_pipeline_get_bus (GST_PIPELINE (receiver_entry->pipeline));
  gst_bus_add_watch (bus, bus_watch_cb, NULL);
  gst_object_unref (bus);

  if (gst_element_set_state (receiver_entry->pipeline, GST_STATE_PLAYING) ==
      GST_STATE_CHANGE_FAILURE)
    g_error ("Could not start pipeline");
Jan Schmidt's avatar
Jan Schmidt committed
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347

  return receiver_entry;

cleanup:
  destroy_receiver_entry ((gpointer) receiver_entry);
  return NULL;
}

void
destroy_receiver_entry (gpointer receiver_entry_ptr)
{
  ReceiverEntry *receiver_entry = (ReceiverEntry *) receiver_entry_ptr;

  g_assert (receiver_entry != NULL);

  if (receiver_entry->pipeline != NULL) {
    gst_element_set_state (GST_ELEMENT (receiver_entry->pipeline),
        GST_STATE_NULL);

    gst_object_unref (GST_OBJECT (receiver_entry->webrtcbin));
    gst_object_unref (GST_OBJECT (receiver_entry->pipeline));
  }

  if (receiver_entry->connection != NULL)
    g_object_unref (G_OBJECT (receiver_entry->connection));

  g_slice_free1 (sizeof (ReceiverEntry), receiver_entry);
}


void
on_offer_created_cb (GstPromise * promise, gpointer user_data)
{
  gchar *sdp_string;
  gchar *json_string;
  JsonObject *sdp_json;
  JsonObject *sdp_data_json;
  GstStructure const *reply;
  GstPromise *local_desc_promise;
  GstWebRTCSessionDescription *offer = NULL;
  ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;

  reply = gst_promise_get_reply (promise);
  gst_structure_get (reply, "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
      &offer, NULL);
  gst_promise_unref (promise);

  local_desc_promise = gst_promise_new ();
  g_signal_emit_by_name (receiver_entry->webrtcbin, "set-local-description",
      offer, local_desc_promise);
  gst_promise_interrupt (local_desc_promise);
  gst_promise_unref (local_desc_promise);

  sdp_string = gst_sdp_message_as_text (offer->sdp);
Seungha Yang's avatar
Seungha Yang committed
348
  gst_print ("Negotiation offer created:\n%s\n", sdp_string);
Jan Schmidt's avatar
Jan Schmidt committed
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374

  sdp_json = json_object_new ();
  json_object_set_string_member (sdp_json, "type", "sdp");

  sdp_data_json = json_object_new ();
  json_object_set_string_member (sdp_data_json, "type", "offer");
  json_object_set_string_member (sdp_data_json, "sdp", sdp_string);
  json_object_set_object_member (sdp_json, "data", sdp_data_json);

  json_string = get_string_from_json_object (sdp_json);
  json_object_unref (sdp_json);

  soup_websocket_connection_send_text (receiver_entry->connection, json_string);
  g_free (json_string);
  g_free (sdp_string);

  gst_webrtc_session_description_free (offer);
}


void
on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data)
{
  GstPromise *promise;
  ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;

Seungha Yang's avatar
Seungha Yang committed
375
  gst_print ("Creating negotiation offer\n");
Jan Schmidt's avatar
Jan Schmidt committed
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486

  promise = gst_promise_new_with_change_func (on_offer_created_cb,
      (gpointer) receiver_entry, NULL);
  g_signal_emit_by_name (G_OBJECT (webrtcbin), "create-offer", NULL, promise);
}


void
on_ice_candidate_cb (G_GNUC_UNUSED GstElement * webrtcbin, guint mline_index,
    gchar * candidate, gpointer user_data)
{
  JsonObject *ice_json;
  JsonObject *ice_data_json;
  gchar *json_string;
  ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;

  ice_json = json_object_new ();
  json_object_set_string_member (ice_json, "type", "ice");

  ice_data_json = json_object_new ();
  json_object_set_int_member (ice_data_json, "sdpMLineIndex", mline_index);
  json_object_set_string_member (ice_data_json, "candidate", candidate);
  json_object_set_object_member (ice_json, "data", ice_data_json);

  json_string = get_string_from_json_object (ice_json);
  json_object_unref (ice_json);

  soup_websocket_connection_send_text (receiver_entry->connection, json_string);
  g_free (json_string);
}


void
soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
    SoupWebsocketDataType data_type, GBytes * message, gpointer user_data)
{
  gsize size;
  gchar *data;
  gchar *data_string;
  const gchar *type_string;
  JsonNode *root_json;
  JsonObject *root_json_object;
  JsonObject *data_json_object;
  JsonParser *json_parser = NULL;
  ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;

  switch (data_type) {
    case SOUP_WEBSOCKET_DATA_BINARY:
      g_error ("Received unknown binary message, ignoring\n");
      g_bytes_unref (message);
      return;

    case SOUP_WEBSOCKET_DATA_TEXT:
      data = g_bytes_unref_to_data (message, &size);
      /* Convert to NULL-terminated string */
      data_string = g_strndup (data, size);
      g_free (data);
      break;

    default:
      g_assert_not_reached ();
  }

  json_parser = json_parser_new ();
  if (!json_parser_load_from_data (json_parser, data_string, -1, NULL))
    goto unknown_message;

  root_json = json_parser_get_root (json_parser);
  if (!JSON_NODE_HOLDS_OBJECT (root_json))
    goto unknown_message;

  root_json_object = json_node_get_object (root_json);

  if (!json_object_has_member (root_json_object, "type")) {
    g_error ("Received message without type field\n");
    goto cleanup;
  }
  type_string = json_object_get_string_member (root_json_object, "type");

  if (!json_object_has_member (root_json_object, "data")) {
    g_error ("Received message without data field\n");
    goto cleanup;
  }
  data_json_object = json_object_get_object_member (root_json_object, "data");

  if (g_strcmp0 (type_string, "sdp") == 0) {
    const gchar *sdp_type_string;
    const gchar *sdp_string;
    GstPromise *promise;
    GstSDPMessage *sdp;
    GstWebRTCSessionDescription *answer;
    int ret;

    if (!json_object_has_member (data_json_object, "type")) {
      g_error ("Received SDP message without type field\n");
      goto cleanup;
    }
    sdp_type_string = json_object_get_string_member (data_json_object, "type");

    if (g_strcmp0 (sdp_type_string, "answer") != 0) {
      g_error ("Expected SDP message type \"answer\", got \"%s\"\n",
          sdp_type_string);
      goto cleanup;
    }

    if (!json_object_has_member (data_json_object, "sdp")) {
      g_error ("Received SDP message without SDP string\n");
      goto cleanup;
    }
    sdp_string = json_object_get_string_member (data_json_object, "sdp");

Seungha Yang's avatar
Seungha Yang committed
487
    gst_print ("Received SDP:\n%s\n", sdp_string);
Jan Schmidt's avatar
Jan Schmidt committed
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527

    ret = gst_sdp_message_new (&sdp);
    g_assert_cmphex (ret, ==, GST_SDP_OK);

    ret =
        gst_sdp_message_parse_buffer ((guint8 *) sdp_string,
        strlen (sdp_string), sdp);
    if (ret != GST_SDP_OK) {
      g_error ("Could not parse SDP string\n");
      goto cleanup;
    }

    answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
        sdp);
    g_assert_nonnull (answer);

    promise = gst_promise_new ();
    g_signal_emit_by_name (receiver_entry->webrtcbin, "set-remote-description",
        answer, promise);
    gst_promise_interrupt (promise);
    gst_promise_unref (promise);
    gst_webrtc_session_description_free (answer);
  } else if (g_strcmp0 (type_string, "ice") == 0) {
    guint mline_index;
    const gchar *candidate_string;

    if (!json_object_has_member (data_json_object, "sdpMLineIndex")) {
      g_error ("Received ICE message without mline index\n");
      goto cleanup;
    }
    mline_index =
        json_object_get_int_member (data_json_object, "sdpMLineIndex");

    if (!json_object_has_member (data_json_object, "candidate")) {
      g_error ("Received ICE message without ICE candidate string\n");
      goto cleanup;
    }
    candidate_string = json_object_get_string_member (data_json_object,
        "candidate");

Seungha Yang's avatar
Seungha Yang committed
528
    gst_print ("Received ICE candidate with mline index %u; candidate: %s\n",
Jan Schmidt's avatar
Jan Schmidt committed
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
        mline_index, candidate_string);

    g_signal_emit_by_name (receiver_entry->webrtcbin, "add-ice-candidate",
        mline_index, candidate_string);
  } else
    goto unknown_message;

cleanup:
  if (json_parser != NULL)
    g_object_unref (G_OBJECT (json_parser));
  g_free (data_string);
  return;

unknown_message:
  g_error ("Unknown message \"%s\", ignoring", data_string);
  goto cleanup;
}


void
soup_websocket_closed_cb (SoupWebsocketConnection * connection,
    gpointer user_data)
{
  GHashTable *receiver_entry_table = (GHashTable *) user_data;
  g_hash_table_remove (receiver_entry_table, connection);
Seungha Yang's avatar
Seungha Yang committed
554
  gst_print ("Closed websocket connection %p\n", (gpointer) connection);
Jan Schmidt's avatar
Jan Schmidt committed
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
}


void
soup_http_handler (G_GNUC_UNUSED SoupServer * soup_server,
    SoupMessage * message, const char *path, G_GNUC_UNUSED GHashTable * query,
    G_GNUC_UNUSED SoupClientContext * client_context,
    G_GNUC_UNUSED gpointer user_data)
{
  SoupBuffer *soup_buffer;

  if ((g_strcmp0 (path, "/") != 0) && (g_strcmp0 (path, "/index.html") != 0)) {
    soup_message_set_status (message, SOUP_STATUS_NOT_FOUND);
    return;
  }

  soup_buffer =
      soup_buffer_new (SOUP_MEMORY_STATIC, html_source, strlen (html_source));

  soup_message_headers_set_content_type (message->response_headers, "text/html",
      NULL);
  soup_message_body_append_buffer (message->response_body, soup_buffer);
  soup_buffer_free (soup_buffer);

  soup_message_set_status (message, SOUP_STATUS_OK);
}


void
soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
    SoupWebsocketConnection * connection, G_GNUC_UNUSED const char *path,
    G_GNUC_UNUSED SoupClientContext * client_context, gpointer user_data)
{
  ReceiverEntry *receiver_entry;
  GHashTable *receiver_entry_table = (GHashTable *) user_data;

Seungha Yang's avatar
Seungha Yang committed
591
  gst_print ("Processing new websocket connection %p", (gpointer) connection);
Jan Schmidt's avatar
Jan Schmidt committed
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619

  g_signal_connect (G_OBJECT (connection), "closed",
      G_CALLBACK (soup_websocket_closed_cb), (gpointer) receiver_entry_table);

  receiver_entry = create_receiver_entry (connection);
  g_hash_table_replace (receiver_entry_table, connection, receiver_entry);
}


static gchar *
get_string_from_json_object (JsonObject * object)
{
  JsonNode *root;
  JsonGenerator *generator;
  gchar *text;

  /* Make it the root node */
  root = json_node_init_object (json_node_alloc (), object);
  generator = json_generator_new ();
  json_generator_set_root (generator, root);
  text = json_generator_to_data (generator, NULL);

  /* Release everything */
  g_object_unref (generator);
  json_node_free (root);
  return text;
}

620
#ifdef G_OS_UNIX
Jan Schmidt's avatar
Jan Schmidt committed
621
622
623
gboolean
exit_sighandler (gpointer user_data)
{
Seungha Yang's avatar
Seungha Yang committed
624
  gst_print ("Caught signal, stopping mainloop\n");
Jan Schmidt's avatar
Jan Schmidt committed
625
626
627
628
  GMainLoop *mainloop = (GMainLoop *) user_data;
  g_main_loop_quit (mainloop);
  return TRUE;
}
629
#endif
Jan Schmidt's avatar
Jan Schmidt committed
630

631
632
633
634
635
636
637
638
639
640
641

static GOptionEntry entries[] = {
  {"video-priority", 0, 0, G_OPTION_ARG_STRING, &video_priority,
        "Priority of the video stream (very-low, low, medium or high)",
      "PRIORITY"},
  {"audio-priority", 0, 0, G_OPTION_ARG_STRING, &audio_priority,
        "Priority of the audio stream (very-low, low, medium or high)",
      "PRIORITY"},
  {NULL},
};

Jan Schmidt's avatar
Jan Schmidt committed
642
643
644
645
646
647
int
main (int argc, char *argv[])
{
  GMainLoop *mainloop;
  SoupServer *soup_server;
  GHashTable *receiver_entry_table;
648
649
  GOptionContext *context;
  GError *error = NULL;
Jan Schmidt's avatar
Jan Schmidt committed
650
651

  setlocale (LC_ALL, "");
652
653
654
655
656
657
658
659

  context = g_option_context_new ("- gstreamer webrtc sendonly demo");
  g_option_context_add_main_entries (context, entries, NULL);
  g_option_context_add_group (context, gst_init_get_option_group ());
  if (!g_option_context_parse (context, &argc, &argv, &error)) {
    g_printerr ("Error initializing: %s\n", error->message);
    return -1;
  }
Jan Schmidt's avatar
Jan Schmidt committed
660
661
662
663
664
665
666
667

  receiver_entry_table =
      g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
      destroy_receiver_entry);

  mainloop = g_main_loop_new (NULL, FALSE);
  g_assert (mainloop != NULL);

668
#ifdef G_OS_UNIX
Jan Schmidt's avatar
Jan Schmidt committed
669
670
  g_unix_signal_add (SIGINT, exit_sighandler, mainloop);
  g_unix_signal_add (SIGTERM, exit_sighandler, mainloop);
671
#endif
Jan Schmidt's avatar
Jan Schmidt committed
672
673
674
675
676
677
678
679
680

  soup_server =
      soup_server_new (SOUP_SERVER_SERVER_HEADER, "webrtc-soup-server", NULL);
  soup_server_add_handler (soup_server, "/", soup_http_handler, NULL, NULL);
  soup_server_add_websocket_handler (soup_server, "/ws", NULL, NULL,
      soup_websocket_handler, (gpointer) receiver_entry_table, NULL);
  soup_server_listen_all (soup_server, SOUP_HTTP_PORT,
      (SoupServerListenOptions) 0, NULL);

Seungha Yang's avatar
Seungha Yang committed
681
  gst_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
Jan Schmidt's avatar
Jan Schmidt committed
682
683
684
685
686
687
688
689
690
691
692

  g_main_loop_run (mainloop);

  g_object_unref (G_OBJECT (soup_server));
  g_hash_table_destroy (receiver_entry_table);
  g_main_loop_unref (mainloop);

  gst_deinit ();

  return 0;
}