transport.c File Reference

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Data Structures
struct	webrtc_recv_msg_t
	Receive queue element (variable-length message) More...
struct	webrtc_transport_data_t
	WebRTC transport implementation data. More...

Macros
#define	WEBRTC_RECV_QUEUE_SIZE   512
	Maximum receive queue size (messages buffered before recv())

Functions
acip_transport_t *	acip_webrtc_transport_create (webrtc_peer_connection_t *peer_conn, webrtc_data_channel_t *data_channel, crypto_context_t *crypto_ctx)

Macro Definition Documentation

WEBRTC_RECV_QUEUE_SIZE

#define WEBRTC_RECV_QUEUE_SIZE   512

Maximum receive queue size (messages buffered before recv())

Power of 2 for ringbuffer optimization. 512 messages provides adequate buffering for large video frames (921KB each) with WebRTC DataChannel fragmentation. At 30 FPS: 512 / 30 = ~17 seconds of buffering for burst traffic and processing delays.

Definition at line 47 of file webrtc/transport.c.

Function Documentation

acip_webrtc_transport_create()

acip_transport_t * acip_webrtc_transport_create	(	webrtc_peer_connection_t *	peer_conn,
		webrtc_data_channel_t *	data_channel,
		crypto_context_t *	crypto_ctx
	)

Definition at line 367 of file webrtc/transport.c.

                                                                             {
  if (!peer_conn || !data_channel) {
    SET_ERRNO(ERROR_INVALID_PARAM, "peer_conn and data_channel are required");
    return NULL;
  }
 
  // Allocate transport structure
  acip_transport_t *transport = SAFE_MALLOC(sizeof(acip_transport_t), acip_transport_t *);
  if (!transport) {
    SET_ERRNO(ERROR_MEMORY, "Failed to allocate WebRTC transport");
    return NULL;
  }
 
  // Allocate WebRTC-specific data
  webrtc_transport_data_t *wrtc_data = SAFE_MALLOC(sizeof(webrtc_transport_data_t), webrtc_transport_data_t *);
  if (!wrtc_data) {
    SAFE_FREE(transport);
    SET_ERRNO(ERROR_MEMORY, "Failed to allocate WebRTC transport data");
    return NULL;
  }
 
  // Create receive queue
  wrtc_data->recv_queue = ringbuffer_create(sizeof(webrtc_recv_msg_t), WEBRTC_RECV_QUEUE_SIZE);
  if (!wrtc_data->recv_queue) {
    SAFE_FREE(wrtc_data);
    SAFE_FREE(transport);
    SET_ERRNO(ERROR_MEMORY, "Failed to create receive queue");
    return NULL;
  }
 
  // Initialize synchronization primitives
  if (mutex_init(&wrtc_data->queue_mutex) != 0) {
    ringbuffer_destroy(wrtc_data->recv_queue);
    SAFE_FREE(wrtc_data);
    SAFE_FREE(transport);
    SET_ERRNO(ERROR_INTERNAL, "Failed to initialize queue mutex");
    return NULL;
  }
 
  if (cond_init(&wrtc_data->queue_cond) != 0) {
    mutex_destroy(&wrtc_data->queue_mutex);
    ringbuffer_destroy(wrtc_data->recv_queue);
    SAFE_FREE(wrtc_data);
    SAFE_FREE(transport);
    SET_ERRNO(ERROR_INTERNAL, "Failed to initialize queue condition variable");
    return NULL;
  }
 
  if (mutex_init(&wrtc_data->state_mutex) != 0) {
    cond_destroy(&wrtc_data->queue_cond);
    mutex_destroy(&wrtc_data->queue_mutex);
    ringbuffer_destroy(wrtc_data->recv_queue);
    SAFE_FREE(wrtc_data);
    SAFE_FREE(transport);
    SET_ERRNO(ERROR_INTERNAL, "Failed to initialize state mutex");
    return NULL;
  }
 
  // Initialize WebRTC data
  wrtc_data->peer_conn = peer_conn;
  wrtc_data->data_channel = data_channel;
  wrtc_data->is_connected = false; // Will be set to true in on_open callback
 
  // Register DataChannel callbacks
  webrtc_datachannel_callbacks_t callbacks = {
      .on_open = webrtc_on_open,
      .on_close = webrtc_on_close,
      .on_error = webrtc_on_error,
      .on_message = webrtc_on_message,
      .user_data = wrtc_data,
  };
 
  asciichat_error_t result = webrtc_datachannel_set_callbacks(data_channel, &callbacks);
  if (result != ASCIICHAT_OK) {
    mutex_destroy(&wrtc_data->state_mutex);
    cond_destroy(&wrtc_data->queue_cond);
    mutex_destroy(&wrtc_data->queue_mutex);
    ringbuffer_destroy(wrtc_data->recv_queue);
    SAFE_FREE(wrtc_data);
    SAFE_FREE(transport);
    SET_ERRNO(ERROR_INTERNAL, "Failed to set DataChannel callbacks");
    return NULL;
  }
 
  // IMPORTANT: The transport is always created from peer_manager's on_datachannel_open callback,
  // which means the DataChannel is ALREADY OPEN when we get here. However, by setting our own
  // callbacks above (webrtc_datachannel_set_callbacks), we replaced the callbacks that would
  // have set dc->is_open=true. So we need to manually mark both the transport AND the DataChannel
  // as open/connected now.
  //
  // We cannot rely on webrtc_on_open being called later because:
  // 1. The DataChannel is already open
  // 2. libdatachannel won't fire the open event again
  // 3. Setting callbacks after open doesn't trigger a retroactive open event
 
  // Mark DataChannel as open (needed for webrtc_datachannel_send() check)
  webrtc_datachannel_set_open_state(data_channel, true);
 
  // Mark transport as connected
  mutex_lock(&wrtc_data->state_mutex);
  wrtc_data->is_connected = true;
  mutex_unlock(&wrtc_data->state_mutex);
  log_debug("Transport and DataChannel marked as connected/open (already open from peer_manager callback)");
 
  // Initialize transport
  transport->methods = &webrtc_methods;
  transport->crypto_ctx = crypto_ctx;
  transport->impl_data = wrtc_data;
 
  log_info("Created WebRTC transport (crypto: %s)", crypto_ctx ? "enabled" : "disabled");
 
  return transport;
}

References webrtc_transport_data_t::data_channel, webrtc_transport_data_t::is_connected, mutex_destroy(), mutex_init(), webrtc_transport_data_t::peer_conn, webrtc_transport_data_t::queue_cond, webrtc_transport_data_t::queue_mutex, webrtc_transport_data_t::recv_queue, ringbuffer_create(), ringbuffer_destroy(), webrtc_transport_data_t::state_mutex, webrtc_datachannel_set_callbacks(), webrtc_datachannel_set_open_state(), and WEBRTC_RECV_QUEUE_SIZE.