mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 09:55:36 +00:00
rtspsrc: Use a mutex for protecting against concurrent send/receives
We currently send data to the RTSP connection from multiple threads: whenever a command is to be handled and whenever RTCP is generated. This can cause data corruption or worse if both happen at the same time. As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive() calls with a mutex. While this means that we hold a mutex during the IO operation, this is not actually a problem as the IO operation can be interrupted (gst_rtsp_connection_flush()) at any time and is blocking by itself anyway.
This commit is contained in:
Sebastian Dröge
parent
deb9c62cd9
commit
a722f6e832
2 changed files with 61 additions and 51 deletions
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@ -2067,29 +2067,35 @@ gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
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}
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static GstRTSPResult
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gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
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gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * message, GTimeVal * timeout)
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{
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GstRTSPResult ret;
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if (conn)
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ret = gst_rtsp_connection_send (conn, message, timeout);
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else
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if (conninfo->connection) {
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g_mutex_lock (&conninfo->send_lock);
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ret = gst_rtsp_connection_send (conninfo->connection, message, timeout);
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g_mutex_unlock (&conninfo->send_lock);
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} else {
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ret = GST_RTSP_ERROR;
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}
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return ret;
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}
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static GstRTSPResult
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gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
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gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * message, GTimeVal * timeout)
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{
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GstRTSPResult ret;
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if (conn)
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ret = gst_rtsp_connection_receive (conn, message, timeout);
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else
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if (conninfo->connection) {
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g_mutex_lock (&conninfo->send_lock);
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ret = gst_rtsp_connection_receive (conninfo->connection, message, timeout);
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g_mutex_unlock (&conninfo->send_lock);
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} else {
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ret = GST_RTSP_ERROR;
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}
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return ret;
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}
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@ -2503,7 +2509,7 @@ gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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guint size;
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GstRTSPResult ret;
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GstRTSPMessage message = { 0 };
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GstRTSPConnection *conn;
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GstRTSPConnInfo *conninfo;
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stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
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src = stream->parent;
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@ -2518,12 +2524,12 @@ gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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gst_rtsp_message_take_body (&message, data, size);
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if (stream->conninfo.connection)
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conn = stream->conninfo.connection;
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conninfo = &stream->conninfo;
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else
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conn = src->conninfo.connection;
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conninfo = &src->conninfo;
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GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
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ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
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ret = gst_rtspsrc_connection_send (src, conninfo, &message, NULL);
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GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
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/* and steal it away again because we will free it when unreffing the
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@ -4213,6 +4219,10 @@ gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
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goto could_not_connect;
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}
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} while (!info->connected && retry);
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g_mutex_init (&info->send_lock);
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g_mutex_init (&info->recv_lock);
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gst_rtsp_message_unset (&response);
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return GST_RTSP_OK;
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@ -4257,6 +4267,9 @@ gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
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gst_rtsp_connection_free (info->connection);
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info->connection = NULL;
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info->flushing = FALSE;
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g_mutex_clear (&info->send_lock);
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g_mutex_clear (&info->recv_lock);
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}
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GST_RTSP_STATE_UNLOCK (src);
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return GST_RTSP_OK;
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@ -4317,7 +4330,7 @@ gst_rtspsrc_init_request (GstRTSPSrc * src, GstRTSPMessage * msg,
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/* FIXME, handle server request, reply with OK, for now */
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static GstRTSPResult
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gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
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gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * request)
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{
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GstRTSPMessage response = { 0 };
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@ -4346,7 +4359,7 @@ gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
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if (src->debug)
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gst_rtsp_message_dump (&response);
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res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
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res = gst_rtspsrc_connection_send (src, conninfo, &response, NULL);
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if (res < 0)
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goto send_error;
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@ -4398,9 +4411,7 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
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if (src->debug)
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gst_rtsp_message_dump (&request);
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res =
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gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
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NULL);
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res = gst_rtspsrc_connection_send (src, &src->conninfo, &request, NULL);
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if (res < 0)
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goto send_error;
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@ -4681,7 +4692,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
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/* protect the connection with the connection lock so that we can see when
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* we are finished doing server communication */
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res =
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gst_rtspsrc_connection_receive (src, src->conninfo.connection,
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gst_rtspsrc_connection_receive (src, &src->conninfo,
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&message, src->ptcp_timeout);
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switch (res) {
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@ -4707,9 +4718,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
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switch (message.type) {
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case GST_RTSP_MESSAGE_REQUEST:
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/* server sends us a request message, handle it */
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res =
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gst_rtspsrc_handle_request (src, src->conninfo.connection,
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&message);
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res = gst_rtspsrc_handle_request (src, &src->conninfo, &message);
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if (res == GST_RTSP_EEOF)
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goto server_eof;
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else if (res < 0)
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@ -4800,7 +4809,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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/* we should continue reading the TCP socket because the server might
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* send us requests. When the session timeout expires, we need to send a
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* keep-alive request to keep the session open. */
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res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
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res = gst_rtspsrc_connection_receive (src, &src->conninfo,
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&message, &tv_timeout);
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switch (res) {
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@ -4840,9 +4849,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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switch (message.type) {
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case GST_RTSP_MESSAGE_REQUEST:
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/* server sends us a request message, handle it */
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res =
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gst_rtspsrc_handle_request (src, src->conninfo.connection,
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&message);
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res = gst_rtspsrc_handle_request (src, &src->conninfo, &message);
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if (res == GST_RTSP_EEOF)
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goto server_eof;
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else if (res < 0)
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@ -5384,7 +5391,7 @@ no_user_pass:
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}
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static GstRTSPResult
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gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
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gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * request, GstRTSPMessage * response,
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GstRTSPStatusCode * code)
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{
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@ -5402,14 +5409,16 @@ again:
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if (src->debug)
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gst_rtsp_message_dump (request);
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res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
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res = gst_rtspsrc_connection_send (src, conninfo, request, src->ptcp_timeout);
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if (res < 0)
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goto send_error;
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gst_rtsp_connection_reset_timeout (conn);
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gst_rtsp_connection_reset_timeout (conninfo->connection);
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next:
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res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
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res =
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gst_rtspsrc_connection_receive (src, conninfo, response,
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src->ptcp_timeout);
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if (res < 0)
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goto receive_error;
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@ -5418,7 +5427,7 @@ next:
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switch (response->type) {
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case GST_RTSP_MESSAGE_REQUEST:
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res = gst_rtspsrc_handle_request (src, conn, response);
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res = gst_rtspsrc_handle_request (src, conninfo, response);
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if (res == GST_RTSP_EEOF)
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goto server_eof;
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else if (res < 0)
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@ -5546,7 +5555,7 @@ server_eof:
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* Returns: #GST_RTSP_OK if the processing was successful.
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*/
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static GstRTSPResult
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gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
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gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnInfo * conninfo,
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GstRTSPMessage * request, GstRTSPMessage * response,
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GstRTSPStatusCode * code)
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{
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@ -5568,7 +5577,8 @@ gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
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method = request->type_data.request.method;
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if ((res =
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gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
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gst_rtspsrc_try_send (src, conninfo, request, response,
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&int_code)) < 0)
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goto error;
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switch (int_code) {
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@ -5671,8 +5681,7 @@ static GstRTSPResult
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gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
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GstRTSPMessage * response, GstRTSPSrc * src)
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{
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return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
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NULL);
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return gst_rtspsrc_send (src, &src->conninfo, request, response, NULL);
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}
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@ -6041,7 +6050,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
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goto no_streams;
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for (walk = src->streams; walk; walk = g_list_next (walk)) {
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GstRTSPConnection *conn;
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GstRTSPConnInfo *conninfo;
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gchar *transports;
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gint retry = 0;
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guint mask = 0;
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@ -6106,9 +6115,9 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
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GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
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continue;
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}
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conn = stream->conninfo.connection;
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conninfo = &stream->conninfo;
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} else {
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conn = src->conninfo.connection;
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conninfo = &src->conninfo;
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}
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GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
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stream->conninfo.location);
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@ -6185,7 +6194,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
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stream->id));
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/* handle the code ourselves */
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res = gst_rtspsrc_send (src, conn, &request, &response, &code);
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res = gst_rtspsrc_send (src, conninfo, &request, &response, &code);
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if (res < 0)
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goto send_error;
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@ -6705,7 +6714,7 @@ restart:
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GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
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if ((res =
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gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
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gst_rtspsrc_send (src, &src->conninfo, &request, &response,
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NULL)) < 0)
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goto send_error;
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@ -6732,7 +6741,7 @@ restart:
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GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
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if ((res =
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gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
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gst_rtspsrc_send (src, &src->conninfo, &request, &response,
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NULL)) < 0)
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goto send_error;
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@ -6955,9 +6964,7 @@ gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
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if (async)
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GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
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if ((res =
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gst_rtspsrc_send (src, info->connection, &request, &response,
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NULL)) < 0)
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if ((res = gst_rtspsrc_send (src, info, &request, &response, NULL)) < 0)
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goto send_error;
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/* FIXME, parse result? */
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@ -7249,7 +7256,7 @@ restart:
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for (walk = src->streams; walk; walk = g_list_next (walk)) {
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GstRTSPStream *stream = (GstRTSPStream *) walk->data;
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const gchar *setup_url;
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GstRTSPConnection *conn;
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GstRTSPConnInfo *conninfo;
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/* try aggregate control first but do non-aggregate control otherwise */
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if (control)
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@ -7258,9 +7265,9 @@ restart:
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continue;
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if (src->conninfo.connection) {
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conn = src->conninfo.connection;
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conninfo = &src->conninfo;
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} else if (stream->conninfo.connection) {
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conn = stream->conninfo.connection;
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conninfo = &stream->conninfo;
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} else {
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continue;
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}
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@ -7292,7 +7299,7 @@ restart:
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if (async)
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GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
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if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
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if ((res = gst_rtspsrc_send (src, conninfo, &request, &response, NULL)) < 0)
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goto send_error;
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if (src->need_redirect) {
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@ -7470,7 +7477,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
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* aggregate control */
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for (walk = src->streams; walk; walk = g_list_next (walk)) {
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GstRTSPStream *stream = (GstRTSPStream *) walk->data;
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GstRTSPConnection *conn;
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GstRTSPConnInfo *conninfo;
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const gchar *setup_url;
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/* try aggregate control first but do non-aggregate control otherwise */
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@ -7480,9 +7487,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
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continue;
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if (src->conninfo.connection) {
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conn = src->conninfo.connection;
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conninfo = &src->conninfo;
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} else if (stream->conninfo.connection) {
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conn = stream->conninfo.connection;
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conninfo = &stream->conninfo;
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} else {
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continue;
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}
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@ -7496,7 +7503,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
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setup_url)) < 0)
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goto create_request_failed;
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if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
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if ((res = gst_rtspsrc_send (src, conninfo, &request, &response, NULL)) < 0)
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goto send_error;
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gst_rtsp_message_unset (&request);
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@ -86,6 +86,9 @@ struct _GstRTSPConnInfo {
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GstRTSPConnection *connection;
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gboolean connected;
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gboolean flushing;
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GMutex send_lock;
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GMutex recv_lock;
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};
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typedef struct _GstRTSPStream GstRTSPStream;
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