mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-01 20:30:41 +00:00
rtspsrc: add non-aggregate control
Add non-aggregate control. Separate retrieving thr SDP from parsing and setting up the streaming from the SDP.
This commit is contained in:
Wim Taymans
parent
cf095fb9bc
commit
ddc214d322
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@ -3104,6 +3104,7 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
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GstRTSPMessage request = { 0 };
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GstRTSPResult res;
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GstRTSPMethod method;
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gchar *control;
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GST_DEBUG_OBJECT (src, "creating server keep-alive");
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@ -3113,7 +3114,15 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
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else
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method = GST_RTSP_OPTIONS;
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res = gst_rtsp_message_init_request (&request, method, src->req_location);
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if (src->control)
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control = src->control;
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else
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control = src->req_location;
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if (control == NULL)
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goto no_control;
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res = gst_rtsp_message_init_request (&request, method, control);
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if (res < 0)
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goto send_error;
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@ -3130,6 +3139,11 @@ gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
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return GST_RTSP_OK;
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/* ERRORS */
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no_control:
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{
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GST_WARNING_OBJECT (src, "no control url to send keepalive");
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return GST_RTSP_OK;
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}
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send_error:
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{
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gchar *str = gst_rtsp_strresult (res);
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@ -4870,6 +4884,77 @@ gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
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return TRUE;
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}
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static gboolean
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gst_rtspsrc_from_sdp (GstRTSPSrc * src, guint8 * data, guint size)
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{
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GstSDPMessage sdp = { 0 };
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gint i, n_streams;
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GST_DEBUG_OBJECT (src, "parse SDP...");
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gst_sdp_message_init (&sdp);
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gst_sdp_message_parse_buffer (data, size, &sdp);
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if (src->debug)
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gst_sdp_message_dump (&sdp);
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gst_rtsp_ext_list_parse_sdp (src->extensions, &sdp, src->props);
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/* parse range for duration reporting. */
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{
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const gchar *range;
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for (i = 0;; i++) {
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range = gst_sdp_message_get_attribute_val_n (&sdp, "range", i);
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if (range == NULL)
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break;
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/* keep track of the range and configure it in the segment */
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if (gst_rtspsrc_parse_range (src, range, &src->segment))
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break;
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}
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}
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/* try to find a global control attribute */
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{
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const gchar *control;
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for (i = 0;; i++) {
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control = gst_sdp_message_get_attribute_val_n (&sdp, "control", i);
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if (control == NULL)
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break;
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/* only take fully qualified urls */
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if (g_str_has_prefix (control, "rtsp://"))
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break;
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}
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g_free (src->control);
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src->control = g_strdup (control);
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}
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/* create streams */
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n_streams = gst_sdp_message_medias_len (&sdp);
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for (i = 0; i < n_streams; i++) {
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gst_rtspsrc_create_stream (src, &sdp, i);
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}
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src->state = GST_RTSP_STATE_INIT;
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/* setup streams */
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if (!gst_rtspsrc_setup_streams (src))
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goto setup_failed;
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src->state = GST_RTSP_STATE_READY;
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gst_sdp_message_uninit (&sdp);
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return TRUE;
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setup_failed:
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{
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gst_sdp_message_uninit (&sdp);
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return FALSE;
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}
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}
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static gboolean
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gst_rtspsrc_open (GstRTSPSrc * src)
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{
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@ -4878,8 +4963,6 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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GstRTSPMessage response = { 0 };
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guint8 *data;
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guint size;
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gint i, n_streams;
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GstSDPMessage sdp = { 0 };
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gchar *respcont = NULL;
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GstRTSPUrl *url;
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@ -4998,66 +5081,12 @@ restart:
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if (data == NULL)
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goto no_describe;
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GST_DEBUG_OBJECT (src, "parse SDP...");
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gst_sdp_message_init (&sdp);
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gst_sdp_message_parse_buffer (data, size, &sdp);
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if (src->debug)
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gst_sdp_message_dump (&sdp);
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gst_rtsp_ext_list_parse_sdp (src->extensions, &sdp, src->props);
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/* parse range for duration reporting. */
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{
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const gchar *range;
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for (i = 0;; i++) {
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range = gst_sdp_message_get_attribute_val_n (&sdp, "range", i);
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if (range == NULL)
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break;
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/* keep track of the range and configure it in the segment */
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if (gst_rtspsrc_parse_range (src, range, &src->segment))
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break;
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}
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}
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/* try to find a global control attribute */
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g_free (src->control);
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src->control = NULL;
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{
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const gchar *control;
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for (i = 0;; i++) {
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control = gst_sdp_message_get_attribute_val_n (&sdp, "control", i);
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if (control == NULL)
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break;
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if (g_str_has_prefix (control, "rtsp://")) {
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src->control = g_strdup (control);
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break;
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}
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}
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}
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/* create streams */
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n_streams = gst_sdp_message_medias_len (&sdp);
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for (i = 0; i < n_streams; i++) {
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gst_rtspsrc_create_stream (src, &sdp, i);
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}
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src->state = GST_RTSP_STATE_INIT;
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/* setup streams */
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if (!gst_rtspsrc_setup_streams (src))
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goto setup_failed;
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src->state = GST_RTSP_STATE_READY;
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GST_RTSP_STATE_UNLOCK (src);
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if (!gst_rtspsrc_from_sdp (src, data, size))
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goto sdp_failed;
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/* clean up any messages */
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gst_rtsp_message_unset (&request);
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gst_rtsp_message_unset (&response);
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gst_sdp_message_uninit (&sdp);
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return TRUE;
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@ -5118,7 +5147,7 @@ no_describe:
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("Server can not provide an SDP."));
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goto cleanup_error;
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}
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setup_failed:
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sdp_failed:
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{
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gst_rtspsrc_close (src);
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/* error was posted */
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@ -5135,7 +5164,6 @@ cleanup_error:
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GST_RTSP_STATE_UNLOCK (src);
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gst_rtsp_message_unset (&request);
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gst_rtsp_message_unset (&response);
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gst_sdp_message_uninit (&sdp);
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return FALSE;
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}
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}
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@ -5163,6 +5191,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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GstRTSPMessage request = { 0 };
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GstRTSPMessage response = { 0 };
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GstRTSPResult res;
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GList *walk;
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gboolean ret = FALSE;
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gchar *control;
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@ -5214,9 +5243,22 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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else
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control = src->req_location;
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if (src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)) {
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if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
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goto not_supported;
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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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gchar *setup_url;
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/* try aggregate control first but do non-aggregate control otherwise */
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if (control)
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setup_url = control;
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else if ((setup_url = stream->setup_url) == NULL)
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continue;
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/* do TEARDOWN */
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res = gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, control);
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res =
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gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
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if (res < 0)
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goto create_request_failed;
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@ -5226,9 +5268,10 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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/* FIXME, parse result? */
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gst_rtsp_message_unset (&request);
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gst_rtsp_message_unset (&response);
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} else {
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GST_DEBUG_OBJECT (src,
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"TEARDOWN and PLAY not supported, can't do TEARDOWN");
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/* early exit when we did aggregate control */
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if (control)
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break;
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}
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close:
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@ -5267,6 +5310,12 @@ send_error:
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ret = FALSE;
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goto close;
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}
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not_supported:
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{
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GST_DEBUG_OBJECT (src,
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"TEARDOWN and PLAY not supported, can't do TEARDOWN");
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goto close;
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}
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}
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/* RTP-Info is of the format:
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@ -5390,6 +5439,7 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment)
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GstRTSPMessage request = { 0 };
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GstRTSPMessage response = { 0 };
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GstRTSPResult res;
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GList *walk;
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gchar *hval;
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gfloat fval;
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gint hval_idx;
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@ -5408,12 +5458,6 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment)
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if (!src->connection || !src->connected)
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goto done;
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/* construct a control url */
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if (src->control)
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control = src->control;
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else
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control = src->req_location;
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/* waiting for connection idle, we were flushing so any attempt at doing data
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* transfer will result in pausing the tasks. */
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GST_DEBUG_OBJECT (src, "wait for connection idle");
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@ -5424,64 +5468,85 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment)
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GST_DEBUG_OBJECT (src, "stop connection flush");
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gst_rtsp_connection_flush (src->connection, FALSE);
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/* do play */
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res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, control);
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if (res < 0)
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goto create_request_failed;
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/* construct a control url */
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if (src->control)
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control = src->control;
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else
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control = src->req_location;
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if (src->need_range) {
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hval = gen_range_header (src, segment);
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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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gchar *setup_url;
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
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g_free (hval);
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src->need_range = FALSE;
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/* try aggregate control first but do non-aggregate control otherwise */
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if (control)
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setup_url = control;
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else if ((setup_url = stream->setup_url) == NULL)
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continue;
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/* do play */
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res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
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if (res < 0)
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goto create_request_failed;
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if (src->need_range) {
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hval = gen_range_header (src, segment);
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
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g_free (hval);
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src->need_range = FALSE;
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}
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if (segment->rate != 1.0) {
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hval = gst_rtspsrc_dup_printf ("%f", segment->rate);
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if (src->skip)
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
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else
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
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g_free (hval);
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}
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if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
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goto send_error;
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gst_rtsp_message_unset (&request);
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/* parse RTP npt field. This is the current position in the stream (Normal
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* Play Time) and should be put in the NEWSEGMENT position field. */
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if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
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0) == GST_RTSP_OK)
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gst_rtspsrc_parse_range (src, hval, segment);
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/* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
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segment->rate = 1.0;
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/* parse Speed header. This is the intended playback rate of the stream
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* and should be put in the NEWSEGMENT rate field. */
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if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
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0) == GST_RTSP_OK) {
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if (gst_rtspsrc_get_float (hval, &fval) > 0)
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segment->rate = fval;
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} else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
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&hval, 0) == GST_RTSP_OK) {
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if (gst_rtspsrc_get_float (hval, &fval) > 0)
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segment->rate = fval;
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}
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/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
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* for the RTP packets. If this is not present, we assume all starts from 0...
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* This is info for the RTP session manager that we pass to it in caps. */
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hval_idx = 0;
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while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
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&hval, hval_idx++) == GST_RTSP_OK)
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gst_rtspsrc_parse_rtpinfo (src, hval);
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gst_rtsp_message_unset (&response);
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/* early exit when we did aggregate control */
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if (control)
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break;
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}
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if (segment->rate != 1.0) {
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hval = gst_rtspsrc_dup_printf ("%f", segment->rate);
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if (src->skip)
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
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else
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gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
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g_free (hval);
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}
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if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
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goto send_error;
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gst_rtsp_message_unset (&request);
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/* parse RTP npt field. This is the current position in the stream (Normal
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* Play Time) and should be put in the NEWSEGMENT position field. */
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if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
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0) == GST_RTSP_OK)
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gst_rtspsrc_parse_range (src, hval, segment);
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/* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
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segment->rate = 1.0;
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/* parse Speed header. This is the intended playback rate of the stream
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* and should be put in the NEWSEGMENT rate field. */
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if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
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0) == GST_RTSP_OK) {
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if (gst_rtspsrc_get_float (hval, &fval) > 0)
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segment->rate = fval;
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} else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE, &hval,
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0) == GST_RTSP_OK) {
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if (gst_rtspsrc_get_float (hval, &fval) > 0)
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segment->rate = fval;
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}
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/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
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* for the RTP packets. If this is not present, we assume all starts from 0...
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* This is info for the RTP session manager that we pass to it in caps. */
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hval_idx = 0;
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while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
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&hval, hval_idx++) == GST_RTSP_OK)
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gst_rtspsrc_parse_rtpinfo (src, hval);
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gst_rtsp_message_unset (&response);
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/* configure the caps of the streams after we parsed all headers. */
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gst_rtspsrc_configure_caps (src, segment);
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@ -5536,6 +5601,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle)
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{
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GstRTSPMessage request = { 0 };
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GstRTSPMessage response = { 0 };
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GList *walk;
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gchar *control;
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GST_RTSP_STATE_LOCK (src);
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@ -5567,15 +5633,31 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle)
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else
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control = src->req_location;
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/* do pause */
|
||||
if (gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE, control) < 0)
|
||||
goto create_request_failed;
|
||||
/* loop over the streams. We might exit the loop early when we could do an
|
||||
* aggregate control */
|
||||
for (walk = src->streams; walk; walk = g_list_next (walk)) {
|
||||
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
|
||||
gchar *setup_url;
|
||||
|
||||
if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
|
||||
goto send_error;
|
||||
/* try aggregate control first but do non-aggregate control otherwise */
|
||||
if (control)
|
||||
setup_url = control;
|
||||
else if ((setup_url = stream->setup_url) == NULL)
|
||||
continue;
|
||||
|
||||
gst_rtsp_message_unset (&request);
|
||||
gst_rtsp_message_unset (&response);
|
||||
if (gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE, setup_url) < 0)
|
||||
goto create_request_failed;
|
||||
|
||||
if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
|
||||
goto send_error;
|
||||
|
||||
gst_rtsp_message_unset (&request);
|
||||
gst_rtsp_message_unset (&response);
|
||||
|
||||
/* exit early when we did agregate control */
|
||||
if (control)
|
||||
break;
|
||||
}
|
||||
|
||||
if (idle && src->task) {
|
||||
GST_DEBUG_OBJECT (src, "starting idle task again");
|
||||
|
|
|
|||
Loading…
Reference in a new issue