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-rw-r--r--gst/rtpmanager/rtpsource.c118
1 files changed, 88 insertions, 30 deletions
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index ed080717..209c17b5 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -26,7 +26,7 @@
GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
#define GST_CAT_DEFAULT rtp_source_debug
-#define RTP_MAX_PROBATION_LEN 32
+#define RTP_MAX_PROBATION_LEN 32
/* signals and args */
enum
@@ -1091,41 +1091,73 @@ rtp_source_process_bye (RTPSource * src, const gchar * reason)
src->received_bye = TRUE;
}
+static GstBufferListItem
+set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src)
+{
+ *buffer = gst_buffer_make_writable (*buffer);
+ gst_rtp_buffer_set_ssrc (*buffer, src->ssrc);
+ return GST_BUFFER_LIST_SKIP_GROUP;
+}
+
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
- * @buffer: an RTP buffer
+ * @data: an RTP buffer or a list of RTP buffers
+ * @is_list: if @data is a buffer or list
* @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This
* is the buffer timestamp converted to NTP time.
*
- * Send an RTP @buffer originating from @src. This will make @src a sender.
- * This function takes ownership of @buffer and modifies the SSRC in the RTP
- * packet to that of @src when needed.
+ * Send @data (an RTP buffer or list of buffers) originating from @src.
+ * This will make @src a sender. This function takes ownership of @data and
+ * modifies the SSRC in the RTP packet to that of @src when needed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
-rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
+rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
+ guint64 ntpnstime)
{
- GstFlowReturn result = GST_FLOW_OK;
+ GstFlowReturn result;
guint len;
guint32 rtptime;
guint64 ext_rtptime;
guint64 ntp_diff, rtp_diff;
guint64 elapsed;
+ GstBufferList *list = NULL;
+ GstBuffer *buffer = NULL;
+ guint packets;
+ guint32 ssrc;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
- g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+ g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
- len = gst_rtp_buffer_get_payload_len (buffer);
+ if (is_list) {
+ list = GST_BUFFER_LIST_CAST (data);
+ /* We can grab the caps from the first group, since all
+ * groups of a buffer list have same caps. */
+ buffer = gst_buffer_list_get (list, 0, 0);
+ if (!buffer)
+ goto no_buffer;
+ } else {
+ buffer = GST_BUFFER_CAST (data);
+ }
rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
/* we are a sender now */
src->is_sender = TRUE;
+ if (is_list) {
+ /* Each group makes up a network packet. */
+ packets = gst_buffer_list_n_groups (list);
+ len = gst_rtp_buffer_list_get_payload_len (list);
+ } else {
+ packets = 1;
+ len = gst_rtp_buffer_get_payload_len (buffer);
+ }
+
/* update stats for the SR */
- src->stats.packets_sent++;
+ src->stats.packets_sent += packets;
src->stats.octets_sent += len;
src->bytes_sent += len;
@@ -1156,7 +1188,11 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
src->bitrate = 0;
}
- rtptime = gst_rtp_buffer_get_timestamp (buffer);
+ if (is_list) {
+ rtptime = gst_rtp_buffer_list_get_timestamp (list);
+ } else {
+ rtptime = gst_rtp_buffer_get_timestamp (buffer);
+ }
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
@@ -1180,31 +1216,53 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
src->last_ntpnstime = ntpnstime;
/* push packet */
- if (src->callbacks.push_rtp) {
- guint32 ssrc;
+ if (!src->callbacks.push_rtp)
+ goto no_callback;
+ if (is_list) {
+ ssrc = gst_rtp_buffer_list_get_ssrc (list);
+ } else {
ssrc = gst_rtp_buffer_get_ssrc (buffer);
- if (ssrc != src->ssrc) {
- /* the SSRC of the packet is not correct, make a writable buffer and
- * update the SSRC. This could involve a complete copy of the packet when
- * it is not writable. Usually the payloader will use caps negotiation to
- * get the correct SSRC from the session manager before pushing anything. */
- buffer = gst_buffer_make_writable (buffer);
-
- /* FIXME, we don't want to warn yet because we can't inform any payloader
- * of the changes SSRC yet because we don't implement pad-alloc. */
- GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
- src->ssrc);
- gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
+ }
+
+ if (ssrc != src->ssrc) {
+ /* the SSRC of the packet is not correct, make a writable buffer and
+ * update the SSRC. This could involve a complete copy of the packet when
+ * it is not writable. Usually the payloader will use caps negotiation to
+ * get the correct SSRC from the session manager before pushing anything. */
+
+ /* FIXME, we don't want to warn yet because we can't inform any payloader
+ * of the changes SSRC yet because we don't implement pad-alloc. */
+ GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
+ src->ssrc);
+
+ if (is_list) {
+ list = gst_buffer_list_make_writable (list);
+ gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src);
+ } else {
+ set_ssrc (&buffer, 0, 0, src);
}
- GST_LOG ("pushing RTP packet %" G_GUINT64_FORMAT, src->stats.packets_sent);
- result = src->callbacks.push_rtp (src, buffer, src->user_data);
- } else {
- GST_WARNING ("no callback installed, dropping packet");
- gst_buffer_unref (buffer);
}
+ GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
+ src->stats.packets_sent);
+
+ result = src->callbacks.push_rtp (src, data, src->user_data);
return result;
+
+ /* ERRORS */
+no_buffer:
+ {
+ GST_WARNING ("no buffers in buffer list");
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+ return GST_FLOW_OK;
+ }
+no_callback:
+ {
+ GST_WARNING ("no callback installed, dropping packet");
+ gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
+ return GST_FLOW_OK;
+ }
}
/**