mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
rtph264pay: Clean up whitespace and syntax
This commit is contained in:
parent
37d22186ff
commit
28d6dfa51f
1 changed files with 6 additions and 6 deletions
|
@ -68,13 +68,13 @@ GST_STATIC_PAD_TEMPLATE ("src",
|
|||
);
|
||||
|
||||
#define DEFAULT_SPROP_PARAMETER_SETS NULL
|
||||
#define DEFAULT_CONFIG_INTERVAL 0
|
||||
#define DEFAULT_CONFIG_INTERVAL 0
|
||||
|
||||
enum
|
||||
{
|
||||
PROP_0,
|
||||
PROP_SPROP_PARAMETER_SETS,
|
||||
PROP_CONFIG_INTERVAL
|
||||
PROP_CONFIG_INTERVAL,
|
||||
};
|
||||
|
||||
static void gst_rtp_h264_pay_finalize (GObject * object);
|
||||
|
@ -1091,7 +1091,7 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|||
|
||||
/* now loop over all NAL units and put them in a packet
|
||||
* FIXME, we should really try to pack multiple NAL units into one RTP packet
|
||||
* if we can, especially for the config packets that wont't cause decoder
|
||||
* if we can, especially for the config packets that wont't cause decoder
|
||||
* latency. */
|
||||
if (avc) {
|
||||
guint nal_length_size;
|
||||
|
@ -1177,10 +1177,10 @@ gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|||
size -= 3;
|
||||
|
||||
/* use next_start_code() to scan buffer.
|
||||
* next_start_code() returns the offset in data,
|
||||
* next_start_code() returns the offset in data,
|
||||
* starting from zero to the first byte of 0.0.0.1
|
||||
* If no start code is found, it returns the value of the
|
||||
* 'size' parameter.
|
||||
* If no start code is found, it returns the value of the
|
||||
* 'size' parameter.
|
||||
* data is unchanged by the call to next_start_code()
|
||||
*/
|
||||
next = next_start_code (data, size);
|
||||
|
|
Loading…
Reference in a new issue