docs: fix typo s/incomming/incoming/

diff --git a/gst/debugutils/gsttaginject.c b/gst/debugutils/gsttaginject.c
index abdb797..46b0d9a 100644 (file)
--- a/gst/debugutils/gsttaginject.c
+++ b/gst/debugutils/gsttaginject.c
@@ -21,7 +21,7 @@
 /**
  * SECTION:element-taginject
  *
- * Element that injects new metadata tags, but passes incomming data through
+ * Element that injects new metadata tags, but passes incoming data through
  * unmodified.
  *
  * <refsect2>

diff --git a/gst/goom2k1/gstgoom.c b/gst/goom2k1/gstgoom.c
index cfbfcbc..19eda10 100644 (file)
--- a/gst/goom2k1/gstgoom.c
+++ b/gst/goom2k1/gstgoom.c
@@ -24,7 +24,7 @@
  * @see_also: goom, synaesthesia
  *
  * Goom2k1 is an audio visualisation element. It creates warping structures
- * based on the incomming audio signal. Goom2k1 is the older version of the
+ * based on the incoming audio signal. Goom2k1 is the older version of the
  * visualisation. Also available is goom2k4, with a different look.
  *
  * <refsect2>

diff --git a/gst/monoscope/gstmonoscope.c b/gst/monoscope/gstmonoscope.c
index 26c4338..1c8f9a8 100644 (file)
--- a/gst/monoscope/gstmonoscope.c
+++ b/gst/monoscope/gstmonoscope.c
@@ -477,7 +477,7 @@ gst_monoscope_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
     case GST_EVENT_SEGMENT:
     {
       /* the newsegment values are used to clip the input samples
-       * and to convert the incomming timestamps to running time so
+       * and to convert the incoming timestamps to running time so
        * we can do QoS */
       gst_event_copy_segment (event, &monoscope->segment);
 

diff --git a/gst/rtp/README b/gst/rtp/README
index 814311e..11bd90d 100644 (file)
--- a/gst/rtp/README
+++ b/gst/rtp/README
@@ -186,7 +186,7 @@ Some pipelines to illustrate the process:
   v4l2src puts a GStreamer timestamp on the video frames base on the current
   running_time. The encoder encodes and passed the timestamp on. The payloader
   generates an RTP timestamp using the above formula and puts it in the RTP
-  packet. It also copies the incomming GStreamer timestamp on the output RTP
+  packet. It also copies the incoming GStreamer timestamp on the output RTP
   packet. udpsink synchronizes on the gstreamer timestamp before pushing out the
   packet. 
 
@@ -208,7 +208,7 @@ following pipeline:
       clock-rate=(int)90000, encoding-name=(string)H263-1998" ! rtph263pdepay !
       avdec_h263 ! autovideosink
 
-It is important that the depayloader copies the incomming GStreamer timestamp
+It is important that the depayloader copies the incoming GStreamer timestamp
 directly to the depayloaded output buffer. It should never attempt to perform
 any logic with the RTP timestamp, this task is for the jitterbuffer as we will
 see next.

diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c
index 57020dc..7c797b7 100644 (file)
--- a/gst/rtp/gstrtpac3pay.c
+++ b/gst/rtp/gstrtpac3pay.c
@@ -362,7 +362,7 @@ gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * basepayload,
     gst_rtp_ac3_pay_reset (rtpac3pay);
   }
 
-  /* count the amount of incomming packets */
+  /* count the amount of incoming packets */
   NF = 0;
   left = map.size;
   p = map.data;

diff --git a/gst/rtp/gstrtpmp4apay.c b/gst/rtp/gstrtpmp4apay.c
index 79db0c4..2f01b21 100644 (file)
--- a/gst/rtp/gstrtpmp4apay.c
+++ b/gst/rtp/gstrtpmp4apay.c
@@ -442,7 +442,7 @@ gst_rtp_mp4a_pay_handle_buffer (GstRTPBasePayload * basepayload,
     offset += payload_len;
     size -= payload_len;
 
-    /* copy incomming timestamp (if any) to outgoing buffers */
+    /* copy incoming timestamp (if any) to outgoing buffers */
     GST_BUFFER_PTS (outbuf) = timestamp;
 
     fragmented = TRUE;

diff --git a/gst/rtp/gstrtpmp4vpay.c b/gst/rtp/gstrtpmp4vpay.c
index 0184f60..362ff65 100644 (file)
--- a/gst/rtp/gstrtpmp4vpay.c
+++ b/gst/rtp/gstrtpmp4vpay.c
@@ -450,7 +450,7 @@ gst_rtp_mp4v_pay_handle_buffer (GstRTPBasePayload * basepayload,
     rtpmp4vpay->duration = 0;
   }
 
-  /* depay incomming data and see if we need to start a new RTP
+  /* depay incoming data and see if we need to start a new RTP
    * packet */
   flush =
       gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, map.data, size, &strip, &vopi);

diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index d07fbed..7847f1a 100644 (file)
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -47,7 +47,7 @@
  * depayloader or other element to create concealment data or some other logic
  * to gracefully handle the missing packets.
  *
- * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
+ * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incoming
  * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
  * buffer.
  *

diff --git a/gst/rtpmanager/rtpjitterbuffer.c b/gst/rtpmanager/rtpjitterbuffer.c
index 0308a53..309d68d 100644 (file)
--- a/gst/rtpmanager/rtpjitterbuffer.c
+++ b/gst/rtpmanager/rtpjitterbuffer.c
@@ -1046,7 +1046,7 @@ duplicate:
  * @percent: the buffering percent
  *
  * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
- * have its timestamp adjusted with the incomming running_time and the detected
+ * have its timestamp adjusted with the incoming running_time and the detected
  * clock skew.
  *
  * Returns: a #GstBuffer or %NULL when there was no packet in the queue.

diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index c381b4f..52fe81d 100644 (file)
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -1220,7 +1220,7 @@ probation_seqnum:
  * @src: an #RTPSource
  * @pinfo: an #RTPPacketInfo
  *
- * Let @src handle the incomming RTP packet described in @pinfo.
+ * Let @src handle the incoming RTP packet described in @pinfo.
  *
  * Returns: a #GstFlowReturn.
  */

diff --git a/gst/smpte/gstsmpte.c b/gst/smpte/gstsmpte.c
index 18c522c..7e88641 100644 (file)
--- a/gst/smpte/gstsmpte.c
+++ b/gst/smpte/gstsmpte.c
@@ -21,7 +21,7 @@
  * SECTION:element-smpte
  *
  * smpte can accept I420 video streams with the same width, height and
- * framerate. The two incomming buffers are blended together using an effect
+ * framerate. The two incoming buffers are blended together using an effect
  * specific alpha mask. 
  *
  * The #GstSmpte:depth property defines the presision in bits of the mask. A