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gst-libs/gst/rtp/: Moved some documentation into .c file

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Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/README:
Moved some documentation into .c file
This commit is contained in:
Philippe Kalaf 2006-09-29 23:50:53 +00:00
parent ad4586e590
commit 5ba46c0866
3 changed files with 47 additions and 23 deletions
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6
ChangeLog
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@ -1,3 +1,9 @@
2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/README:
Moved some documentation into .c file
2006-09-29 Wim Taymans <wim@fluendo.com>
* gst/playback/gstdecodebin.c: (no_more_pads):

23
gst-libs/gst/rtp/README
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@ -39,29 +39,6 @@ The RTP libraries
RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
-------------------------------------------------------
This class derives from GstBaseRTPPayload.
It can be used for payloading audio codecs. It will only work with constant
bitrate codecs. It supports both frame based and sample based codecs. It takes
care of packing up the audio data into RTP packets and filling up the headers
accordingly. The payloading is done based on the maximum MTU (mtu) and the
maximum time per packet (max-ptime). The general idea is to divide large data
buffers into smaller RTP packets. The RTP packet size is the minimum of either
the MTU, max-ptime (if set) or available data. Any residual data is always
sent in a last RTP packet (no minimum RTP packet size). The idea is that since
this is a real time protocol, data should never be delayed. In the case of
frame based codecs, the resulting RTP packets always contain full frames.
To use this base class, your child element needs to call either
gst_basertpaudiopayload_set_frame_based() or
gst_basertpaudiopayload_set_sample_based(). This is usually done in the
element's _init() function. Then, the child element must call either
gst_basertpaudiopayload_set_frame_options() or
gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
derives from GstBaseRTPPayload, the child element must set any variables or
call/override any functions required by that base class. The child element
does not need to override any other functions specific to
GstBaseRTPAudioPayload.
This base class can be tested through it's children classes. Here is an
example using the iLBC payloader (frame based).

41
gst-libs/gst/rtp/gstbasertpaudiopayload.c
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@ -17,6 +17,47 @@
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstbasertpaudiopayload
* @short_description: Base class for audio RTP payloader
*
* <refsect2>
* <para>
* Provides a base class for audio RTP payloaders for frame or sample based
* audio codecs (constant bitrate)
* </para>
*
* <para>
* This class derives from GstBaseRTPPayload. It can be used for payloading
* audio codecs. It will only work with constant bitrate codecs. It supports
* both frame based and sample based codecs. It takes care of packing up the
* audio data into RTP packets and filling up the headers accordingly. The
* payloading is done based on the maximum MTU (mtu) and the maximum time per
* packet (max-ptime). The general idea is to divide large data buffers into
* smaller RTP packets. The RTP packet size is the minimum of either the MTU,
* max-ptime (if set) or available data. Any residual data is always sent in a
* last RTP packet (no minimum RTP packet size). A minimum packet size might be
* added in future versions if the need arises. In the case of frame
* based codecs, the resulting RTP packets always contain full frames.
* </para>
*
* <title>Usage</title>
* <para>
* To use this base class, your child element needs to call either
* gst_basertpaudiopayload_set_frame_based() or
* gst_basertpaudiopayload_set_sample_based(). This is usually done in the
* element's _init() function. Then, the child element must call either
* gst_basertpaudiopayload_set_frame_options() or
* gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
* derives from GstBaseRTPPayload, the child element must set any variables or
* call/override any functions required by that base class. The child element
* does not need to override any other functions specific to
* GstBaseRTPAudioPayload.
* </para>
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif