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gst-libs/gst/rtp/: Moved some documentation into .c file
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
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ad4586e590
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3 changed files with 47 additions and 23 deletions
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@ -1,3 +1,9 @@
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2006-09-29 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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* gst-libs/gst/rtp/README:
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Moved some documentation into .c file
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2006-09-29 Wim Taymans <wim@fluendo.com>
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2006-09-29 Wim Taymans <wim@fluendo.com>
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* gst/playback/gstdecodebin.c: (no_more_pads):
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* gst/playback/gstdecodebin.c: (no_more_pads):
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@ -39,29 +39,6 @@ The RTP libraries
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RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
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RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
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-------------------------------------------------------
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-------------------------------------------------------
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This class derives from GstBaseRTPPayload.
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It can be used for payloading audio codecs. It will only work with constant
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bitrate codecs. It supports both frame based and sample based codecs. It takes
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care of packing up the audio data into RTP packets and filling up the headers
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accordingly. The payloading is done based on the maximum MTU (mtu) and the
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maximum time per packet (max-ptime). The general idea is to divide large data
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buffers into smaller RTP packets. The RTP packet size is the minimum of either
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the MTU, max-ptime (if set) or available data. Any residual data is always
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sent in a last RTP packet (no minimum RTP packet size). The idea is that since
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this is a real time protocol, data should never be delayed. In the case of
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frame based codecs, the resulting RTP packets always contain full frames.
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To use this base class, your child element needs to call either
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gst_basertpaudiopayload_set_frame_based() or
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gst_basertpaudiopayload_set_sample_based(). This is usually done in the
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element's _init() function. Then, the child element must call either
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gst_basertpaudiopayload_set_frame_options() or
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gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
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derives from GstBaseRTPPayload, the child element must set any variables or
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call/override any functions required by that base class. The child element
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does not need to override any other functions specific to
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GstBaseRTPAudioPayload.
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This base class can be tested through it's children classes. Here is an
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This base class can be tested through it's children classes. Here is an
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example using the iLBC payloader (frame based).
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example using the iLBC payloader (frame based).
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@ -17,6 +17,47 @@
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* Boston, MA 02111-1307, USA.
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* Boston, MA 02111-1307, USA.
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*/
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*/
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/**
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* SECTION:gstbasertpaudiopayload
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* @short_description: Base class for audio RTP payloader
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*
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* <refsect2>
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* <para>
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* Provides a base class for audio RTP payloaders for frame or sample based
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* audio codecs (constant bitrate)
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* </para>
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*
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* <para>
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* This class derives from GstBaseRTPPayload. It can be used for payloading
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* audio codecs. It will only work with constant bitrate codecs. It supports
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* both frame based and sample based codecs. It takes care of packing up the
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* audio data into RTP packets and filling up the headers accordingly. The
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* payloading is done based on the maximum MTU (mtu) and the maximum time per
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* packet (max-ptime). The general idea is to divide large data buffers into
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* smaller RTP packets. The RTP packet size is the minimum of either the MTU,
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* max-ptime (if set) or available data. Any residual data is always sent in a
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* last RTP packet (no minimum RTP packet size). A minimum packet size might be
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* added in future versions if the need arises. In the case of frame
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* based codecs, the resulting RTP packets always contain full frames.
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* </para>
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*
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* <title>Usage</title>
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* <para>
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* To use this base class, your child element needs to call either
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* gst_basertpaudiopayload_set_frame_based() or
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* gst_basertpaudiopayload_set_sample_based(). This is usually done in the
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* element's _init() function. Then, the child element must call either
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* gst_basertpaudiopayload_set_frame_options() or
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* gst_basertpaudiopayload_set_sample_options(). Since GstBaseRTPAudioPayload
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* derives from GstBaseRTPPayload, the child element must set any variables or
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* call/override any functions required by that base class. The child element
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* does not need to override any other functions specific to
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* GstBaseRTPAudioPayload.
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* </para>
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* </refsect2>
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*
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*/
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#ifdef HAVE_CONFIG_H
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#include "config.h"
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#endif
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#endif
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