From 5ba46c0866dac6f3aa0c461276eab8a5f6874f7b Mon Sep 17 00:00:00 2001 From: Philippe Kalaf Date: Fri, 29 Sep 2006 23:50:53 +0000 Subject: [PATCH] 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 --- ChangeLog | 6 +++++ gst-libs/gst/rtp/README | 23 ----------------- gst-libs/gst/rtp/gstbasertpaudiopayload.c | 41 +++++++++++++++++++++++++++++++ 3 files changed, 47 insertions(+), 23 deletions(-) diff --git a/ChangeLog b/ChangeLog index b3ab21a..567057e 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,9 @@ +2006-09-29 Philippe Kalaf + + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/README: + Moved some documentation into .c file + 2006-09-29 Wim Taymans * gst/playback/gstdecodebin.c: (no_more_pads): diff --git a/gst-libs/gst/rtp/README b/gst-libs/gst/rtp/README index 4507325..77253b7 100644 --- a/gst-libs/gst/rtp/README +++ b/gst-libs/gst/rtp/README @@ -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). diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c index 7727c48..1f700a0 100644 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c +++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c @@ -17,6 +17,47 @@ * Boston, MA 02111-1307, USA. */ +/** + * SECTION:gstbasertpaudiopayload + * @short_description: Base class for audio RTP payloader + * + * + * + * Provides a base class for audio RTP payloaders for frame or sample based + * audio codecs (constant bitrate) + * + * + * + * 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. + * + * + * Usage + * + * 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. + * + * + * + */ + #ifdef HAVE_CONFIG_H #include "config.h" #endif -- 2.7.4