diff options
author | Philippe Kalaf <philippe.kalaf@collabora.co.uk> | 2006-09-29 23:50:53 +0000 |
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committer | Philippe Kalaf <philippe.kalaf@collabora.co.uk> | 2006-09-29 23:50:53 +0000 |
commit | 5ba46c0866dac6f3aa0c461276eab8a5f6874f7b (patch) | |
tree | 9527b0d8d14bf9e0187b093d1dbc3797a562cc2c /gst-libs | |
parent | ad4586e590c182ae7ac671e4491054b88913539f (diff) |
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
Diffstat (limited to 'gst-libs')
-rw-r--r-- | gst-libs/gst/rtp/README | 23 | ||||
-rw-r--r-- | gst-libs/gst/rtp/gstbasertpaudiopayload.c | 41 |
2 files changed, 41 insertions, 23 deletions
diff --git a/gst-libs/gst/rtp/README b/gst-libs/gst/rtp/README index 4507325f8c..77253b783f 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 7727c484d0..1f700a011f 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 + * + * <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 |