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<html>
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<head>
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<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
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<title>GstRTPBaseAudioPayload: GStreamer Base Plugins 1.0 Library Reference Manual</title>
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<meta name="generator" content="DocBook XSL Stylesheets V1.79.1">
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<link rel="home" href="index.html" title="GStreamer Base Plugins 1.0 Library Reference Manual">
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<link rel="up" href="gstreamer-rtp.html" title="RTP Library">
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<link rel="prev" href="gstreamer-rtp.html" title="RTP Library">
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<link rel="next" href="GstRTPBaseDepayload.html" title="GstRTPBaseDepayload">
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<meta name="generator" content="GTK-Doc V1.27 (XML mode)">
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</head>
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<body bgcolor="white" text="black" link="#0000FF" vlink="#840084" alink="#0000FF">
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Top |
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Description |
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Object Hierarchy |
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Properties
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GstRTPBaseAudioPayload
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GstRTPBaseAudioPayload — Base class for audio RTP payloader
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Functions
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void
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gst_rtp_base_audio_payload_set_frame_based ()
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void
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gst_rtp_base_audio_payload_set_frame_options ()
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void
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gst_rtp_base_audio_payload_set_sample_based ()
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void
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gst_rtp_base_audio_payload_set_sample_options ()
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GstAdapter *
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gst_rtp_base_audio_payload_get_adapter ()
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GstFlowReturn
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gst_rtp_base_audio_payload_push ()
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GstFlowReturn
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gst_rtp_base_audio_payload_flush ()
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void
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gst_rtp_base_audio_payload_set_samplebits_options ()
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Properties
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gboolean
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buffer-list
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Read / Write
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Types and Values
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struct
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GstRTPBaseAudioPayload
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struct
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GstRTPBaseAudioPayloadClass
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Object Hierarchy
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GObject
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╰── GInitiallyUnowned
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╰── GstObject
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╰── GstElement
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╰── GstRTPBasePayload
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╰── GstRTPBaseAudioPayload
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Includes
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#include <gst/rtp/gstrtpbaseaudiopayload.h>
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Description
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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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This class derives from GstRTPBasePayload. 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. The RTP packet size is always larger or
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equal to min-ptime (if set). If min-ptime is not set, any residual data is
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sent in a last RTP packet. In the case of frame based codecs, the resulting
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RTP packets always contain full frames.
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Usage
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To use this base class, your child element needs to call either
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gst_rtp_base_audio_payload_set_frame_based() or
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gst_rtp_base_audio_payload_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_rtp_base_audio_payload_set_frame_options(),
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gst_rtp_base_audio_payload_set_sample_options() or
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gst_rtp_base_audio_payload_set_samplebits_options. Since
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GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element
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must set any variables or call/override any functions required by that base
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class. The child element does not need to override any other functions
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specific to GstRTPBaseAudioPayload.
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Functions
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gst_rtp_base_audio_payload_set_frame_based ()
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void
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gst_rtp_base_audio_payload_set_frame_based
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(GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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Tells GstRTPBaseAudioPayload that the child element is for a frame based
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audio codec
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Parameters
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rtpbaseaudiopayload
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a pointer to the element.
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gst_rtp_base_audio_payload_set_frame_options ()
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void
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gst_rtp_base_audio_payload_set_frame_options
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(GstRTPBaseAudioPayload *rtpbaseaudiopayload,
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gint frame_duration,
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gint frame_size);
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Sets the options for frame based audio codecs.
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Parameters
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rtpbaseaudiopayload
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a pointer to the element.
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frame_duration
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The duraction of an audio frame in milliseconds.
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frame_size
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The size of an audio frame in bytes.
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gst_rtp_base_audio_payload_set_sample_based ()
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void
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gst_rtp_base_audio_payload_set_sample_based
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(GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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Tells GstRTPBaseAudioPayload that the child element is for a sample based
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audio codec
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Parameters
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rtpbaseaudiopayload
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a pointer to the element.
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gst_rtp_base_audio_payload_set_sample_options ()
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void
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gst_rtp_base_audio_payload_set_sample_options
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(GstRTPBaseAudioPayload *rtpbaseaudiopayload,
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gint sample_size);
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Sets the options for sample based audio codecs.
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Parameters
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rtpbaseaudiopayload
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a pointer to the element.
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sample_size
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Size per sample in bytes.
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gst_rtp_base_audio_payload_get_adapter ()
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GstAdapter *
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gst_rtp_base_audio_payload_get_adapter
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(GstRTPBaseAudioPayload *rtpbaseaudiopayload);
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Gets the internal adapter used by the depayloader.
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Parameters
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rtpbaseaudiopayload
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a GstRTPBaseAudioPayload
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Returns
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a GstAdapter.
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[transfer full]
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gst_rtp_base_audio_payload_push ()
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GstFlowReturn
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gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload *baseaudiopayload,
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const guint8 *data,
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guint payload_len,
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GstClockTime timestamp);
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Create an RTP buffer and store payload_len
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bytes of data
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as the
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payload. Set the timestamp on the new buffer to timestamp
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before pushing
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the buffer downstream.
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Parameters
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baseaudiopayload
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a GstRTPBasePayload
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data
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data to set as payload.
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[array length=payload_len]
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payload_len
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length of payload
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timestamp
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a GstClockTime
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Returns
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a GstFlowReturn
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gst_rtp_base_audio_payload_flush ()
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GstFlowReturn
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gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload *baseaudiopayload,
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guint payload_len,
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GstClockTime timestamp);
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Create an RTP buffer and store payload_len
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bytes of the adapter as the
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payload. Set the timestamp on the new buffer to timestamp
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before pushing
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the buffer downstream.
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If payload_len
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is -1, all pending bytes will be flushed. If timestamp
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is
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-1, the timestamp will be calculated automatically.
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Parameters
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baseaudiopayload
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a GstRTPBasePayload
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payload_len
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length of payload
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timestamp
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a GstClockTime
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Returns
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a GstFlowReturn
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gst_rtp_base_audio_payload_set_samplebits_options ()
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void
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gst_rtp_base_audio_payload_set_samplebits_options
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(GstRTPBaseAudioPayload *rtpbaseaudiopayload,
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gint sample_size);
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Sets the options for sample based audio codecs.
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Parameters
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rtpbaseaudiopayload
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a pointer to the element.
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sample_size
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Size per sample in bits.
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Types and Values
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struct GstRTPBaseAudioPayload
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struct GstRTPBaseAudioPayload;
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struct GstRTPBaseAudioPayloadClass
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struct GstRTPBaseAudioPayloadClass {
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GstRTPBasePayloadClass parent_class;
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};
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Base class for audio RTP payloader.
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Members
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Property Details
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The “buffer-list” property
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“buffer-list” gboolean
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Use Buffer Lists.
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Flags: Read / Write
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Default value: FALSE
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Generated by GTK-Doc V1.27
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</body>
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</html>
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