manpagez: man pages & more
html files: gst-plugins-base-libs-1.0
Home | html | info | man

GstRTPBaseAudioPayload

GstRTPBaseAudioPayload — Base class for audio RTP payloader

Properties

gboolean buffer-list Read / Write

Types and Values

Object Hierarchy

    GObject
    ╰── GInitiallyUnowned
        ╰── GstObject
            ╰── GstElement
                ╰── GstRTPBasePayload
                    ╰── GstRTPBaseAudioPayload

Includes

#include <gst/rtp/gstrtpbaseaudiopayload.h>

Description

Provides a base class for audio RTP payloaders for frame or sample based audio codecs (constant bitrate)

This class derives from GstRTPBasePayload. 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. The RTP packet size is always larger or equal to min-ptime (if set). If min-ptime is not set, any residual data is sent in a last RTP packet. 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_rtp_base_audio_payload_set_frame_based() or gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the element's _init() function. Then, the child element must call either gst_rtp_base_audio_payload_set_frame_options(), gst_rtp_base_audio_payload_set_sample_options() or gst_rtp_base_audio_payload_set_samplebits_options. Since GstRTPBaseAudioPayload derives from GstRTPBasePayload, 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 GstRTPBaseAudioPayload.

Functions

gst_rtp_base_audio_payload_set_frame_based ()

void
gst_rtp_base_audio_payload_set_frame_based
                               (GstRTPBaseAudioPayload *rtpbaseaudiopayload);

Tells GstRTPBaseAudioPayload that the child element is for a frame based audio codec

Parameters

rtpbaseaudiopayload

a pointer to the element.

 

gst_rtp_base_audio_payload_set_frame_options ()

void
gst_rtp_base_audio_payload_set_frame_options
                               (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
                                gint frame_duration,
                                gint frame_size);

Sets the options for frame based audio codecs.

Parameters

rtpbaseaudiopayload

a pointer to the element.

 

frame_duration

The duraction of an audio frame in milliseconds.

 

frame_size

The size of an audio frame in bytes.

 

gst_rtp_base_audio_payload_set_sample_based ()

void
gst_rtp_base_audio_payload_set_sample_based
                               (GstRTPBaseAudioPayload *rtpbaseaudiopayload);

Tells GstRTPBaseAudioPayload that the child element is for a sample based audio codec

Parameters

rtpbaseaudiopayload

a pointer to the element.

 

gst_rtp_base_audio_payload_set_sample_options ()

void
gst_rtp_base_audio_payload_set_sample_options
                               (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
                                gint sample_size);

Sets the options for sample based audio codecs.

Parameters

rtpbaseaudiopayload

a pointer to the element.

 

sample_size

Size per sample in bytes.

 

gst_rtp_base_audio_payload_get_adapter ()

GstAdapter *
gst_rtp_base_audio_payload_get_adapter
                               (GstRTPBaseAudioPayload *rtpbaseaudiopayload);

Gets the internal adapter used by the depayloader.

Parameters

rtpbaseaudiopayload

a GstRTPBaseAudioPayload

 

Returns

a GstAdapter.

[transfer full]


gst_rtp_base_audio_payload_push ()

GstFlowReturn
gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload *baseaudiopayload,
                                 const guint8 *data,
                                 guint payload_len,
                                 GstClockTime timestamp);

Create an RTP buffer and store payload_len bytes of data as the payload. Set the timestamp on the new buffer to timestamp before pushing the buffer downstream.

Parameters

baseaudiopayload

a GstRTPBasePayload

 

data

data to set as payload

 

payload_len

length of payload

 

timestamp

a GstClockTime

 

Returns

a GstFlowReturn


gst_rtp_base_audio_payload_flush ()

GstFlowReturn
gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload *baseaudiopayload,
                                  guint payload_len,
                                  GstClockTime timestamp);

Create an RTP buffer and store payload_len bytes of the adapter as the payload. Set the timestamp on the new buffer to timestamp before pushing the buffer downstream.

If payload_len is -1, all pending bytes will be flushed. If timestamp is -1, the timestamp will be calculated automatically.

Parameters

baseaudiopayload

a GstRTPBasePayload

 

payload_len

length of payload

 

timestamp

a GstClockTime

 

Returns

a GstFlowReturn


gst_rtp_base_audio_payload_set_samplebits_options ()

void
gst_rtp_base_audio_payload_set_samplebits_options
                               (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
                                gint sample_size);

Sets the options for sample based audio codecs.

Parameters

rtpbaseaudiopayload

a pointer to the element.

 

sample_size

Size per sample in bits.

 

Types and Values

struct GstRTPBaseAudioPayload

struct GstRTPBaseAudioPayload;

struct GstRTPBaseAudioPayloadClass

struct GstRTPBaseAudioPayloadClass {
  GstRTPBasePayloadClass parent_class;
};

Base class for audio RTP payloader.

Members

Property Details

The “buffer-list” property

  “buffer-list”              gboolean

Use Buffer Lists.

Flags: Read / Write

Default value: FALSE

© manpagez.com 2000-2017
Individual documents may contain additional copyright information.