gstbasertpaudiopayload

gstbasertpaudiopayload — Base class for audio RTP payloader

Functions

Properties

gboolean buffer-list Read / Write

Types and Values

Object Hierarchy

    GObject
    ╰── GstObject
        ╰── GstElement
            ╰── GstBaseRTPPayload
                ╰── GstBaseRTPAudioPayload

Includes

#include <gst/rtp/gstbasertpaudiopayload.h>

Description

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. 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_base_rtp_audio_payload_set_frame_based() or gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the element's _init() function. Then, the child element must call either gst_base_rtp_audio_payload_set_frame_options(), gst_base_rtp_audio_payload_set_sample_options() or gst_base_rtp_audio_payload_set_samplebits_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.

Functions

gst_base_rtp_audio_payload_set_frame_based ()

void
gst_base_rtp_audio_payload_set_frame_based
                               (GstBaseRTPAudioPayload *basertpaudiopayload);

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

Parameters

basertpaudiopayload

a pointer to the element.

 

gst_base_rtp_audio_payload_set_frame_options ()

void
gst_base_rtp_audio_payload_set_frame_options
                               (GstBaseRTPAudioPayload *basertpaudiopayload,
                                gint frame_duration,
                                gint frame_size);

Sets the options for frame based audio codecs.

Parameters

basertpaudiopayload

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_base_rtp_audio_payload_set_sample_based ()

void
gst_base_rtp_audio_payload_set_sample_based
                               (GstBaseRTPAudioPayload *basertpaudiopayload);

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

Parameters

basertpaudiopayload

a pointer to the element.

 

gst_base_rtp_audio_payload_set_sample_options ()

void
gst_base_rtp_audio_payload_set_sample_options
                               (GstBaseRTPAudioPayload *basertpaudiopayload,
                                gint sample_size);

Sets the options for sample based audio codecs.

Parameters

basertpaudiopayload

a pointer to the element.

 

sample_size

Size per sample in bytes.

 

gst_base_rtp_audio_payload_get_adapter ()

GstAdapter *
gst_base_rtp_audio_payload_get_adapter
                               (GstBaseRTPAudioPayload *basertpaudiopayload);

Gets the internal adapter used by the depayloader.

Parameters

basertpaudiopayload

a GstBaseRTPAudioPayload

 

Returns

a GstAdapter.

Since 0.10.13


gst_base_rtp_audio_payload_push ()

GstFlowReturn
gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload *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 GstBaseRTPPayload

 

data

data to set as payload

 

payload_len

length of payload

 

timestamp

a GstClockTime

 

Returns

a GstFlowReturn

Since 0.10.13


gst_base_rtp_audio_payload_flush ()

GstFlowReturn
gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload *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 GstBaseRTPPayload

 

payload_len

length of payload

 

timestamp

a GstClockTime

 

Returns

a GstFlowReturn

Since 0.10.25


gst_base_rtp_audio_payload_set_samplebits_options ()

void
gst_base_rtp_audio_payload_set_samplebits_options
                               (GstBaseRTPAudioPayload *basertpaudiopayload,
                                gint sample_size);

Sets the options for sample based audio codecs.

Parameters

basertpaudiopayload

a pointer to the element.

 

sample_size

Size per sample in bits.

 

Since 0.10.18

Types and Values

struct GstBaseRTPAudioPayload

struct GstBaseRTPAudioPayload;

struct GstBaseRTPAudioPayloadClass

struct GstBaseRTPAudioPayloadClass {
  GstBaseRTPPayloadClass parent_class;
};

Base class for audio RTP payloader.

Members

GstBaseRTPPayloadClass parent_class;

the parent class

 

Property Details

The “buffer-list” property

  “buffer-list”              gboolean

Use Buffer Lists.

Flags: Read / Write

Default value: FALSE