gstrtpbaseaudiopayload: GStreamer Base Plugins 1.0 Library Reference Manual

## gstrtpbaseaudiopayload gstrtpbaseaudiopayload — Base class for audio RTP payloader |

## Functions

void | gst_rtp_base_audio_payload_set_frame_based ()
void | gst_rtp_base_audio_payload_set_frame_options ()
void | gst_rtp_base_audio_payload_set_sample_based ()
void | gst_rtp_base_audio_payload_set_sample_options ()
GstAdapter * | gst_rtp_base_audio_payload_get_adapter ()
GstFlowReturn | gst_rtp_base_audio_payload_push ()
GstFlowReturn | gst_rtp_base_audio_payload_flush ()
void | gst_rtp_base_audio_payload_set_samplebits_options ()

## Properties

gboolean | buffer-list | Read / Write

## Types and Values

struct | GstRTPBaseAudioPayload
struct | GstRTPBaseAudioPayloadClass

## 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