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GStreamer Base Plugins 0.10 Library Reference Manual | ![]() |
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#include <gst/rtp/gstbasertpaudiopayload.h> GstBaseRTPAudioPayload; GstBaseRTPAudioPayloadClass; void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload); void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload *basertpaudiopayload, gint frame_duration, gint frame_size); void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload); void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload, gint sample_size);
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. 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.
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.
typedef struct { GstBaseRTPPayload payload; GstBaseRTPAudioPayloadPrivate *priv; GstClockTime base_ts; gint frame_size; gint frame_duration; gint sample_size; gpointer _gst_reserved[GST_PADDING]; } GstBaseRTPAudioPayload;
typedef struct { GstBaseRTPPayloadClass parent_class; gpointer _gst_reserved[GST_PADDING]; } GstBaseRTPAudioPayloadClass;
void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
Tells GstBaseRTPAudioPayload that the child element is for a frame based audio codec
basertpaudiopayload : | a pointer to the element. |
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.
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. |
void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
Tells GstBaseRTPAudioPayload that the child element is for a sample based audio codec
basertpaudiopayload : | a pointer to the element. |
void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload, gint sample_size);
Sets the options for sample based audio codecs.
basertpaudiopayload : | a pointer to the element. |
sample_size : | Size per sample in bytes. |