diff -r 000000000000 -r 0e761a78d257 gst_plugins_base/gst-libs/gst/rtp/gstbasertpaudiopayload.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/gst_plugins_base/gst-libs/gst/rtp/gstbasertpaudiopayload.c Thu Dec 17 08:53:32 2009 +0200 @@ -0,0 +1,728 @@ +/* GStreamer + * Copyright (C) <2006> Philippe Khalaf + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:gstbasertpaudiopayload + * @short_description: Base class for audio RTP payloader + * + * + * + * 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. + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include + +#include "gstbasertpaudiopayload.h" + +GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug); +#define GST_CAT_DEFAULT (basertpaudiopayload_debug) + +typedef enum +{ + AUDIO_CODEC_TYPE_NONE, + AUDIO_CODEC_TYPE_FRAME_BASED, + AUDIO_CODEC_TYPE_SAMPLE_BASED +} AudioCodecType; + +struct _GstBaseRTPAudioPayloadPrivate +{ + AudioCodecType type; + GstAdapter *adapter; + guint64 min_ptime; +}; + + +#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \ + GstBaseRTPAudioPayloadPrivate)) + +static void gst_base_rtp_audio_payload_finalize (GObject * object); + +static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload + * payload, GstBuffer * buffer); + +static GstFlowReturn +gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer); + +static GstFlowReturn +gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer); + +static GstStateChangeReturn +gst_base_rtp_payload_audio_change_state (GstElement * element, + GstStateChange transition); +static gboolean +gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event); + +GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload, + GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD); +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + + +static void +gst_base_rtp_audio_payload_base_init (gpointer klass) +{ +} + +static void +gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate)); + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + gobject_class->finalize = + GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_finalize); + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state); + + gstbasertppayload_class->handle_buffer = + GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer); + gstbasertppayload_class->handle_event = + GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event); + + GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0, + "base audio RTP payloader"); +} + +static void +gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload, + GstBaseRTPAudioPayloadClass * klass) +{ + basertpaudiopayload->priv = + GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload); + + basertpaudiopayload->base_ts = 0; + + basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_NONE; + + /* these need to be set by child object if frame based */ + basertpaudiopayload->frame_size = 0; + basertpaudiopayload->frame_duration = 0; + + /* these need to be set by child object if sample based */ + basertpaudiopayload->sample_size = 0; + + basertpaudiopayload->priv->adapter = gst_adapter_new (); +} + +static void +gst_base_rtp_audio_payload_finalize (GObject * object) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object); + + g_object_unref (basertpaudiopayload->priv->adapter); + + GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object)); +} + +/** + * gst_base_rtp_audio_payload_set_frame_based: + * @basertpaudiopayload: a pointer to the element. + * + * Tells #GstBaseRTPAudioPayload that the child element is for a frame based + * audio codec + * + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +void +gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload * + basertpaudiopayload) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE); + + basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED; +} + +/** + * gst_base_rtp_audio_payload_set_sample_based: + * @basertpaudiopayload: a pointer to the element. + * + * Tells #GstBaseRTPAudioPayload that the child element is for a sample based + * audio codec + * + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +void +gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * + basertpaudiopayload) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE); + + basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED; +} + +/** + * gst_base_rtp_audio_payload_set_frame_options: + * @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. + * + * Sets the options for frame based audio codecs. + * + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +void +gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload + * basertpaudiopayload, gint frame_duration, gint frame_size) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + basertpaudiopayload->frame_size = frame_size; + basertpaudiopayload->frame_duration = frame_duration; + + if (basertpaudiopayload->priv->adapter) { + gst_adapter_clear (basertpaudiopayload->priv->adapter); + } +} + +/** + * gst_base_rtp_audio_payload_set_sample_options: + * @basertpaudiopayload: a pointer to the element. + * @sample_size: Size per sample in bytes. + * + * Sets the options for sample based audio codecs. + * + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +void +gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload + * basertpaudiopayload, gint sample_size) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + /* sample_size is in bits internally */ + basertpaudiopayload->sample_size = sample_size * 8; + + if (basertpaudiopayload->priv->adapter) { + gst_adapter_clear (basertpaudiopayload->priv->adapter); + } +} + +/** + * gst_base_rtp_audio_payload_set_samplebits_options: + * @basertpaudiopayload: a pointer to the element. + * @sample_size: Size per sample in bits. + * + * Sets the options for sample based audio codecs. + * + * Since: 0.10.18 + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +void +gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload + * basertpaudiopayload, gint sample_size) +{ + g_return_if_fail (basertpaudiopayload != NULL); + + basertpaudiopayload->sample_size = sample_size; + + if (basertpaudiopayload->priv->adapter) { + gst_adapter_clear (basertpaudiopayload->priv->adapter); + } +} + +static GstFlowReturn +gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload, + GstBuffer * buffer) +{ + GstFlowReturn ret; + GstBaseRTPAudioPayload *basertpaudiopayload; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + + ret = GST_FLOW_ERROR; + + if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) { + ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload, + buffer); + } else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) { + ret = gst_base_rtp_audio_payload_handle_sample_based_buffer (basepayload, + buffer); + } else { + GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set"); + } + + return ret; +} + +/* this assumes all frames have a constant duration and a constant size */ +static GstFlowReturn +gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + guint payload_len; + const guint8 *data = NULL; + GstFlowReturn ret; + guint available; + gint frame_size, frame_duration; + + guint maxptime_octets = G_MAXUINT; + guint minptime_octets = 0; + guint min_payload_len; + guint max_payload_len; + gboolean use_adapter = FALSE; + guint minptime_ms; + + ret = GST_FLOW_OK; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + + if (basertpaudiopayload->frame_size == 0 || + basertpaudiopayload->frame_duration == 0) { + GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); + gst_buffer_unref (buffer); + return GST_FLOW_ERROR; + } + frame_size = basertpaudiopayload->frame_size; + frame_duration = basertpaudiopayload->frame_duration; + + /* max number of bytes based on given ptime, has to be multiple of + * frame_duration */ + if (basepayload->max_ptime != -1) { + guint ptime_ms = basepayload->max_ptime / 1000000; + + maxptime_octets = frame_size * (int) (ptime_ms / frame_duration); + if (maxptime_octets == 0) { + GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than" + " minimum %d ms, overwriting to minimum", ptime_ms, frame_duration); + maxptime_octets = frame_size; + } + } + + max_payload_len = MIN ( + /* MTU max */ + (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU + (basertpaudiopayload), 0, 0) / frame_size) * frame_size, + /* ptime max */ + maxptime_octets); + + /* min number of bytes based on a given ptime, has to be a multiple + of frame duration */ + minptime_ms = basepayload->min_ptime / 1000000; + + minptime_octets = frame_size * (int) (minptime_ms / frame_duration); + + min_payload_len = MAX (minptime_octets, frame_size); + + if (min_payload_len > max_payload_len) { + min_payload_len = max_payload_len; + } + + GST_DEBUG_OBJECT (basertpaudiopayload, + "Calculated min_payload_len %u and max_payload_len %u", + min_payload_len, max_payload_len); + + if (basertpaudiopayload->priv->adapter && + gst_adapter_available (basertpaudiopayload->priv->adapter)) { + /* If there is always data in the adapter, we have to use it */ + gst_adapter_push (basertpaudiopayload->priv->adapter, buffer); + available = gst_adapter_available (basertpaudiopayload->priv->adapter); + use_adapter = TRUE; + } else { + /* let's set the base timestamp */ + basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer); + + /* If buffer fits on an RTP packet, let's just push it through */ + /* this will check against max_ptime and max_mtu */ + if (GST_BUFFER_SIZE (buffer) >= min_payload_len && + GST_BUFFER_SIZE (buffer) <= max_payload_len) { + ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, + GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), + GST_BUFFER_TIMESTAMP (buffer)); + gst_buffer_unref (buffer); + + return ret; + } + + available = GST_BUFFER_SIZE (buffer); + data = (guint8 *) GST_BUFFER_DATA (buffer); + } + + /* as long as we have full frames */ + while (available >= min_payload_len) { + gfloat ts_inc; + + /* We send as much as we can */ + payload_len = MIN (max_payload_len, (available / frame_size) * frame_size); + + if (use_adapter) { + data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len); + } + + ret = + gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len, + basertpaudiopayload->base_ts); + + ts_inc = (payload_len * frame_duration) / frame_size; + + ts_inc = ts_inc * GST_MSECOND; + basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc); + + if (use_adapter) { + gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len); + available = gst_adapter_available (basertpaudiopayload->priv->adapter); + } else { + available -= payload_len; + data += payload_len; + } + } + + if (!use_adapter) { + if (available != 0 && basertpaudiopayload->priv->adapter) { + GstBuffer *buf; + + buf = gst_buffer_create_sub (buffer, + GST_BUFFER_SIZE (buffer) - available, available); + gst_adapter_push (basertpaudiopayload->priv->adapter, buf); + } + gst_buffer_unref (buffer); + } + + return ret; +} + +static GstFlowReturn +gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload * + basepayload, GstBuffer * buffer) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + guint payload_len; + const guint8 *data = NULL; + GstFlowReturn ret; + guint available; + + guint maxptime_octets = G_MAXUINT; + guint minptime_octets = 0; + guint min_payload_len; + guint max_payload_len; + gboolean use_adapter = FALSE; + + guint fragment_size; + + ret = GST_FLOW_OK; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload); + + if (basertpaudiopayload->sample_size == 0) { + GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set"); + gst_buffer_unref (buffer); + return GST_FLOW_ERROR; + } + + /* sample_size is in bits and is converted into multiple bytes */ + fragment_size = basertpaudiopayload->sample_size; + while ((fragment_size % 8) != 0) + fragment_size += fragment_size; + fragment_size /= 8; + + /* max number of bytes based on given ptime */ + if (basepayload->max_ptime != -1) { + maxptime_octets = 8 * basepayload->max_ptime * basepayload->clock_rate / + (basertpaudiopayload->sample_size * GST_SECOND); + } + + max_payload_len = MIN ( + /* MTU max */ + gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU + (basertpaudiopayload), 0, 0), + /* ptime max */ + maxptime_octets); + + /* min number of bytes based on a given ptime, has to be a multiple + of sample rate */ + minptime_octets = 8 * basepayload->min_ptime * basepayload->clock_rate / + (basertpaudiopayload->sample_size * GST_SECOND); + + min_payload_len = MAX (minptime_octets, fragment_size); + + if (min_payload_len > max_payload_len) { + min_payload_len = max_payload_len; + } + + GST_DEBUG_OBJECT (basertpaudiopayload, + "Calculated min_payload_len %u and max_payload_len %u", + min_payload_len, max_payload_len); + + if (basertpaudiopayload->priv->adapter && + gst_adapter_available (basertpaudiopayload->priv->adapter)) { + /* If there is always data in the adapter, we have to use it */ + gst_adapter_push (basertpaudiopayload->priv->adapter, buffer); + available = gst_adapter_available (basertpaudiopayload->priv->adapter); + use_adapter = TRUE; + } else { + /* let's set the base timestamp */ + basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer); + + /* If buffer fits on an RTP packet, let's just push it through */ + /* this will check against max_ptime and max_mtu */ + if (GST_BUFFER_SIZE (buffer) >= min_payload_len && + GST_BUFFER_SIZE (buffer) <= max_payload_len) { + ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, + GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer), + GST_BUFFER_TIMESTAMP (buffer)); + gst_buffer_unref (buffer); + + return ret; + } + + available = GST_BUFFER_SIZE (buffer); + data = (guint8 *) GST_BUFFER_DATA (buffer); + } + + while (available >= min_payload_len) { + gfloat num, datarate; + + payload_len = + MIN (max_payload_len, (available / fragment_size) * fragment_size); + + if (use_adapter) { + data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len); + } + + ret = + gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len, + basertpaudiopayload->base_ts); + + num = payload_len * 8; + datarate = (basertpaudiopayload->sample_size * basepayload->clock_rate); + + basertpaudiopayload->base_ts += + /* payload_len (bits) * nsecs/sec / datarate (bits*sec) */ + gst_gdouble_to_guint64 (num / datarate * GST_SECOND); + GST_DEBUG_OBJECT (basertpaudiopayload, "New ts is %" GST_TIME_FORMAT, + GST_TIME_ARGS (basertpaudiopayload->base_ts)); + + if (use_adapter) { + gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len); + available = gst_adapter_available (basertpaudiopayload->priv->adapter); + } else { + available -= payload_len; + data += payload_len; + } + } + + if (!use_adapter) { + if (available != 0 && basertpaudiopayload->priv->adapter) { + GstBuffer *buf; + + buf = gst_buffer_create_sub (buffer, + GST_BUFFER_SIZE (buffer) - available, available); + gst_adapter_push (basertpaudiopayload->priv->adapter, buf); + } + gst_buffer_unref (buffer); + } + + return ret; +} + +/** + * gst_base_rtp_audio_payload_push: + * @baseaudiopayload: a #GstBaseRTPPayload + * @data: data to set as payload + * @payload_len: length of payload + * @timestamp: a #GstClockTime + * + * 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. + * + * Returns: a #GstFlowReturn + * + * Since: 0.10.13 + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +GstFlowReturn +gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, + const guint8 * data, guint payload_len, GstClockTime timestamp) +{ + GstBaseRTPPayload *basepayload; + GstBuffer *outbuf; + guint8 *payload; + GstFlowReturn ret; + + basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); + + GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, + payload_len, GST_TIME_ARGS (timestamp)); + + /* create buffer to hold the payload */ + outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); + + /* copy payload */ + gst_rtp_buffer_set_payload_type (outbuf, basepayload->pt); + payload = gst_rtp_buffer_get_payload (outbuf); + memcpy (payload, data, payload_len); + + GST_BUFFER_TIMESTAMP (outbuf) = timestamp; + ret = gst_basertppayload_push (basepayload, outbuf); + + return ret; +} + +static GstStateChangeReturn +gst_base_rtp_payload_audio_change_state (GstElement * element, + GstStateChange transition) +{ + GstBaseRTPAudioPayload *basertppayload; + GstStateChangeReturn ret; + + basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element); + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_READY: + if (basertppayload->priv->adapter) { + gst_adapter_clear (basertppayload->priv->adapter); + } + break; + default: + break; + } + + return ret; +} + +static gboolean +gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event) +{ + GstBaseRTPAudioPayload *basertpaudiopayload; + gboolean res = FALSE; + + basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_EOS: + if (basertpaudiopayload->priv->adapter) { + gst_adapter_clear (basertpaudiopayload->priv->adapter); + } + break; + case GST_EVENT_FLUSH_STOP: + if (basertpaudiopayload->priv->adapter) { + gst_adapter_clear (basertpaudiopayload->priv->adapter); + } + break; + default: + break; + } + + gst_object_unref (basertpaudiopayload); + + /* return FALSE to let parent handle the remainder of the event */ + return res; +} + +/** + * gst_base_rtp_audio_payload_get_adapter: + * @basertpaudiopayload: a #GstBaseRTPAudioPayload + * + * Gets the internal adapter used by the depayloader. + * + * Returns: a #GstAdapter. + * + * Since: 0.10.13 + */ +#ifdef __SYMBIAN32__ +EXPORT_C +#endif + +GstAdapter * +gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload + * basertpaudiopayload) +{ + GstAdapter *adapter; + + if ((adapter = basertpaudiopayload->priv->adapter)) + g_object_ref (adapter); + + return adapter; +}