/* 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) /* function to convert bytes to a time */ typedef GstClockTime (*GetBytesToTimeFunc) (GstBaseRTPAudioPayload * payload, guint64 bytes); /* function to convert bytes to a RTP time */ typedef guint32 (*GetBytesToRTPTimeFunc) (GstBaseRTPAudioPayload * payload, guint64 bytes); /* function to convert time to bytes */ typedef guint64 (*GetTimeToBytesFunc) (GstBaseRTPAudioPayload * payload, GstClockTime time); struct _GstBaseRTPAudioPayloadPrivate { GetBytesToTimeFunc bytes_to_time; GetBytesToRTPTimeFunc bytes_to_rtptime; GetTimeToBytesFunc time_to_bytes; GstAdapter *adapter; guint fragment_size; GstClockTime frame_duration_ns; gboolean discont; guint64 offset; GstClockTime last_timestamp; guint32 last_rtptime; guint align; guint cached_mtu; guint cached_min_ptime; guint cached_max_ptime; guint cached_min_length; guint cached_max_length; }; #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); /* bytes to time functions */ static GstClockTime gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * payload, guint64 bytes); static GstClockTime gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * payload, guint64 bytes); /* bytes to RTP time functions */ static guint32 gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * payload, guint64 bytes); static guint32 gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * payload, guint64 bytes); /* time to bytes functions */ static guint64 gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * payload, GstClockTime time); static guint64 gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * payload, GstClockTime time); static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * payload, 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); 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 * payload, GstBaseRTPAudioPayloadClass * klass) { payload->priv = GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (payload); /* these need to be set by child object if frame based */ payload->frame_size = 0; payload->frame_duration = 0; /* these need to be set by child object if sample based */ payload->sample_size = 0; payload->priv->adapter = gst_adapter_new (); } static void gst_base_rtp_audio_payload_finalize (GObject * object) { GstBaseRTPAudioPayload *payload; payload = GST_BASE_RTP_AUDIO_PAYLOAD (object); g_object_unref (payload->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 */ void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL); g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL); g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); basertpaudiopayload->priv->bytes_to_time = gst_base_rtp_audio_payload_frame_bytes_to_time; basertpaudiopayload->priv->bytes_to_rtptime = gst_base_rtp_audio_payload_frame_bytes_to_rtptime; basertpaudiopayload->priv->time_to_bytes = gst_base_rtp_audio_payload_frame_time_to_bytes; } /** * 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 */ void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload * basertpaudiopayload) { g_return_if_fail (basertpaudiopayload != NULL); g_return_if_fail (basertpaudiopayload->priv->time_to_bytes == NULL); g_return_if_fail (basertpaudiopayload->priv->bytes_to_time == NULL); g_return_if_fail (basertpaudiopayload->priv->bytes_to_rtptime == NULL); basertpaudiopayload->priv->bytes_to_time = gst_base_rtp_audio_payload_sample_bytes_to_time; basertpaudiopayload->priv->bytes_to_rtptime = gst_base_rtp_audio_payload_sample_bytes_to_rtptime; basertpaudiopayload->priv->time_to_bytes = gst_base_rtp_audio_payload_sample_time_to_bytes; } /** * 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. * */ void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint frame_duration, gint frame_size) { GstBaseRTPAudioPayloadPrivate *priv; g_return_if_fail (basertpaudiopayload != NULL); priv = basertpaudiopayload->priv; basertpaudiopayload->frame_duration = frame_duration; priv->frame_duration_ns = frame_duration * GST_MSECOND; basertpaudiopayload->frame_size = frame_size; priv->align = frame_size; gst_adapter_clear (priv->adapter); GST_DEBUG_OBJECT (basertpaudiopayload, "frame set to %d ms and size %d", frame_duration, frame_size); } /** * 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. */ 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 */ gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload, sample_size * 8); } /** * 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 */ void gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload * basertpaudiopayload, gint sample_size) { guint fragment_size; GstBaseRTPAudioPayloadPrivate *priv; g_return_if_fail (basertpaudiopayload != NULL); priv = basertpaudiopayload->priv; basertpaudiopayload->sample_size = sample_size; /* sample_size is in bits and is converted into multiple bytes */ fragment_size = sample_size; while ((fragment_size % 8) != 0) fragment_size += fragment_size; priv->fragment_size = fragment_size / 8; priv->align = priv->fragment_size; gst_adapter_clear (priv->adapter); GST_DEBUG_OBJECT (basertpaudiopayload, "Samplebits set to sample size %d bits", sample_size); } static void gst_base_rtp_audio_payload_set_meta (GstBaseRTPAudioPayload * payload, GstBuffer * buffer, guint payload_len, GstClockTime timestamp) { GstBaseRTPPayload *basepayload; GstBaseRTPAudioPayloadPrivate *priv; basepayload = GST_BASE_RTP_PAYLOAD_CAST (payload); priv = payload->priv; /* set payload type */ gst_rtp_buffer_set_payload_type (buffer, basepayload->pt); /* set marker bit for disconts */ if (priv->discont) { GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT"); gst_rtp_buffer_set_marker (buffer, TRUE); GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); priv->discont = FALSE; } GST_BUFFER_TIMESTAMP (buffer) = timestamp; /* get the offset in RTP time */ GST_BUFFER_OFFSET (buffer) = priv->bytes_to_rtptime (payload, priv->offset); priv->offset += payload_len; /* remember the last rtptime/timestamp pair. We will use this to realign our * RTP timestamp after a buffer discont */ priv->last_rtptime = GST_BUFFER_OFFSET (buffer); priv->last_timestamp = timestamp; } /** * 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 */ 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 */ payload = gst_rtp_buffer_get_payload (outbuf); memcpy (payload, data, payload_len); /* set metadata */ gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); ret = gst_basertppayload_push (basepayload, outbuf); return ret; } /** * gst_base_rtp_audio_payload_flush: * @baseaudiopayload: a #GstBaseRTPPayload * @payload_len: length of payload * @timestamp: a #GstClockTime * * 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. * * Returns: a #GstFlowReturn * * Since: 0.10.25 */ GstFlowReturn gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload, guint payload_len, GstClockTime timestamp) { GstBaseRTPPayload *basepayload; GstBaseRTPAudioPayloadPrivate *priv; GstBuffer *outbuf; guint8 *payload; GstFlowReturn ret; GstAdapter *adapter; guint64 distance; priv = baseaudiopayload->priv; adapter = priv->adapter; basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload); if (payload_len == -1) payload_len = gst_adapter_available (adapter); /* nothing to do, just return */ if (payload_len == 0) return GST_FLOW_OK; if (timestamp == -1) { /* calculate the timestamp */ timestamp = gst_adapter_prev_timestamp (adapter, &distance); GST_LOG_OBJECT (baseaudiopayload, "last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT, GST_TIME_ARGS (timestamp), distance); if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) { /* convert the number of bytes since the last timestamp to time and add to * the last seen timestamp */ timestamp += priv->bytes_to_time (baseaudiopayload, distance); } } 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); payload = gst_rtp_buffer_get_payload (outbuf); gst_adapter_copy (adapter, payload, 0, payload_len); gst_adapter_flush (adapter, payload_len); /* set metadata */ gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); ret = gst_basertppayload_push (basepayload, outbuf); return ret; } #define ALIGN_DOWN(val,len) ((val) - ((val) % (len))) /* calculate the min and max length of a packet. This depends on the configured * mtu and min/max_ptime values. We cache those so that we don't have to redo * all the calculations */ static gboolean gst_base_rtp_audio_payload_get_lengths (GstBaseRTPPayload * basepayload, guint * min_payload_len, guint * max_payload_len, guint * align) { GstBaseRTPAudioPayload *payload; GstBaseRTPAudioPayloadPrivate *priv; guint max_mtu, mtu; guint maxptime_octets; guint minptime_octets; payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); priv = payload->priv; if (priv->align == 0) return FALSE; *align = priv->align; mtu = GST_BASE_RTP_PAYLOAD_MTU (payload); /* check cached values */ if (G_LIKELY (priv->cached_mtu == mtu && priv->cached_max_ptime == basepayload->max_ptime && priv->cached_min_ptime == basepayload->min_ptime)) { /* if nothing changed, return cached values */ *min_payload_len = priv->cached_min_length; *max_payload_len = priv->cached_max_length; return TRUE; } /* ptime max */ if (basepayload->max_ptime != -1) { maxptime_octets = priv->time_to_bytes (payload, basepayload->max_ptime); } else { maxptime_octets = G_MAXUINT; } /* MTU max */ max_mtu = gst_rtp_buffer_calc_payload_len (mtu, 0, 0); /* round down to alignment */ max_mtu = ALIGN_DOWN (max_mtu, *align); /* combine max ptime and max payload length */ *max_payload_len = MIN (max_mtu, maxptime_octets); /* min number of bytes based on a given ptime */ minptime_octets = priv->time_to_bytes (payload, basepayload->min_ptime); /* must be at least one frame size */ *min_payload_len = MAX (minptime_octets, *align); if (*min_payload_len > *max_payload_len) *min_payload_len = *max_payload_len; /* cache values */ priv->cached_mtu = mtu; priv->cached_min_ptime = basepayload->min_ptime; priv->cached_max_ptime = basepayload->max_ptime; priv->cached_min_length = *min_payload_len; priv->cached_max_length = *max_payload_len; return TRUE; } /* frame conversions functions */ static GstClockTime gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload * payload, guint64 bytes) { return (bytes / payload->frame_size) * (payload->priv->frame_duration_ns); } static guint32 gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload * payload, guint64 bytes) { guint64 time; time = (bytes / payload->frame_size) * (payload->priv->frame_duration_ns); return gst_util_uint64_scale_int (time, GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND); } static guint64 gst_base_rtp_audio_payload_frame_time_to_bytes (GstBaseRTPAudioPayload * payload, GstClockTime time) { return gst_util_uint64_scale (time, payload->frame_size, payload->priv->frame_duration_ns); } /* sample conversion functions */ static GstClockTime gst_base_rtp_audio_payload_sample_bytes_to_time (GstBaseRTPAudioPayload * payload, guint64 bytes) { guint64 rtptime; /* avoid division when we can */ if (G_LIKELY (payload->sample_size != 8)) rtptime = gst_util_uint64_scale_int (bytes, 8, payload->sample_size); else rtptime = bytes; return gst_util_uint64_scale_int (rtptime, GST_SECOND, GST_BASE_RTP_PAYLOAD (payload)->clock_rate); } static guint32 gst_base_rtp_audio_payload_sample_bytes_to_rtptime (GstBaseRTPAudioPayload * payload, guint64 bytes) { /* avoid division when we can */ if (G_LIKELY (payload->sample_size != 8)) return gst_util_uint64_scale_int (bytes, 8, payload->sample_size); else return bytes; } static guint64 gst_base_rtp_audio_payload_sample_time_to_bytes (GstBaseRTPAudioPayload * payload, guint64 time) { guint64 samples; samples = gst_util_uint64_scale_int (time, GST_BASE_RTP_PAYLOAD (payload)->clock_rate, GST_SECOND); /* avoid multiplication when we can */ if (G_LIKELY (payload->sample_size != 8)) return gst_util_uint64_scale_int (samples, payload->sample_size, 8); else return samples; } static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload, GstBuffer * buffer) { GstBaseRTPAudioPayload *payload; GstBaseRTPAudioPayloadPrivate *priv; guint payload_len; GstFlowReturn ret; guint available; guint min_payload_len; guint max_payload_len; guint align; guint size; gboolean discont; ret = GST_FLOW_OK; payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload); priv = payload->priv; discont = GST_BUFFER_IS_DISCONT (buffer); if (discont) { GstClockTime timestamp; GST_DEBUG_OBJECT (payload, "Got DISCONT"); /* flush everything out of the adapter, mark DISCONT */ ret = gst_base_rtp_audio_payload_flush (payload, -1, -1); priv->discont = TRUE; timestamp = GST_BUFFER_TIMESTAMP (buffer); /* get the distance between the timestamp gap and produce the same gap in * the RTP timestamps */ if (priv->last_timestamp != -1 && timestamp != -1) { /* we had a last timestamp, compare it to the new timestamp and update the * offset counter for RTP timestamps. The effect is that we will produce * output buffers containing the same RTP timestamp gap as the gap * between the GST timestamps. */ if (timestamp > priv->last_timestamp) { GstClockTime diff; guint64 bytes; /* we're only going to apply a positive gap, otherwise we let the marker * bit do its thing. simply convert to bytes and add the the current * offset */ diff = timestamp - priv->last_timestamp; bytes = priv->time_to_bytes (payload, diff); priv->offset += bytes; GST_DEBUG_OBJECT (payload, "elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT ", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes, priv->offset); } } } if (!gst_base_rtp_audio_payload_get_lengths (basepayload, &min_payload_len, &max_payload_len, &align)) goto config_error; GST_DEBUG_OBJECT (payload, "Calculated min_payload_len %u and max_payload_len %u", min_payload_len, max_payload_len); size = GST_BUFFER_SIZE (buffer); /* shortcut, we don't need to use the adapter when the packet can be pushed * through directly. */ available = gst_adapter_available (priv->adapter); GST_DEBUG_OBJECT (payload, "got buffer size %u, available %u", size, available); if (available == 0 && (size >= min_payload_len && size <= max_payload_len)) { /* If buffer fits on an RTP packet, let's just push it through * this will check against max_ptime and max_mtu */ GST_DEBUG_OBJECT (payload, "Fast packet push"); ret = gst_base_rtp_audio_payload_push (payload, GST_BUFFER_DATA (buffer), size, GST_BUFFER_TIMESTAMP (buffer)); gst_buffer_unref (buffer); } else { /* push the buffer in the adapter */ gst_adapter_push (priv->adapter, buffer); available += size; GST_DEBUG_OBJECT (payload, "available now %u", available); /* as long as we have full frames */ while (available >= min_payload_len) { /* get multiple of alignment */ payload_len = ALIGN_DOWN (available, align); payload_len = MIN (max_payload_len, payload_len); /* and flush out the bytes from the adapter, automatically set the * timestamp. */ ret = gst_base_rtp_audio_payload_flush (payload, payload_len, -1); available -= payload_len; GST_DEBUG_OBJECT (payload, "available after push %u", available); } } return ret; /* ERRORS */ config_error: { GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL), ("subclass did not configure us properly")); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } } static GstStateChangeReturn gst_base_rtp_payload_audio_change_state (GstElement * element, GstStateChange transition) { GstBaseRTPAudioPayload *basertppayload; GstStateChangeReturn ret; basertppayload = GST_BASE_RTP_AUDIO_PAYLOAD (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: basertppayload->priv->cached_mtu = -1; basertppayload->priv->last_rtptime = -1; basertppayload->priv->last_timestamp = -1; break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: 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 *payload; gboolean res = FALSE; payload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* flush remaining bytes in the adapter */ gst_base_rtp_audio_payload_flush (payload, -1, -1); break; case GST_EVENT_FLUSH_STOP: gst_adapter_clear (payload->priv->adapter); break; default: break; } gst_object_unref (payload); /* 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 */ GstAdapter * gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload * basertpaudiopayload) { GstAdapter *adapter; if ((adapter = basertpaudiopayload->priv->adapter)) g_object_ref (adapter); return adapter; }