--- /dev/null
+/* GStreamer
+ * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
+ *
+ * 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
+ *
+ * <refsect2>
+ * <para>
+ * Provides a base class for audio RTP payloaders for frame or sample based
+ * audio codecs (constant bitrate)
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <title>Usage</title>
+ * <para>
+ * 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.
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstbasertpaudiopayload.h"
+
+GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);
+#define GST_CAT_DEFAULT (basertpaudiopayload_debug)
+
+#define DEFAULT_BUFFER_LIST FALSE
+
+enum
+{
+ PROP_0,
+ PROP_BUFFER_LIST,
+ PROP_LAST
+};
+
+/* 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_ptime;
+ guint cached_min_length;
+ guint cached_max_length;
+ guint cached_ptime_multiple;
+ guint cached_align;
+
+ gboolean buffer_list;
+};
+
+
+#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 void gst_base_rtp_audio_payload_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_base_rtp_audio_payload_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+/* 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_base_rtp_audio_payload_finalize;
+ gobject_class->set_property = gst_base_rtp_audio_payload_set_property;
+ gobject_class->get_property = gst_base_rtp_audio_payload_get_property;
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
+ g_param_spec_boolean ("buffer-list", "Buffer List",
+ "Use Buffer Lists",
+ DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ 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 ();
+
+ payload->priv->buffer_list = DEFAULT_BUFFER_LIST;
+}
+
+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));
+}
+
+static void
+gst_base_rtp_audio_payload_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstBaseRTPAudioPayload *payload;
+
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+
+ switch (prop_id) {
+ case PROP_BUFFER_LIST:
+ payload->priv->buffer_list = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_base_rtp_audio_payload_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstBaseRTPAudioPayload *payload;
+
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD (object);
+
+ switch (prop_id) {
+ case PROP_BUFFER_LIST:
+ g_value_set_boolean (value, payload->priv->buffer_list);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/**
+ * 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;
+
+ /* Set the duration from the size */
+ GST_BUFFER_DURATION (buffer) = priv->bytes_to_time (payload, 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;
+}
+
+static GstFlowReturn
+gst_base_rtp_audio_payload_push_buffer (GstBaseRTPAudioPayload *
+ baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
+{
+ GstBaseRTPPayload *basepayload;
+ GstBaseRTPAudioPayloadPrivate *priv;
+ GstBuffer *outbuf;
+ guint8 *payload;
+ guint payload_len;
+ GstFlowReturn ret;
+
+ priv = baseaudiopayload->priv;
+ basepayload = GST_BASE_RTP_PAYLOAD (baseaudiopayload);
+
+ payload_len = GST_BUFFER_SIZE (buffer);
+
+ GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
+ payload_len, GST_TIME_ARGS (timestamp));
+
+ if (priv->buffer_list) {
+ /* create just the RTP header buffer */
+ outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
+ } else {
+ /* create buffer to hold the payload */
+ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+ }
+
+ /* set metadata */
+ gst_base_rtp_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
+ timestamp);
+
+ if (priv->buffer_list) {
+ GstBufferList *list;
+ GstBufferListIterator *it;
+
+ list = gst_buffer_list_new ();
+ it = gst_buffer_list_iterate (list);
+
+ /* add both buffers to the buffer list */
+ gst_buffer_list_iterator_add_group (it);
+ gst_buffer_list_iterator_add (it, outbuf);
+ gst_buffer_list_iterator_add (it, buffer);
+
+ gst_buffer_list_iterator_free (it);
+
+ GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
+ ret = gst_basertppayload_push_list (basepayload, list);
+ } else {
+ /* copy payload */
+ payload = gst_rtp_buffer_get_payload (outbuf);
+ memcpy (payload, GST_BUFFER_DATA (buffer), payload_len);
+ gst_buffer_unref (buffer);
+
+ GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
+ 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));
+
+ if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) {
+ GstBuffer *buffer;
+ /* we can quickly take a buffer out of the adapter without having to copy
+ * anything. */
+ buffer = gst_adapter_take_buffer (adapter, payload_len);
+
+ ret =
+ gst_base_rtp_audio_payload_push_buffer (baseaudiopayload, buffer,
+ timestamp);
+ } else {
+ /* 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);
+ 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;
+ guint ptime_mult_octets;
+
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
+ priv = payload->priv;
+
+ if (priv->align == 0)
+ return FALSE;
+
+ mtu = GST_BASE_RTP_PAYLOAD_MTU (payload);
+
+ /* check cached values */
+ if (G_LIKELY (priv->cached_mtu == mtu
+ && priv->cached_ptime_multiple ==
+ basepayload->abidata.ABI.ptime_multiple
+ && priv->cached_ptime == basepayload->abidata.ABI.ptime
+ && 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;
+ *align = priv->cached_align;
+ return TRUE;
+ }
+
+ ptime_mult_octets = priv->time_to_bytes (payload,
+ basepayload->abidata.ABI.ptime_multiple);
+ *align = ALIGN_DOWN (MAX (priv->align, ptime_mult_octets), priv->align);
+
+ /* 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;
+
+ /* If the ptime is specified in the caps, tried to adhere to it exactly */
+ if (basepayload->abidata.ABI.ptime) {
+ guint ptime_in_bytes = priv->time_to_bytes (payload,
+ basepayload->abidata.ABI.ptime);
+
+ /* clip to computed min and max lengths */
+ ptime_in_bytes = MAX (*min_payload_len, ptime_in_bytes);
+ ptime_in_bytes = MIN (*max_payload_len, ptime_in_bytes);
+
+ *min_payload_len = *max_payload_len = ptime_in_bytes;
+ }
+
+ /* cache values */
+ priv->cached_mtu = mtu;
+ priv->cached_ptime = basepayload->abidata.ABI.ptime;
+ priv->cached_min_ptime = basepayload->min_ptime;
+ priv->cached_max_ptime = basepayload->max_ptime;
+ priv->cached_ptime_multiple = basepayload->abidata.ABI.ptime_multiple;
+ priv->cached_min_length = *min_payload_len;
+ priv->cached_max_length = *max_payload_len;
+ priv->cached_align = *align;
+
+ return TRUE;
+}
+
+/* frame conversions functions */
+static GstClockTime
+gst_base_rtp_audio_payload_frame_bytes_to_time (GstBaseRTPAudioPayload *
+ payload, guint64 bytes)
+{
+ guint64 framecount;
+
+ framecount = bytes / payload->frame_size;
+ if (G_UNLIKELY (bytes % payload->frame_size))
+ framecount++;
+
+ return framecount * payload->priv->frame_duration_ns;
+}
+
+static guint32
+gst_base_rtp_audio_payload_frame_bytes_to_rtptime (GstBaseRTPAudioPayload *
+ payload, guint64 bytes)
+{
+ guint64 framecount;
+ guint64 time;
+
+ framecount = bytes / payload->frame_size;
+ if (G_UNLIKELY (bytes % payload->frame_size))
+ framecount++;
+
+ time = framecount * 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;
+ GstClockTime timestamp;
+
+ ret = GST_FLOW_OK;
+
+ payload = GST_BASE_RTP_AUDIO_PAYLOAD_CAST (basepayload);
+ priv = payload->priv;
+
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ discont = GST_BUFFER_IS_DISCONT (buffer);
+ if (discont) {
+
+ 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;
+
+ /* 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) &&
+ (size % align == 0)) {
+ /* 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_buffer (payload, buffer, timestamp);
+ } 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 = MIN (max_payload_len, available);
+ payload_len = ALIGN_DOWN (payload_len, align);
+
+ /* 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;
+}
projects / mafwsubrenderer / blobdiff