/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:gstrtpbaseaudiopayload
* @title: GstRTPBaseAudioPayload
* @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 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.
*
*/
#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 <gst/audio/audio.h>
#include "gstrtpbaseaudiopayload.h"
GST_DEBUG_CATEGORY_STATIC (rtpbaseaudiopayload_debug);
#define GST_CAT_DEFAULT (rtpbaseaudiopayload_debug)
#define DEFAULT_BUFFER_LIST FALSE
enum
{
PROP_0,
PROP_BUFFER_LIST,
PROP_LAST
};
/* function to convert bytes to a time */
typedef GstClockTime (*GetBytesToTimeFunc) (GstRTPBaseAudioPayload * payload,
guint64 bytes);
/* function to convert bytes to a RTP time */
typedef guint32 (*GetBytesToRTPTimeFunc) (GstRTPBaseAudioPayload * payload,
guint64 bytes);
/* function to convert time to bytes */
typedef guint64 (*GetTimeToBytesFunc) (GstRTPBaseAudioPayload * payload,
GstClockTime time);
struct _GstRTPBaseAudioPayloadPrivate
{
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_RTP_BASE_AUDIO_PAYLOAD_GET_PRIVATE(o) \
(G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_BASE_AUDIO_PAYLOAD, \
GstRTPBaseAudioPayloadPrivate))
static void gst_rtp_base_audio_payload_finalize (GObject * object);
static void gst_rtp_base_audio_payload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_rtp_base_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* bytes to time functions */
static GstClockTime
gst_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *
payload, guint64 bytes);
static GstClockTime
gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
payload, guint64 bytes);
/* bytes to RTP time functions */
static guint32
gst_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
payload, guint64 bytes);
static guint32
gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *
payload, guint64 bytes);
/* time to bytes functions */
static guint64
gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *
payload, GstClockTime time);
static guint64
gst_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
payload, GstClockTime time);
static GstFlowReturn gst_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload
* payload, GstBuffer * buffer);
static GstStateChangeReturn gst_rtp_base_payload_audio_change_state (GstElement
* element, GstStateChange transition);
static gboolean gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload
* payload, GstEvent * event);
#define gst_rtp_base_audio_payload_parent_class parent_class
G_DEFINE_TYPE (GstRTPBaseAudioPayload, gst_rtp_base_audio_payload,
GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_base_audio_payload_class_init (GstRTPBaseAudioPayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
g_type_class_add_private (klass, sizeof (GstRTPBaseAudioPayloadPrivate));
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_base_audio_payload_finalize;
gobject_class->set_property = gst_rtp_base_audio_payload_set_property;
gobject_class->get_property = gst_rtp_base_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_rtp_base_payload_audio_change_state);
gstrtpbasepayload_class->handle_buffer =
GST_DEBUG_FUNCPTR (gst_rtp_base_audio_payload_handle_buffer);
gstrtpbasepayload_class->sink_event =
GST_DEBUG_FUNCPTR (gst_rtp_base_payload_audio_sink_event);
GST_DEBUG_CATEGORY_INIT (rtpbaseaudiopayload_debug, "rtpbaseaudiopayload", 0,
"base audio RTP payloader");
}
static void
gst_rtp_base_audio_payload_init (GstRTPBaseAudioPayload * payload)
{
payload->priv = GST_RTP_BASE_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_rtp_base_audio_payload_finalize (GObject * object)
{
GstRTPBaseAudioPayload *payload;
payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
g_object_unref (payload->priv->adapter);
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
}
static void
gst_rtp_base_audio_payload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRTPBaseAudioPayload *payload;
payload = GST_RTP_BASE_AUDIO_PAYLOAD (object);
switch (prop_id) {
case PROP_BUFFER_LIST:
#if 0
payload->priv->buffer_list = g_value_get_boolean (value);
#endif
payload->priv->buffer_list = FALSE;
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_base_audio_payload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRTPBaseAudioPayload *payload;
payload = GST_RTP_BASE_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_rtp_base_audio_payload_set_frame_based:
* @rtpbaseaudiopayload: a pointer to the element.
*
* Tells #GstRTPBaseAudioPayload that the child element is for a frame based
* audio codec
*/
void
gst_rtp_base_audio_payload_set_frame_based (GstRTPBaseAudioPayload *
rtpbaseaudiopayload)
{
g_return_if_fail (rtpbaseaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);
rtpbaseaudiopayload->priv->bytes_to_time =
gst_rtp_base_audio_payload_frame_bytes_to_time;
rtpbaseaudiopayload->priv->bytes_to_rtptime =
gst_rtp_base_audio_payload_frame_bytes_to_rtptime;
rtpbaseaudiopayload->priv->time_to_bytes =
gst_rtp_base_audio_payload_frame_time_to_bytes;
}
/**
* gst_rtp_base_audio_payload_set_sample_based:
* @rtpbaseaudiopayload: a pointer to the element.
*
* Tells #GstRTPBaseAudioPayload that the child element is for a sample based
* audio codec
*/
void
gst_rtp_base_audio_payload_set_sample_based (GstRTPBaseAudioPayload *
rtpbaseaudiopayload)
{
g_return_if_fail (rtpbaseaudiopayload != NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->time_to_bytes == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_time == NULL);
g_return_if_fail (rtpbaseaudiopayload->priv->bytes_to_rtptime == NULL);
rtpbaseaudiopayload->priv->bytes_to_time =
gst_rtp_base_audio_payload_sample_bytes_to_time;
rtpbaseaudiopayload->priv->bytes_to_rtptime =
gst_rtp_base_audio_payload_sample_bytes_to_rtptime;
rtpbaseaudiopayload->priv->time_to_bytes =
gst_rtp_base_audio_payload_sample_time_to_bytes;
}
/**
* gst_rtp_base_audio_payload_set_frame_options:
* @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.
*
* Sets the options for frame based audio codecs.
*
*/
void
gst_rtp_base_audio_payload_set_frame_options (GstRTPBaseAudioPayload
* rtpbaseaudiopayload, gint frame_duration, gint frame_size)
{
GstRTPBaseAudioPayloadPrivate *priv;
g_return_if_fail (rtpbaseaudiopayload != NULL);
priv = rtpbaseaudiopayload->priv;
rtpbaseaudiopayload->frame_duration = frame_duration;
priv->frame_duration_ns = frame_duration * GST_MSECOND;
rtpbaseaudiopayload->frame_size = frame_size;
priv->align = frame_size;
gst_adapter_clear (priv->adapter);
GST_DEBUG_OBJECT (rtpbaseaudiopayload, "frame set to %d ms and size %d",
frame_duration, frame_size);
}
/**
* gst_rtp_base_audio_payload_set_sample_options:
* @rtpbaseaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bytes.
*
* Sets the options for sample based audio codecs.
*/
void
gst_rtp_base_audio_payload_set_sample_options (GstRTPBaseAudioPayload
* rtpbaseaudiopayload, gint sample_size)
{
g_return_if_fail (rtpbaseaudiopayload != NULL);
/* sample_size is in bits internally */
gst_rtp_base_audio_payload_set_samplebits_options (rtpbaseaudiopayload,
sample_size * 8);
}
/**
* gst_rtp_base_audio_payload_set_samplebits_options:
* @rtpbaseaudiopayload: a pointer to the element.
* @sample_size: Size per sample in bits.
*
* Sets the options for sample based audio codecs.
*/
void
gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload
* rtpbaseaudiopayload, gint sample_size)
{
guint fragment_size;
GstRTPBaseAudioPayloadPrivate *priv;
g_return_if_fail (rtpbaseaudiopayload != NULL);
priv = rtpbaseaudiopayload->priv;
rtpbaseaudiopayload->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 (rtpbaseaudiopayload,
"Samplebits set to sample size %d bits", sample_size);
}
static void
gst_rtp_base_audio_payload_set_meta (GstRTPBaseAudioPayload * payload,
GstBuffer * buffer, guint payload_len, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstRTPBuffer rtp = { NULL };
basepayload = GST_RTP_BASE_PAYLOAD_CAST (payload);
priv = payload->priv;
/* set payload type */
gst_rtp_buffer_map (buffer, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_payload_type (&rtp, basepayload->pt);
/* set marker bit for disconts */
if (priv->discont) {
GST_DEBUG_OBJECT (payload, "Setting marker and DISCONT");
gst_rtp_buffer_set_marker (&rtp, TRUE);
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
gst_rtp_buffer_unmap (&rtp);
GST_BUFFER_PTS (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_rtp_base_audio_payload_push:
* @baseaudiopayload: a #GstRTPBasePayload
* @data: (array length=payload_len): 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
*/
GstFlowReturn
gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstBuffer *outbuf;
guint8 *payload;
GstFlowReturn ret;
GstRTPBuffer rtp = { NULL };
basepayload = GST_RTP_BASE_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_map (outbuf, GST_MAP_WRITE, &rtp);
payload = gst_rtp_buffer_get_payload (&rtp);
memcpy (payload, data, payload_len);
gst_rtp_buffer_unmap (&rtp);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
}
typedef struct
{
GstRTPBaseAudioPayload *pay;
GstBuffer *outbuf;
} CopyMetaData;
static gboolean
foreach_metadata (GstBuffer * inbuf, GstMeta ** meta, gpointer user_data)
{
CopyMetaData *data = user_data;
GstRTPBaseAudioPayload *pay = data->pay;
GstBuffer *outbuf = data->outbuf;
const GstMetaInfo *info = (*meta)->info;
const gchar *const *tags = gst_meta_api_type_get_tags (info->api);
if (info->transform_func && (!tags || (g_strv_length ((gchar **) tags) == 1
&& gst_meta_api_type_has_tag (info->api,
g_quark_from_string (GST_META_TAG_AUDIO_STR))))) {
GstMetaTransformCopy copy_data = { FALSE, 0, -1 };
GST_DEBUG_OBJECT (pay, "copy metadata %s", g_type_name (info->api));
/* simply copy then */
info->transform_func (outbuf, *meta, inbuf,
_gst_meta_transform_copy, ©_data);
} else {
GST_DEBUG_OBJECT (pay, "not copying metadata %s", g_type_name (info->api));
}
return TRUE;
}
static GstFlowReturn
gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
guint payload_len;
GstFlowReturn ret;
priv = baseaudiopayload->priv;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
payload_len = gst_buffer_get_size (buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
/* create just the RTP header buffer */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
if (priv->buffer_list) {
GstBufferList *list;
guint i, len;
list = gst_buffer_list_new ();
len = gst_buffer_list_length (list);
for (i = 0; i < len; i++) {
/* FIXME */
g_warning ("bufferlist not implemented");
gst_buffer_list_add (list, outbuf);
gst_buffer_list_add (list, buffer);
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list);
} else {
CopyMetaData data;
/* copy payload */
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (buffer, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, buffer);
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
/**
* gst_rtp_base_audio_payload_flush:
* @baseaudiopayload: a #GstRTPBasePayload
* @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
*/
GstFlowReturn
gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn ret;
GstAdapter *adapter;
guint64 distance;
priv = baseaudiopayload->priv;
adapter = priv->adapter;
basepayload = GST_RTP_BASE_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_pts (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_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer,
timestamp);
} else {
GstBuffer *paybuf;
CopyMetaData data;
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (paybuf, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, paybuf);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_rtp_base_payload_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_rtp_base_audio_payload_get_lengths (GstRTPBasePayload *
basepayload, guint * min_payload_len, guint * max_payload_len,
guint * align)
{
GstRTPBaseAudioPayload *payload;
GstRTPBaseAudioPayloadPrivate *priv;
guint max_mtu, mtu;
guint maxptime_octets;
guint minptime_octets;
guint ptime_mult_octets;
payload = GST_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv;
if (priv->align == 0)
return FALSE;
mtu = GST_RTP_BASE_PAYLOAD_MTU (payload);
/* check cached values */
if (G_LIKELY (priv->cached_mtu == mtu
&& priv->cached_ptime_multiple ==
basepayload->ptime_multiple
&& priv->cached_ptime == basepayload->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->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->ptime) {
guint ptime_in_bytes = priv->time_to_bytes (payload,
basepayload->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->ptime;
priv->cached_min_ptime = basepayload->min_ptime;
priv->cached_max_ptime = basepayload->max_ptime;
priv->cached_ptime_multiple = basepayload->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_rtp_base_audio_payload_frame_bytes_to_time (GstRTPBaseAudioPayload *
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_rtp_base_audio_payload_frame_bytes_to_rtptime (GstRTPBaseAudioPayload *
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_RTP_BASE_PAYLOAD (payload)->clock_rate, GST_SECOND);
}
static guint64
gst_rtp_base_audio_payload_frame_time_to_bytes (GstRTPBaseAudioPayload *
payload, GstClockTime time)
{
return gst_util_uint64_scale (time, payload->frame_size,
payload->priv->frame_duration_ns);
}
/* sample conversion functions */
static GstClockTime
gst_rtp_base_audio_payload_sample_bytes_to_time (GstRTPBaseAudioPayload *
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_RTP_BASE_PAYLOAD (payload)->clock_rate);
}
static guint32
gst_rtp_base_audio_payload_sample_bytes_to_rtptime (GstRTPBaseAudioPayload *
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_rtp_base_audio_payload_sample_time_to_bytes (GstRTPBaseAudioPayload *
payload, guint64 time)
{
guint64 samples;
samples = gst_util_uint64_scale_int (time,
GST_RTP_BASE_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_rtp_base_audio_payload_handle_buffer (GstRTPBasePayload *
basepayload, GstBuffer * buffer)
{
GstRTPBaseAudioPayload *payload;
GstRTPBaseAudioPayloadPrivate *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_RTP_BASE_AUDIO_PAYLOAD_CAST (basepayload);
priv = payload->priv;
timestamp = GST_BUFFER_PTS (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_rtp_base_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 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_rtp_base_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_get_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_rtp_base_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 */
/* TODO: Use buffer lists here */
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_rtp_base_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_rtp_base_payload_audio_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBaseAudioPayload *rtpbasepayload;
GstStateChangeReturn ret;
rtpbasepayload = GST_RTP_BASE_AUDIO_PAYLOAD (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtpbasepayload->priv->cached_mtu = -1;
rtpbasepayload->priv->last_rtptime = -1;
rtpbasepayload->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 (rtpbasepayload->priv->adapter);
break;
default:
break;
}
return ret;
}
static gboolean
gst_rtp_base_payload_audio_sink_event (GstRTPBasePayload * basep,
GstEvent * event)
{
GstRTPBaseAudioPayload *payload;
gboolean res = FALSE;
payload = GST_RTP_BASE_AUDIO_PAYLOAD (basep);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* flush remaining bytes in the adapter */
gst_rtp_base_audio_payload_flush (payload, -1, -1);
break;
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (payload->priv->adapter);
break;
default:
break;
}
/* let parent handle the remainder of the event */
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (basep, event);
return res;
}
/**
* gst_rtp_base_audio_payload_get_adapter:
* @rtpbaseaudiopayload: a #GstRTPBaseAudioPayload
*
* Gets the internal adapter used by the depayloader.
*
* Returns: (transfer full): a #GstAdapter.
*/
GstAdapter *
gst_rtp_base_audio_payload_get_adapter (GstRTPBaseAudioPayload
* rtpbaseaudiopayload)
{
GstAdapter *adapter;
if ((adapter = rtpbaseaudiopayload->priv->adapter))
g_object_ref (adapter);
return adapter;
}