static void
gst_rtp_qcelp_depay_init (GstRtpQCELPDepay * rtpqcelpdepay,
GstRtpQCELPDepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtpqcelpdepay);
}
static gboolean
gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadClass *bclass;
GstBaseRTPDepayloadPrivate *priv;
gboolean res;
GstStructure *caps_struct;
const GValue *value;
filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
priv = filter->priv;
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
GST_DEBUG_OBJECT (filter, "Set caps");
caps_struct = gst_caps_get_structure (caps, 0);
/* get other values for newsegment */
value = gst_structure_get_value (caps_struct, "npt-start");
if (value && G_VALUE_HOLDS_UINT64 (value))
priv->npt_start = g_value_get_uint64 (value);
else
priv->npt_start = 0;
GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
value = gst_structure_get_value (caps_struct, "npt-stop");
if (value && G_VALUE_HOLDS_UINT64 (value))
priv->npt_stop = g_value_get_uint64 (value);
else
priv->npt_stop = -1;
GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
value = gst_structure_get_value (caps_struct, "play-speed");
if (value && G_VALUE_HOLDS_DOUBLE (value))
priv->play_speed = g_value_get_double (value);
else
priv->play_speed = 1.0;
value = gst_structure_get_value (caps_struct, "play-scale");
if (value && G_VALUE_HOLDS_DOUBLE (value))
priv->play_scale = g_value_get_double (value);
else
priv->play_scale = 1.0;
if (bclass->set_caps)
res = bclass->set_caps (filter, caps);
else
res = TRUE;
priv->negotiated = res;
gst_object_unref (filter);
return res;
}
static void
gst_base_rtp_depayload_finalize (GObject * object)
{
GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
g_queue_free (filter->queue);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay,
GstRtpAMRDepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);
gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
static void
gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
GstRTPDTMFPayload payload, GstBuffer * buffer)
{
gint16 *p;
gint tone_size;
double i = 0;
double amplitude, f1, f2;
double volume_factor;
DTMF_KEY key = DTMF_KEYS[payload.event];
guint32 clock_rate = 8000 /* default */ ;
GstBaseRTPDepayload *depayload = GST_BASE_RTP_DEPAYLOAD (rtpdtmfdepay);
gint volume;
clock_rate = depayload->clock_rate;
/* Create a buffer for the tone */
tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
GST_BUFFER_SIZE (buffer) = tone_size;
GST_BUFFER_MALLOCDATA (buffer) = g_malloc (tone_size);
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
GST_BUFFER_DURATION (buffer) = payload.duration * GST_SECOND / clock_rate;
volume = payload.volume;
p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
volume_factor = pow (10, (-volume) / 20);
/*
* For each sample point we calculate 'x' as the
* the amplitude value.
*/
for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
/*
* We add the fundamental frequencies together.
*/
f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
clock_rate));
f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
clock_rate));
amplitude = (f1 + f2) / 2;
/* Adjust the volume */
amplitude *= volume_factor;
/* Make the [-1:1] interval into a [-32767:32767] interval */
amplitude *= 32767;
/* Store it in the data buffer */
*(p++) = (gint16) amplitude;
(rtpdtmfdepay->sample)++;
}
}
static void
gst_rtp_ilbc_depay_init (GstRTPiLBCDepay * rtpilbcdepay,
GstRTPiLBCDepayClass * klass)
{
GstBaseRTPDepayload *depayload;
depayload = GST_BASE_RTP_DEPAYLOAD (rtpilbcdepay);
/* Set default mode */
rtpilbcdepay->mode = DEFAULT_MODE;
}
static void
gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstBaseRTPDepayload *filter;
filter = GST_BASE_RTP_DEPAYLOAD (object);
switch (prop_id) {
case PROP_QUEUE_DELAY:
g_value_set_uint (value, filter->queue_delay);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_base_rtp_depayload_change_state (GstElement * element,
GstStateChange transition)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstStateChangeReturn ret;
filter = GST_BASE_RTP_DEPAYLOAD (element);
priv = filter->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
filter->need_newsegment = TRUE;
priv->npt_start = 0;
priv->npt_stop = -1;
priv->play_speed = 1.0;
priv->play_scale = 1.0;
priv->next_seqnum = -1;
priv->negotiated = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static void
flush_packets (GstRtpQCELPDepay * depay)
{
guint i, size;
GST_DEBUG_OBJECT (depay, "flushing packets");
size = depay->packets->len;
for (i = 0; i < size; i++) {
GstBuffer *outbuf;
outbuf = g_ptr_array_index (depay->packets, i);
g_ptr_array_index (depay->packets, i) = NULL;
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (depay), outbuf);
}
/* and reset interleaving state */
depay->interleaved = FALSE;
depay->bundling = 0;
}
static gboolean
gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
{
GstBaseRTPDepayload *filter;
gboolean res = TRUE;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
res = gst_pad_push_event (filter->srcpad, event);
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
filter->need_newsegment = TRUE;
filter->priv->next_seqnum = -1;
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
gdouble rate;
GstFormat fmt;
gint64 start, stop, position;
gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
&position);
gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
start, stop, position);
/* don't pass the event downstream, we generate our own segment including
* the NTP time and other things we receive in caps */
gst_event_unref (event);
break;
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
GstBaseRTPDepayloadClass *bclass;
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (gst_event_has_name (event, "GstRTPPacketLost")) {
/* we get this event from the jitterbuffer when it considers a packet as
* being lost. We send it to our packet_lost vmethod. The default
* implementation will make time progress by pushing out a NEWSEGMENT
* update event. Subclasses can override and to one of the following:
* - Adjust timestamp/duration to something more accurate before
* calling the parent (default) packet_lost method.
* - do some more advanced error concealing on the already received
* (fragmented) packets.
* - ignore the packet lost.
*/
if (bclass->packet_lost)
res = bclass->packet_lost (filter, event);
}
gst_event_unref (event);
break;
}
default:
/* pass other events forward */
res = gst_pad_push_event (filter->srcpad, event);
break;
}
return res;
}
static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime timestamp;
guint16 seqnum;
guint32 rtptime;
gboolean reset_seq, discont;
gint gap;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
priv = filter->priv;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
/* we must validate, it's possible that this element is plugged right after a
* network receiver and we don't want to operate on invalid data */
if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
goto invalid_buffer;
priv->discont = GST_BUFFER_IS_DISCONT (in);
timestamp = GST_BUFFER_TIMESTAMP (in);
/* convert to running_time and save the timestamp, this is the timestamp
* we put on outgoing buffers. */
timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
timestamp);
priv->timestamp = timestamp;
priv->duration = GST_BUFFER_DURATION (in);
seqnum = gst_rtp_buffer_get_seq (in);
rtptime = gst_rtp_buffer_get_timestamp (in);
reset_seq = TRUE;
discont = FALSE;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
GST_TIME_ARGS (timestamp));
/* Check seqnum. This is a very simple check that makes sure that the seqnums
* are striclty increasing, dropping anything that is out of the ordinary. We
* can only do this when the next_seqnum is known. */
if (G_LIKELY (priv->next_seqnum != -1)) {
gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
/* if we have no gap, all is fine */
if (G_UNLIKELY (gap != 0)) {
GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
priv->next_seqnum, gap);
if (gap < 0) {
/* seqnum > next_seqnum, we are missing some packets, this is always a
* DISCONT. */
GST_LOG_OBJECT (filter, "%d missing packets", gap);
discont = TRUE;
} else {
/* seqnum < next_seqnum, we have seen this packet before or the sender
* could be restarted. If the packet is not too old, we throw it away as
* a duplicate, otherwise we mark discont and continue. 100 misordered
* packets is a good threshold. See also RFC 4737. */
if (gap < 100)
goto dropping;
GST_LOG_OBJECT (filter,
"%d > 100, packet too old, sender likely restarted", gap);
discont = TRUE;
}
}
}
priv->next_seqnum = (seqnum + 1) & 0xffff;
if (G_UNLIKELY (discont && !priv->discont)) {
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
/* we detected a seqnum discont but the buffer was not flagged with a discont,
* set the discont flag so that the subclass can throw away old data. */
priv->discont = TRUE;
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
}
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (G_UNLIKELY (bclass->process == NULL))
goto no_process;
/* let's send it out to processing */
out_buf = bclass->process (filter, in);
if (out_buf) {
/* we pass rtptime as backward compatibility, in reality, the incomming
* buffer timestamp is always applied to the outgoing packet. */
ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
}
gst_buffer_unref (in);
return ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, (NULL),
("Not RTP format was negotiated"));
gst_buffer_unref (in);
return GST_FLOW_NOT_NEGOTIATED;
}
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (in);
return GST_FLOW_OK;
}
dropping:
{
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
gst_buffer_unref (in);
return GST_FLOW_OK;
}
no_process:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
("The subclass does not have a process method"));
gst_buffer_unref (in);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_rtp_celt_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpCELTDepay *rtpceltdepay;
gint clock_rate, nb_channels, frame_size;
GstBuffer *buf;
guint8 *data;
const gchar *params;
GstCaps *srccaps;
gboolean res;
rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
goto no_clockrate;
depayload->clock_rate = clock_rate;
if (!(params = gst_structure_get_string (structure, "encoding-params")))
nb_channels = 1;
else {
nb_channels = atoi (params);
}
if (!(params = gst_structure_get_string (structure, "frame-size")))
frame_size = 480;
else {
frame_size = atoi (params);
}
/* construct minimal header and comment packet for the decoder */
buf = gst_buffer_new_and_alloc (60);
data = GST_BUFFER_DATA (buf);
memcpy (data, "CELT ", 8);
data += 8;
memcpy (data, "1.1.12", 7);
data += 20;
GST_WRITE_UINT32_LE (data, 0x80000006); /* version */
data += 4;
GST_WRITE_UINT32_LE (data, 56); /* header_size */
data += 4;
GST_WRITE_UINT32_LE (data, clock_rate); /* rate */
data += 4;
GST_WRITE_UINT32_LE (data, nb_channels); /* channels */
data += 4;
GST_WRITE_UINT32_LE (data, frame_size); /* frame-size */
data += 4;
GST_WRITE_UINT32_LE (data, -1); /* overlap */
data += 4;
GST_WRITE_UINT32_LE (data, -1); /* bytes_per_packet */
data += 4;
GST_WRITE_UINT32_LE (data, 0); /* extra headers */
srccaps = gst_caps_new_simple ("audio/x-celt", NULL);
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
gst_buffer_set_caps (buf, GST_PAD_CAPS (depayload->srcpad));
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpceltdepay), buf);
buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_celt_comment));
memcpy (GST_BUFFER_DATA (buf), gst_rtp_celt_comment,
sizeof (gst_rtp_celt_comment));
gst_buffer_set_caps (buf, GST_PAD_CAPS (depayload->srcpad));
gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpceltdepay), buf);
return res;
/* ERRORS */
no_clockrate:
{
GST_DEBUG_OBJECT (depayload, "no clock-rate specified");
return FALSE;
}
}
