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gstreamer/gst/dtmf/gstrtpdtmfsrc.c
Olivier Crête bc44628831 dtmf: port to 0.11
2012-01-25 11:55:13 +01:00

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/* GStreamer RTP DTMF source
*
* gstrtpdtmfsrc.c:
*
* Copyright (C) <2007> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* 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:element-rtpdtmfsrc
* @see_also: dtmfsrc, rtpdtmfdepay, rtpdtmfmux
*
* The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event packets on request
* from application. The application communicates the beginning and end of a
* DTMF event using custom upstream gstreamer events. To report a DTMF event, an
* application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
* structure of name "dtmf-event" with fields set according to the following
* table:
*
* <informaltable>
* <tgroup cols='4'>
* <colspec colname='Name' />
* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
* <thead>
* <row>
* <entry>Name</entry>
* <entry>GType</entry>
* <entry>Possible values</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
* <tbody>
* <row>
* <entry>type</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-1</entry>
* <entry>The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output.
* </entry>
* </row>
* <row>
* <entry>number</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-15</entry>
* <entry>The event number.</entry>
* </row>
* <row>
* <entry>volume</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-36</entry>
* <entry>This field describes the power level of the tone, expressed in dBm0
* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
* </entry>
* </row>
* <row>
* <entry>start</entry>
* <entry>G_TYPE_BOOLEAN</entry>
* <entry>True or False</entry>
* <entry>Whether the event is starting or ending.</entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>1</entry>
* <entry>The method used for sending event, this element will react if this
* field is absent or 1.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
*
* For example, the following code informs the pipeline (and in turn, the
* RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
* event '1' of volume -25 dBm0:
*
* <programlisting>
* structure = gst_structure_new ("dtmf-event",
* "type", G_TYPE_INT, 1,
* "number", G_TYPE_INT, 1,
* "volume", G_TYPE_INT, 25,
* "start", G_TYPE_BOOLEAN, TRUE, NULL);
*
* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
* gst_element_send_event (pipeline, event);
* </programlisting>
*
* When a DTMF tone actually starts or stop, a "dtmf-event-processed"
* element #GstMessage with the same fields as the "dtmf-event"
* #GstEvent that was used to request the event. Also, if any event
* has not been processed when the element goes from the PAUSED to the
* READY state, then a "dtmf-event-dropped" message is posted on the
* #GstBus in the order that they were received.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <glib.h>
#include "gstrtpdtmfsrc.h"
#define GST_RTP_DTMF_TYPE_EVENT 1
#define DEFAULT_PTIME 40 /* ms */
#define DEFAULT_SSRC -1
#define DEFAULT_PT 96
#define DEFAULT_TIMESTAMP_OFFSET -1
#define DEFAULT_SEQNUM_OFFSET -1
#define DEFAULT_CLOCK_RATE 8000
#define DEFAULT_PACKET_REDUNDANCY 1
#define MIN_PACKET_REDUNDANCY 1
#define MAX_PACKET_REDUNDANCY 5
GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug);
#define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug
/* signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_SSRC,
PROP_TIMESTAMP_OFFSET,
PROP_SEQNUM_OFFSET,
PROP_PT,
PROP_CLOCK_RATE,
PROP_TIMESTAMP,
PROP_SEQNUM,
PROP_REDUNDANCY
};
static GstStaticPadTemplate gst_rtp_dtmf_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) [ 0, MAX ], "
"ssrc = (int) [ 0, MAX ], "
"encoding-name = (string) \"TELEPHONE-EVENT\"")
/* "events = (string) \"0-15\" */
);
G_DEFINE_TYPE (GstRTPDTMFSrc, gst_rtp_dtmf_src, GST_TYPE_BASE_SRC);
static void gst_rtp_dtmf_src_finalize (GObject * object);
static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_dtmf_src_handle_event (GstBaseSrc * basesrc,
GstEvent * event);
static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc * dtmfsrc,
gint event_number, gint event_volume);
static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc * dtmfsrc);
static gboolean gst_rtp_dtmf_src_unlock (GstBaseSrc * src);
static gboolean gst_rtp_dtmf_src_unlock_stop (GstBaseSrc * src);
static GstFlowReturn gst_rtp_dtmf_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
static gboolean gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc);
static void
gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
{
GObjectClass *gobject_class;
GstBaseSrcClass *gstbasesrc_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
"rtpdtmfsrc", 0, "rtpdtmfsrc element");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
gst_element_class_set_metadata (gstelement_class,
"RTP DTMF packet generator", "Source/Network",
"Generates RTP DTMF packets", "Zeeshan Ali <zeeshan.ali@nokia.com>");
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
g_param_spec_uint ("timestamp", "Timestamp",
"The RTP timestamp of the last processed packet",
0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
g_param_spec_uint ("seqnum", "Sequence number",
"The RTP sequence number of the last processed packet",
0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
"Timestamp Offset",
"Offset to add to all outgoing timestamps (-1 = random)", -1,
G_MAXINT, DEFAULT_TIMESTAMP_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
g_param_spec_int ("seqnum-offset", "Sequence number Offset",
"Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
g_param_spec_uint ("clock-rate", "clockrate",
"The clock-rate at which to generate the dtmf packets",
0, G_MAXUINT, DEFAULT_CLOCK_RATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
g_param_spec_uint ("ssrc", "SSRC",
"The SSRC of the packets (-1 == random)",
0, G_MAXUINT, DEFAULT_SSRC,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
g_param_spec_uint ("pt", "payload type",
"The payload type of the packets",
0, 0x80, DEFAULT_PT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
g_param_spec_uint ("packet-redundancy", "Packet Redundancy",
"Number of packets to send to indicate start and stop dtmf events",
MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
DEFAULT_PACKET_REDUNDANCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock);
gstbasesrc_class->unlock_stop =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock_stop);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_handle_event);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_create);
gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_negotiate);
}
static void
gst_rtp_dtmf_src_event_free (GstRTPDTMFSrcEvent * event)
{
if (event) {
if (event->payload)
g_slice_free (GstRTPDTMFPayload, event->payload);
g_slice_free (GstRTPDTMFSrcEvent, event);
}
}
static void
gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object)
{
gst_base_src_set_format (GST_BASE_SRC (object), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (object), TRUE);
object->ssrc = DEFAULT_SSRC;
object->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
object->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
object->pt = DEFAULT_PT;
object->clock_rate = DEFAULT_CLOCK_RATE;
object->ptime = DEFAULT_PTIME;
object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
object->event_queue =
g_async_queue_new_full ((GDestroyNotify) gst_rtp_dtmf_src_event_free);
object->payload = NULL;
GST_DEBUG_OBJECT (object, "init done");
}
static void
gst_rtp_dtmf_src_finalize (GObject * object)
{
GstRTPDTMFSrc *dtmfsrc;
dtmfsrc = GST_RTP_DTMF_SRC (object);
if (dtmfsrc->event_queue) {
g_async_queue_unref (dtmfsrc->event_queue);
dtmfsrc->event_queue = NULL;
}
G_OBJECT_CLASS (gst_rtp_dtmf_src_parent_class)->finalize (object);
}
static gboolean
gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc * dtmfsrc,
const GstStructure * event_structure)
{
gint event_type;
gboolean start;
gint method;
GstClockTime last_stop;
gint event_number;
gint event_volume;
gboolean correct_order;
if (!gst_structure_get_int (event_structure, "type", &event_type) ||
!gst_structure_get_boolean (event_structure, "start", &start) ||
event_type != GST_RTP_DTMF_TYPE_EVENT)
goto failure;
if (gst_structure_get_int (event_structure, "method", &method)) {
if (method != 1) {
goto failure;
}
}
if (start)
if (!gst_structure_get_int (event_structure, "number", &event_number) ||
!gst_structure_get_int (event_structure, "volume", &event_volume))
goto failure;
GST_OBJECT_LOCK (dtmfsrc);
if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop))
dtmfsrc->last_stop = last_stop;
else
dtmfsrc->last_stop = GST_CLOCK_TIME_NONE;
correct_order = (start != dtmfsrc->last_event_was_start);
dtmfsrc->last_event_was_start = start;
GST_OBJECT_UNLOCK (dtmfsrc);
if (!correct_order)
goto failure;
if (start) {
if (!gst_structure_get_int (event_structure, "number", &event_number) ||
!gst_structure_get_int (event_structure, "volume", &event_volume))
goto failure;
GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
event_number, event_volume);
gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
}
else {
GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
}
return TRUE;
failure:
return FALSE;
}
static gboolean
gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc * dtmfsrc,
GstEvent * event)
{
gboolean result = FALSE;
gchar *struct_str;
const GstStructure *structure;
GstState state;
GstStateChangeReturn ret;
ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
goto ret;
}
GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
structure = gst_event_get_structure (event);
struct_str = gst_structure_to_string (structure);
GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
g_free (struct_str);
if (structure && gst_structure_has_name (structure, "dtmf-event"))
result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
ret:
return result;
}
static gboolean
gst_rtp_dtmf_src_handle_event (GstBaseSrc * basesrc, GstEvent * event)
{
GstRTPDTMFSrc *dtmfsrc;
gboolean result = FALSE;
dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
}
return result;
}
static void
gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPDTMFSrc *dtmfsrc;
dtmfsrc = GST_RTP_DTMF_SRC (object);
switch (prop_id) {
case PROP_TIMESTAMP_OFFSET:
dtmfsrc->ts_offset = g_value_get_int (value);
break;
case PROP_SEQNUM_OFFSET:
dtmfsrc->seqnum_offset = g_value_get_int (value);
break;
case PROP_CLOCK_RATE:
dtmfsrc->clock_rate = g_value_get_uint (value);
dtmfsrc->dirty = TRUE;
break;
case PROP_SSRC:
dtmfsrc->ssrc = g_value_get_uint (value);
break;
case PROP_PT:
dtmfsrc->pt = g_value_get_uint (value);
dtmfsrc->dirty = TRUE;
break;
case PROP_REDUNDANCY:
dtmfsrc->packet_redundancy = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTPDTMFSrc *dtmfsrc;
dtmfsrc = GST_RTP_DTMF_SRC (object);
switch (prop_id) {
case PROP_TIMESTAMP_OFFSET:
g_value_set_int (value, dtmfsrc->ts_offset);
break;
case PROP_SEQNUM_OFFSET:
g_value_set_int (value, dtmfsrc->seqnum_offset);
break;
case PROP_CLOCK_RATE:
g_value_set_uint (value, dtmfsrc->clock_rate);
break;
case PROP_SSRC:
g_value_set_uint (value, dtmfsrc->ssrc);
break;
case PROP_PT:
g_value_set_uint (value, dtmfsrc->pt);
break;
case PROP_TIMESTAMP:
g_value_set_uint (value, dtmfsrc->rtp_timestamp);
break;
case PROP_SEQNUM:
g_value_set_uint (value, dtmfsrc->seqnum);
break;
case PROP_REDUNDANCY:
g_value_set_uint (value, dtmfsrc->packet_redundancy);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc * dtmfsrc)
{
GstClockTime last_stop;
GST_OBJECT_LOCK (dtmfsrc);
last_stop = dtmfsrc->last_stop;
GST_OBJECT_UNLOCK (dtmfsrc);
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
dtmfsrc->start_timestamp = last_stop;
} else {
GstClock *clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock == NULL)
return FALSE;
dtmfsrc->start_timestamp = gst_clock_get_time (clock)
- gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
gst_object_unref (clock);
}
/* If the last stop was in the past, then lets add the buffers together */
if (dtmfsrc->start_timestamp < dtmfsrc->timestamp)
dtmfsrc->start_timestamp = dtmfsrc->timestamp;
dtmfsrc->timestamp = dtmfsrc->start_timestamp;
dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
gst_util_uint64_scale_int (gst_segment_to_running_time (&GST_BASE_SRC
(dtmfsrc)->segment, GST_FORMAT_TIME, dtmfsrc->timestamp),
dtmfsrc->clock_rate, GST_SECOND);
return TRUE;
}
static void
gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc * dtmfsrc, gint event_number,
gint event_volume)
{
GstRTPDTMFSrcEvent *event = g_slice_new0 (GstRTPDTMFSrcEvent);
event->event_type = RTP_DTMF_EVENT_TYPE_START;
event->payload = g_slice_new0 (GstRTPDTMFPayload);
event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
event->payload->duration = dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
g_async_queue_push (dtmfsrc->event_queue, event);
}
static void
gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc * dtmfsrc)
{
GstRTPDTMFSrcEvent *event = g_slice_new0 (GstRTPDTMFSrcEvent);
event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
g_async_queue_push (dtmfsrc->event_queue, event);
}
static void
gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc * dtmfsrc,
GstRTPBuffer * rtpbuf)
{
gst_rtp_buffer_set_ssrc (rtpbuf, dtmfsrc->current_ssrc);
gst_rtp_buffer_set_payload_type (rtpbuf, dtmfsrc->pt);
/* Only the very first packet gets a marker */
if (dtmfsrc->first_packet) {
gst_rtp_buffer_set_marker (rtpbuf, TRUE);
} else if (dtmfsrc->last_packet) {
dtmfsrc->payload->e = 1;
}
dtmfsrc->seqnum++;
gst_rtp_buffer_set_seq (rtpbuf, dtmfsrc->seqnum);
/* timestamp of RTP header */
gst_rtp_buffer_set_timestamp (rtpbuf, dtmfsrc->rtp_timestamp);
}
static GstBuffer *
gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc * dtmfsrc)
{
GstBuffer *buf;
GstRTPDTMFPayload *payload;
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtpbuffer);
gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, &rtpbuffer);
/* timestamp and duration of GstBuffer */
/* Redundant buffer have no duration ... */
if (dtmfsrc->redundancy_count > 1)
GST_BUFFER_DURATION (buf) = 0;
else
GST_BUFFER_DURATION (buf) = dtmfsrc->ptime * GST_MSECOND;
GST_BUFFER_PTS (buf) = dtmfsrc->timestamp;
payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (&rtpbuffer);
/* copy payload and convert to network-byte order */
g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
payload->duration = g_htons (payload->duration);
if (dtmfsrc->redundancy_count <= 1 && dtmfsrc->last_packet) {
GstClockTime inter_digit_interval = MIN_INTER_DIGIT_INTERVAL;
if (inter_digit_interval % dtmfsrc->ptime != 0)
inter_digit_interval += dtmfsrc->ptime -
(MIN_INTER_DIGIT_INTERVAL % dtmfsrc->ptime);
GST_BUFFER_DURATION (buf) += inter_digit_interval * GST_MSECOND;
}
GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration "
" gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT "(rtp ts:%u dur:%u)",
dtmfsrc->payload->event, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), dtmfsrc->rtp_timestamp,
dtmfsrc->payload->duration);
/* duration of DTMF payloadfor the NEXT packet */
/* not updated for redundant packets */
if (dtmfsrc->redundancy_count <= 1)
dtmfsrc->payload->duration += dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
if (GST_CLOCK_TIME_IS_VALID (dtmfsrc->timestamp))
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
gst_rtp_buffer_unmap (&rtpbuffer);
return buf;
}
static void
gst_dtmf_src_post_message (GstRTPDTMFSrc * dtmfsrc, const gchar * message_name,
GstRTPDTMFSrcEvent * event)
{
GstStructure *s = NULL;
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_START:
s = gst_structure_new (message_name,
"type", G_TYPE_INT, 1,
"method", G_TYPE_INT, 1,
"start", G_TYPE_BOOLEAN, TRUE,
"number", G_TYPE_INT, event->payload->event,
"volume", G_TYPE_INT, event->payload->volume, NULL);
break;
case RTP_DTMF_EVENT_TYPE_STOP:
s = gst_structure_new (message_name,
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
"start", G_TYPE_BOOLEAN, FALSE, NULL);
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
return;
}
if (s)
gst_element_post_message (GST_ELEMENT (dtmfsrc),
gst_message_new_element (GST_OBJECT (dtmfsrc), s));
}
static GstFlowReturn
gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstRTPDTMFSrcEvent *event;
GstRTPDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
do {
if (dtmfsrc->payload == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_START:
dtmfsrc->first_packet = TRUE;
dtmfsrc->last_packet = FALSE;
/* Set the redundancy on the first packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
if (!gst_rtp_dtmf_prepare_timestamps (dtmfsrc))
goto no_clock;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
dtmfsrc->payload = event->payload;
event->payload = NULL;
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed
*/
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
} else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
(dtmfsrc->timestamp - dtmfsrc->start_timestamp) / GST_MSECOND >=
MIN_PULSE_DURATION) {
GST_DEBUG_OBJECT (dtmfsrc, "try popping");
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case RTP_DTMF_EVENT_TYPE_STOP:
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundancy on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
gst_rtp_dtmf_src_event_free (event);
}
}
} while (dtmfsrc->payload == NULL);
GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
if (!clock)
goto no_clock;
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
send_last:
if (dtmfsrc->dirty)
if (!gst_rtp_dtmf_src_negotiate (basesrc))
return GST_FLOW_NOT_NEGOTIATED;
/* create buffer to hold the payload */
*buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
if (dtmfsrc->redundancy_count)
dtmfsrc->redundancy_count--;
/* Only the very first one has a marker */
dtmfsrc->first_packet = FALSE;
/* This is the end of the event */
if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
g_slice_free (GstRTPDTMFPayload, dtmfsrc->payload);
dtmfsrc->payload = NULL;
dtmfsrc->last_packet = FALSE;
}
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->payload) {
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
/* Set the redundanc on the last packet */
dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
goto send_last;
} else {
return GST_FLOW_WRONG_STATE;
}
no_clock:
GST_ELEMENT_ERROR (dtmfsrc, STREAM, MUX, ("No available clock"),
("No available clock"));
gst_pad_pause_task (GST_BASE_SRC_PAD (dtmfsrc));
return GST_FLOW_ERROR;
}
static gboolean
gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
{
GstCaps *srccaps, *peercaps;
GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
gboolean ret;
/* fill in the defaults, there properties cannot be negotiated. */
srccaps = gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, "audio",
"encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT", NULL);
/* the peer caps can override some of the defaults */
peercaps = gst_pad_peer_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
if (peercaps == NULL) {
/* no peer caps, just add the other properties */
gst_caps_set_simple (srccaps,
"payload", G_TYPE_INT, dtmfsrc->pt,
"ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
"clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
"clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
"seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
GST_DEBUG_OBJECT (dtmfsrc, "no peer caps: %" GST_PTR_FORMAT, srccaps);
} else {
GstCaps *temp;
GstStructure *s;
const GValue *value;
gint pt;
gint clock_rate;
/* peer provides caps we can use to fixate, intersect. This always returns a
* writable caps. */
temp = gst_caps_intersect (srccaps, peercaps);
gst_caps_unref (srccaps);
gst_caps_unref (peercaps);
if (!temp) {
GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
return FALSE;
}
if (gst_caps_is_empty (temp)) {
GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
gst_caps_unref (temp);
return FALSE;
}
/* now fixate, start by taking the first caps */
gst_caps_truncate (temp);
srccaps = temp;
/* get first structure */
s = gst_caps_get_structure (srccaps, 0);
if (gst_structure_get_int (s, "payload", &pt)) {
/* use peer pt */
dtmfsrc->pt = pt;
GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
} else {
if (gst_structure_has_field (s, "payload")) {
/* can only fixate if there is a field */
gst_structure_fixate_field_nearest_int (s, "payload", dtmfsrc->pt);
gst_structure_get_int (s, "payload", &pt);
GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
} else {
/* no pt field, use the internal pt */
pt = dtmfsrc->pt;
gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
GST_LOG_OBJECT (dtmfsrc, "using internal pt %d", pt);
}
}
if (gst_structure_get_int (s, "clock-rate", &clock_rate)) {
dtmfsrc->clock_rate = clock_rate;
GST_LOG_OBJECT (dtmfsrc, "using clock-rate from caps %d",
dtmfsrc->clock_rate);
} else {
GST_LOG_OBJECT (dtmfsrc, "using existing clock-rate %d",
dtmfsrc->clock_rate);
}
gst_structure_set (s, "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate, NULL);
if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "ssrc");
dtmfsrc->current_ssrc = g_value_get_uint (value);
GST_LOG_OBJECT (dtmfsrc, "using peer ssrc %08x", dtmfsrc->current_ssrc);
} else {
/* FIXME, fixate_nearest_uint would be even better */
gst_structure_set (s, "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
GST_LOG_OBJECT (dtmfsrc, "using internal ssrc %08x",
dtmfsrc->current_ssrc);
}
if (gst_structure_has_field_typed (s, "clock-base", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "clock-base");
dtmfsrc->ts_base = g_value_get_uint (value);
GST_LOG_OBJECT (dtmfsrc, "using peer clock-base %u", dtmfsrc->ts_base);
} else {
/* FIXME, fixate_nearest_uint would be even better */
gst_structure_set (s, "clock-base", G_TYPE_UINT, dtmfsrc->ts_base, NULL);
GST_LOG_OBJECT (dtmfsrc, "using internal clock-base %u",
dtmfsrc->ts_base);
}
if (gst_structure_has_field_typed (s, "seqnum-base", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "seqnum-base");
dtmfsrc->seqnum_base = g_value_get_uint (value);
GST_LOG_OBJECT (dtmfsrc, "using peer seqnum-base %u",
dtmfsrc->seqnum_base);
} else {
/* FIXME, fixate_nearest_uint would be even better */
gst_structure_set (s, "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base,
NULL);
GST_LOG_OBJECT (dtmfsrc, "using internal seqnum-base %u",
dtmfsrc->seqnum_base);
}
if (gst_structure_has_field_typed (s, "ptime", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "ptime");
dtmfsrc->ptime = g_value_get_uint (value);
GST_LOG_OBJECT (dtmfsrc, "using peer ptime %u", dtmfsrc->ptime);
} else if (gst_structure_has_field_typed (s, "maxptime", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "maxptime");
dtmfsrc->ptime = g_value_get_uint (value);
GST_LOG_OBJECT (dtmfsrc, "using peer maxptime as ptime %u",
dtmfsrc->ptime);
} else {
/* FIXME, fixate_nearest_uint would be even better */
gst_structure_set (s, "ptime", G_TYPE_UINT, dtmfsrc->ptime, NULL);
GST_LOG_OBJECT (dtmfsrc, "using internal ptime %u", dtmfsrc->ptime);
}
GST_DEBUG_OBJECT (dtmfsrc, "with peer caps: %" GST_PTR_FORMAT, srccaps);
}
ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
gst_caps_unref (srccaps);
dtmfsrc->dirty = FALSE;
return ret;
}
static void
gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc * dtmfsrc)
{
if (dtmfsrc->ssrc == -1)
dtmfsrc->current_ssrc = g_random_int ();
else
dtmfsrc->current_ssrc = dtmfsrc->ssrc;
if (dtmfsrc->seqnum_offset == -1)
dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
else
dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
dtmfsrc->seqnum = dtmfsrc->seqnum_base;
if (dtmfsrc->ts_offset == -1)
dtmfsrc->ts_base = g_random_int ();
else
dtmfsrc->ts_base = dtmfsrc->ts_offset;
dtmfsrc->timestamp = 0;
}
static GstStateChangeReturn
gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
GstRTPDTMFSrc *dtmfsrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
GstRTPDTMFSrcEvent *event = NULL;
dtmfsrc = GST_RTP_DTMF_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
/* Flushing the event queue */
while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL) {
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
gst_rtp_dtmf_src_event_free (event);
}
dtmfsrc->last_event_was_start = FALSE;
no_preroll = TRUE;
break;
default:
break;
}
if ((result =
GST_ELEMENT_CLASS (gst_rtp_dtmf_src_parent_class)->change_state
(element, transition)) == GST_STATE_CHANGE_FAILURE)
goto failure;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* Flushing the event queue */
while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL) {
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
gst_rtp_dtmf_src_event_free (event);
}
dtmfsrc->last_event_was_start = FALSE;
/* Indicate that we don't do PRE_ROLL */
break;
default:
break;
}
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
result = GST_STATE_CHANGE_NO_PREROLL;
return result;
/* ERRORS */
failure:
{
GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
return result;
}
}
static gboolean
gst_rtp_dtmf_src_unlock (GstBaseSrc * src)
{
GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
GstRTPDTMFSrcEvent *event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = TRUE;
if (dtmfsrc->clockid) {
gst_clock_id_unschedule (dtmfsrc->clockid);
}
GST_OBJECT_UNLOCK (dtmfsrc);
GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
event = g_slice_new0 (GstRTPDTMFSrcEvent);
event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
g_async_queue_push (dtmfsrc->event_queue, event);
return TRUE;
}
static gboolean
gst_rtp_dtmf_src_unlock_stop (GstBaseSrc * src)
{
GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = FALSE;
GST_OBJECT_UNLOCK (dtmfsrc);
return TRUE;
}
gboolean
gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpdtmfsrc",
GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);
}
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