#include <string.h>
#include <math.h>
+#include <glib.h>
+
#ifndef M_PI
# define M_PI 3.14159265358979323846 /* pi */
#endif
#define MAX_EVENT 16
#define MIN_VOLUME 0
#define MAX_VOLUME 36
-
+#define MIN_INTER_DIGIT_INTERVAL 50
+#define MIN_PULSE_DURATION 70
+#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
typedef struct st_dtmf_key {
static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
-static void gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration,
+static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
GstBuffer * buffer);
static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
-static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc, gint event_number,
- gint event_volume);
+static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
+static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
+ gint event_volume);
+static void gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc);
static void
gst_dtmf_src_base_init (gpointer g_class)
dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
- dtmfsrc->sample = 0;
+ dtmfsrc->event_queue = g_async_queue_new ();
+ dtmfsrc->last_event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
dtmfsrc = GST_DTMF_SRC (object);
+
+ gst_dtmf_src_stop (dtmfsrc);
+
+ if (dtmfsrc->event_queue) {
+ g_async_queue_unref (dtmfsrc->event_queue);
+ dtmfsrc->event_queue = NULL;
+ }
+
G_OBJECT_CLASS (parent_class)->finalize (object);
}
GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
event_number, event_volume);
- gst_dtmf_src_start (dtmfsrc, event_number, event_volume);
+ gst_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
}
else {
GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
- gst_dtmf_src_stop (dtmfsrc);
+ gst_dtmf_src_add_stop_event (dtmfsrc);
}
return TRUE;
}
static void
-gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc)
+gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
{
GstClock *clock;
clock = GST_ELEMENT_CLOCK (dtmfsrc);
if (clock != NULL)
- dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
+ event->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
else {
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
GST_ELEMENT_NAME (dtmfsrc));
- dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
+ event->timestamp = GST_CLOCK_TIME_NONE;
}
}
static void
-gst_dtmf_src_start (GstDTMFSrc *dtmfsrc,
- gint event_number, gint event_volume)
+gst_dtmf_src_start (GstDTMFSrc *dtmfsrc)
{
GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
- dtmfsrc->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
- dtmfsrc->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
-
- gst_dtmf_prepare_timestamps (dtmfsrc);
-
- /* Don't forget to get exclusive access to the stream */
- gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
-
if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
GST_ERROR_OBJECT (dtmfsrc,
"Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
/* Don't forget to release the stream lock */
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+
+ /* Flushing the event queue */
+ GstDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
+
+ while (event != NULL) {
+ g_free (event);
+ event = g_async_queue_try_pop (dtmfsrc->event_queue);
+ }
+
+ if (dtmfsrc->last_event) {
+ g_free (dtmfsrc->last_event);
+ dtmfsrc->last_event = NULL;
+ }
+
if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
return;
}
static void
-gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration, GstBuffer * buffer)
+gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
+ gint event_volume)
+{
+
+ GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
+ event->event_type = DTMF_EVENT_TYPE_START;
+ event->sample = 0;
+ event->event_number = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
+ event->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
+
+ g_async_queue_push (dtmfsrc->event_queue, event);
+}
+
+static void
+gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc)
+{
+
+ GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
+ event->event_type = DTMF_EVENT_TYPE_STOP;
+ event->sample = 0;
+ event->event_number = 0;
+ event->volume = 0;
+
+ g_async_queue_push (dtmfsrc->event_queue, event);
+}
+
+static void
+gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
+{
+ gint buf_size;
+
+ /* Create a buffer with data set to 0 */
+ buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
+ GST_BUFFER_SIZE (buffer) = buf_size;
+ GST_BUFFER_MALLOCDATA (buffer) = g_malloc0(buf_size);
+ GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+
+}
+
+static void
+gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration, GstBuffer * buffer)
{
gint16 *p;
gint tone_size;
/*
* We add the fundamental frequencies together.
*/
- f1 = sin(2 * M_PI * key.low_frequency * (dtmfsrc->sample / SAMPLE_RATE));
- f2 = sin(2 * M_PI * key.high_frequency * (dtmfsrc->sample / SAMPLE_RATE));
+ f1 = sin(2 * M_PI * key.low_frequency * (event->sample / SAMPLE_RATE));
+ f2 = sin(2 * M_PI * key.high_frequency * (event->sample / SAMPLE_RATE));
amplitude = (f1 + f2) / 2;
/* Store it in the data buffer */
*(p++) = (gint16) amplitude;
- (dtmfsrc->sample)++;
+ (event->sample)++;
}
}
static GstBuffer *
-gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc)
+gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
{
GstBuffer *buf = NULL;
+ guint32 duration;
GST_DEBUG_OBJECT (dtmfsrc,
/* create buffer to hold the tone */
buf = gst_buffer_new ();
- /* Generate the tone */
- gst_dtmf_src_generate_tone(dtmfsrc, DTMF_KEYS[dtmfsrc->event], dtmfsrc->interval, buf);
+ /* The first packet must be inter digit silence, then the second and third must be the
+ * minimal pulse duration divided into two packets to make it small
+ */
+ switch(event->packet_count) {
+ case 0:
+ duration = MIN_INTER_DIGIT_INTERVAL;
+ gst_dtmf_src_generate_silence (buf, duration);
+ break;
+ case 1:
+ case 2:
+ /* Generate the tone */
+ duration = MIN_PULSE_DURATION / 2;
+ gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
+ break;
+ default:
+ duration = dtmfsrc->interval;
+ gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
+ break;
+ }
+ event->packet_count++;
/* timestamp and duration of GstBuffer */
- GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
- GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
- dtmfsrc->timestamp += GST_BUFFER_DURATION (buf) /2;
+ GST_BUFFER_DURATION (buf) = duration * GST_MSECOND;
+ GST_BUFFER_TIMESTAMP (buf) = event->timestamp;
+ event->timestamp += GST_BUFFER_DURATION (buf);
/* FIXME: Should we sync to clock ourselves or leave it to sink */
gst_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
{
GstBuffer *buf = NULL;
GstFlowReturn ret;
+ GstDTMFSrcEvent *event;
- buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc);
+ g_async_queue_ref (dtmfsrc->event_queue);
- gst_buffer_ref(buf);
+ if (dtmfsrc->last_event == NULL) {
+ event = g_async_queue_pop (dtmfsrc->event_queue);
- GST_DEBUG_OBJECT (dtmfsrc,
- "pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
- ret = gst_pad_push (dtmfsrc->srcpad, buf);
- if (ret != GST_FLOW_OK) {
- GST_ERROR_OBJECT (dtmfsrc, "Failed to push buffer on src pad", GST_BUFFER_SIZE (buf));
+ if (event->event_type == DTMF_EVENT_TYPE_STOP) {
+ GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped", GST_BUFFER_SIZE (buf));
+ } else if (event->event_type == DTMF_EVENT_TYPE_START) {
+ gst_dtmf_prepare_timestamps (dtmfsrc, event);
+
+ /* Don't forget to get exclusive access to the stream */
+ gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
+
+ event->packet_count = 0;
+ dtmfsrc->last_event = event;
+ }
+ } else if (dtmfsrc->last_event->packet_count >= 3) {
+ event = g_async_queue_try_pop (dtmfsrc->event_queue);
+
+ if (event != NULL) {
+ if (event->event_type == DTMF_EVENT_TYPE_START) {
+ GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events", GST_BUFFER_SIZE (buf));
+ } else if (event->event_type == DTMF_EVENT_TYPE_STOP) {
+ gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+ g_free (dtmfsrc->last_event);
+ dtmfsrc->last_event = NULL;
+ }
+ }
}
+ g_async_queue_unref (dtmfsrc->event_queue);
+
+ if (dtmfsrc->last_event) {
+ buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
- gst_buffer_unref(buf);
- GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
+ gst_buffer_ref(buf);
+
+ GST_DEBUG_OBJECT (dtmfsrc,
+ "pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
+ ret = gst_pad_push (dtmfsrc->srcpad, buf);
+ if (ret != GST_FLOW_OK) {
+ GST_ERROR_OBJECT (dtmfsrc, "Failed to push buffer on src pad", GST_BUFFER_SIZE (buf));
+ }
+
+ gst_buffer_unref(buf);
+ GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
+ }
}
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- /* gst_dtmf_src_start (dtmfsrc, 6, 30); */
+ gst_dtmf_src_start (dtmfsrc);
break;
default:
break;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* Indicate that we don't do PRE_ROLL */
- /* gst_dtmf_src_stop (dtmfsrc); */
+ gst_dtmf_src_stop (dtmfsrc);
no_preroll = TRUE;
break;
default:
return gst_element_register (plugin, "dtmfsrc",
GST_RANK_NONE, GST_TYPE_DTMF_SRC);
}
+
projects / platform / upstream / gstreamer.git / commitdiff