* <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. This element is only capable of generating tones.
+ * 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>
* <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 2.
+ * <entry>2</entry>
+ * <entry>The method used for sending event, this element will react if this
+ * field is absent or 2.
* </entry>
* </row>
* </tbody>
* <para>
* <programlisting>
* structure = gst_structure_new ("dtmf-event",
- * "type", G_TYPE_INT, 0,
+ * "type", G_TYPE_INT, 1,
* "number", G_TYPE_INT, 1,
* "volume", G_TYPE_INT, 25,
* "start", G_TYPE_BOOLEAN, TRUE, NULL);
#include "gstdtmfsrc.h"
-#define GST_TONE_DTMF_TYPE_EVENT 0
+#define GST_TONE_DTMF_TYPE_EVENT 1
#define DEFAULT_PACKET_INTERVAL 50 /* ms */
#define MIN_PACKET_INTERVAL 10 /* ms */
#define MAX_PACKET_INTERVAL 50 /* ms */
-#define SAMPLE_RATE 8000
+#define DEFAULT_SAMPLE_RATE 8000
#define SAMPLE_SIZE 16
#define CHANNELS 1
#define MIN_EVENT 0
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) 16, "
"depth = (int) 16, "
- "endianness = (int) 1234, "
+ "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
"signed = (bool) true, "
"rate = (int) 8000, "
"channels = (int) 1")
static gboolean gst_dtmf_src_handle_event (GstBaseSrc *src, GstEvent * event);
static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
-static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key,
- float duration, GstBuffer * buffer);
static GstFlowReturn gst_dtmf_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
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_get_times (GstBaseSrc * basesrc,
- GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
-
+static gboolean gst_dtmf_src_unlock (GstBaseSrc *src);
+static gboolean gst_dtmf_src_unlock_stop (GstBaseSrc *src);
+static gboolean gst_dtmf_src_negotiate (GstBaseSrc * basesrc);
static void
gst_dtmf_src_base_init (gpointer g_class)
GST_DEBUG_FUNCPTR (gst_dtmf_src_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
- g_param_spec_int ("interval", "Interval between tone packets",
+ g_param_spec_uint ("interval", "Interval between tone packets",
"Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
+ gstbasesrc_class->unlock =
+ GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock);
+ gstbasesrc_class->unlock_stop =
+ GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock_stop);
gstbasesrc_class->event =
GST_DEBUG_FUNCPTR (gst_dtmf_src_handle_event);
- gstbasesrc_class->get_times =
- GST_DEBUG_FUNCPTR (gst_dtmf_src_get_times);
gstbasesrc_class->create =
GST_DEBUG_FUNCPTR (gst_dtmf_src_create);
+ gstbasesrc_class->negotiate =
+ GST_DEBUG_FUNCPTR (gst_dtmf_src_negotiate);
}
dtmfsrc->event_queue = g_async_queue_new ();
dtmfsrc->last_event = NULL;
- dtmfsrc->clock_id = NULL;
+ dtmfsrc->sample_rate = DEFAULT_SAMPLE_RATE;
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
switch (prop_id) {
case PROP_INTERVAL:
- dtmfsrc->interval = g_value_get_int (value);
+ dtmfsrc->interval = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
GstClock *clock;
GstClockTime base_time;
- base_time = GST_ELEMENT_CAST (dtmfsrc)->base_time;
+ base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
+#ifdef MAEMO_BROKEN
+ dtmfsrc->timestamp = gst_clock_get_time (clock);
+#else
dtmfsrc->timestamp = gst_clock_get_time (clock) - base_time;
+#endif
gst_object_unref (clock);
} else {
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
}
static void
-gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
+gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration,
+ gint sample_rate)
{
gint buf_size;
/* Create a buffer with data set to 0 */
- buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
+ 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)
+ GstBuffer * buffer, gint sample_rate)
{
gint16 *p;
gint tone_size;
double volume_factor;
/* Create a buffer for the tone */
- tone_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
+ tone_size = ((duration/1000)*sample_rate*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);
/*
* We add the fundamental frequencies together.
*/
- f1 = sin(2 * M_PI * key.low_frequency * (event->sample / SAMPLE_RATE));
- f2 = sin(2 * M_PI * key.high_frequency * (event->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;
}
}
-static void
-gst_dtmf_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
- GstClockTime * start, GstClockTime * end)
-{
- /* for live sources, sync on the timestamp of the buffer */
- if (gst_base_src_is_live (basesrc)) {
- GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
-
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- /* get duration to calculate end time */
- GstClockTime duration = GST_BUFFER_DURATION (buffer);
-
- *start = timestamp;
- if (GST_CLOCK_TIME_IS_VALID (duration)) {
- *end = *start + duration;
- }
- }
- } else {
- *start = -1;
- *end = -1;
- }
-}
static GstBuffer *
if (send_silence) {
GST_DEBUG_OBJECT (dtmfsrc, "Generating silence");
- gst_dtmf_src_generate_silence (buf, dtmfsrc->interval);
+ gst_dtmf_src_generate_silence (buf, dtmfsrc->interval,
+ dtmfsrc->sample_rate);
} else {
GST_DEBUG_OBJECT (dtmfsrc, "Generating tone");
gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number],
- dtmfsrc->interval, buf);
+ dtmfsrc->interval, buf, dtmfsrc->sample_rate);
}
event->packet_count++;
guint length, GstBuffer ** buffer)
{
GstBuffer *buf = NULL;
- GstFlowReturn ret;
GstDTMFSrcEvent *event;
GstDTMFSrc * dtmfsrc;
+ GstClock *clock;
+ GstClockID *clockid;
+ GstClockReturn clockret;
dtmfsrc = GST_DTMF_SRC (basesrc);
- g_async_queue_ref (dtmfsrc->event_queue);
-
- start:
- if (dtmfsrc->last_event == NULL) {
- GST_DEBUG_OBJECT (dtmfsrc, "popping");
- event = g_async_queue_pop (dtmfsrc->event_queue);
-
- GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
-
- if (event->event_type == DTMF_EVENT_TYPE_STOP) {
- GST_WARNING_OBJECT (dtmfsrc,
- "Received a DTMF stop event when already stopped");
- } else if (event->event_type == DTMF_EVENT_TYPE_START) {
- gst_dtmf_prepare_timestamps (dtmfsrc);
-
- /* 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 (event->event_type == 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...");
- g_async_queue_push (dtmfsrc->event_queue, event);
- g_async_queue_unref (dtmfsrc->event_queue);
- }
- } else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
- MIN_DUTY_CYCLE) {
- 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");
- } 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;
- goto start;
- } else if (event->event_type == 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...");
- g_async_queue_push (dtmfsrc->event_queue, event);
- g_async_queue_unref (dtmfsrc->event_queue);
+ do {
+
+ if (dtmfsrc->last_event == 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 DTMF_EVENT_TYPE_STOP:
+ GST_WARNING_OBJECT (dtmfsrc,
+ "Received a DTMF stop event when already stopped");
+ break;
+ case DTMF_EVENT_TYPE_START:
+ gst_dtmf_prepare_timestamps (dtmfsrc);
+
+ /* 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;
+ event = NULL;
+ break;
+ case 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;
+ }
+ if (event)
+ g_free (event);
+ } else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
+ MIN_DUTY_CYCLE) {
+ event = g_async_queue_try_pop (dtmfsrc->event_queue);
+
+ if (event != NULL) {
+
+ switch (event->event_type) {
+ case DTMF_EVENT_TYPE_START:
+ GST_WARNING_OBJECT (dtmfsrc,
+ "Received two consecutive DTMF start events");
+ break;
+ case DTMF_EVENT_TYPE_STOP:
+ gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+
+ g_free (dtmfsrc->last_event);
+ dtmfsrc->last_event = NULL;
+ break;
+ case 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;
+ }
+ g_free (event);
}
}
+ } while (dtmfsrc->last_event == NULL);
+
+ GST_DEBUG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
+
+ clock = gst_element_get_clock (GST_ELEMENT (basesrc));
+
+#ifdef MAEMO_BROKEN
+ clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp);
+#else
+ clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
+ gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
+#endif
+ 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;
}
- g_async_queue_unref (dtmfsrc->event_queue);
- GST_DEBUG_OBJECT (dtmfsrc, "end event check");
+ buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
+
+ GST_DEBUG_OBJECT (dtmfsrc, "Created buffer of size %d", GST_BUFFER_SIZE (buf));
+ *buffer = buf;
+
+ GST_DEBUG_OBJECT (dtmfsrc, "returning a buffer");
+ return GST_FLOW_OK;
+
+ paused_locked:
+ GST_OBJECT_UNLOCK (dtmfsrc);
+
+ paused:
if (dtmfsrc->last_event) {
- buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
+ GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
+ /* Don't forget to release the stream lock */
+ gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+ g_free (dtmfsrc->last_event);
+ dtmfsrc->last_event = NULL;
+ }
+
+ return GST_FLOW_WRONG_STATE;
+
+}
+
+static gboolean
+gst_dtmf_src_unlock (GstBaseSrc *src) {
+ GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
+ GstDTMFSrcEvent *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_malloc (sizeof(GstDTMFSrcEvent));
+ event->event_type = DTMF_EVENT_TYPE_PAUSE_TASK;
+ g_async_queue_push (dtmfsrc->event_queue, event);
+
+ return TRUE;
+}
+
- GST_DEBUG_OBJECT (dtmfsrc, "Created buffer of size %d", GST_BUFFER_SIZE (buf));
- *buffer = buf;
- ret = GST_FLOW_OK;
+static gboolean
+gst_dtmf_src_unlock_stop (GstBaseSrc *src) {
+ GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
+
+ GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
+
+ GST_OBJECT_LOCK (dtmfsrc);
+ dtmfsrc->paused = FALSE;
+ GST_OBJECT_UNLOCK (dtmfsrc);
+
+ return TRUE;
+}
+
+
+static gboolean
+gst_dtmf_src_negotiate (GstBaseSrc * basesrc)
+{
+ GstCaps *srccaps, *peercaps;
+ GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
+ gboolean ret = FALSE;
+
+ srccaps = gst_caps_new_simple ("audio/x-raw-int",
+ "width", G_TYPE_INT, 16,
+ "depth", G_TYPE_INT, 16,
+ "endianness", G_TYPE_INT, G_BYTE_ORDER,
+ "signed", G_TYPE_BOOLEAN, TRUE,
+ "channels", G_TYPE_INT, 1,
+ NULL);
+
+ peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
+
+ if (peercaps == NULL) {
+ /* no peer caps, just add the other properties */
+ gst_caps_set_simple (srccaps,
+ "rate", G_TYPE_INT, dtmfsrc->sample_rate,
+ NULL);
} else {
- *buffer = NULL;
- ret = GST_FLOW_WRONG_STATE;
+ GstStructure *s;
+ gint sample_rate;
+ GstCaps *temp = NULL;
+
+ /* 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, "rate", &sample_rate))
+ {
+ dtmfsrc->sample_rate = sample_rate;
+ GST_LOG_OBJECT (dtmfsrc, "using rate from caps %d",
+ dtmfsrc->sample_rate);
+ } else {
+ GST_LOG_OBJECT (dtmfsrc, "using existing rate %d",
+ dtmfsrc->sample_rate);
+ }
+ gst_structure_set (s, "rate", G_TYPE_INT, dtmfsrc->sample_rate,
+ NULL);
}
- GST_DEBUG_OBJECT (dtmfsrc, "returning");
- return ret;
+ ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
+ gst_caps_unref (srccaps);
+
+ return ret;
}
static GstStateChangeReturn
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* Flushing the event queue */
event = g_async_queue_try_pop (dtmfsrc->event_queue);
g_free (event);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
+ no_preroll = TRUE;
break;
default:
break;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- if (dtmfsrc->last_event) {
- /* Don't forget to release the stream lock */
- gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
- g_free (dtmfsrc->last_event);
- dtmfsrc->last_event = NULL;
- }
-
+ no_preroll = TRUE;
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ GST_DEBUG_OBJECT (dtmfsrc, "Flushing event queue");
/* Flushing the event queue */
event = g_async_queue_try_pop (dtmfsrc->event_queue);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
- /* Indicate that we don't do PRE_ROLL */
- no_preroll = TRUE;
break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- event = g_malloc (sizeof(GstDTMFSrcEvent));
- event->event_type = DTMF_EVENT_TYPE_PAUSE_TASK;
- g_async_queue_push (dtmfsrc->event_queue, event);
-
- event = NULL;
default:
break;
}
projects / platform / upstream / gstreamer.git / blobdiff