* <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 GstElementClass *parent_class = NULL;
+GST_BOILERPLATE (GstDTMFSrc, gst_dtmf_src, GstBaseSrc, GST_TYPE_BASE_SRC);
-static void gst_dtmf_src_base_init (gpointer g_class);
-static void gst_dtmf_src_class_init (GstDTMFSrcClass * klass);
-static void gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class);
static void gst_dtmf_src_finalize (GObject * object);
-GType
-gst_dtmf_src_get_type (void)
-{
- static GType base_src_type = 0;
-
- if (G_UNLIKELY (base_src_type == 0)) {
- static const GTypeInfo base_src_info = {
- sizeof (GstDTMFSrcClass),
- (GBaseInitFunc) gst_dtmf_src_base_init,
- NULL,
- (GClassInitFunc) gst_dtmf_src_class_init,
- NULL,
- NULL,
- sizeof (GstDTMFSrc),
- 0,
- (GInstanceInitFunc) gst_dtmf_src_init,
- };
-
- base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
- "GstDTMFSrc", &base_src_info, 0);
- }
- return base_src_type;
-}
-
static void gst_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
+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 void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
-static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
-static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
+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 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_dtmf_src_class_init (GstDTMFSrcClass * 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);
- parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_dtmf_src_finalize);
gobject_class->set_property =
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->create =
+ GST_DEBUG_FUNCPTR (gst_dtmf_src_create);
+ gstbasesrc_class->negotiate =
+ GST_DEBUG_FUNCPTR (gst_dtmf_src_negotiate);
}
+
static void
-gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class)
+gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, GstDTMFSrcClass *g_class)
{
- dtmfsrc->srcpad =
- gst_pad_new_from_static_template (&gst_dtmf_src_template, "src");
- GST_DEBUG_OBJECT (dtmfsrc, "adding src pad");
- gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
-
- gst_pad_set_event_function (dtmfsrc->srcpad, gst_dtmf_src_handle_event);
+ /* we operate in time */
+ gst_base_src_set_format (GST_BASE_SRC (dtmfsrc), GST_FORMAT_TIME);
+ gst_base_src_set_live (GST_BASE_SRC (dtmfsrc), TRUE);
dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
dtmfsrc->event_queue = g_async_queue_new ();
dtmfsrc->last_event = NULL;
+ dtmfsrc->sample_rate = DEFAULT_SAMPLE_RATE;
+
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;
GstState state;
GstStateChangeReturn ret;
- ret = gst_element_get_state (dtmfsrc, &state, NULL, 0);
+ 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;
}
static gboolean
-gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
+gst_dtmf_src_handle_event (GstBaseSrc * src, GstEvent * event)
{
GstDTMFSrc *dtmfsrc;
gboolean result = FALSE;
- GstElement *parent = gst_pad_get_parent_element (pad);
- dtmfsrc = GST_DTMF_SRC (parent);
+
+ dtmfsrc = GST_DTMF_SRC (src);
GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CUSTOM_UPSTREAM:
- {
- result = gst_dtmf_src_handle_custom_upstream (dtmfsrc, event);
- break;
- }
- /* Ideally this element should not be flushed but let's handle the event
- * just in case it is */
- case GST_EVENT_FLUSH_START:
- gst_dtmf_src_stop (dtmfsrc);
- result = TRUE;
- break;
- case GST_EVENT_FLUSH_STOP:
- gst_segment_init (&dtmfsrc->segment, GST_FORMAT_TIME);
- 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 (&dtmfsrc->segment, update, rate, fmt,
- start, stop, position);
- }
- /* fallthrough */
- default:
- result = gst_pad_event_default (pad, event);
- break;
+ if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
+ result = gst_dtmf_src_handle_custom_upstream (dtmfsrc, event);
}
- gst_object_unref (parent);
- gst_event_unref (event);
return result;
}
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);
static void
gst_dtmf_src_set_stream_lock (GstDTMFSrc *dtmfsrc, gboolean lock)
{
+ GstPad *srcpad = GST_BASE_SRC_PAD (dtmfsrc);
GstEvent *event;
GstStructure *structure;
"lock", G_TYPE_BOOLEAN, lock, NULL);
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
- if (!gst_pad_push_event (dtmfsrc->srcpad, event)) {
+ if (!gst_pad_push_event (srcpad, event)) {
GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
}
}
gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc)
{
GstClock *clock;
+ GstClockTime base_time;
+
+ base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
- clock = gst_element_get_clock (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_start (GstDTMFSrc *dtmfsrc)
-{
- GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
-
- if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
- GST_ERROR_OBJECT (dtmfsrc,
- "Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
- else
- GST_DEBUG_OBJECT (dtmfsrc,
- "caps %" GST_PTR_FORMAT " set on src pad", caps);
-
-
- if (!gst_pad_start_task (dtmfsrc->srcpad,
- (GstTaskFunction) gst_dtmf_src_push_next_tone_packet, dtmfsrc)) {
- GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
- }
-}
-
-static void
-gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
-{
- /* 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_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
gint event_volume)
{
}
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_wait_for_buffer_ts (GstDTMFSrc *dtmfsrc, GstBuffer * buf)
-{
- GstClock *clock;
-
- clock = gst_element_get_clock (dtmfsrc);
- if (clock != NULL) {
- GstClockID clock_id;
- GstClockReturn clock_ret;
-
- clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf));
- clock_ret = gst_clock_id_wait (clock_id, NULL);
- if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) {
- gchar *clock_name = gst_element_get_name (clock);
- GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s", clock_name);
- g_free (clock_name);
- }
- gst_clock_id_unref (clock_id);
- gst_object_unref (clock);
- }
-
- else {
- gchar *dtmf_name = gst_element_get_name (dtmfsrc);
- GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
- g_free (dtmf_name);
- }
-}
static GstBuffer *
{
GstBuffer *buf = NULL;
gboolean send_silence = FALSE;
+ GstPad *srcpad = GST_BASE_SRC_PAD (dtmfsrc);
GST_DEBUG_OBJECT (dtmfsrc, "Creating buffer for tone %s",
DTMF_KEYS[event->event_number].event_name);
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++;
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
dtmfsrc->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);
-
/* Set caps on the buffer before pushing it */
- gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (srcpad));
return buf;
}
-static void
-gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
+static GstFlowReturn
+gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
+ guint length, GstBuffer ** buffer)
{
GstBuffer *buf = NULL;
- GstFlowReturn ret;
GstDTMFSrcEvent *event;
+ GstDTMFSrc * dtmfsrc;
+ GstClock *clock;
+ GstClockID *clockid;
+ GstClockReturn clockret;
+
+ dtmfsrc = GST_DTMF_SRC (basesrc);
+
+ 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);
- g_async_queue_ref (dtmfsrc->event_queue);
+ GST_DEBUG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
- if (dtmfsrc->last_event == NULL) {
- event = g_async_queue_pop (dtmfsrc->event_queue);
+ clock = gst_element_get_clock (GST_ELEMENT (basesrc));
- 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);
+#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);
- /* Don't forget to get exclusive access to the stream */
- gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
+ GST_OBJECT_LOCK (dtmfsrc);
+ if (!dtmfsrc->paused) {
+ dtmfsrc->clockid = clockid;
+ GST_OBJECT_UNLOCK (dtmfsrc);
- event->packet_count = 0;
- dtmfsrc->last_event = 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) {
- 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;
- }
- }
+ clockret = gst_clock_id_wait (clockid, NULL);
+
+ GST_OBJECT_LOCK (dtmfsrc);
+ if (dtmfsrc->paused)
+ clockret = GST_CLOCK_UNSCHEDULED;
+ } else {
+ clockret = GST_CLOCK_UNSCHEDULED;
}
- g_async_queue_unref (dtmfsrc->event_queue);
+ gst_clock_id_unref (clockid);
+ dtmfsrc->clockid = NULL;
+ GST_OBJECT_UNLOCK (dtmfsrc);
+
+ if (clockret == GST_CLOCK_UNSCHEDULED) {
+ goto paused;
+ }
+
+ 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_buffer_ref(buf);
+ 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;
+}
+
+
+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 {
+ 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;
+ }
- 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");
+ if (gst_caps_is_empty (temp)) {
+ GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
+ gst_caps_unref (temp);
+ return FALSE;
}
- gst_buffer_unref(buf);
- GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
+ /* 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);
}
+ ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
+
+ gst_caps_unref (srccaps);
+
+ return ret;
}
static GstStateChangeReturn
GstDTMFSrc *dtmfsrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
+ GstDTMFSrcEvent *event = NULL;
dtmfsrc = GST_DTMF_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
- gst_segment_init (&dtmfsrc->segment, GST_FORMAT_TIME);
- gst_pad_push_event (dtmfsrc->srcpad, gst_event_new_new_segment (FALSE,
- dtmfsrc->segment.rate, dtmfsrc->segment.format,
- dtmfsrc->segment.start, dtmfsrc->segment.stop,
- dtmfsrc->segment.time));
- /* Indicate that we don't do PRE_ROLL */
+ /* Flushing the event queue */
+ event = g_async_queue_try_pop (dtmfsrc->event_queue);
+
+ while (event != NULL) {
+ g_free (event);
+ event = g_async_queue_try_pop (dtmfsrc->event_queue);
+ }
no_preroll = TRUE;
break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- 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);
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);
+
+ while (event != NULL) {
+ g_free (event);
+ event = g_async_queue_try_pop (dtmfsrc->event_queue);
+ }
+
+ break;
default:
break;
}
projects / platform / upstream / gstreamer.git / blobdiff