* <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 named events.
+ * 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>
#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, "
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
- "endianness = (int) 1234, "
-#else
- "endianness = (int) 4321, "
-#endif
+ "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,
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_handle_event);
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->sample_rate = DEFAULT_SAMPLE_RATE;
+
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
}
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;
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++;
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;
+ }
+
+ 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);
+ }
+
+ ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
+
+ gst_caps_unref (srccaps);
+
+ return ret;
+}
+
static GstStateChangeReturn
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
projects / platform / upstream / gstreamer.git / blobdiff