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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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[MOVED FROM GST-P-FARSIGHT] Added a queue based system for the dtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each tone, including inter-digit silence.
20070820203826-4f0f6-750a22b612a5e495e767666934465c34fe32074b.gz
This commit is contained in:
Youness Alaoui
Edward Hervey
parent
4aff57476c
commit
e7d5bdc4e8
2 changed files with 177 additions and 45 deletions
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@ -130,6 +130,8 @@
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#include <string.h>
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#include <math.h>
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#include <glib.h>
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#ifndef M_PI
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# define M_PI 3.14159265358979323846 /* pi */
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#endif
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@ -148,7 +150,9 @@
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#define MAX_EVENT 16
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#define MIN_VOLUME 0
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#define MAX_VOLUME 36
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#define MIN_INTER_DIGIT_INTERVAL 50
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#define MIN_PULSE_DURATION 70
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#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
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typedef struct st_dtmf_key {
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@ -265,12 +269,14 @@ static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
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static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration,
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static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
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GstBuffer * buffer);
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static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume);
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static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume);
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static void gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc);
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static void
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gst_dtmf_src_base_init (gpointer g_class)
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@ -324,7 +330,8 @@ gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class)
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dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
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dtmfsrc->sample = 0;
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dtmfsrc->event_queue = g_async_queue_new ();
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dtmfsrc->last_event = NULL;
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GST_DEBUG_OBJECT (dtmfsrc, "init done");
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}
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@ -336,6 +343,14 @@ gst_dtmf_src_finalize (GObject * object)
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dtmfsrc = GST_DTMF_SRC (object);
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gst_dtmf_src_stop (dtmfsrc);
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if (dtmfsrc->event_queue) {
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g_async_queue_unref (dtmfsrc->event_queue);
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dtmfsrc->event_queue = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -368,12 +383,12 @@ gst_dtmf_src_handle_dtmf_event (GstDTMFSrc *dtmfsrc,
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GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
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event_number, event_volume);
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gst_dtmf_src_start (dtmfsrc, event_number, event_volume);
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gst_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
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}
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else {
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GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
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gst_dtmf_src_stop (dtmfsrc);
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gst_dtmf_src_add_stop_event (dtmfsrc);
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}
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return TRUE;
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@ -498,35 +513,26 @@ gst_dtmf_src_set_stream_lock (GstDTMFSrc *dtmfsrc, gboolean lock)
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}
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static void
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gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc)
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gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
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{
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GstClock *clock;
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clock = GST_ELEMENT_CLOCK (dtmfsrc);
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if (clock != NULL)
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dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
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event->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
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else {
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GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
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GST_ELEMENT_NAME (dtmfsrc));
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dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
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event->timestamp = GST_CLOCK_TIME_NONE;
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}
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}
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static void
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gst_dtmf_src_start (GstDTMFSrc *dtmfsrc,
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gint event_number, gint event_volume)
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gst_dtmf_src_start (GstDTMFSrc *dtmfsrc)
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{
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GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
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dtmfsrc->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
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dtmfsrc->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
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gst_dtmf_prepare_timestamps (dtmfsrc);
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/* Don't forget to get exclusive access to the stream */
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gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
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if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
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GST_ERROR_OBJECT (dtmfsrc,
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"Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
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@ -547,6 +553,20 @@ gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
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/* Don't forget to release the stream lock */
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gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
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/* Flushing the event queue */
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GstDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
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while (event != NULL) {
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g_free (event);
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event = g_async_queue_try_pop (dtmfsrc->event_queue);
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}
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if (dtmfsrc->last_event) {
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g_free (dtmfsrc->last_event);
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dtmfsrc->last_event = NULL;
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}
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if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
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GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
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return;
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@ -555,7 +575,47 @@ gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
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}
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static void
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gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration, GstBuffer * buffer)
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gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume)
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{
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GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
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event->event_type = DTMF_EVENT_TYPE_START;
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event->sample = 0;
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event->event_number = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
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event->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
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g_async_queue_push (dtmfsrc->event_queue, event);
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}
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static void
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gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc)
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{
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GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
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event->event_type = DTMF_EVENT_TYPE_STOP;
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event->sample = 0;
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event->event_number = 0;
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event->volume = 0;
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g_async_queue_push (dtmfsrc->event_queue, event);
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}
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static void
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gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
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{
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gint buf_size;
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/* Create a buffer with data set to 0 */
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buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
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GST_BUFFER_SIZE (buffer) = buf_size;
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GST_BUFFER_MALLOCDATA (buffer) = g_malloc0(buf_size);
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GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
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}
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static void
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gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration, GstBuffer * buffer)
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{
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gint16 *p;
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gint tone_size;
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@ -578,8 +638,8 @@ gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration, Gs
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/*
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* We add the fundamental frequencies together.
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*/
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f1 = sin(2 * M_PI * key.low_frequency * (dtmfsrc->sample / SAMPLE_RATE));
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f2 = sin(2 * M_PI * key.high_frequency * (dtmfsrc->sample / SAMPLE_RATE));
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f1 = sin(2 * M_PI * key.low_frequency * (event->sample / SAMPLE_RATE));
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f2 = sin(2 * M_PI * key.high_frequency * (event->sample / SAMPLE_RATE));
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amplitude = (f1 + f2) / 2;
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@ -589,7 +649,7 @@ gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration, Gs
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/* Store it in the data buffer */
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*(p++) = (gint16) amplitude;
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(dtmfsrc->sample)++;
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(event->sample)++;
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}
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}
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@ -620,9 +680,10 @@ gst_dtmf_src_wait_for_buffer_ts (GstDTMFSrc *dtmfsrc, GstBuffer * buf)
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static GstBuffer *
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gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc)
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gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
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{
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GstBuffer *buf = NULL;
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guint32 duration;
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GST_DEBUG_OBJECT (dtmfsrc,
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@ -631,14 +692,32 @@ gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc)
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/* create buffer to hold the tone */
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buf = gst_buffer_new ();
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/* The first packet must be inter digit silence, then the second and third must be the
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* minimal pulse duration divided into two packets to make it small
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*/
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switch(event->packet_count) {
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case 0:
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duration = MIN_INTER_DIGIT_INTERVAL;
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gst_dtmf_src_generate_silence (buf, duration);
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break;
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case 1:
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case 2:
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/* Generate the tone */
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gst_dtmf_src_generate_tone(dtmfsrc, DTMF_KEYS[dtmfsrc->event], dtmfsrc->interval, buf);
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duration = MIN_PULSE_DURATION / 2;
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gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
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break;
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default:
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duration = dtmfsrc->interval;
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gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
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break;
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}
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event->packet_count++;
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/* timestamp and duration of GstBuffer */
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GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
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GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
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dtmfsrc->timestamp += GST_BUFFER_DURATION (buf) /2;
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GST_BUFFER_DURATION (buf) = duration * GST_MSECOND;
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GST_BUFFER_TIMESTAMP (buf) = event->timestamp;
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event->timestamp += GST_BUFFER_DURATION (buf);
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/* FIXME: Should we sync to clock ourselves or leave it to sink */
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gst_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
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@ -654,8 +733,41 @@ gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
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{
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GstBuffer *buf = NULL;
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GstFlowReturn ret;
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GstDTMFSrcEvent *event;
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buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc);
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g_async_queue_ref (dtmfsrc->event_queue);
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if (dtmfsrc->last_event == NULL) {
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event = g_async_queue_pop (dtmfsrc->event_queue);
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if (event->event_type == DTMF_EVENT_TYPE_STOP) {
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GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped", GST_BUFFER_SIZE (buf));
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} else if (event->event_type == DTMF_EVENT_TYPE_START) {
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gst_dtmf_prepare_timestamps (dtmfsrc, event);
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/* Don't forget to get exclusive access to the stream */
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gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
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event->packet_count = 0;
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dtmfsrc->last_event = event;
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}
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} else if (dtmfsrc->last_event->packet_count >= 3) {
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event = g_async_queue_try_pop (dtmfsrc->event_queue);
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if (event != NULL) {
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if (event->event_type == DTMF_EVENT_TYPE_START) {
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GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events", GST_BUFFER_SIZE (buf));
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} else if (event->event_type == DTMF_EVENT_TYPE_STOP) {
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gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
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g_free (dtmfsrc->last_event);
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dtmfsrc->last_event = NULL;
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}
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}
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}
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g_async_queue_unref (dtmfsrc->event_queue);
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if (dtmfsrc->last_event) {
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buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
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gst_buffer_ref(buf);
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@ -668,6 +780,7 @@ gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
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gst_buffer_unref(buf);
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GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
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}
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}
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@ -687,7 +800,7 @@ gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
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no_preroll = TRUE;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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/* gst_dtmf_src_start (dtmfsrc, 6, 30); */
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gst_dtmf_src_start (dtmfsrc);
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break;
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default:
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break;
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@ -701,7 +814,7 @@ gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
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switch (transition) {
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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/* Indicate that we don't do PRE_ROLL */
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/* gst_dtmf_src_stop (dtmfsrc); */
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gst_dtmf_src_stop (dtmfsrc);
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no_preroll = TRUE;
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break;
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default:
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@ -727,3 +840,4 @@ gst_dtmf_src_plugin_init (GstPlugin * plugin)
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return gst_element_register (plugin, "dtmfsrc",
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GST_RANK_NONE, GST_TYPE_DTMF_SRC);
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}
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@ -49,18 +49,36 @@ typedef struct _GstDTMFSrcClass GstDTMFSrcClass;
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*
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* The opaque #GstDTMFSrc data structure.
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*/
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enum _GstDTMFEventType {
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DTMF_EVENT_TYPE_START,
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DTMF_EVENT_TYPE_STOP
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};
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typedef enum _GstDTMFEventType GstDTMFEventType;
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struct _GstDTMFSrcEvent {
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GstClockTime timestamp;
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GstDTMFEventType event_type;
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double sample;
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guint16 event_number;
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guint16 volume;
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guint32 packet_count;
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};
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typedef struct _GstDTMFSrcEvent GstDTMFSrcEvent;
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struct _GstDTMFSrc {
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GstElement element;
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GstPad *srcpad;
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GstClockTime timestamp;
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GstSegment segment;
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double sample;
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GAsyncQueue* event_queue;
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GstDTMFSrcEvent* last_event;
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guint16 event;
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guint16 volume;
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guint16 interval;
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};
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struct _GstDTMFSrcClass {
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GstElementClass parent_class;
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};
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