| /* GStreamer DTMF source |
| * |
| * gstdtmfsrc.c: |
| * |
| * Copyright (C) <2007> Collabora. |
| * Contact: Youness Alaoui <youness.alaoui@collabora.co.uk> |
| * Copyright (C) <2007> Nokia Corporation. |
| * Contact: Zeeshan Ali <zeeshan.ali@nokia.com> |
| * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
| * 2000,2005 Wim Taymans <wim@fluendo.com> |
| * |
| * This library is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Library General Public |
| * License as published by the Free Software Foundation; either |
| * version 2 of the License, or (at your option) any later version. |
| * |
| * This library is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Library General Public License for more details. |
| * |
| * You should have received a copy of the GNU Library General Public |
| * License along with this library; if not, write to the |
| * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, USA. |
| */ |
| |
| /** |
| * SECTION:element-dtmfsrc |
| * @see_also: rtpdtmsrc, rtpdtmfmuxx |
| * |
| * The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone packets on request |
| * from application. The application communicates the beginning and end of a |
| * DTMF event using custom upstream gstreamer events. To report a DTMF event, an |
| * application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a |
| * structure of name "dtmf-event" with fields set according to the following |
| * table: |
| * |
| * <informaltable> |
| * <tgroup cols='4'> |
| * <colspec colname='Name' /> |
| * <colspec colname='Type' /> |
| * <colspec colname='Possible values' /> |
| * <colspec colname='Purpose' /> |
| * <thead> |
| * <row> |
| * <entry>Name</entry> |
| * <entry>GType</entry> |
| * <entry>Possible values</entry> |
| * <entry>Purpose</entry> |
| * </row> |
| * </thead> |
| * <tbody> |
| * <row> |
| * <entry>type</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <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. 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> |
| * <entry>number</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>0-15</entry> |
| * <entry>The event number.</entry> |
| * </row> |
| * <row> |
| * <entry>volume</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>0-36</entry> |
| * <entry>This field describes the power level of the tone, expressed in dBm0 |
| * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of |
| * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE. |
| * </entry> |
| * </row> |
| * <row> |
| * <entry>start</entry> |
| * <entry>G_TYPE_BOOLEAN</entry> |
| * <entry>True or False</entry> |
| * <entry>Whether the event is starting or ending.</entry> |
| * </row> |
| * <row> |
| * <entry>method</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>2</entry> |
| * <entry>The method used for sending event, this element will react if this |
| * field is absent or 2. |
| * </entry> |
| * </row> |
| * </tbody> |
| * </tgroup> |
| * </informaltable> |
| * |
| * For example, the following code informs the pipeline (and in turn, the |
| * DTMFSrc element inside the pipeline) about the start of a DTMF named |
| * event '1' of volume -25 dBm0: |
| * |
| * <programlisting> |
| * structure = gst_structure_new ("dtmf-event", |
| * "type", G_TYPE_INT, 1, |
| * "number", G_TYPE_INT, 1, |
| * "volume", G_TYPE_INT, 25, |
| * "start", G_TYPE_BOOLEAN, TRUE, NULL); |
| * |
| * event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure); |
| * gst_element_send_event (pipeline, event); |
| * </programlisting> |
| * |
| * When a DTMF tone actually starts or stop, a "dtmf-event-processed" |
| * element #GstMessage with the same fields as the "dtmf-event" |
| * #GstEvent that was used to request the event. Also, if any event |
| * has not been processed when the element goes from the PAUSED to the |
| * READY state, then a "dtmf-event-dropped" message is posted on the |
| * #GstBus in the order that they were received. |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include <stdlib.h> |
| #include <string.h> |
| #include <math.h> |
| |
| #include <glib.h> |
| |
| #include "gstdtmfcommon.h" |
| |
| #include "gstdtmfsrc.h" |
| |
| #include <gst/audio/audio.h> |
| |
| #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 DEFAULT_SAMPLE_RATE 8000 |
| #define SAMPLE_SIZE 16 |
| #define CHANNELS 1 |
| #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION) |
| |
| |
| typedef struct st_dtmf_key |
| { |
| const char *event_name; |
| int event_encoding; |
| float low_frequency; |
| float high_frequency; |
| } DTMF_KEY; |
| |
| static const DTMF_KEY DTMF_KEYS[] = { |
| {"DTMF_KEY_EVENT_0", 0, 941, 1336}, |
| {"DTMF_KEY_EVENT_1", 1, 697, 1209}, |
| {"DTMF_KEY_EVENT_2", 2, 697, 1336}, |
| {"DTMF_KEY_EVENT_3", 3, 697, 1477}, |
| {"DTMF_KEY_EVENT_4", 4, 770, 1209}, |
| {"DTMF_KEY_EVENT_5", 5, 770, 1336}, |
| {"DTMF_KEY_EVENT_6", 6, 770, 1477}, |
| {"DTMF_KEY_EVENT_7", 7, 852, 1209}, |
| {"DTMF_KEY_EVENT_8", 8, 852, 1336}, |
| {"DTMF_KEY_EVENT_9", 9, 852, 1477}, |
| {"DTMF_KEY_EVENT_S", 10, 941, 1209}, |
| {"DTMF_KEY_EVENT_P", 11, 941, 1477}, |
| {"DTMF_KEY_EVENT_A", 12, 697, 1633}, |
| {"DTMF_KEY_EVENT_B", 13, 770, 1633}, |
| {"DTMF_KEY_EVENT_C", 14, 852, 1633}, |
| {"DTMF_KEY_EVENT_D", 15, 941, 1633}, |
| }; |
| |
| #define MAX_DTMF_EVENTS 16 |
| |
| enum |
| { |
| DTMF_KEY_EVENT_1 = 1, |
| DTMF_KEY_EVENT_2 = 2, |
| DTMF_KEY_EVENT_3 = 3, |
| DTMF_KEY_EVENT_4 = 4, |
| DTMF_KEY_EVENT_5 = 5, |
| DTMF_KEY_EVENT_6 = 6, |
| DTMF_KEY_EVENT_7 = 7, |
| DTMF_KEY_EVENT_8 = 8, |
| DTMF_KEY_EVENT_9 = 9, |
| DTMF_KEY_EVENT_0 = 0, |
| DTMF_KEY_EVENT_STAR = 10, |
| DTMF_KEY_EVENT_POUND = 11, |
| DTMF_KEY_EVENT_A = 12, |
| DTMF_KEY_EVENT_B = 13, |
| DTMF_KEY_EVENT_C = 14, |
| DTMF_KEY_EVENT_D = 15, |
| }; |
| |
| GST_DEBUG_CATEGORY_STATIC (gst_dtmf_src_debug); |
| #define GST_CAT_DEFAULT gst_dtmf_src_debug |
| |
| enum |
| { |
| PROP_0, |
| PROP_INTERVAL, |
| }; |
| |
| static GstStaticPadTemplate gst_dtmf_src_template = |
| GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, " |
| "format = (string) \"" GST_AUDIO_NE (S16) "\", " |
| "rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1") |
| ); |
| |
| #define parent_class gst_dtmf_src_parent_class |
| G_DEFINE_TYPE (GstDTMFSrc, gst_dtmf_src, GST_TYPE_BASE_SRC); |
| |
| static void gst_dtmf_src_finalize (GObject * object); |
| |
| 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 (GstBaseSrc * src, GstEvent * event); |
| static gboolean gst_dtmf_src_send_event (GstElement * src, GstEvent * event); |
| static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element, |
| GstStateChange transition); |
| 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 gboolean gst_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query); |
| |
| |
| static void |
| 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); |
| |
| |
| GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug, "dtmfsrc", 0, "dtmfsrc element"); |
| |
| gst_element_class_add_static_pad_template (gstelement_class, |
| &gst_dtmf_src_template); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "DTMF tone generator", "Source/Audio", "Generates DTMF tones", |
| "Youness Alaoui <youness.alaoui@collabora.co.uk>"); |
| |
| |
| gobject_class->finalize = gst_dtmf_src_finalize; |
| gobject_class->set_property = gst_dtmf_src_set_property; |
| gobject_class->get_property = gst_dtmf_src_get_property; |
| |
| g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL, |
| 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 | G_PARAM_STATIC_STRINGS)); |
| |
| gstelement_class->change_state = |
| GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state); |
| gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_dtmf_src_send_event); |
| 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); |
| gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_dtmf_src_query); |
| } |
| |
| static void |
| event_free (GstDTMFSrcEvent * event) |
| { |
| if (event) |
| g_slice_free (GstDTMFSrcEvent, event); |
| } |
| |
| static void |
| gst_dtmf_src_init (GstDTMFSrc * dtmfsrc) |
| { |
| /* 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_full ((GDestroyNotify) event_free); |
| dtmfsrc->last_event = NULL; |
| |
| dtmfsrc->sample_rate = DEFAULT_SAMPLE_RATE; |
| |
| GST_DEBUG_OBJECT (dtmfsrc, "init done"); |
| } |
| |
| static void |
| gst_dtmf_src_finalize (GObject * object) |
| { |
| GstDTMFSrc *dtmfsrc; |
| |
| dtmfsrc = GST_DTMF_SRC (object); |
| |
| if (dtmfsrc->event_queue) { |
| g_async_queue_unref (dtmfsrc->event_queue); |
| dtmfsrc->event_queue = NULL; |
| } |
| |
| G_OBJECT_CLASS (gst_dtmf_src_parent_class)->finalize (object); |
| } |
| |
| static gboolean |
| gst_dtmf_src_handle_dtmf_event (GstDTMFSrc * dtmfsrc, GstEvent * event) |
| { |
| const GstStructure *event_structure; |
| GstStateChangeReturn sret; |
| GstState state; |
| gint event_type; |
| gboolean start; |
| gint method; |
| GstClockTime last_stop; |
| gint event_number; |
| gint event_volume; |
| gboolean correct_order; |
| |
| sret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0); |
| if (sret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) { |
| GST_DEBUG_OBJECT (dtmfsrc, "dtmf-event, but not in PLAYING state"); |
| goto failure; |
| } |
| |
| event_structure = gst_event_get_structure (event); |
| |
| if (!gst_structure_get_int (event_structure, "type", &event_type) || |
| !gst_structure_get_boolean (event_structure, "start", &start) || |
| (start == TRUE && event_type != GST_TONE_DTMF_TYPE_EVENT)) |
| goto failure; |
| |
| if (gst_structure_get_int (event_structure, "method", &method)) { |
| if (method != 2) { |
| goto failure; |
| } |
| } |
| |
| if (start) |
| if (!gst_structure_get_int (event_structure, "number", &event_number) || |
| !gst_structure_get_int (event_structure, "volume", &event_volume)) |
| goto failure; |
| |
| |
| GST_OBJECT_LOCK (dtmfsrc); |
| if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop)) |
| dtmfsrc->last_stop = last_stop; |
| else |
| dtmfsrc->last_stop = GST_CLOCK_TIME_NONE; |
| correct_order = (start != dtmfsrc->last_event_was_start); |
| dtmfsrc->last_event_was_start = start; |
| GST_OBJECT_UNLOCK (dtmfsrc); |
| |
| if (!correct_order) |
| goto failure; |
| |
| if (start) { |
| GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d", |
| 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_add_stop_event (dtmfsrc); |
| } |
| |
| return TRUE; |
| failure: |
| return FALSE; |
| } |
| |
| static gboolean |
| gst_dtmf_src_handle_event (GstBaseSrc * src, GstEvent * event) |
| { |
| GstDTMFSrc *dtmfsrc; |
| gboolean result = FALSE; |
| |
| dtmfsrc = GST_DTMF_SRC (src); |
| |
| GST_LOG_OBJECT (dtmfsrc, "Received an %s event on the src pad", |
| GST_EVENT_TYPE_NAME (event)); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| case GST_EVENT_CUSTOM_UPSTREAM: |
| if (gst_event_has_name (event, "dtmf-event")) { |
| result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event); |
| break; |
| } |
| /* fall through */ |
| default: |
| result = GST_BASE_SRC_CLASS (parent_class)->event (src, event); |
| break; |
| } |
| |
| return result; |
| } |
| |
| |
| static gboolean |
| gst_dtmf_src_send_event (GstElement * element, GstEvent * event) |
| { |
| GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (element); |
| gboolean ret; |
| |
| GST_LOG_OBJECT (dtmfsrc, "Received an %s event via send_event", |
| GST_EVENT_TYPE_NAME (event)); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| case GST_EVENT_CUSTOM_BOTH: |
| case GST_EVENT_CUSTOM_BOTH_OOB: |
| case GST_EVENT_CUSTOM_UPSTREAM: |
| case GST_EVENT_CUSTOM_DOWNSTREAM: |
| case GST_EVENT_CUSTOM_DOWNSTREAM_OOB: |
| if (gst_event_has_name (event, "dtmf-event")) { |
| ret = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event); |
| break; |
| } |
| /* fall through */ |
| default: |
| ret = GST_ELEMENT_CLASS (parent_class)->send_event (element, event); |
| break; |
| } |
| |
| return ret; |
| } |
| |
| static void |
| gst_dtmf_src_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec) |
| { |
| GstDTMFSrc *dtmfsrc; |
| |
| dtmfsrc = GST_DTMF_SRC (object); |
| |
| switch (prop_id) { |
| case PROP_INTERVAL: |
| dtmfsrc->interval = g_value_get_uint (value); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value, |
| GParamSpec * pspec) |
| { |
| GstDTMFSrc *dtmfsrc; |
| |
| dtmfsrc = GST_DTMF_SRC (object); |
| |
| switch (prop_id) { |
| case PROP_INTERVAL: |
| g_value_set_uint (value, dtmfsrc->interval); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_dtmf_prepare_timestamps (GstDTMFSrc * dtmfsrc) |
| { |
| GstClockTime last_stop; |
| GstClockTime timestamp; |
| |
| GST_OBJECT_LOCK (dtmfsrc); |
| last_stop = dtmfsrc->last_stop; |
| GST_OBJECT_UNLOCK (dtmfsrc); |
| |
| if (GST_CLOCK_TIME_IS_VALID (last_stop)) { |
| timestamp = last_stop; |
| } else { |
| GstClock *clock; |
| |
| /* If there is no valid start time, lets use now as the start time */ |
| |
| clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc)); |
| if (clock != NULL) { |
| timestamp = gst_clock_get_time (clock) |
| - gst_element_get_base_time (GST_ELEMENT (dtmfsrc)); |
| 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); |
| dtmfsrc->timestamp = GST_CLOCK_TIME_NONE; |
| g_free (dtmf_name); |
| return; |
| } |
| } |
| |
| /* Make sure the timestamp always goes forward */ |
| if (timestamp > dtmfsrc->timestamp) |
| dtmfsrc->timestamp = timestamp; |
| } |
| |
| static void |
| gst_dtmf_src_add_start_event (GstDTMFSrc * dtmfsrc, gint event_number, |
| gint event_volume) |
| { |
| |
| GstDTMFSrcEvent *event = g_slice_new0 (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_slice_new0 (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 GstBuffer * |
| gst_dtmf_src_generate_silence (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; |
| |
| return gst_buffer_new_wrapped (g_malloc0 (buf_size), buf_size); |
| } |
| |
| static GstBuffer * |
| gst_dtmf_src_generate_tone (GstDTMFSrcEvent * event, DTMF_KEY key, |
| float duration, gint sample_rate) |
| { |
| GstBuffer *buffer; |
| GstMapInfo map; |
| gint16 *p; |
| gint tone_size; |
| double i = 0; |
| double amplitude, f1, f2; |
| double volume_factor; |
| static GstAllocationParams params = { 0, 1, 0, 0, }; |
| |
| /* Create a buffer for the tone */ |
| tone_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8; |
| |
| buffer = gst_buffer_new_allocate (NULL, tone_size, ¶ms); |
| |
| gst_buffer_map (buffer, &map, GST_MAP_READWRITE); |
| p = (gint16 *) map.data; |
| |
| volume_factor = pow (10, (-event->volume) / 20); |
| |
| /* |
| * For each sample point we calculate 'x' as the |
| * the amplitude value. |
| */ |
| for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) { |
| /* |
| * 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)); |
| |
| amplitude = (f1 + f2) / 2; |
| |
| /* Adjust the volume */ |
| amplitude *= volume_factor; |
| |
| /* Make the [-1:1] interval into a [-32767:32767] interval */ |
| amplitude *= 32767; |
| |
| /* Store it in the data buffer */ |
| *(p++) = (gint16) amplitude; |
| |
| (event->sample)++; |
| } |
| |
| gst_buffer_unmap (buffer, &map); |
| |
| return buffer; |
| } |
| |
| |
| |
| static GstBuffer * |
| gst_dtmf_src_create_next_tone_packet (GstDTMFSrc * dtmfsrc, |
| GstDTMFSrcEvent * event) |
| { |
| GstBuffer *buf = NULL; |
| gboolean send_silence = FALSE; |
| |
| GST_LOG_OBJECT (dtmfsrc, "Creating buffer for tone %s", |
| DTMF_KEYS[event->event_number].event_name); |
| |
| if (event->packet_count * dtmfsrc->interval < MIN_INTER_DIGIT_INTERVAL) { |
| send_silence = TRUE; |
| } |
| |
| if (send_silence) { |
| GST_LOG_OBJECT (dtmfsrc, "Generating silence"); |
| buf = gst_dtmf_src_generate_silence (dtmfsrc->interval, |
| dtmfsrc->sample_rate); |
| } else { |
| GST_LOG_OBJECT (dtmfsrc, "Generating tone"); |
| buf = gst_dtmf_src_generate_tone (event, DTMF_KEYS[event->event_number], |
| dtmfsrc->interval, dtmfsrc->sample_rate); |
| } |
| event->packet_count++; |
| |
| |
| /* timestamp and duration of GstBuffer */ |
| GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND; |
| GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp; |
| |
| GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration " |
| " gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT, |
| event->event_number, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), |
| GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); |
| |
| dtmfsrc->timestamp += GST_BUFFER_DURATION (buf); |
| |
| return buf; |
| } |
| |
| static void |
| gst_dtmf_src_post_message (GstDTMFSrc * dtmfsrc, const gchar * message_name, |
| GstDTMFSrcEvent * event) |
| { |
| GstStructure *s = NULL; |
| |
| switch (event->event_type) { |
| case DTMF_EVENT_TYPE_START: |
| s = gst_structure_new (message_name, |
| "type", G_TYPE_INT, 1, |
| "method", G_TYPE_INT, 2, |
| "start", G_TYPE_BOOLEAN, TRUE, |
| "number", G_TYPE_INT, event->event_number, |
| "volume", G_TYPE_INT, event->volume, NULL); |
| break; |
| case DTMF_EVENT_TYPE_STOP: |
| s = gst_structure_new (message_name, |
| "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 2, |
| "start", G_TYPE_BOOLEAN, FALSE, NULL); |
| break; |
| case DTMF_EVENT_TYPE_PAUSE_TASK: |
| return; |
| } |
| |
| if (s) |
| gst_element_post_message (GST_ELEMENT (dtmfsrc), |
| gst_message_new_element (GST_OBJECT (dtmfsrc), s)); |
| } |
| |
| static GstFlowReturn |
| gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset, |
| guint length, GstBuffer ** buffer) |
| { |
| GstBuffer *buf = NULL; |
| 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"); |
| gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event); |
| break; |
| case DTMF_EVENT_TYPE_START: |
| gst_dtmf_prepare_timestamps (dtmfsrc); |
| |
| event->packet_count = 0; |
| dtmfsrc->last_event = event; |
| event = NULL; |
| gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", |
| dtmfsrc->last_event); |
| 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_slice_free (GstDTMFSrcEvent, 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"); |
| gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event); |
| break; |
| case DTMF_EVENT_TYPE_STOP: |
| g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event); |
| dtmfsrc->last_event = NULL; |
| gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event); |
| 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_slice_free (GstDTMFSrcEvent, event); |
| } |
| } |
| } while (dtmfsrc->last_event == NULL); |
| |
| GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time"); |
| |
| clock = gst_element_get_clock (GST_ELEMENT (basesrc)); |
| |
| clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp + |
| gst_element_get_base_time (GST_ELEMENT (dtmfsrc))); |
| 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; |
| } |
| |
| buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event); |
| |
| GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT, |
| gst_buffer_get_size (buf)); |
| *buffer = buf; |
| |
| return GST_FLOW_OK; |
| |
| paused_locked: |
| GST_OBJECT_UNLOCK (dtmfsrc); |
| |
| paused: |
| |
| if (dtmfsrc->last_event) { |
| GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event"); |
| /* Don't forget to release the stream lock */ |
| g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event); |
| dtmfsrc->last_event = NULL; |
| } |
| |
| return GST_FLOW_FLUSHING; |
| |
| } |
| |
| 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_slice_new0 (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) |
| { |
| GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc); |
| GstCaps *caps; |
| GstStructure *s; |
| gboolean ret; |
| |
| caps = gst_pad_get_allowed_caps (GST_BASE_SRC_PAD (basesrc)); |
| |
| if (!caps) |
| caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (basesrc)); |
| |
| if (gst_caps_is_empty (caps)) { |
| gst_caps_unref (caps); |
| return FALSE; |
| } |
| |
| caps = gst_caps_truncate (caps); |
| |
| caps = gst_caps_make_writable (caps); |
| s = gst_caps_get_structure (caps, 0); |
| |
| gst_structure_fixate_field_nearest_int (s, "rate", DEFAULT_SAMPLE_RATE); |
| |
| if (!gst_structure_get_int (s, "rate", &dtmfsrc->sample_rate)) { |
| GST_ERROR_OBJECT (dtmfsrc, "Could not get rate"); |
| gst_caps_unref (caps); |
| return FALSE; |
| } |
| |
| ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps); |
| |
| gst_caps_unref (caps); |
| |
| return ret; |
| } |
| |
| static gboolean |
| gst_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query) |
| { |
| GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc); |
| gboolean res = FALSE; |
| |
| switch (GST_QUERY_TYPE (query)) { |
| case GST_QUERY_LATENCY: |
| { |
| GstClockTime latency; |
| |
| latency = dtmfsrc->interval * GST_MSECOND; |
| gst_query_set_latency (query, gst_base_src_is_live (basesrc), latency, |
| GST_CLOCK_TIME_NONE); |
| GST_DEBUG_OBJECT (dtmfsrc, "Reporting latency of %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (latency)); |
| res = TRUE; |
| } |
| break; |
| default: |
| res = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query); |
| break; |
| } |
| |
| return res; |
| } |
| |
| static GstStateChangeReturn |
| gst_dtmf_src_change_state (GstElement * element, GstStateChange transition) |
| { |
| 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: |
| /* Flushing the event queue */ |
| event = g_async_queue_try_pop (dtmfsrc->event_queue); |
| |
| while (event != NULL) { |
| gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event); |
| g_slice_free (GstDTMFSrcEvent, event); |
| event = g_async_queue_try_pop (dtmfsrc->event_queue); |
| } |
| dtmfsrc->last_event_was_start = FALSE; |
| dtmfsrc->timestamp = 0; |
| no_preroll = TRUE; |
| break; |
| default: |
| break; |
| } |
| |
| if ((result = |
| GST_ELEMENT_CLASS (gst_dtmf_src_parent_class)->change_state (element, |
| transition)) == GST_STATE_CHANGE_FAILURE) |
| goto failure; |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_PLAYING_TO_PAUSED: |
| 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) { |
| gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event); |
| g_slice_free (GstDTMFSrcEvent, event); |
| event = g_async_queue_try_pop (dtmfsrc->event_queue); |
| } |
| dtmfsrc->last_event_was_start = FALSE; |
| |
| break; |
| default: |
| break; |
| } |
| |
| if (no_preroll && result == GST_STATE_CHANGE_SUCCESS) |
| result = GST_STATE_CHANGE_NO_PREROLL; |
| |
| return result; |
| |
| /* ERRORS */ |
| failure: |
| { |
| GST_ERROR_OBJECT (dtmfsrc, "parent failed state change"); |
| return result; |
| } |
| } |
| |
| gboolean |
| gst_dtmf_src_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "dtmfsrc", |
| GST_RANK_NONE, GST_TYPE_DTMF_SRC); |
| } |