From 50dbdcc4e143a50e278fb634592cf01670bfa449 Mon Sep 17 00:00:00 2001 From: "zeeshan.ali@nokia.com" Date: Mon, 2 Apr 2007 12:46:35 +0000 Subject: [PATCH] [MOVED FROM GST-P-FARSIGHT] Clean-up and refactorize dtmfsrc code 20070402124635-65035-3d13244461c1dd1fcc96b74124ad7a74d2ff0144.gz --- gst/rtpdtmf/gstrtpdtmfsrc.c | 246 ++++++++++++++++++++++++++------------------ 1 file changed, 146 insertions(+), 100 deletions(-) diff --git a/gst/rtpdtmf/gstrtpdtmfsrc.c b/gst/rtpdtmf/gstrtpdtmfsrc.c index 72b4e1a..cecacb9 100644 --- a/gst/rtpdtmf/gstrtpdtmfsrc.c +++ b/gst/rtpdtmf/gstrtpdtmfsrc.c @@ -316,6 +316,61 @@ gst_rtp_dtmf_src_finalize (GObject * object) } static gboolean +gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc, + const GstStructure * event_structure) +{ + gint event_type; + gboolean start; + + if (!gst_structure_get_int (event_structure, "type", &event_type) || + !gst_structure_get_boolean (event_structure, "start", &start) || + event_type != GST_RTP_DTMF_TYPE_EVENT) + goto failure; + + if (start) { + gint event_number; + gint event_volume; + + if (!gst_structure_get_int (event_structure, "number", &event_number) || + !gst_structure_get_int (event_structure, "volume", &event_volume)) + goto failure; + + GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d", + event_number, event_volume); + gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume); + } + + else { + GST_DEBUG_OBJECT (dtmfsrc, "Received stop event"); + gst_rtp_dtmf_src_stop (dtmfsrc); + } + + return TRUE; +failure: + return FALSE; +} + +static gboolean +gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc, GstEvent * event) +{ + gboolean result = FALSE; + const GstStructure *structure; + + if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) { + GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state"); + goto ret; + } + + GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest"); + structure = gst_event_get_structure (event); + if (structure && gst_structure_has_name (structure, "dtmf-event")) + result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure); + +ret: + return result; +} + +static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event) { GstRTPDTMFSrc *dtmfsrc; @@ -327,44 +382,7 @@ gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event) switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CUSTOM_UPSTREAM: { - const GstStructure *structure; - - if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) { - GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state"); - break; - } - - GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest"); - structure = gst_event_get_structure (event); - if (structure && gst_structure_has_name (structure, "dtmf-event")) { - gint event_type; - gboolean start; - - if (!gst_structure_get_int (structure, "type", &event_type) || - !gst_structure_get_boolean (structure, "start", &start) || - event_type != GST_RTP_DTMF_TYPE_EVENT) - break; - - if (start) { - gint event_number; - gint event_volume; - - if (!gst_structure_get_int (structure, "number", &event_number) || - !gst_structure_get_int (structure, "volume", &event_volume)) - break; - - GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d", - event_number, event_volume); - gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume); - } - - else { - GST_DEBUG_OBJECT (dtmfsrc, "Received stop event"); - gst_rtp_dtmf_src_stop (dtmfsrc); - } - } - - result = TRUE; + result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event); break; } /* Ideally this element should not be flushed but let's handle the event @@ -464,18 +482,10 @@ gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock) } static void -gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc, - gint event_number, gint event_volume) +gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc) { GstClock *clock; - g_return_if_fail (dtmfsrc->payload == NULL); - - dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1); - dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT); - dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME); - dtmfsrc->first_packet = TRUE; - clock = GST_ELEMENT_CLOCK (dtmfsrc); if (clock != NULL) dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc)); @@ -489,8 +499,20 @@ gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc, gst_util_uint64_scale_int ( dtmfsrc->timestamp - gst_element_get_base_time (GST_ELEMENT (dtmfsrc)), dtmfsrc->clock_rate, GST_SECOND); - +} +static void +gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc, + gint event_number, gint event_volume) +{ + g_return_if_fail (dtmfsrc->payload == NULL); + + dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1); + dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT); + dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME); + dtmfsrc->first_packet = TRUE; + + gst_rtp_dtmf_prepare_timestamps (dtmfsrc); gst_rtp_dtmf_src_set_caps (dtmfsrc); /* Don't forget to get exclusive access to the stream */ @@ -524,16 +546,32 @@ gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc) } static void -gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) +gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf) { - GstBuffer *buf = NULL; - GstFlowReturn ret; - GstRTPDTMFPayload *payload; - GstClock * clock; + GstClock *clock; + + clock = GST_ELEMENT_CLOCK (dtmfsrc); + if (clock != NULL) { + GstClockID clock_id; + GstClockReturn clock_ret; - /* create buffer to hold the payload */ - buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0); + clock_id = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp); + clock_ret = gst_clock_id_wait (clock_id, NULL); + if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) { + GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s", + GST_ELEMENT_NAME (clock)); + } + gst_clock_id_unref (clock_id); + } + else { + GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", GST_ELEMENT_NAME (dtmfsrc)); + } +} + +static void +gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf) +{ gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc); gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt); if (dtmfsrc->first_packet) { @@ -547,12 +585,19 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp); dtmfsrc->rtp_timestamp += DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000; +} +static void +gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf) +{ + GstRTPDTMFPayload *payload; + + gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf); /* duration of DTMF payload */ dtmfsrc->payload->duration += DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000; - + /* timestamp and duration of GstBuffer */ GST_BUFFER_DURATION (buf) = DEFAULT_PACKET_INTERVAL * GST_MSECOND; GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp; @@ -563,28 +608,36 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) /* copy payload and convert to network-byte order */ g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload)); payload->duration = g_htons (payload->duration); +} - /* FIXME: Should we sync to clock ourselves or leave it to sink */ - clock = GST_ELEMENT_CLOCK (dtmfsrc); - if (clock != NULL) { - GstClockID clock_id; - GstClockReturn clock_ret; +static GstBuffer * +gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) +{ + GstBuffer *buf = NULL; + + /* create buffer to hold the payload */ + buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0); - clock_id = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp); - clock_ret = gst_clock_id_wait (clock_id, NULL); - if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) { - GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s", - GST_ELEMENT_NAME (clock)); - } - gst_clock_id_unref (clock_id); - } + gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf); - else { - GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", GST_ELEMENT_NAME (dtmfsrc)); - } + /* FIXME: Should we sync to clock ourselves or leave it to sink */ + gst_rtp_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)); + + return buf; +} + +static void +gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) +{ + GstBuffer *buf = NULL; + GstFlowReturn ret; + + /* create buffer to hold the payload */ + buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc); + GST_DEBUG_OBJECT (dtmfsrc, "pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf)); ret = gst_pad_push (dtmfsrc->srcpad, buf); @@ -620,6 +673,26 @@ gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc) gst_caps_unref (caps); } +static void +gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc) +{ + if (dtmfsrc->ssrc == -1) + dtmfsrc->current_ssrc = g_random_int (); + else + dtmfsrc->current_ssrc = dtmfsrc->ssrc; + + if (dtmfsrc->seqnum_offset == -1) + dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16); + else + dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset; + dtmfsrc->seqnum = dtmfsrc->seqnum_base; + + if (dtmfsrc->ts_offset == -1) + dtmfsrc->ts_base = g_random_int (); + else + dtmfsrc->ts_base = dtmfsrc->ts_offset; +} + static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition) { @@ -630,33 +703,10 @@ gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition) dtmfsrc = GST_RTP_DTMF_SRC (element); switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; case GST_STATE_CHANGE_READY_TO_PAUSED: - - if (dtmfsrc->ssrc == -1) - dtmfsrc->current_ssrc = g_random_int (); - else - dtmfsrc->current_ssrc = dtmfsrc->ssrc; - - if (dtmfsrc->seqnum_offset == -1) - dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16); - else - dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset; - dtmfsrc->seqnum = dtmfsrc->seqnum_base; - - if (dtmfsrc->ts_offset == -1) - dtmfsrc->ts_base = g_random_int (); - else - dtmfsrc->ts_base = dtmfsrc->ts_offset; - - - + gst_rtp_dtmf_src_ready_to_paused (dtmfsrc); /* Indicate that we don't do PRE_ROLL */ no_preroll = TRUE; - - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; default: break; @@ -672,10 +722,6 @@ gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition) /* Indicate that we don't do PRE_ROLL */ no_preroll = TRUE; break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; default: break; } -- 2.7.4