/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2001 Thomas * 2005,2006 Wim Taymans * * adder.c: Adder element, N in, one out, samples are added * * 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-adder * * The adder allows to mix several streams into one by adding the data. * Mixed data is clamped to the min/max values of the data format. * * The adder currently mixes all data received on the sinkpads as soon as * possible without trying to synchronize the streams. * * * Example launch line * |[ * gst-launch audiotestsrc freq=100 ! adder name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix. * ]| This pipeline produces two sine waves mixed together. * */ /* Element-Checklist-Version: 5 */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstadder.h" #include #include /* strcmp */ #include "gstadderorc.h" #define GST_CAT_DEFAULT gst_adder_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); #define DEFAULT_PAD_VOLUME (1.0) #define DEFAULT_PAD_MUTE (FALSE) /* some defines for audio processing */ /* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0 * we map 1.0 to VOLUME_UNITY_INT* */ #define VOLUME_UNITY_INT8 8 /* internal int for unity 2^(8-5) */ #define VOLUME_UNITY_INT8_BIT_SHIFT 3 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT16 2048 /* internal int for unity 2^(16-5) */ #define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT24 524288 /* internal int for unity 2^(24-5) */ #define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */ #define VOLUME_UNITY_INT32_BIT_SHIFT 27 enum { PROP_PAD_0, PROP_PAD_VOLUME, PROP_PAD_MUTE }; G_DEFINE_TYPE (GstAdderPad, gst_adder_pad, GST_TYPE_PAD); static void gst_adder_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAdderPad *pad = GST_ADDER_PAD (object); switch (prop_id) { case PROP_PAD_VOLUME: g_value_set_double (value, pad->volume); break; case PROP_PAD_MUTE: g_value_set_boolean (value, pad->mute); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_adder_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAdderPad *pad = GST_ADDER_PAD (object); switch (prop_id) { case PROP_PAD_VOLUME: GST_OBJECT_LOCK (pad); pad->volume = g_value_get_double (value); pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8; pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16; pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32; GST_OBJECT_UNLOCK (pad); break; case PROP_PAD_MUTE: GST_OBJECT_LOCK (pad); pad->mute = g_value_get_boolean (value); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_adder_pad_class_init (GstAdderPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->set_property = gst_adder_pad_set_property; gobject_class->get_property = gst_adder_pad_get_property; g_object_class_install_property (gobject_class, PROP_PAD_VOLUME, g_param_spec_double ("volume", "Volume", "Volume of this pad", 0.0, 10.0, DEFAULT_PAD_VOLUME, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PAD_MUTE, g_param_spec_boolean ("mute", "Mute", "Mute this pad", DEFAULT_PAD_MUTE, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); } static void gst_adder_pad_init (GstAdderPad * pad) { pad->volume = DEFAULT_PAD_VOLUME; pad->mute = DEFAULT_PAD_MUTE; } enum { PROP_0, PROP_FILTER_CAPS }; /* elementfactory information */ #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define CAPS \ GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \ ", layout = (string) { interleaved, non-interleaved }" #else #define CAPS \ GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \ ", layout = (string) { interleaved, non-interleaved }" #endif static GstStaticPadTemplate gst_adder_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (CAPS) ); static GstStaticPadTemplate gst_adder_sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS (CAPS) ); static void gst_adder_child_proxy_init (gpointer g_iface, gpointer iface_data); #define gst_adder_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAdder, gst_adder, GST_TYPE_ELEMENT, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, gst_adder_child_proxy_init)); static void gst_adder_dispose (GObject * object); static void gst_adder_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_adder_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_adder_setcaps (GstAdder * adder, GstPad * pad, GstCaps * caps); static gboolean gst_adder_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static gboolean gst_adder_sink_query (GstCollectPads * pads, GstCollectData * pad, GstQuery * query, gpointer user_data); static gboolean gst_adder_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_adder_sink_event (GstCollectPads * pads, GstCollectData * pad, GstEvent * event, gpointer user_data); static GstPad *gst_adder_request_new_pad (GstElement * element, GstPadTemplate * temp, const gchar * unused, const GstCaps * caps); static void gst_adder_release_pad (GstElement * element, GstPad * pad); static GstStateChangeReturn gst_adder_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_adder_do_clip (GstCollectPads * pads, GstCollectData * data, GstBuffer * buffer, GstBuffer ** out, gpointer user_data); static GstFlowReturn gst_adder_collected (GstCollectPads * pads, gpointer user_data); /* we can only accept caps that we and downstream can handle. * if we have filtercaps set, use those to constrain the target caps. */ static GstCaps * gst_adder_sink_getcaps (GstPad * pad, GstCaps * filter) { GstAdder *adder; GstCaps *result, *peercaps, *current_caps, *filter_caps; GstStructure *s; gint i, n; adder = GST_ADDER (GST_PAD_PARENT (pad)); GST_OBJECT_LOCK (adder); /* take filter */ if ((filter_caps = adder->filter_caps)) { if (filter) filter_caps = gst_caps_intersect_full (filter, filter_caps, GST_CAPS_INTERSECT_FIRST); else gst_caps_ref (filter_caps); } else { filter_caps = filter ? gst_caps_ref (filter) : NULL; } GST_OBJECT_UNLOCK (adder); if (filter_caps && gst_caps_is_empty (filter_caps)) { GST_WARNING_OBJECT (pad, "Empty filter caps"); return filter_caps; } /* get the downstream possible caps */ peercaps = gst_pad_peer_query_caps (adder->srcpad, filter_caps); /* get the allowed caps on this sinkpad */ GST_OBJECT_LOCK (adder); current_caps = adder->current_caps ? gst_caps_ref (adder->current_caps) : NULL; if (current_caps == NULL) { current_caps = gst_pad_get_pad_template_caps (pad); if (!current_caps) current_caps = gst_caps_new_any (); } GST_OBJECT_UNLOCK (adder); if (peercaps) { /* if the peer has caps, intersect */ GST_DEBUG_OBJECT (adder, "intersecting peer and our caps"); result = gst_caps_intersect_full (peercaps, current_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (peercaps); gst_caps_unref (current_caps); } else { /* the peer has no caps (or there is no peer), just use the allowed caps * of this sinkpad. */ /* restrict with filter-caps if any */ if (filter_caps) { GST_DEBUG_OBJECT (adder, "no peer caps, using filtered caps"); result = gst_caps_intersect_full (filter_caps, current_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (current_caps); } else { GST_DEBUG_OBJECT (adder, "no peer caps, using our caps"); result = current_caps; } } result = gst_caps_make_writable (result); n = gst_caps_get_size (result); for (i = 0; i < n; i++) { GstStructure *sref; s = gst_caps_get_structure (result, i); sref = gst_structure_copy (s); gst_structure_set (sref, "channels", GST_TYPE_INT_RANGE, 0, 2, NULL); if (gst_structure_is_subset (s, sref)) { /* This field is irrelevant when in mono or stereo */ gst_structure_remove_field (s, "channel-mask"); } gst_structure_free (sref); } if (filter_caps) gst_caps_unref (filter_caps); GST_LOG_OBJECT (adder, "getting caps on pad %p,%s to %" GST_PTR_FORMAT, pad, GST_PAD_NAME (pad), result); return result; } static gboolean gst_adder_sink_query (GstCollectPads * pads, GstCollectData * pad, GstQuery * query, gpointer user_data) { gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_adder_sink_getcaps (pad->pad, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: res = gst_collect_pads_query_default (pads, pad, query, FALSE); break; } return res; } /* the first caps we receive on any of the sinkpads will define the caps for all * the other sinkpads because we can only mix streams with the same caps. */ static gboolean gst_adder_setcaps (GstAdder * adder, GstPad * pad, GstCaps * orig_caps) { GstCaps *caps; GstAudioInfo info; GstStructure *s; gint channels; caps = gst_caps_copy (orig_caps); s = gst_caps_get_structure (caps, 0); if (gst_structure_get_int (s, "channels", &channels)) if (channels <= 2) gst_structure_remove_field (s, "channel-mask"); if (!gst_audio_info_from_caps (&info, caps)) goto invalid_format; GST_OBJECT_LOCK (adder); /* don't allow reconfiguration for now; there's still a race between the * different upstream threads doing query_caps + accept_caps + sending * (possibly different) CAPS events, but there's not much we can do about * that, upstream needs to deal with it. */ if (adder->current_caps != NULL) { if (gst_audio_info_is_equal (&info, &adder->info)) { GST_OBJECT_UNLOCK (adder); gst_caps_unref (caps); return TRUE; } else { GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but " "current caps are %" GST_PTR_FORMAT, caps, adder->current_caps); GST_OBJECT_UNLOCK (adder); gst_pad_push_event (pad, gst_event_new_reconfigure ()); gst_caps_unref (caps); return FALSE; } } GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps); adder->current_caps = gst_caps_ref (caps); memcpy (&adder->info, &info, sizeof (info)); GST_OBJECT_UNLOCK (adder); /* send caps event later, after stream-start event */ GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps); gst_caps_unref (caps); return TRUE; /* ERRORS */ invalid_format: { gst_caps_unref (caps); GST_WARNING_OBJECT (adder, "invalid format set as caps"); return FALSE; } } /* FIXME, the duration query should reflect how long you will produce * data, that is the amount of stream time until you will emit EOS. * * For synchronized mixing this is always the max of all the durations * of upstream since we emit EOS when all of them finished. * * We don't do synchronized mixing so this really depends on where the * streams where punched in and what their relative offsets are against * eachother which we can get from the first timestamps we see. * * When we add a new stream (or remove a stream) the duration might * also become invalid again and we need to post a new DURATION * message to notify this fact to the parent. * For now we take the max of all the upstream elements so the simple * cases work at least somewhat. */ static gboolean gst_adder_query_duration (GstAdder * adder, GstQuery * query) { gint64 max; gboolean res; GstFormat format; GstIterator *it; gboolean done; GValue item = { 0, }; /* parse format */ gst_query_parse_duration (query, &format, NULL); max = -1; res = TRUE; done = FALSE; it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder)); while (!done) { GstIteratorResult ires; ires = gst_iterator_next (it, &item); switch (ires) { case GST_ITERATOR_DONE: done = TRUE; break; case GST_ITERATOR_OK: { GstPad *pad = g_value_get_object (&item); gint64 duration; /* ask sink peer for duration */ res &= gst_pad_peer_query_duration (pad, format, &duration); /* take max from all valid return values */ if (res) { /* valid unknown length, stop searching */ if (duration == -1) { max = duration; done = TRUE; } /* else see if bigger than current max */ else if (duration > max) max = duration; } g_value_reset (&item); break; } case GST_ITERATOR_RESYNC: max = -1; res = TRUE; gst_iterator_resync (it); break; default: res = FALSE; done = TRUE; break; } } g_value_unset (&item); gst_iterator_free (it); if (res) { /* and store the max */ GST_DEBUG_OBJECT (adder, "Total duration in format %s: %" GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max)); gst_query_set_duration (query, format, max); } return res; } static gboolean gst_adder_query_latency (GstAdder * adder, GstQuery * query) { GstClockTime min, max; gboolean live; gboolean res; GstIterator *it; gboolean done; GValue item = { 0, }; res = TRUE; done = FALSE; live = FALSE; min = 0; max = GST_CLOCK_TIME_NONE; /* Take maximum of all latency values */ it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder)); while (!done) { GstIteratorResult ires; ires = gst_iterator_next (it, &item); switch (ires) { case GST_ITERATOR_DONE: done = TRUE; break; case GST_ITERATOR_OK: { GstPad *pad = g_value_get_object (&item); GstQuery *peerquery; GstClockTime min_cur, max_cur; gboolean live_cur; peerquery = gst_query_new_latency (); /* Ask peer for latency */ res &= gst_pad_peer_query (pad, peerquery); /* take max from all valid return values */ if (res) { gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur); if (live_cur) { if (min_cur > min) min = min_cur; if (max == GST_CLOCK_TIME_NONE) max = max_cur; else if (max_cur < max) max = max_cur; live = TRUE; } } gst_query_unref (peerquery); g_value_reset (&item); break; } case GST_ITERATOR_RESYNC: live = FALSE; min = 0; max = GST_CLOCK_TIME_NONE; res = TRUE; gst_iterator_resync (it); break; default: res = FALSE; done = TRUE; break; } } g_value_unset (&item); gst_iterator_free (it); if (res) { /* store the results */ GST_DEBUG_OBJECT (adder, "Calculated total latency: live %s, min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, (live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max)); gst_query_set_latency (query, live, min, max); } return res; } static gboolean gst_adder_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstAdder *adder = GST_ADDER (parent); gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { GstFormat format; gst_query_parse_position (query, &format, NULL); switch (format) { case GST_FORMAT_TIME: /* FIXME, bring to stream time, might be tricky */ gst_query_set_position (query, format, adder->segment.position); res = TRUE; break; case GST_FORMAT_DEFAULT: gst_query_set_position (query, format, adder->offset); res = TRUE; break; default: break; } break; } case GST_QUERY_DURATION: res = gst_adder_query_duration (adder, query); break; case GST_QUERY_LATENCY: res = gst_adder_query_latency (adder, query); break; default: /* FIXME, needs a custom query handler because we have multiple * sinkpads */ res = gst_pad_query_default (pad, parent, query); break; } return res; } /* event handling */ typedef struct { GstEvent *event; gboolean flush; } EventData; static gboolean forward_event_func (const GValue * val, GValue * ret, EventData * data) { GstPad *pad = g_value_get_object (val); GstEvent *event = data->event; GstPad *peer; gst_event_ref (event); GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event)); peer = gst_pad_get_peer (pad); /* collect pad might have been set flushing, * so bypass core checking that and send directly to peer */ if (!peer || !gst_pad_send_event (peer, event)) { if (!peer) gst_event_unref (event); GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.", event, GST_EVENT_TYPE_NAME (event)); /* quick hack to unflush the pads, ideally we need a way to just unflush * this single collect pad */ if (data->flush) gst_pad_send_event (pad, gst_event_new_flush_stop (TRUE)); } else { g_value_set_boolean (ret, TRUE); GST_LOG_OBJECT (pad, "Sent event %p (%s).", event, GST_EVENT_TYPE_NAME (event)); } if (peer) gst_object_unref (peer); /* continue on other pads, even if one failed */ return TRUE; } /* forwards the event to all sinkpads, takes ownership of the * event * * Returns: TRUE if the event could be forwarded on all * sinkpads. */ static gboolean forward_event (GstAdder * adder, GstEvent * event, gboolean flush) { gboolean ret; GstIterator *it; GstIteratorResult ires; GValue vret = { 0 }; EventData data; GST_LOG_OBJECT (adder, "Forwarding event %p (%s)", event, GST_EVENT_TYPE_NAME (event)); data.event = event; data.flush = flush; g_value_init (&vret, G_TYPE_BOOLEAN); g_value_set_boolean (&vret, FALSE); it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder)); while (TRUE) { ires = gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func, &vret, &data); switch (ires) { case GST_ITERATOR_RESYNC: GST_WARNING ("resync"); gst_iterator_resync (it); g_value_set_boolean (&vret, TRUE); break; case GST_ITERATOR_OK: case GST_ITERATOR_DONE: ret = g_value_get_boolean (&vret); goto done; default: ret = FALSE; goto done; } } done: gst_iterator_free (it); GST_LOG_OBJECT (adder, "Forwarded event %p (%s), ret=%d", event, GST_EVENT_TYPE_NAME (event), ret); gst_event_unref (event); return ret; } static gboolean gst_adder_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstAdder *adder; gboolean result; adder = GST_ADDER (parent); GST_DEBUG_OBJECT (pad, "Got %s event on src pad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: { GstSeekFlags flags; gdouble rate; GstSeekType start_type, stop_type; gint64 start, stop; GstFormat seek_format, dest_format; gboolean flush; /* parse the seek parameters */ gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, &start, &stop_type, &stop); if ((start_type != GST_SEEK_TYPE_NONE) && (start_type != GST_SEEK_TYPE_SET)) { result = FALSE; GST_DEBUG_OBJECT (adder, "seeking failed, unhandled seek type for start: %d", start_type); goto done; } if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) { result = FALSE; GST_DEBUG_OBJECT (adder, "seeking failed, unhandled seek type for end: %d", stop_type); goto done; } dest_format = adder->segment.format; if (seek_format != dest_format) { result = FALSE; GST_DEBUG_OBJECT (adder, "seeking failed, unhandled seek format: %d", seek_format); goto done; } flush = (flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH; /* check if we are flushing */ if (flush) { /* flushing seek, start flush downstream, the flush will be done * when all pads received a FLUSH_STOP. * Make sure we accept nothing anymore and return WRONG_STATE. * We send a flush-start before, to ensure no streaming is done * as we need to take the stream lock. */ gst_pad_push_event (adder->srcpad, gst_event_new_flush_start ()); gst_collect_pads_set_flushing (adder->collect, TRUE); /* We can't send FLUSH_STOP here since upstream could start pushing data * after we unlock adder->collect. * We set flush_stop_pending to TRUE instead and send FLUSH_STOP after * forwarding the seek upstream or from gst_adder_collected, * whichever happens first. */ GST_COLLECT_PADS_STREAM_LOCK (adder->collect); adder->flush_stop_pending = TRUE; GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect); GST_DEBUG_OBJECT (adder, "mark pending flush stop event"); } GST_DEBUG_OBJECT (adder, "handling seek event: %" GST_PTR_FORMAT, event); /* now wait for the collected to be finished and mark a new * segment. After we have the lock, no collect function is running and no * new collect function will be called for as long as we're flushing. */ GST_COLLECT_PADS_STREAM_LOCK (adder->collect); /* clip position and update our segment */ if (adder->segment.stop != -1) { adder->segment.position = adder->segment.stop; } gst_segment_do_seek (&adder->segment, rate, seek_format, flags, start_type, start, stop_type, stop, NULL); if (flush) { /* Yes, we need to call _set_flushing again *WHEN* the streaming threads * have stopped so that the cookie gets properly updated. */ gst_collect_pads_set_flushing (adder->collect, TRUE); } GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect); GST_DEBUG_OBJECT (adder, "forwarding seek event: %" GST_PTR_FORMAT, event); GST_DEBUG_OBJECT (adder, "updated segment: %" GST_SEGMENT_FORMAT, &adder->segment); /* we're forwarding seek to all upstream peers and wait for one to reply * with a newsegment-event before we send a newsegment-event downstream */ g_atomic_int_set (&adder->new_segment_pending, TRUE); result = forward_event (adder, event, flush); if (!result) { /* seek failed. maybe source is a live source. */ GST_DEBUG_OBJECT (adder, "seeking failed"); } if (g_atomic_int_compare_and_exchange (&adder->flush_stop_pending, TRUE, FALSE)) { GST_DEBUG_OBJECT (adder, "pending flush stop"); if (!gst_pad_push_event (adder->srcpad, gst_event_new_flush_stop (TRUE))) { GST_WARNING_OBJECT (adder, "Sending flush stop event failed"); } } break; } case GST_EVENT_QOS: /* QoS might be tricky */ result = FALSE; gst_event_unref (event); break; case GST_EVENT_NAVIGATION: /* navigation is rather pointless. */ result = FALSE; gst_event_unref (event); break; default: /* just forward the rest for now */ GST_DEBUG_OBJECT (adder, "forward unhandled event: %s", GST_EVENT_TYPE_NAME (event)); result = forward_event (adder, event, FALSE); break; } done: return result; } static gboolean gst_adder_sink_event (GstCollectPads * pads, GstCollectData * pad, GstEvent * event, gpointer user_data) { GstAdder *adder = GST_ADDER (user_data); gboolean res = TRUE, discard = FALSE; GST_DEBUG_OBJECT (pad->pad, "Got %s event on sink pad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); res = gst_adder_setcaps (adder, pad->pad, caps); gst_event_unref (event); event = NULL; break; } case GST_EVENT_FLUSH_START: /* ensure that we will send a flush stop */ res = gst_collect_pads_event_default (pads, pad, event, discard); event = NULL; GST_COLLECT_PADS_STREAM_LOCK (adder->collect); adder->flush_stop_pending = TRUE; GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect); break; case GST_EVENT_FLUSH_STOP: /* we received a flush-stop. We will only forward it when * flush_stop_pending is set, and we will unset it then. */ g_atomic_int_set (&adder->new_segment_pending, TRUE); GST_COLLECT_PADS_STREAM_LOCK (adder->collect); if (adder->flush_stop_pending) { GST_DEBUG_OBJECT (pad->pad, "forwarding flush stop"); res = gst_collect_pads_event_default (pads, pad, event, discard); adder->flush_stop_pending = FALSE; event = NULL; } else { discard = TRUE; GST_DEBUG_OBJECT (pad->pad, "eating flush stop"); } GST_COLLECT_PADS_STREAM_UNLOCK (adder->collect); /* Clear pending tags */ if (adder->pending_events) { g_list_foreach (adder->pending_events, (GFunc) gst_event_unref, NULL); g_list_free (adder->pending_events); adder->pending_events = NULL; } break; case GST_EVENT_TAG: /* collect tags here so we can push them out when we collect data */ adder->pending_events = g_list_append (adder->pending_events, event); event = NULL; break; case GST_EVENT_SEGMENT:{ const GstSegment *segment; gst_event_parse_segment (event, &segment); if (segment->rate != adder->segment.rate) { GST_ERROR_OBJECT (pad->pad, "Got segment event with wrong rate %lf, expected %lf", segment->rate, adder->segment.rate); res = FALSE; gst_event_unref (event); event = NULL; } discard = TRUE; break; } default: break; } if (G_LIKELY (event)) return gst_collect_pads_event_default (pads, pad, event, discard); else return res; } static void gst_adder_class_init (GstAdderClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_adder_set_property; gobject_class->get_property = gst_adder_get_property; gobject_class->dispose = gst_adder_dispose; g_object_class_install_property (gobject_class, PROP_FILTER_CAPS, g_param_spec_boxed ("caps", "Target caps", "Set target format for mixing (NULL means ANY). " "Setting this property takes a reference to the supplied GstCaps " "object.", GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_adder_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&gst_adder_sink_template)); gst_element_class_set_static_metadata (gstelement_class, "Adder", "Generic/Audio", "Add N audio channels together", "Thomas Vander Stichele "); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_adder_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_adder_release_pad); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_adder_change_state); } static void gst_adder_init (GstAdder * adder) { GstPadTemplate *template; template = gst_static_pad_template_get (&gst_adder_src_template); adder->srcpad = gst_pad_new_from_template (template, "src"); gst_object_unref (template); gst_pad_set_query_function (adder->srcpad, GST_DEBUG_FUNCPTR (gst_adder_src_query)); gst_pad_set_event_function (adder->srcpad, GST_DEBUG_FUNCPTR (gst_adder_src_event)); GST_PAD_SET_PROXY_CAPS (adder->srcpad); gst_element_add_pad (GST_ELEMENT (adder), adder->srcpad); adder->current_caps = NULL; gst_audio_info_init (&adder->info); adder->padcount = 0; adder->filter_caps = NULL; /* keep track of the sinkpads requested */ adder->collect = gst_collect_pads_new (); gst_collect_pads_set_function (adder->collect, GST_DEBUG_FUNCPTR (gst_adder_collected), adder); gst_collect_pads_set_clip_function (adder->collect, GST_DEBUG_FUNCPTR (gst_adder_do_clip), adder); gst_collect_pads_set_event_function (adder->collect, GST_DEBUG_FUNCPTR (gst_adder_sink_event), adder); gst_collect_pads_set_query_function (adder->collect, GST_DEBUG_FUNCPTR (gst_adder_sink_query), adder); } static void gst_adder_dispose (GObject * object) { GstAdder *adder = GST_ADDER (object); if (adder->collect) { gst_object_unref (adder->collect); adder->collect = NULL; } gst_caps_replace (&adder->filter_caps, NULL); gst_caps_replace (&adder->current_caps, NULL); if (adder->pending_events) { g_list_foreach (adder->pending_events, (GFunc) gst_event_unref, NULL); g_list_free (adder->pending_events); adder->pending_events = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_adder_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAdder *adder = GST_ADDER (object); switch (prop_id) { case PROP_FILTER_CAPS:{ GstCaps *new_caps = NULL; GstCaps *old_caps; const GstCaps *new_caps_val = gst_value_get_caps (value); if (new_caps_val != NULL) { new_caps = (GstCaps *) new_caps_val; gst_caps_ref (new_caps); } GST_OBJECT_LOCK (adder); old_caps = adder->filter_caps; adder->filter_caps = new_caps; GST_OBJECT_UNLOCK (adder); if (old_caps) gst_caps_unref (old_caps); GST_DEBUG_OBJECT (adder, "set new caps %" GST_PTR_FORMAT, new_caps); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_adder_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAdder *adder = GST_ADDER (object); switch (prop_id) { case PROP_FILTER_CAPS: GST_OBJECT_LOCK (adder); gst_value_set_caps (value, adder->filter_caps); GST_OBJECT_UNLOCK (adder); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstPad * gst_adder_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * unused, const GstCaps * caps) { gchar *name; GstAdder *adder; GstPad *newpad; gint padcount; if (templ->direction != GST_PAD_SINK) goto not_sink; adder = GST_ADDER (element); /* increment pad counter */ padcount = g_atomic_int_add (&adder->padcount, 1); name = g_strdup_printf ("sink_%u", padcount); newpad = g_object_new (GST_TYPE_ADDER_PAD, "name", name, "direction", templ->direction, "template", templ, NULL); GST_DEBUG_OBJECT (adder, "request new pad %s", name); g_free (name); gst_collect_pads_add_pad (adder->collect, newpad, sizeof (GstCollectData), NULL, TRUE); /* takes ownership of the pad */ if (!gst_element_add_pad (GST_ELEMENT (adder), newpad)) goto could_not_add; gst_child_proxy_child_added (GST_CHILD_PROXY (adder), G_OBJECT (newpad), GST_OBJECT_NAME (newpad)); return newpad; /* errors */ not_sink: { g_warning ("gstadder: request new pad that is not a SINK pad\n"); return NULL; } could_not_add: { GST_DEBUG_OBJECT (adder, "could not add pad"); gst_collect_pads_remove_pad (adder->collect, newpad); gst_object_unref (newpad); return NULL; } } static void gst_adder_release_pad (GstElement * element, GstPad * pad) { GstAdder *adder; adder = GST_ADDER (element); GST_DEBUG_OBJECT (adder, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad)); gst_child_proxy_child_removed (GST_CHILD_PROXY (adder), G_OBJECT (pad), GST_OBJECT_NAME (pad)); if (adder->collect) gst_collect_pads_remove_pad (adder->collect, pad); gst_element_remove_pad (element, pad); } static GstFlowReturn gst_adder_do_clip (GstCollectPads * pads, GstCollectData * data, GstBuffer * buffer, GstBuffer ** out, gpointer user_data) { GstAdder *adder = GST_ADDER (user_data); gint rate, bpf; rate = GST_AUDIO_INFO_RATE (&adder->info); bpf = GST_AUDIO_INFO_BPF (&adder->info); buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf); *out = buffer; return GST_FLOW_OK; } static GstFlowReturn gst_adder_collected (GstCollectPads * pads, gpointer user_data) { /* * combine streams by adding data values * basic algorithm : * - this function is called when all pads have a buffer * - get available bytes on all pads. * - repeat for each input pad : * - read available bytes, copy or add to target buffer * - if there's an EOS event, remove the input channel * - push out the output buffer * * todo: * - would be nice to have a mixing mode, where instead of adding we mix * - for float we could downscale after collect loop * - for int we need to downscale each input to avoid clipping or * mix into a temp (float) buffer and scale afterwards as well */ GstAdder *adder; GSList *collected, *next = NULL; GstFlowReturn ret; GstBuffer *outbuf = NULL, *gapbuf = NULL; GstMapInfo outmap = { NULL }; guint outsize; gint64 next_offset; gint64 next_timestamp; gint rate, bps, bpf; gboolean had_mute = FALSE; gboolean is_eos = TRUE; adder = GST_ADDER (user_data); /* this is fatal */ if (G_UNLIKELY (adder->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) goto not_negotiated; if (adder->flush_stop_pending) { GST_INFO_OBJECT (adder->srcpad, "send pending flush stop event"); if (!gst_pad_push_event (adder->srcpad, gst_event_new_flush_stop (TRUE))) { GST_WARNING_OBJECT (adder->srcpad, "Sending flush stop event failed"); } adder->flush_stop_pending = FALSE; } if (adder->send_stream_start) { gchar s_id[32]; GstEvent *event; GST_INFO_OBJECT (adder->srcpad, "send pending stream start event"); /* FIXME: create id based on input ids, we can't use * gst_pad_create_stream_id() though as that only handles 0..1 sink-pad */ g_snprintf (s_id, sizeof (s_id), "adder-%08x", g_random_int ()); event = gst_event_new_stream_start (s_id); gst_event_set_group_id (event, gst_util_group_id_next ()); if (!gst_pad_push_event (adder->srcpad, event)) { GST_WARNING_OBJECT (adder->srcpad, "Sending stream start event failed"); } adder->send_stream_start = FALSE; } if (adder->send_caps) { GstEvent *caps_event; caps_event = gst_event_new_caps (adder->current_caps); GST_INFO_OBJECT (adder->srcpad, "send pending caps event %" GST_PTR_FORMAT, caps_event); if (!gst_pad_push_event (adder->srcpad, caps_event)) { GST_WARNING_OBJECT (adder->srcpad, "Sending caps event failed"); } adder->send_caps = FALSE; } rate = GST_AUDIO_INFO_RATE (&adder->info); bps = GST_AUDIO_INFO_BPS (&adder->info); bpf = GST_AUDIO_INFO_BPF (&adder->info); if (g_atomic_int_compare_and_exchange (&adder->new_segment_pending, TRUE, FALSE)) { GstEvent *event; /* * When seeking we set the start and stop positions as given in the seek * event. We also adjust offset & timestamp accordingly. * This basically ignores all newsegments sent by upstream. */ event = gst_event_new_segment (&adder->segment); if (adder->segment.rate > 0.0) { adder->segment.position = adder->segment.start; } else { adder->segment.position = adder->segment.stop; } adder->offset = gst_util_uint64_scale (adder->segment.position, rate, GST_SECOND); GST_INFO_OBJECT (adder->srcpad, "sending pending new segment event %" GST_SEGMENT_FORMAT, &adder->segment); if (event) { if (!gst_pad_push_event (adder->srcpad, event)) { GST_WARNING_OBJECT (adder->srcpad, "Sending new segment event failed"); } } else { GST_WARNING_OBJECT (adder->srcpad, "Creating new segment event for " "start:%" G_GINT64_FORMAT " end:%" G_GINT64_FORMAT " failed", adder->segment.start, adder->segment.stop); } } /* get available bytes for reading, this can be 0 which could mean empty * buffers or EOS, which we will catch when we loop over the pads. */ outsize = gst_collect_pads_available (pads); GST_LOG_OBJECT (adder, "starting to cycle through channels, %d bytes available (bps = %d, bpf = %d)", outsize, bps, bpf); for (collected = pads->data; collected; collected = next) { GstCollectData *collect_data; GstBuffer *inbuf; gboolean is_gap; GstAdderPad *pad; GstClockTime timestamp, stream_time; /* take next to see if this is the last collectdata */ next = g_slist_next (collected); collect_data = (GstCollectData *) collected->data; pad = GST_ADDER_PAD (collect_data->pad); /* get a buffer of size bytes, if we get a buffer, it is at least outsize * bytes big. */ inbuf = gst_collect_pads_take_buffer (pads, collect_data, outsize); if (!GST_COLLECT_PADS_STATE_IS_SET (collect_data, GST_COLLECT_PADS_STATE_EOS)) is_eos = FALSE; /* NULL means EOS or an empty buffer so we still need to flush in * case of an empty buffer. */ if (inbuf == NULL) { GST_LOG_OBJECT (adder, "channel %p: no bytes available", collect_data); continue; } timestamp = GST_BUFFER_TIMESTAMP (inbuf); stream_time = gst_segment_to_stream_time (&collect_data->segment, GST_FORMAT_TIME, timestamp); /* sync object properties on stream time */ if (GST_CLOCK_TIME_IS_VALID (stream_time)) gst_object_sync_values (GST_OBJECT (pad), stream_time); GST_OBJECT_LOCK (pad); if (pad->mute || pad->volume < G_MINDOUBLE) { had_mute = TRUE; GST_DEBUG_OBJECT (adder, "channel %p: skipping muted pad", collect_data); gst_buffer_unref (inbuf); GST_OBJECT_UNLOCK (pad); continue; } is_gap = GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP); /* Try to make an output buffer */ if (outbuf == NULL) { /* if this is a gap buffer but we have some more pads to check, skip it. * If we are at the last buffer, take it, regardless if it is a GAP * buffer or not. */ if (is_gap && next) { GST_DEBUG_OBJECT (adder, "skipping, non-last GAP buffer"); /* we keep the GAP buffer, if we don't have anymore buffers (all pads * EOS, we can use this one as the output buffer. */ if (gapbuf == NULL) gapbuf = inbuf; else gst_buffer_unref (inbuf); GST_OBJECT_UNLOCK (pad); continue; } GST_LOG_OBJECT (adder, "channel %p: preparing output buffer of %d bytes", collect_data, outsize); /* make data and metadata writable, can simply return the inbuf when we * are the only one referencing this buffer. If this is the last (and * only) GAP buffer, it will automatically copy the GAP flag. */ outbuf = gst_buffer_make_writable (inbuf); gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE); if (pad->volume != 1.0) { switch (adder->info.finfo->format) { case GST_AUDIO_FORMAT_U8: adder_orc_volume_u8 ((gpointer) outmap.data, pad->volume_i8, outmap.size / bps); break; case GST_AUDIO_FORMAT_S8: adder_orc_volume_s8 ((gpointer) outmap.data, pad->volume_i8, outmap.size / bps); break; case GST_AUDIO_FORMAT_U16: adder_orc_volume_u16 ((gpointer) outmap.data, pad->volume_i16, outmap.size / bps); break; case GST_AUDIO_FORMAT_S16: adder_orc_volume_s16 ((gpointer) outmap.data, pad->volume_i16, outmap.size / bps); break; case GST_AUDIO_FORMAT_U32: adder_orc_volume_u32 ((gpointer) outmap.data, pad->volume_i32, outmap.size / bps); break; case GST_AUDIO_FORMAT_S32: adder_orc_volume_s32 ((gpointer) outmap.data, pad->volume_i32, outmap.size / bps); break; case GST_AUDIO_FORMAT_F32: adder_orc_volume_f32 ((gpointer) outmap.data, pad->volume, outmap.size / bps); break; case GST_AUDIO_FORMAT_F64: adder_orc_volume_f64 ((gpointer) outmap.data, pad->volume, outmap.size / bps); break; default: g_assert_not_reached (); break; } } } else { if (!is_gap) { /* we had a previous output buffer, mix this non-GAP buffer */ GstMapInfo inmap; gst_buffer_map (inbuf, &inmap, GST_MAP_READ); /* all buffers should have outsize, there are no short buffers because we * asked for the max size above */ g_assert (inmap.size == outmap.size); GST_LOG_OBJECT (adder, "channel %p: mixing %" G_GSIZE_FORMAT " bytes" " from data %p", collect_data, inmap.size, inmap.data); /* further buffers, need to add them */ if (pad->volume == 1.0) { switch (adder->info.finfo->format) { case GST_AUDIO_FORMAT_U8: adder_orc_add_u8 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_S8: adder_orc_add_s8 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_U16: adder_orc_add_u16 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_S16: adder_orc_add_s16 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_U32: adder_orc_add_u32 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_S32: adder_orc_add_s32 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_F32: adder_orc_add_f32 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; case GST_AUDIO_FORMAT_F64: adder_orc_add_f64 ((gpointer) outmap.data, (gpointer) inmap.data, inmap.size / bps); break; default: g_assert_not_reached (); break; } } else { switch (adder->info.finfo->format) { case GST_AUDIO_FORMAT_U8: adder_orc_add_volume_u8 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume_i8, inmap.size / bps); break; case GST_AUDIO_FORMAT_S8: adder_orc_add_volume_s8 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume_i8, inmap.size / bps); break; case GST_AUDIO_FORMAT_U16: adder_orc_add_volume_u16 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume_i16, inmap.size / bps); break; case GST_AUDIO_FORMAT_S16: adder_orc_add_volume_s16 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume_i16, inmap.size / bps); break; case GST_AUDIO_FORMAT_U32: adder_orc_add_volume_u32 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume_i32, inmap.size / bps); break; case GST_AUDIO_FORMAT_S32: adder_orc_add_volume_s32 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume_i32, inmap.size / bps); break; case GST_AUDIO_FORMAT_F32: adder_orc_add_volume_f32 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume, inmap.size / bps); break; case GST_AUDIO_FORMAT_F64: adder_orc_add_volume_f64 ((gpointer) outmap.data, (gpointer) inmap.data, pad->volume, inmap.size / bps); break; default: g_assert_not_reached (); break; } } gst_buffer_unmap (inbuf, &inmap); } else { /* skip gap buffer */ GST_LOG_OBJECT (adder, "channel %p: skipping GAP buffer", collect_data); } gst_buffer_unref (inbuf); } GST_OBJECT_UNLOCK (pad); } if (outbuf) gst_buffer_unmap (outbuf, &outmap); if (is_eos) goto eos; if (outbuf == NULL) { /* no output buffer, reuse one of the GAP buffers then if we have one */ if (gapbuf) { GST_LOG_OBJECT (adder, "reusing GAP buffer %p", gapbuf); outbuf = gapbuf; } else if (had_mute) { GstMapInfo map; /* Means we had all pads muted, create some silence */ outbuf = gst_buffer_new_allocate (NULL, outsize, NULL); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); gst_audio_format_fill_silence (adder->info.finfo, map.data, outsize); gst_buffer_unmap (outbuf, &map); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); } else { /* assume EOS otherwise, this should not happen, really */ goto eos; } } else if (gapbuf) { /* we had an output buffer, unref the gapbuffer we kept */ gst_buffer_unref (gapbuf); } if (G_UNLIKELY (adder->pending_events)) { GList *tmp = adder->pending_events; while (tmp) { GstEvent *ev = (GstEvent *) tmp->data; gst_pad_push_event (adder->srcpad, ev); tmp = g_list_next (tmp); } g_list_free (adder->pending_events); adder->pending_events = NULL; } /* for the next timestamp, use the sample counter, which will * never accumulate rounding errors */ if (adder->segment.rate > 0.0) { next_offset = adder->offset + outsize / bpf; } else { next_offset = adder->offset - outsize / bpf; } next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate); /* set timestamps on the output buffer */ GST_BUFFER_DTS (outbuf) = GST_CLOCK_TIME_NONE; if (adder->segment.rate > 0.0) { GST_BUFFER_PTS (outbuf) = adder->segment.position; GST_BUFFER_OFFSET (outbuf) = adder->offset; GST_BUFFER_OFFSET_END (outbuf) = next_offset; GST_BUFFER_DURATION (outbuf) = next_timestamp - adder->segment.position; } else { GST_BUFFER_PTS (outbuf) = next_timestamp; GST_BUFFER_OFFSET (outbuf) = next_offset; GST_BUFFER_OFFSET_END (outbuf) = adder->offset; GST_BUFFER_DURATION (outbuf) = adder->segment.position - next_timestamp; } adder->offset = next_offset; adder->segment.position = next_timestamp; /* send it out */ GST_LOG_OBJECT (adder, "pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %" G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), GST_BUFFER_OFFSET (outbuf)); ret = gst_pad_push (adder->srcpad, outbuf); GST_LOG_OBJECT (adder, "pushed outbuf, result = %s", gst_flow_get_name (ret)); return ret; /* ERRORS */ not_negotiated: { GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL), ("Unknown data received, not negotiated")); return GST_FLOW_NOT_NEGOTIATED; } eos: { GST_DEBUG_OBJECT (adder, "no data available, must be EOS"); gst_pad_push_event (adder->srcpad, gst_event_new_eos ()); return GST_FLOW_EOS; } } static GstStateChangeReturn gst_adder_change_state (GstElement * element, GstStateChange transition) { GstAdder *adder; GstStateChangeReturn ret; adder = GST_ADDER (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: adder->offset = 0; adder->flush_stop_pending = FALSE; adder->new_segment_pending = TRUE; adder->send_stream_start = TRUE; adder->send_caps = TRUE; gst_caps_replace (&adder->current_caps, NULL); gst_segment_init (&adder->segment, GST_FORMAT_TIME); gst_collect_pads_start (adder->collect); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: /* need to unblock the collectpads before calling the * parent change_state so that streaming can finish */ gst_collect_pads_stop (adder->collect); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { default: break; } return ret; } /* GstChildProxy implementation */ static GObject * gst_adder_child_proxy_get_child_by_index (GstChildProxy * child_proxy, guint index) { GstAdder *adder = GST_ADDER (child_proxy); GObject *obj = NULL; GST_OBJECT_LOCK (adder); obj = g_list_nth_data (GST_ELEMENT_CAST (adder)->sinkpads, index); if (obj) gst_object_ref (obj); GST_OBJECT_UNLOCK (adder); return obj; } static guint gst_adder_child_proxy_get_children_count (GstChildProxy * child_proxy) { guint count = 0; GstAdder *adder = GST_ADDER (child_proxy); GST_OBJECT_LOCK (adder); count = GST_ELEMENT_CAST (adder)->numsinkpads; GST_OBJECT_UNLOCK (adder); GST_INFO_OBJECT (adder, "Children Count: %d", count); return count; } static void gst_adder_child_proxy_init (gpointer g_iface, gpointer iface_data) { GstChildProxyInterface *iface = g_iface; GST_INFO ("intializing child proxy interface"); iface->get_child_by_index = gst_adder_child_proxy_get_child_by_index; iface->get_children_count = gst_adder_child_proxy_get_children_count; } static gboolean plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "adder", 0, "audio channel mixing element"); if (!gst_element_register (plugin, "adder", GST_RANK_NONE, GST_TYPE_ADDER)) { return FALSE; } return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, adder, "Adds multiple streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)