/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000 Wim Taymans * 2005 Wim Taymans * 2007 Andy Wingo * 2008 Sebastian Dröge * 2014 Collabora * Olivier Crete * * gstaudiointerleave.c: audiointerleave 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-audiointerleave * @title: audiointerleave * */ /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray * with newer GLib versions (>= 2.31.0) */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstaudiomixerelements.h" #include "gstaudiointerleave.h" #include "gst/glib-compat-private.h" #define GST_CAT_DEFAULT gst_audio_interleave_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); enum { PROP_PAD_0, PROP_PAD_CHANNEL }; G_DEFINE_TYPE (GstAudioInterleavePad, gst_audio_interleave_pad, GST_TYPE_AUDIO_AGGREGATOR_PAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (audiointerleave, "audiointerleave", GST_RANK_NONE, GST_TYPE_AUDIO_INTERLEAVE, audiomixer_element_init (plugin)); static void gst_audio_interleave_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (object); switch (prop_id) { case PROP_PAD_CHANNEL: g_value_set_uint (value, pad->channel); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_interleave_pad_class_init (GstAudioInterleavePadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->get_property = gst_audio_interleave_pad_get_property; g_object_class_install_property (gobject_class, PROP_PAD_CHANNEL, g_param_spec_uint ("channel", "Channel number", "Number of the channel of this pad in the output", 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); } static void gst_audio_interleave_pad_init (GstAudioInterleavePad * pad) { } enum { PROP_0, PROP_CHANNEL_POSITIONS, PROP_CHANNEL_POSITIONS_FROM_INPUT }; /* 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_audio_interleave_sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("audio/x-raw, " "rate = (int) [ 1, MAX ], " "channels = (int) 1, " "format = (string) " GST_AUDIO_FORMATS_ALL ", " "layout = (string) {non-interleaved, interleaved}") ); static GstStaticPadTemplate gst_audio_interleave_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], " "format = (string) " GST_AUDIO_FORMATS_ALL ", " "layout = (string) interleaved") ); static void gst_audio_interleave_child_proxy_init (gpointer g_iface, gpointer iface_data); #define gst_audio_interleave_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAudioInterleave, gst_audio_interleave, GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, gst_audio_interleave_child_proxy_init)); static void gst_audio_interleave_finalize (GObject * object); static void gst_audio_interleave_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_interleave_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_audio_interleave_setcaps (GstAudioInterleave * self, GstPad * pad, GstCaps * caps); static GstPad *gst_audio_interleave_request_new_pad (GstElement * element, GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps); static void gst_audio_interleave_release_pad (GstElement * element, GstPad * pad); static gboolean gst_audio_interleave_stop (GstAggregator * agg); static gboolean gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset, GstBuffer * outbuf, guint out_offset, guint num_samples); static void __remove_channels (GstCaps * caps) { GstStructure *s; gint i, size; size = gst_caps_get_size (caps); for (i = 0; i < size; i++) { s = gst_caps_get_structure (caps, i); gst_structure_remove_field (s, "channel-mask"); gst_structure_remove_field (s, "channels"); } } static void __set_channels (GstCaps * caps, gint channels) { GstStructure *s; gint i, size; size = gst_caps_get_size (caps); for (i = 0; i < size; i++) { s = gst_caps_get_structure (caps, i); if (channels > 0) gst_structure_set_static_str (s, "channels", G_TYPE_INT, channels, NULL); else gst_structure_set_static_str (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); } } /* 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_audio_interleave_sink_getcaps (GstAggregator * agg, GstPad * pad, GstCaps * filter) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); GstCaps *result = NULL, *peercaps, *sinkcaps; GST_OBJECT_LOCK (self); /* If we already have caps on one of the sink pads return them */ if (self->sinkcaps) result = gst_caps_copy (self->sinkcaps); GST_OBJECT_UNLOCK (self); if (result == NULL) { /* get the downstream possible caps */ peercaps = gst_pad_peer_query_caps (agg->srcpad, NULL); /* get the allowed caps on this sinkpad */ sinkcaps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); __remove_channels (sinkcaps); if (peercaps) { peercaps = gst_caps_make_writable (peercaps); __remove_channels (peercaps); /* if the peer has caps, intersect */ GST_DEBUG_OBJECT (pad, "intersecting peer and template caps"); result = gst_caps_intersect (peercaps, sinkcaps); gst_caps_unref (peercaps); gst_caps_unref (sinkcaps); } else { /* the peer has no caps (or there is no peer), just use the allowed caps * of this sinkpad. */ GST_DEBUG_OBJECT (pad, "no peer caps, using sinkcaps"); result = sinkcaps; } __set_channels (result, 1); } if (filter != NULL) { GstCaps *caps = result; GST_LOG_OBJECT (pad, "intersecting filter caps %" GST_PTR_FORMAT " with " "preliminary result %" GST_PTR_FORMAT, filter, caps); result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); } GST_DEBUG_OBJECT (pad, "Returning caps %" GST_PTR_FORMAT, result); return result; } static gboolean gst_audio_interleave_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad, GstQuery * query) { gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_audio_interleave_sink_getcaps (agg, GST_PAD (aggpad), filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: res = GST_AGGREGATOR_CLASS (parent_class)->sink_query (agg, aggpad, query); break; } return res; } static gint compare_positions (gconstpointer a, gconstpointer b, gpointer user_data) { const gint i = *(const gint *) a; const gint j = *(const gint *) b; const gint *pos = (const gint *) user_data; if (pos[i] < pos[j]) return -1; else if (pos[i] > pos[j]) return 1; else return 0; } static gboolean gst_audio_interleave_channel_positions_to_mask (GValueArray * positions, gint default_ordering_map[64], guint64 * mask) { gint i; guint channels; GstAudioChannelPosition *pos; gboolean ret; channels = positions->n_values; pos = g_new (GstAudioChannelPosition, channels); for (i = 0; i < channels; i++) { GValue *val; val = g_value_array_get_nth (positions, i); pos[i] = g_value_get_enum (val); } /* sort the default ordering map according to the position order */ for (i = 0; i < channels; i++) { default_ordering_map[i] = i; } g_sort_array (default_ordering_map, channels, sizeof (*default_ordering_map), compare_positions, pos); ret = gst_audio_channel_positions_to_mask (pos, channels, FALSE, mask); g_free (pos); return ret; } /* Must be called with the object lock held */ static guint64 gst_audio_interleave_get_channel_mask (GstAudioInterleave * self) { guint64 channel_mask = 0; if (self->channels <= 64 && self->channel_positions != NULL && self->channels == self->channel_positions->n_values) { if (!gst_audio_interleave_channel_positions_to_mask (self->channel_positions, self->default_channels_ordering_map, &channel_mask)) { GST_WARNING_OBJECT (self, "Invalid channel positions, using NONE"); channel_mask = 0; } } else if (self->channels <= 64) { GST_WARNING_OBJECT (self, "Using NONE channel positions"); } return channel_mask; } #define MAKE_FUNC(type) \ static void interleave_##type (guint##type *out, guint##type *in, \ guint stride, guint nframes) \ { \ gint i; \ \ for (i = 0; i < nframes; i++) { \ *out = in[i]; \ out += stride; \ } \ } MAKE_FUNC (8); MAKE_FUNC (16); MAKE_FUNC (32); MAKE_FUNC (64); static void interleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes) { gint i; for (i = 0; i < nframes; i++) { memcpy (out, in, 3); out += stride * 3; in += 3; } } static void gst_audio_interleave_set_process_function (GstAudioInterleave * self, GstAudioInfo * info) { switch (GST_AUDIO_INFO_WIDTH (info)) { case 8: self->func = (GstInterleaveFunc) interleave_8; break; case 16: self->func = (GstInterleaveFunc) interleave_16; break; case 24: self->func = (GstInterleaveFunc) interleave_24; break; case 32: self->func = (GstInterleaveFunc) interleave_32; break; case 64: self->func = (GstInterleaveFunc) interleave_64; break; default: g_assert_not_reached (); break; } } /* 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_audio_interleave_setcaps (GstAudioInterleave * self, GstPad * pad, GstCaps * caps) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self); GstAudioInfo info; GValue *val; guint channel; gboolean new = FALSE; if (!gst_audio_info_from_caps (&info, caps)) goto invalid_format; GST_OBJECT_LOCK (self); if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps)) goto cannot_change_caps; if (!self->sinkcaps) { GstCaps *sinkcaps = gst_caps_copy (caps); GstStructure *s = gst_caps_get_structure (sinkcaps, 0); gst_structure_remove_field (s, "channel-mask"); GST_DEBUG_OBJECT (self, "setting sinkcaps %" GST_PTR_FORMAT, sinkcaps); gst_caps_replace (&self->sinkcaps, sinkcaps); gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (aagg)); gst_caps_unref (sinkcaps); new = TRUE; } if (self->channel_positions_from_input && GST_AUDIO_INFO_CHANNELS (&info) == 1) { channel = GST_AUDIO_INTERLEAVE_PAD (pad)->channel; val = g_value_array_get_nth (self->input_channel_positions, channel); g_value_set_enum (val, GST_AUDIO_INFO_POSITION (&info, 0)); } GST_OBJECT_UNLOCK (self); gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad), caps); if (!new) return TRUE; GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps); return TRUE; /* ERRORS */ invalid_format: { GST_WARNING_OBJECT (self, "invalid format set as caps: %" GST_PTR_FORMAT, caps); return FALSE; } cannot_change_caps: { GST_OBJECT_UNLOCK (self); GST_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't " "change", self->sinkcaps); return FALSE; } } static gboolean gst_audio_interleave_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad, GstEvent * event) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); gboolean res = TRUE; GST_DEBUG_OBJECT (aggpad, "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_audio_interleave_setcaps (self, GST_PAD_CAST (aggpad), caps); gst_event_unref (event); event = NULL; break; } default: break; } if (event != NULL) return GST_AGGREGATOR_CLASS (parent_class)->sink_event (agg, aggpad, event); return res; } static GstFlowReturn gst_audio_interleave_update_src_caps (GstAggregator * agg, GstCaps * caps, GstCaps ** ret) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); GstStructure *s; /* This means that either no caps have been set on the sink pad (if * sinkcaps is NULL) or that there is no sink pad (if channels == 0). */ GST_OBJECT_LOCK (self); if (self->sinkcaps == NULL || self->channels == 0) { GST_OBJECT_UNLOCK (self); return GST_FLOW_NOT_NEGOTIATED; } *ret = gst_caps_copy (self->sinkcaps); s = gst_caps_get_structure (*ret, 0); gst_structure_set_static_str (s, "channels", G_TYPE_INT, self->channels, "layout", G_TYPE_STRING, "interleaved", "channel-mask", GST_TYPE_BITMASK, gst_audio_interleave_get_channel_mask (self), NULL); GST_OBJECT_UNLOCK (self); return GST_FLOW_OK; } static gboolean gst_audio_interleave_negotiated_src_caps (GstAggregator * agg, GstCaps * caps) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad); if (!GST_AGGREGATOR_CLASS (parent_class)->negotiated_src_caps (agg, caps)) return FALSE; gst_audio_interleave_set_process_function (self, &srcpad->info); return TRUE; } static void gst_audio_interleave_class_init (GstAudioInterleaveClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; GstAggregatorClass *agg_class = (GstAggregatorClass *) klass; GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass; GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiointerleave", 0, "audio interleaving element"); gobject_class->set_property = gst_audio_interleave_set_property; gobject_class->get_property = gst_audio_interleave_get_property; gobject_class->finalize = gst_audio_interleave_finalize; gst_element_class_add_static_pad_template_with_gtype (gstelement_class, &gst_audio_interleave_src_template, GST_TYPE_AUDIO_AGGREGATOR_PAD); gst_element_class_add_static_pad_template_with_gtype (gstelement_class, &gst_audio_interleave_sink_template, GST_TYPE_AUDIO_INTERLEAVE_PAD); gst_element_class_set_static_metadata (gstelement_class, "AudioInterleave", "Generic/Audio", "Mixes multiple audio streams", "Olivier Crete "); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_audio_interleave_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_audio_interleave_release_pad); agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audio_interleave_sink_query); agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audio_interleave_sink_event); agg_class->stop = gst_audio_interleave_stop; agg_class->update_src_caps = gst_audio_interleave_update_src_caps; agg_class->negotiated_src_caps = gst_audio_interleave_negotiated_src_caps; aagg_class->aggregate_one_buffer = gst_audio_interleave_aggregate_one_buffer; /** * GstAudioInterleave:channel-positions: * * Channel positions: This property controls the channel positions * that are used on the src caps. The number of elements should be * the same as the number of sink pads and the array should contain * a valid list of channel positions. The n-th element of the array * is the position of the n-th sink pad. * * These channel positions will only be used if they're valid and the * number of elements is the same as the number of channels. If this * is not given a NONE layout will be used. * */ g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS, g_param_spec_value_array ("channel-positions", "Channel positions", "Channel positions used on the output", g_param_spec_enum ("channel-position", "Channel position", "Channel position of the n-th input", GST_TYPE_AUDIO_CHANNEL_POSITION, GST_AUDIO_CHANNEL_POSITION_NONE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS), G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstAudioInterleave:channel-positions-from-input: * * Channel positions from input: If this property is set to %TRUE the channel * positions will be taken from the input caps if valid channel positions for * the output can be constructed from them. If this is set to %TRUE setting the * channel-positions property overwrites this property again. * */ g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS_FROM_INPUT, g_param_spec_boolean ("channel-positions-from-input", "Channel positions from input", "Take channel positions from the input", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_INTERLEAVE_PAD, 0); } static void gst_audio_interleave_init (GstAudioInterleave * self) { self->input_channel_positions = g_value_array_new (0); self->channel_positions_from_input = TRUE; self->channel_positions = self->input_channel_positions; } static void gst_audio_interleave_finalize (GObject * object) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object); if (self->channel_positions && self->channel_positions != self->input_channel_positions) { g_value_array_free (self->channel_positions); self->channel_positions = NULL; } if (self->input_channel_positions) { g_value_array_free (self->input_channel_positions); self->input_channel_positions = NULL; } G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_audio_interleave_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object); switch (prop_id) { case PROP_CHANNEL_POSITIONS: g_return_if_fail ( ((GValueArray *) g_value_get_boxed (value))->n_values > 0); if (self->channel_positions && self->channel_positions != self->input_channel_positions) g_value_array_free (self->channel_positions); self->channel_positions = g_value_dup_boxed (value); self->channel_positions_from_input = FALSE; break; case PROP_CHANNEL_POSITIONS_FROM_INPUT: self->channel_positions_from_input = g_value_get_boolean (value); if (self->channel_positions_from_input) { if (self->channel_positions && self->channel_positions != self->input_channel_positions) g_value_array_free (self->channel_positions); self->channel_positions = self->input_channel_positions; } break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_interleave_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object); switch (prop_id) { case PROP_CHANNEL_POSITIONS: g_value_set_boxed (value, self->channel_positions); break; case PROP_CHANNEL_POSITIONS_FROM_INPUT: g_value_set_boolean (value, self->channel_positions_from_input); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean gst_audio_interleave_stop (GstAggregator * agg) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); if (!GST_AGGREGATOR_CLASS (parent_class)->stop (agg)) return FALSE; gst_caps_replace (&self->sinkcaps, NULL); return TRUE; } static GstPad * gst_audio_interleave_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * req_name, const GstCaps * caps) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (element); GstAudioInterleavePad *newpad; gchar *pad_name; gint channel, padnumber; GValue val = { 0, }; /* FIXME: We ignore req_name, this is evil! */ GST_OBJECT_LOCK (self); padnumber = g_atomic_int_add (&self->padcounter, 1); channel = self->channels++; if (!self->channel_positions_from_input) channel = padnumber; GST_OBJECT_UNLOCK (self); pad_name = g_strdup_printf ("sink_%u", padnumber); newpad = (GstAudioInterleavePad *) GST_ELEMENT_CLASS (parent_class)->request_new_pad (element, templ, pad_name, caps); g_free (pad_name); if (newpad == NULL) goto could_not_create; newpad->channel = channel; gst_pad_use_fixed_caps (GST_PAD (newpad)); gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (newpad), GST_OBJECT_NAME (newpad)); g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION); g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_NONE); self->input_channel_positions = g_value_array_append (self->input_channel_positions, &val); g_value_unset (&val); /* Update the src caps if we already have them */ gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (self)); return GST_PAD_CAST (newpad); could_not_create: { GST_DEBUG_OBJECT (element, "could not create/add pad"); return NULL; } } static void gst_audio_interleave_release_pad (GstElement * element, GstPad * pad) { GstAudioInterleave *self; gint position; GList *l; self = GST_AUDIO_INTERLEAVE (element); /* Take lock to make sure we're not changing this when processing buffers */ GST_OBJECT_LOCK (self); self->channels--; position = GST_AUDIO_INTERLEAVE_PAD (pad)->channel; g_value_array_remove (self->input_channel_positions, position); /* Update channel numbers */ /* Taken above, GST_OBJECT_LOCK (self); */ for (l = GST_ELEMENT_CAST (self)->sinkpads; l != NULL; l = l->next) { GstAudioInterleavePad *ipad = GST_AUDIO_INTERLEAVE_PAD (l->data); if (GST_AUDIO_INTERLEAVE_PAD (pad)->channel < ipad->channel) ipad->channel--; } gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (self)); GST_OBJECT_UNLOCK (self); GST_DEBUG_OBJECT (self, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad)); gst_child_proxy_child_removed (GST_CHILD_PROXY (self), G_OBJECT (pad), GST_OBJECT_NAME (pad)); GST_ELEMENT_CLASS (parent_class)->release_pad (element, pad); } /* Called with object lock and pad object lock held */ static gboolean gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset, GstBuffer * outbuf, guint out_offset, guint num_frames) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (aagg); GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (aaggpad); GstMapInfo inmap; GstMapInfo outmap; gint out_width, in_bpf, out_bpf, out_channels, channel; guint8 *outdata; GstAggregator *agg = GST_AGGREGATOR (aagg); GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad); GST_OBJECT_LOCK (aagg); GST_OBJECT_LOCK (aaggpad); out_width = GST_AUDIO_INFO_WIDTH (&srcpad->info) / 8; in_bpf = GST_AUDIO_INFO_BPF (&aaggpad->info); out_bpf = GST_AUDIO_INFO_BPF (&srcpad->info); out_channels = GST_AUDIO_INFO_CHANNELS (&srcpad->info); gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE); gst_buffer_map (inbuf, &inmap, GST_MAP_READ); GST_LOG_OBJECT (pad, "interleaves %u frames on channel %d/%d at offset %u" " from offset %u", num_frames, pad->channel, out_channels, out_offset * out_bpf, in_offset * in_bpf); if (self->channels > 64) { channel = pad->channel; } else { channel = self->default_channels_ordering_map[pad->channel]; } outdata = outmap.data + (out_offset * out_bpf) + (out_width * channel); self->func (outdata, inmap.data + (in_offset * in_bpf), out_channels, num_frames); gst_buffer_unmap (inbuf, &inmap); gst_buffer_unmap (outbuf, &outmap); GST_OBJECT_UNLOCK (aaggpad); GST_OBJECT_UNLOCK (aagg); return TRUE; } /* GstChildProxy implementation */ static GObject * gst_audio_interleave_child_proxy_get_child_by_index (GstChildProxy * child_proxy, guint index) { GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (child_proxy); GObject *obj = NULL; GST_OBJECT_LOCK (self); obj = g_list_nth_data (GST_ELEMENT_CAST (self)->sinkpads, index); if (obj) gst_object_ref (obj); GST_OBJECT_UNLOCK (self); return obj; } static guint gst_audio_interleave_child_proxy_get_children_count (GstChildProxy * child_proxy) { guint count = 0; GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (child_proxy); GST_OBJECT_LOCK (self); count = GST_ELEMENT_CAST (self)->numsinkpads; GST_OBJECT_UNLOCK (self); GST_INFO_OBJECT (self, "Children Count: %d", count); return count; } static void gst_audio_interleave_child_proxy_init (gpointer g_iface, gpointer iface_data) { GstChildProxyInterface *iface = g_iface; GST_INFO ("initializing child proxy interface"); iface->get_child_by_index = gst_audio_interleave_child_proxy_get_child_by_index; iface->get_children_count = gst_audio_interleave_child_proxy_get_children_count; }