--- /dev/null
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ * 2000 Wim Taymans <wtay@chello.be>
+ * 2005 Wim Taymans <wim@fluendo.com>
+ * 2007 Andy Wingo <wingo at pobox.com>
+ * 2008 Sebastian Dröge <slomo@circular-chaos.org>
+ * 2014 Collabora
+ * Olivier Crete <olivier.crete@collabora.com>
+ *
+ * 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
+ *
+ *
+ */
+
+/* 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 "gstaudiointerleave.h"
+#include <gst/audio/audio.h>
+
+#include <string.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);
+
+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 (s, "channels", G_TYPE_INT, channels, NULL);
+ else
+ gst_structure_set (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_qsort_with_data (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;
+}
+
+static void
+gst_audio_interleave_set_channel_positions (GstAudioInterleave * self,
+ GstStructure * s)
+{
+ guint64 channel_mask = 0;
+
+ GST_OBJECT_LOCK (self);
+
+ if (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 {
+ GST_WARNING_OBJECT (self, "Using NONE channel positions");
+ }
+ gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
+
+ GST_OBJECT_UNLOCK (self);
+}
+
+
+#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)
+{
+ GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self);
+
+ switch (GST_AUDIO_INFO_WIDTH (&aagg->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);
+ GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad);
+ GstAudioInfo info;
+ GValue *val;
+ guint channel;
+ GstCaps *srccaps;
+ GstStructure *s;
+ gboolean ret;
+
+ if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps))
+ goto cannot_change_caps;
+
+ if (!gst_audio_info_from_caps (&info, caps))
+ goto invalid_format;
+
+ if (aaggpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)
+ g_atomic_int_add (&self->configured_sinkpads_counter, 1);
+
+ gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
+ caps);
+
+ 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));
+ }
+
+ if (g_atomic_int_get (&self->configured_sinkpads_counter) < self->channels)
+ return TRUE;
+
+ srccaps = gst_caps_copy (caps);
+ s = gst_caps_get_structure (srccaps, 0);
+
+ gst_structure_remove_field (s, "channel-mask");
+
+ gst_structure_set (s, "channels", G_TYPE_INT, self->channels, "layout",
+ G_TYPE_STRING, "interleaved", NULL);
+ gst_audio_interleave_set_channel_positions (self, s);
+
+ ret = gst_audio_aggregator_set_src_caps (aagg, srccaps);
+
+ gst_caps_unref (srccaps);
+
+ if (!ret)
+ goto src_did_not_accept;
+
+ gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
+ caps);
+
+ gst_audio_interleave_set_process_function (self);
+
+ 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_caps_unref (sinkcaps);
+ }
+
+
+ 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_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't "
+ "change", self->sinkcaps);
+ return FALSE;
+ }
+src_did_not_accept:
+ {
+ GST_WARNING_OBJECT (self, "src did not accept setcaps()");
+ 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 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_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_audio_interleave_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_audio_interleave_sink_template));
+ gst_element_class_set_static_metadata (gstelement_class, "AudioInterleave",
+ "Generic/Audio",
+ "Mixes multiple audio streams",
+ "Olivier Crete <olivier.crete@collabora.com>");
+
+ 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->sinkpads_type = GST_TYPE_AUDIO_INTERLEAVE_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;
+
+ aagg_class->aggregate_one_buffer = gst_audio_interleave_aggregate_one_buffer;
+
+
+ /**
+ * GstInterleave: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));
+
+ /**
+ * GstInterleave: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));
+}
+
+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;
+ self->channels = self->channel_positions->n_values;
+ 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 channels, padnumber;
+ GValue val = { 0, };
+
+ /* FIXME: We ignore req_name, this is evil! */
+
+ padnumber = g_atomic_int_add (&self->padcounter, 1);
+ if (self->channel_positions_from_input)
+ channels = g_atomic_int_add (&self->channels, 1);
+ else
+ channels = padnumber;
+
+ 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 = channels;
+ 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 */
+ if (self->sinkcaps) {
+ GstCaps *srccaps;
+ GstStructure *s;
+
+ /* Take lock to make sure processing finishes first */
+
+ srccaps = gst_caps_copy (self->sinkcaps);
+ s = gst_caps_get_structure (srccaps, 0);
+
+ gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
+
+ gst_audio_interleave_set_channel_positions (self, s);
+
+ gst_audio_aggregator_set_src_caps (GST_AUDIO_AGGREGATOR (self), srccaps);
+ gst_caps_unref (srccaps);
+ }
+
+ 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);
+
+ g_atomic_int_add (&self->channels, -1);
+
+ if (gst_pad_has_current_caps (pad))
+ g_atomic_int_add (&self->configured_sinkpads_counter, -1);
+
+ 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--;
+ }
+
+
+ /* Update the src caps if we already have them */
+ if (self->sinkcaps) {
+ if (self->channels > 0) {
+ GstCaps *srccaps;
+ GstStructure *s;
+
+ srccaps = gst_caps_copy (self->sinkcaps);
+ s = gst_caps_get_structure (srccaps, 0);
+
+ gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
+ gst_audio_interleave_set_channel_positions (self, s);
+
+ GST_OBJECT_UNLOCK (self);
+
+ gst_audio_aggregator_set_src_caps (GST_AUDIO_AGGREGATOR (self), srccaps);
+ gst_caps_unref (srccaps);
+ } else {
+ gst_caps_replace (&self->sinkcaps, NULL);
+ GST_OBJECT_UNLOCK (self);
+ }
+ } else {
+ 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;
+ guint8 *outdata;
+
+ out_width = GST_AUDIO_INFO_WIDTH (&aagg->info) / 8;
+ in_bpf = GST_AUDIO_INFO_BPF (&aaggpad->info);
+ out_bpf = GST_AUDIO_INFO_BPF (&aagg->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, self->channels,
+ out_offset * out_bpf, in_offset * in_bpf);
+
+ outdata = outmap.data + (out_offset * out_bpf) +
+ (out_width * self->default_channels_ordering_map[pad->channel]);
+
+
+ self->func (outdata, inmap.data + (in_offset * in_bpf), self->channels,
+ num_frames);
+
+
+ gst_buffer_unmap (inbuf, &inmap);
+ gst_buffer_unmap (outbuf, &outmap);
+
+ 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 ("intializing 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;
+}
projects / platform / upstream / gstreamer.git / commitdiff