/* 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
* @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 "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;
}
/* 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 (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 <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->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;
/**
* 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;
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 ("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;
}