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902 lines
26 KiB
C
902 lines
26 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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* 2005 Wim Taymans <wim@fluendo.com>
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* 2007 Andy Wingo <wingo at pobox.com>
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* 2008 Sebastian Dröge <slomo@circular-chaos.org>
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* 2014 Collabora
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* Olivier Crete <olivier.crete@collabora.com>
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*
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* gstaudiointerleave.c: audiointerleave element, N in, one out,
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* samples are added
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-audiointerleave
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* @title: audiointerleave
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*
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*/
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/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstaudiointerleave.h"
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#include <gst/audio/audio.h>
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#include <string.h>
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#define GST_CAT_DEFAULT gst_audio_interleave_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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enum
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{
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PROP_PAD_0,
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PROP_PAD_CHANNEL
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};
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G_DEFINE_TYPE (GstAudioInterleavePad, gst_audio_interleave_pad,
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GST_TYPE_AUDIO_AGGREGATOR_PAD);
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static void
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gst_audio_interleave_pad_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (object);
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switch (prop_id) {
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case PROP_PAD_CHANNEL:
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g_value_set_uint (value, pad->channel);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_interleave_pad_class_init (GstAudioInterleavePadClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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gobject_class->get_property = gst_audio_interleave_pad_get_property;
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g_object_class_install_property (gobject_class,
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PROP_PAD_CHANNEL,
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g_param_spec_uint ("channel",
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"Channel number",
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"Number of the channel of this pad in the output", 0, G_MAXUINT, 0,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_audio_interleave_pad_init (GstAudioInterleavePad * pad)
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{
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}
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enum
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{
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PROP_0,
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PROP_CHANNEL_POSITIONS,
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PROP_CHANNEL_POSITIONS_FROM_INPUT
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};
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/* elementfactory information */
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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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#define CAPS \
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GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
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", layout = (string) { interleaved, non-interleaved }"
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#else
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#define CAPS \
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GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
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", layout = (string) { interleaved, non-interleaved }"
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#endif
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static GstStaticPadTemplate gst_audio_interleave_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("audio/x-raw, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) 1, "
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"format = (string) " GST_AUDIO_FORMATS_ALL ", "
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"layout = (string) {non-interleaved, interleaved}")
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);
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static GstStaticPadTemplate gst_audio_interleave_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"rate = (int) [ 1, MAX ], "
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"channels = (int) [ 1, MAX ], "
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"format = (string) " GST_AUDIO_FORMATS_ALL ", "
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"layout = (string) interleaved")
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);
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static void gst_audio_interleave_child_proxy_init (gpointer g_iface,
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gpointer iface_data);
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#define gst_audio_interleave_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAudioInterleave, gst_audio_interleave,
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GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
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gst_audio_interleave_child_proxy_init));
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static void gst_audio_interleave_finalize (GObject * object);
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static void gst_audio_interleave_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_interleave_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audio_interleave_setcaps (GstAudioInterleave * self,
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GstPad * pad, GstCaps * caps);
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static GstPad *gst_audio_interleave_request_new_pad (GstElement * element,
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GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps);
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static void gst_audio_interleave_release_pad (GstElement * element,
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GstPad * pad);
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static gboolean gst_audio_interleave_stop (GstAggregator * agg);
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static gboolean
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gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg,
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GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
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GstBuffer * outbuf, guint out_offset, guint num_samples);
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static void
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__remove_channels (GstCaps * caps)
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{
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GstStructure *s;
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gint i, size;
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size = gst_caps_get_size (caps);
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for (i = 0; i < size; i++) {
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s = gst_caps_get_structure (caps, i);
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gst_structure_remove_field (s, "channel-mask");
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gst_structure_remove_field (s, "channels");
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}
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}
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static void
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__set_channels (GstCaps * caps, gint channels)
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{
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GstStructure *s;
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gint i, size;
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size = gst_caps_get_size (caps);
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for (i = 0; i < size; i++) {
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s = gst_caps_get_structure (caps, i);
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if (channels > 0)
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gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
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else
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gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
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}
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}
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/* we can only accept caps that we and downstream can handle.
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* if we have filtercaps set, use those to constrain the target caps.
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*/
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static GstCaps *
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gst_audio_interleave_sink_getcaps (GstAggregator * agg, GstPad * pad,
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GstCaps * filter)
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{
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GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
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GstCaps *result = NULL, *peercaps, *sinkcaps;
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GST_OBJECT_LOCK (self);
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/* If we already have caps on one of the sink pads return them */
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if (self->sinkcaps)
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result = gst_caps_copy (self->sinkcaps);
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GST_OBJECT_UNLOCK (self);
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if (result == NULL) {
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/* get the downstream possible caps */
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peercaps = gst_pad_peer_query_caps (agg->srcpad, NULL);
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/* get the allowed caps on this sinkpad */
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sinkcaps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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__remove_channels (sinkcaps);
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if (peercaps) {
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peercaps = gst_caps_make_writable (peercaps);
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__remove_channels (peercaps);
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/* if the peer has caps, intersect */
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GST_DEBUG_OBJECT (pad, "intersecting peer and template caps");
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result = gst_caps_intersect (peercaps, sinkcaps);
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gst_caps_unref (peercaps);
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gst_caps_unref (sinkcaps);
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} else {
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/* the peer has no caps (or there is no peer), just use the allowed caps
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* of this sinkpad. */
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GST_DEBUG_OBJECT (pad, "no peer caps, using sinkcaps");
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result = sinkcaps;
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}
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__set_channels (result, 1);
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}
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if (filter != NULL) {
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GstCaps *caps = result;
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GST_LOG_OBJECT (pad, "intersecting filter caps %" GST_PTR_FORMAT " with "
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"preliminary result %" GST_PTR_FORMAT, filter, caps);
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result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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}
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GST_DEBUG_OBJECT (pad, "Returning caps %" GST_PTR_FORMAT, result);
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return result;
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}
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static gboolean
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gst_audio_interleave_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad,
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GstQuery * query)
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{
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gboolean res = FALSE;
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_CAPS:
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{
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GstCaps *filter, *caps;
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gst_query_parse_caps (query, &filter);
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caps = gst_audio_interleave_sink_getcaps (agg, GST_PAD (aggpad), filter);
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gst_query_set_caps_result (query, caps);
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gst_caps_unref (caps);
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res = TRUE;
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break;
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}
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default:
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res =
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GST_AGGREGATOR_CLASS (parent_class)->sink_query (agg, aggpad, query);
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break;
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}
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return res;
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}
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static gint
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compare_positions (gconstpointer a, gconstpointer b, gpointer user_data)
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{
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const gint i = *(const gint *) a;
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const gint j = *(const gint *) b;
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const gint *pos = (const gint *) user_data;
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if (pos[i] < pos[j])
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return -1;
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else if (pos[i] > pos[j])
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return 1;
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else
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return 0;
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}
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static gboolean
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gst_audio_interleave_channel_positions_to_mask (GValueArray * positions,
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gint default_ordering_map[64], guint64 * mask)
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{
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gint i;
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guint channels;
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GstAudioChannelPosition *pos;
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gboolean ret;
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channels = positions->n_values;
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pos = g_new (GstAudioChannelPosition, channels);
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for (i = 0; i < channels; i++) {
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GValue *val;
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val = g_value_array_get_nth (positions, i);
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pos[i] = g_value_get_enum (val);
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}
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/* sort the default ordering map according to the position order */
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for (i = 0; i < channels; i++) {
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default_ordering_map[i] = i;
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}
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g_qsort_with_data (default_ordering_map, channels,
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sizeof (*default_ordering_map), compare_positions, pos);
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ret = gst_audio_channel_positions_to_mask (pos, channels, FALSE, mask);
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g_free (pos);
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return ret;
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}
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/* Must be called with the object lock held */
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static guint64
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gst_audio_interleave_get_channel_mask (GstAudioInterleave * self)
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{
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guint64 channel_mask = 0;
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if (self->channels <= 64 &&
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self->channel_positions != NULL &&
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self->channels == self->channel_positions->n_values) {
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if (!gst_audio_interleave_channel_positions_to_mask
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(self->channel_positions, self->default_channels_ordering_map,
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&channel_mask)) {
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GST_WARNING_OBJECT (self, "Invalid channel positions, using NONE");
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channel_mask = 0;
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}
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} else if (self->channels <= 64) {
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GST_WARNING_OBJECT (self, "Using NONE channel positions");
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}
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return channel_mask;
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}
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#define MAKE_FUNC(type) \
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static void interleave_##type (guint##type *out, guint##type *in, \
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guint stride, guint nframes) \
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{ \
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gint i; \
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\
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for (i = 0; i < nframes; i++) { \
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*out = in[i]; \
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out += stride; \
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} \
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}
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MAKE_FUNC (8);
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MAKE_FUNC (16);
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MAKE_FUNC (32);
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MAKE_FUNC (64);
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static void
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interleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes)
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{
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gint i;
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for (i = 0; i < nframes; i++) {
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memcpy (out, in, 3);
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out += stride * 3;
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in += 3;
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}
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}
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static void
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gst_audio_interleave_set_process_function (GstAudioInterleave * self,
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GstAudioInfo * info)
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{
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switch (GST_AUDIO_INFO_WIDTH (info)) {
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case 8:
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self->func = (GstInterleaveFunc) interleave_8;
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break;
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case 16:
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self->func = (GstInterleaveFunc) interleave_16;
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break;
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case 24:
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self->func = (GstInterleaveFunc) interleave_24;
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break;
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case 32:
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self->func = (GstInterleaveFunc) interleave_32;
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break;
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case 64:
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self->func = (GstInterleaveFunc) interleave_64;
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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}
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/* the first caps we receive on any of the sinkpads will define the caps for all
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* the other sinkpads because we can only mix streams with the same caps.
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*/
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static gboolean
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gst_audio_interleave_setcaps (GstAudioInterleave * self, GstPad * pad,
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GstCaps * caps)
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{
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GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self);
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GstAudioInfo info;
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GValue *val;
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guint channel;
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gboolean new = FALSE;
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if (!gst_audio_info_from_caps (&info, caps))
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goto invalid_format;
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GST_OBJECT_LOCK (self);
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if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps))
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goto cannot_change_caps;
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if (!self->sinkcaps) {
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GstCaps *sinkcaps = gst_caps_copy (caps);
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GstStructure *s = gst_caps_get_structure (sinkcaps, 0);
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gst_structure_remove_field (s, "channel-mask");
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GST_DEBUG_OBJECT (self, "setting sinkcaps %" GST_PTR_FORMAT, sinkcaps);
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gst_caps_replace (&self->sinkcaps, sinkcaps);
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gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (aagg));
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gst_caps_unref (sinkcaps);
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new = TRUE;
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}
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if (self->channel_positions_from_input
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&& GST_AUDIO_INFO_CHANNELS (&info) == 1) {
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channel = GST_AUDIO_INTERLEAVE_PAD (pad)->channel;
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val = g_value_array_get_nth (self->input_channel_positions, channel);
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g_value_set_enum (val, GST_AUDIO_INFO_POSITION (&info, 0));
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}
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GST_OBJECT_UNLOCK (self);
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gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
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caps);
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if (!new)
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return TRUE;
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GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
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return TRUE;
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/* ERRORS */
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invalid_format:
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{
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GST_WARNING_OBJECT (self, "invalid format set as caps: %" GST_PTR_FORMAT,
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caps);
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return FALSE;
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}
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cannot_change_caps:
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{
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GST_OBJECT_UNLOCK (self);
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GST_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't "
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"change", self->sinkcaps);
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return FALSE;
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}
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}
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static gboolean
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gst_audio_interleave_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
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GstEvent * event)
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{
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GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg);
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gboolean res = TRUE;
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GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad",
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GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CAPS:
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{
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GstCaps *caps;
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gst_event_parse_caps (event, &caps);
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res = gst_audio_interleave_setcaps (self, GST_PAD_CAST (aggpad), caps);
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gst_event_unref (event);
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event = NULL;
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break;
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}
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default:
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break;
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}
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if (event != NULL)
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return GST_AGGREGATOR_CLASS (parent_class)->sink_event (agg, aggpad, event);
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return res;
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}
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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);
|
|
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self);
|
|
|
|
if (!GST_AGGREGATOR_CLASS (parent_class)->negotiated_src_caps (agg, caps))
|
|
return FALSE;
|
|
|
|
gst_audio_interleave_set_process_function (self, &aagg->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 (gstelement_class,
|
|
&gst_audio_interleave_src_template);
|
|
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;
|
|
aagg_class->convert_buffer = NULL;
|
|
|
|
/**
|
|
* 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;
|
|
|
|
GST_OBJECT_LOCK (aagg);
|
|
GST_OBJECT_LOCK (aaggpad);
|
|
|
|
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);
|
|
out_channels = GST_AUDIO_INFO_CHANNELS (&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, 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;
|
|
}
|