/* 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>
*
* deinterleave.c: deinterleave samples
*
* 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.
*/
/* TODO:
* - handle changes in number of channels
* - handle changes in channel positions
* - better capsnego by using a buffer alloc function
* and passing downstream caps changes upstream there
*/
/**
* SECTION:element-deinterleave
* @see_also: interleave
*
* Splits one interleaved multichannel audio stream into many mono audio streams.
*
* This element handles all raw audio formats and supports changing the input caps as long as
* all downstream elements can handle the new caps and the number of channels and the channel
* positions stay the same. This restriction will be removed in later versions by adding or
* removing some source pads as required.
*
* In most cases a queue and an audioconvert element should be added after each source pad
* before further processing of the audio data.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch-1.0 filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw,channels=2 ! deinterleave name=d d.src_0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg d.src_1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg
* ]| Decodes an MP3 file and encodes the left and right channel into separate
* Ogg Vorbis files.
* |[
* gst-launch-1.0 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src_0 ! queue ! audioconvert ! i.sink_1 d.src_1 ! queue ! audioconvert ! i.sink_0
* ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and
* then interleaves the channels again to a WAV file with the channel with the
* channels exchanged.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/gst.h>
#include <string.h>
#include "deinterleave.h"
GST_DEBUG_CATEGORY_STATIC (gst_deinterleave_debug);
#define GST_CAT_DEFAULT gst_deinterleave_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_FORMATS_ALL ", "
"rate = (int) [ 1, MAX ], "
"channels = (int) 1, layout = (string) {non-interleaved, interleaved}"));
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_FORMATS_ALL ", "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], layout = (string) interleaved"));
#define MAKE_FUNC(type) \
static void deinterleave_##type (guint##type *out, guint##type *in, \
guint stride, guint nframes) \
{ \
gint i; \
\
for (i = 0; i < nframes; i++) { \
out[i] = *in; \
in += stride; \
} \
}
MAKE_FUNC (8);
MAKE_FUNC (16);
MAKE_FUNC (32);
MAKE_FUNC (64);
static void
deinterleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes)
{
gint i;
for (i = 0; i < nframes; i++) {
memcpy (out, in, 3);
out += 3;
in += stride * 3;
}
}
#define gst_deinterleave_parent_class parent_class
G_DEFINE_TYPE (GstDeinterleave, gst_deinterleave, GST_TYPE_ELEMENT);
enum
{
PROP_0,
PROP_KEEP_POSITIONS
};
static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer);
static gboolean gst_deinterleave_sink_setcaps (GstDeinterleave * self,
GstCaps * caps);
static GstStateChangeReturn
gst_deinterleave_change_state (GstElement * element, GstStateChange transition);
static gboolean gst_deinterleave_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_deinterleave_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_deinterleave_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static void gst_deinterleave_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_deinterleave_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static void
gst_deinterleave_finalize (GObject * obj)
{
GstDeinterleave *self = GST_DEINTERLEAVE (obj);
if (self->pending_events) {
g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL);
g_list_free (self->pending_events);
self->pending_events = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (obj);
}
static void
gst_deinterleave_class_init (GstDeinterleaveClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_deinterleave_debug, "deinterleave", 0,
"deinterleave element");
gst_element_class_set_static_metadata (gstelement_class,
"Audio deinterleaver", "Filter/Converter/Audio",
"Splits one interleaved multichannel audio stream into many mono audio streams",
"Andy Wingo <wingo at pobox.com>, " "Iain <iain@prettypeople.org>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gstelement_class->change_state = gst_deinterleave_change_state;
gobject_class->finalize = gst_deinterleave_finalize;
gobject_class->set_property = gst_deinterleave_set_property;
gobject_class->get_property = gst_deinterleave_get_property;
/**
* GstDeinterleave:keep-positions
*
* Keep positions: When enable the caps on the output buffers will
* contain the original channel positions. This can be used to correctly
* interleave the output again later but can also lead to unwanted effects
* if the output should be handled as Mono.
*
*/
g_object_class_install_property (gobject_class, PROP_KEEP_POSITIONS,
g_param_spec_boolean ("keep-positions", "Keep positions",
"Keep the original channel positions on the output buffers",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_deinterleave_init (GstDeinterleave * self)
{
self->keep_positions = FALSE;
self->func = NULL;
gst_audio_info_init (&self->audio_info);
/* Add sink pad */
self->sink = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (self->sink,
GST_DEBUG_FUNCPTR (gst_deinterleave_chain));
gst_pad_set_event_function (self->sink,
GST_DEBUG_FUNCPTR (gst_deinterleave_sink_event));
gst_pad_set_query_function (self->sink,
GST_DEBUG_FUNCPTR (gst_deinterleave_sink_query));
gst_element_add_pad (GST_ELEMENT (self), self->sink);
}
typedef struct
{
GstCaps *caps;
GstPad *pad;
} CopyStickyEventsData;
static gboolean
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
{
CopyStickyEventsData *data = user_data;
if (GST_EVENT_TYPE (*event) >= GST_EVENT_CAPS && data->caps) {
gst_pad_set_caps (data->pad, data->caps);
data->caps = NULL;
}
if (GST_EVENT_TYPE (*event) != GST_EVENT_CAPS)
gst_pad_push_event (data->pad, gst_event_ref (*event));
return TRUE;
}
static void
gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
{
GstPad *pad;
guint i;
for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&self->audio_info); i++) {
gchar *name = g_strdup_printf ("src_%u", i);
GstCaps *srccaps;
GstAudioInfo info;
GstAudioFormat format = GST_AUDIO_INFO_FORMAT (&self->audio_info);
gint rate = GST_AUDIO_INFO_RATE (&self->audio_info);
GstAudioChannelPosition position = GST_AUDIO_CHANNEL_POSITION_MONO;
CopyStickyEventsData data;
/* Set channel position if we know it */
if (self->keep_positions)
position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, format, rate, 1, &position);
srccaps = gst_audio_info_to_caps (&info);
pad = gst_pad_new_from_static_template (&src_template, name);
g_free (name);
gst_pad_use_fixed_caps (pad);
gst_pad_set_query_function (pad,
GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
gst_pad_set_active (pad, TRUE);
data.pad = pad;
data.caps = srccaps;
gst_pad_sticky_events_foreach (self->sink, copy_sticky_events, &data);
if (data.caps)
gst_pad_set_caps (pad, data.caps);
gst_element_add_pad (GST_ELEMENT (self), pad);
self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
gst_caps_unref (srccaps);
}
gst_element_no_more_pads (GST_ELEMENT (self));
self->srcpads = g_list_reverse (self->srcpads);
}
static gboolean
gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps)
{
GList *l;
gint i;
gboolean ret = TRUE;
for (l = self->srcpads, i = 0; l; l = l->next, i++) {
GstPad *pad = GST_PAD (l->data);
GstCaps *srccaps;
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
ret = FALSE;
continue;
}
if (self->keep_positions)
GST_AUDIO_INFO_POSITION (&info, 0) =
GST_AUDIO_INFO_POSITION (&self->audio_info, i);
srccaps = gst_audio_info_to_caps (&info);
gst_pad_set_caps (pad, srccaps);
gst_caps_unref (srccaps);
}
return ret;
}
static void
gst_deinterleave_remove_pads (GstDeinterleave * self)
{
GList *l;
GST_INFO_OBJECT (self, "removing pads");
for (l = self->srcpads; l; l = l->next) {
GstPad *pad = GST_PAD (l->data);
gst_element_remove_pad (GST_ELEMENT_CAST (self), pad);
gst_object_unref (pad);
}
g_list_free (self->srcpads);
self->srcpads = NULL;
gst_caps_replace (&self->sinkcaps, NULL);
}
static gboolean
gst_deinterleave_set_process_function (GstDeinterleave * self)
{
switch (GST_AUDIO_INFO_WIDTH (&self->audio_info)) {
case 8:
self->func = (GstDeinterleaveFunc) deinterleave_8;
break;
case 16:
self->func = (GstDeinterleaveFunc) deinterleave_16;
break;
case 24:
self->func = (GstDeinterleaveFunc) deinterleave_24;
break;
case 32:
self->func = (GstDeinterleaveFunc) deinterleave_32;
break;
case 64:
self->func = (GstDeinterleaveFunc) deinterleave_64;
break;
default:
return FALSE;
}
return TRUE;
}
static gboolean
gst_deinterleave_check_caps_change (GstDeinterleave * self,
GstAudioInfo * old_info, GstAudioInfo * new_info)
{
gint i;
gboolean same_layout = TRUE;
gboolean was_unpositioned;
gboolean is_unpositioned;
gint new_channels;
gint old_channels;
new_channels = GST_AUDIO_INFO_CHANNELS (new_info);
old_channels = GST_AUDIO_INFO_CHANNELS (old_info);
if (GST_AUDIO_INFO_IS_UNPOSITIONED (new_info) || new_channels == 1)
is_unpositioned = TRUE;
else
is_unpositioned = FALSE;
if (GST_AUDIO_INFO_IS_UNPOSITIONED (old_info) || old_channels == 1)
was_unpositioned = TRUE;
else
was_unpositioned = FALSE;
/* We allow caps changes as long as the number of channels doesn't change
* and the channel positions stay the same. _getcaps() should've cared
* for this already but better be safe.
*/
if (new_channels != old_channels)
goto cannot_change_caps;
/* Now check the channel positions. If we had no channel positions
* and get them or the other way around things have changed.
* If we had channel positions and get different ones things have
* changed too of course
*/
if ((!was_unpositioned && is_unpositioned) || (was_unpositioned
&& !is_unpositioned))
goto cannot_change_caps;
if (!is_unpositioned) {
if (GST_AUDIO_INFO_CHANNELS (old_info) !=
GST_AUDIO_INFO_CHANNELS (new_info))
goto cannot_change_caps;
for (i = 0; i < GST_AUDIO_INFO_CHANNELS (old_info); i++) {
if (new_info->position[i] != old_info->position[i]) {
same_layout = FALSE;
break;
}
}
if (!same_layout)
goto cannot_change_caps;
}
return TRUE;
cannot_change_caps:
return FALSE;
}
static gboolean
gst_deinterleave_sink_setcaps (GstDeinterleave * self, GstCaps * caps)
{
GstCaps *srccaps;
GstStructure *s;
GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps);
if (!gst_audio_info_from_caps (&self->audio_info, caps))
goto invalid_caps;
if (!gst_deinterleave_set_process_function (self))
goto unsupported_caps;
if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) {
GstAudioInfo old_info;
gst_audio_info_init (&old_info);
if (!gst_audio_info_from_caps (&old_info, self->sinkcaps))
goto info_from_caps_failed;
if (gst_deinterleave_check_caps_change (self, &old_info, &self->audio_info)) {
if (!gst_deinterleave_set_process_function (self))
goto cannot_change_caps;
} else
goto cannot_change_caps;
}
gst_caps_replace (&self->sinkcaps, caps);
/* Get srcpad caps */
srccaps = gst_caps_copy (caps);
s = gst_caps_get_structure (srccaps, 0);
gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
gst_structure_remove_field (s, "channel-mask");
/* If we already have pads, update the caps otherwise
* add new pads */
if (self->srcpads) {
if (!gst_deinterleave_set_pads_caps (self, srccaps))
goto set_caps_failed;
} else {
gst_deinterleave_add_new_pads (self, srccaps);
}
gst_caps_unref (srccaps);
return TRUE;
cannot_change_caps:
{
GST_WARNING_OBJECT (self, "caps change from %" GST_PTR_FORMAT
" to %" GST_PTR_FORMAT " not supported: channel number or channel "
"positions change", self->sinkcaps, caps);
return FALSE;
}
unsupported_caps:
{
GST_ERROR_OBJECT (self, "caps not supported: %" GST_PTR_FORMAT, caps);
return FALSE;
}
invalid_caps:
{
GST_ERROR_OBJECT (self, "invalid caps");
return FALSE;
}
set_caps_failed:
{
GST_ERROR_OBJECT (self, "set_caps failed");
gst_caps_unref (srccaps);
return FALSE;
}
info_from_caps_failed:
{
GST_ERROR_OBJECT (self, "coud not get info from caps");
return FALSE;
}
}
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);
}
}
static gboolean
gst_deinterleave_sink_acceptcaps (GstPad * pad, GstObject * parent,
GstCaps * caps)
{
GstDeinterleave *self = GST_DEINTERLEAVE (parent);
GstCaps *templ_caps = gst_pad_get_pad_template_caps (pad);
gboolean ret;
ret = gst_caps_can_intersect (templ_caps, caps);
gst_caps_unref (templ_caps);
if (ret && self->sinkcaps) {
GstAudioInfo new_info;
gst_audio_info_init (&new_info);
if (!gst_audio_info_from_caps (&new_info, caps))
goto info_from_caps_failed;
ret =
gst_deinterleave_check_caps_change (self, &self->audio_info, &new_info);
}
return ret;
info_from_caps_failed:
{
GST_ERROR_OBJECT (self, "coud not get info from caps");
return FALSE;
}
}
static GstCaps *
gst_deinterleave_getcaps (GstPad * pad, GstObject * parent, GstCaps * filter)
{
GstDeinterleave *self = GST_DEINTERLEAVE (parent);
GstCaps *ret;
GstIterator *it;
GstIteratorResult res;
GValue v = G_VALUE_INIT;
if (pad != self->sink) {
ret = gst_pad_get_current_caps (pad);
if (ret) {
if (filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (ret);
ret = tmp;
}
return ret;
}
}
/* Intersect all of our pad template caps with the peer caps of the pad
* to get all formats that are possible up- and downstream.
*
* For the pad for which the caps are requested we don't remove the channel
* informations as they must be in the returned caps and incompatibilities
* will be detected here already
*/
ret = gst_caps_new_any ();
it = gst_element_iterate_pads (GST_ELEMENT_CAST (self));
do {
res = gst_iterator_next (it, &v);
switch (res) {
case GST_ITERATOR_OK:{
GstPad *ourpad = GST_PAD (g_value_get_object (&v));
GstCaps *peercaps = NULL, *ourcaps;
GstCaps *templ_caps = gst_pad_get_pad_template_caps (ourpad);
ourcaps = gst_caps_copy (templ_caps);
gst_caps_unref (templ_caps);
if (pad == ourpad) {
if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK)
__set_channels (ourcaps,
GST_AUDIO_INFO_CHANNELS (&self->audio_info));
else
__set_channels (ourcaps, 1);
} else {
__remove_channels (ourcaps);
/* Only ask for peer caps for other pads than pad
* as otherwise gst_pad_peer_get_caps() might call
* back into this function and deadlock
*/
peercaps = gst_pad_peer_query_caps (ourpad, NULL);
peercaps = gst_caps_make_writable (peercaps);
}
/* If the peer exists and has caps add them to the intersection,
* otherwise assume that the peer accepts everything */
if (peercaps) {
GstCaps *intersection;
GstCaps *oldret = ret;
__remove_channels (peercaps);
intersection = gst_caps_intersect (peercaps, ourcaps);
ret = gst_caps_intersect (ret, intersection);
gst_caps_unref (intersection);
gst_caps_unref (peercaps);
gst_caps_unref (oldret);
} else {
GstCaps *oldret = ret;
ret = gst_caps_intersect (ret, ourcaps);
gst_caps_unref (oldret);
}
gst_caps_unref (ourcaps);
g_value_reset (&v);
break;
}
case GST_ITERATOR_DONE:
break;
case GST_ITERATOR_ERROR:
gst_caps_unref (ret);
ret = gst_caps_new_empty ();
break;
case GST_ITERATOR_RESYNC:
gst_caps_unref (ret);
ret = gst_caps_new_any ();
gst_iterator_resync (it);
break;
}
} while (res != GST_ITERATOR_DONE && res != GST_ITERATOR_ERROR);
g_value_unset (&v);
gst_iterator_free (it);
if (filter) {
GstCaps *aux;
aux = gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (ret);
ret = aux;
}
GST_DEBUG_OBJECT (pad, "Intersected caps to %" GST_PTR_FORMAT, ret);
return ret;
}
static gboolean
gst_deinterleave_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstDeinterleave *self = GST_DEINTERLEAVE (parent);
gboolean ret;
GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event),
GST_DEBUG_PAD_NAME (pad));
/* Send FLUSH_STOP, FLUSH_START and EOS immediately, no matter if
* we have src pads already or not. Queue all other events and
* push them after we have src pads
*/
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
case GST_EVENT_FLUSH_START:
case GST_EVENT_EOS:
ret = gst_pad_event_default (pad, parent, event);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_deinterleave_sink_setcaps (self, caps);
gst_event_unref (event);
break;
}
default:
if (!self->srcpads && !GST_EVENT_IS_STICKY (event)) {
/* Sticky events are copied when creating a new pad */
GST_OBJECT_LOCK (self);
self->pending_events = g_list_append (self->pending_events, event);
GST_OBJECT_UNLOCK (self);
ret = TRUE;
} else {
ret = gst_pad_event_default (pad, parent, event);
}
break;
}
return ret;
}
static gboolean
gst_deinterleave_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
gboolean res;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:{
GstCaps *filter;
GstCaps *caps;
gst_query_parse_caps (query, &filter);
caps = gst_deinterleave_getcaps (pad, parent, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
case GST_QUERY_ACCEPT_CAPS:{
GstCaps *caps;
gboolean ret;
gst_query_parse_accept_caps (query, &caps);
ret = gst_deinterleave_sink_acceptcaps (pad, parent, caps);
gst_query_set_accept_caps_result (query, ret);
res = TRUE;
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static gboolean
gst_deinterleave_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstDeinterleave *self = GST_DEINTERLEAVE (parent);
gboolean res;
res = gst_pad_query_default (pad, parent, query);
if (res && GST_QUERY_TYPE (query) == GST_QUERY_DURATION) {
GstFormat format;
gint64 dur;
gst_query_parse_duration (query, &format, &dur);
/* Need to divide by the number of channels in byte format
* to get the correct value. All other formats should be fine
*/
if (format == GST_FORMAT_BYTES && dur != -1)
gst_query_set_duration (query, format,
dur / GST_AUDIO_INFO_CHANNELS (&self->audio_info));
} else if (res && GST_QUERY_TYPE (query) == GST_QUERY_POSITION) {
GstFormat format;
gint64 pos;
gst_query_parse_position (query, &format, &pos);
/* Need to divide by the number of channels in byte format
* to get the correct value. All other formats should be fine
*/
if (format == GST_FORMAT_BYTES && pos != -1)
gst_query_set_position (query, format,
pos / GST_AUDIO_INFO_CHANNELS (&self->audio_info));
} else if (res && GST_QUERY_TYPE (query) == GST_QUERY_CAPS) {
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_deinterleave_getcaps (pad, parent, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
}
return res;
}
static void
gst_deinterleave_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstDeinterleave *self = GST_DEINTERLEAVE (object);
switch (prop_id) {
case PROP_KEEP_POSITIONS:
self->keep_positions = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_deinterleave_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstDeinterleave *self = GST_DEINTERLEAVE (object);
switch (prop_id) {
case PROP_KEEP_POSITIONS:
g_value_set_boolean (value, self->keep_positions);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstFlowReturn
gst_deinterleave_process (GstDeinterleave * self, GstBuffer * buf)
{
GstFlowReturn ret = GST_FLOW_OK;
guint channels = GST_AUDIO_INFO_CHANNELS (&self->audio_info);
guint pads_pushed = 0, buffers_allocated = 0;
guint nframes =
gst_buffer_get_size (buf) / channels /
(GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
guint bufsize = nframes * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
guint i;
GList *srcs;
GstBuffer **buffers_out = g_new0 (GstBuffer *, channels);
guint8 *in, *out;
GstMapInfo read_info;
GList *pending_events, *l;
/* Send any pending events to all src pads */
GST_OBJECT_LOCK (self);
pending_events = self->pending_events;
self->pending_events = NULL;
GST_OBJECT_UNLOCK (self);
if (pending_events) {
GstEvent *event;
GST_DEBUG_OBJECT (self, "Sending pending events to all src pads");
for (l = pending_events; l; l = l->next) {
event = l->data;
for (srcs = self->srcpads; srcs != NULL; srcs = srcs->next)
gst_pad_push_event (GST_PAD (srcs->data), gst_event_ref (event));
gst_event_unref (event);
}
g_list_free (pending_events);
}
gst_buffer_map (buf, &read_info, GST_MAP_READ);
/* Allocate buffers */
for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
buffers_out[i] = gst_buffer_new_allocate (NULL, bufsize, NULL);
/* Make sure we got a correct buffer. The only other case we allow
* here is an unliked pad */
if (!buffers_out[i])
goto alloc_buffer_failed;
else if (buffers_out[i]
&& gst_buffer_get_size (buffers_out[i]) != bufsize)
goto alloc_buffer_bad_size;
if (buffers_out[i]) {
gst_buffer_copy_into (buffers_out[i], buf, GST_BUFFER_COPY_METADATA, 0,
-1);
buffers_allocated++;
}
}
/* Return NOT_LINKED if no pad was linked */
if (!buffers_allocated) {
GST_WARNING_OBJECT (self,
"Couldn't allocate any buffers because no pad was linked");
ret = GST_FLOW_NOT_LINKED;
goto done;
}
/* deinterleave */
for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
GstPad *pad = (GstPad *) srcs->data;
GstMapInfo write_info;
in = (guint8 *) read_info.data;
in += i * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
if (buffers_out[i]) {
gst_buffer_map (buffers_out[i], &write_info, GST_MAP_WRITE);
out = (guint8 *) write_info.data;
self->func (out, in, channels, nframes);
gst_buffer_unmap (buffers_out[i], &write_info);
ret = gst_pad_push (pad, buffers_out[i]);
buffers_out[i] = NULL;
if (ret == GST_FLOW_OK)
pads_pushed++;
else if (ret == GST_FLOW_NOT_LINKED)
ret = GST_FLOW_OK;
else
goto push_failed;
}
}
/* Return NOT_LINKED if no pad was linked */
if (!pads_pushed)
ret = GST_FLOW_NOT_LINKED;
GST_DEBUG_OBJECT (self, "Pushed on %d pads", pads_pushed);
done:
gst_buffer_unmap (buf, &read_info);
gst_buffer_unref (buf);
g_free (buffers_out);
return ret;
alloc_buffer_failed:
{
GST_WARNING ("gst_pad_alloc_buffer() returned %s", gst_flow_get_name (ret));
goto clean_buffers;
}
alloc_buffer_bad_size:
{
GST_WARNING ("called alloc_buffer(), but didn't get requested bytes");
ret = GST_FLOW_NOT_NEGOTIATED;
goto clean_buffers;
}
push_failed:
{
GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret));
goto clean_buffers;
}
clean_buffers:
{
gst_buffer_unmap (buf, &read_info);
for (i = 0; i < channels; i++) {
if (buffers_out[i])
gst_buffer_unref (buffers_out[i]);
}
gst_buffer_unref (buf);
g_free (buffers_out);
return ret;
}
}
static GstFlowReturn
gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstDeinterleave *self = GST_DEINTERLEAVE (parent);
GstFlowReturn ret;
g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (&self->audio_info) > 0,
GST_FLOW_NOT_NEGOTIATED);
g_return_val_if_fail (GST_AUDIO_INFO_CHANNELS (&self->audio_info) > 0,
GST_FLOW_NOT_NEGOTIATED);
ret = gst_deinterleave_process (self, buffer);
if (ret != GST_FLOW_OK)
GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret));
return ret;
}
static GstStateChangeReturn
gst_deinterleave_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstDeinterleave *self = GST_DEINTERLEAVE (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_deinterleave_remove_pads (self);
self->func = NULL;
if (self->pending_events) {
g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref,
NULL);
g_list_free (self->pending_events);
self->pending_events = NULL;
}
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_deinterleave_remove_pads (self);
self->func = NULL;
if (self->pending_events) {
g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref,
NULL);
g_list_free (self->pending_events);
self->pending_events = NULL;
}
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}