* <refsect2>
* <title>Example launch line</title>
* |[
- * gst-launch filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2 ! deinterleave name=d d.src0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg d.src1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg
+ * gst-launch 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 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src0 ! queue ! audioconvert ! i.sink1 d.src1 ! queue ! audioconvert ! i.sink0
+ * gst-launch 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.
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
- GST_STATIC_CAPS ("audio/x-raw-int, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_FORMATS_ALL ", "
"rate = (int) [ 1, MAX ], "
- "channels = (int) 1, "
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) [ 1, 32 ], "
- "signed = (boolean) { true, false }; "
- "audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) 1, "
- "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
- "width = (int) { 32, 64 }")
- );
+ "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-int, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) [ 1, 32 ], "
- "signed = (boolean) { true, false }; "
- "audio/x-raw-float, "
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_FORMATS_ALL ", "
"rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
- "width = (int) { 32, 64 }")
- );
+ "channels = (int) [ 1, MAX ], layout = (string) interleaved"));
#define MAKE_FUNC(type) \
static void deinterleave_##type (guint##type *out, guint##type *in, \
}
}
-GST_BOILERPLATE (GstDeinterleave, gst_deinterleave, GstElement,
- GST_TYPE_ELEMENT);
+#define gst_deinterleave_parent_class parent_class
+G_DEFINE_TYPE (GstDeinterleave, gst_deinterleave, GST_TYPE_ELEMENT);
enum
{
PROP_KEEP_POSITIONS
};
-static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstBuffer * buffer);
+static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstObject * parent,
+ GstBuffer * buffer);
-static gboolean gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps);
+static gboolean gst_deinterleave_sink_setcaps (GstDeinterleave * self,
+ GstCaps * caps);
-static GstCaps *gst_deinterleave_sink_getcaps (GstPad * pad);
+static GstStateChangeReturn
+gst_deinterleave_change_state (GstElement * element, GstStateChange transition);
-static gboolean gst_deinterleave_sink_activate_push (GstPad * pad,
- gboolean active);
-static gboolean gst_deinterleave_sink_event (GstPad * pad, GstEvent * event);
+static gboolean gst_deinterleave_sink_event (GstPad * pad, GstObject * parent,
+ GstEvent * event);
-static gboolean gst_deinterleave_src_query (GstPad * pad, 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);
{
GstDeinterleave *self = GST_DEINTERLEAVE (obj);
- if (self->pos) {
- g_free (self->pos);
- self->pos = NULL;
- }
-
if (self->pending_events) {
g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL);
g_list_free (self->pending_events);
}
static void
-gst_deinterleave_base_init (gpointer g_class)
+gst_deinterleave_class_init (GstDeinterleaveClass * klass)
{
- GstElementClass *gstelement_class = (GstElementClass *) g_class;
+ 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_details_simple (gstelement_class, "Audio deinterleaver",
"Filter/Converter/Audio",
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
-}
-
-static void
-gst_deinterleave_class_init (GstDeinterleaveClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
- GST_DEBUG_CATEGORY_INIT (gst_deinterleave_debug, "deinterleave", 0,
- "deinterleave element");
+ gstelement_class->change_state = gst_deinterleave_change_state;
gobject_class->finalize = gst_deinterleave_finalize;
gobject_class->set_property = gst_deinterleave_set_property;
}
static void
-gst_deinterleave_init (GstDeinterleave * self, GstDeinterleaveClass * klass)
+gst_deinterleave_init (GstDeinterleave * self)
{
- self->channels = 0;
- self->pos = NULL;
self->keep_positions = FALSE;
- self->width = 0;
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_setcaps_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_setcaps));
- gst_pad_set_getcaps_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_getcaps));
- gst_pad_set_activatepush_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_activate_push));
gst_pad_set_event_function (self->sink,
GST_DEBUG_FUNCPTR (gst_deinterleave_sink_event));
gst_element_add_pad (GST_ELEMENT (self), self->sink);
guint i;
- for (i = 0; i < self->channels; 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 = 0;
- GstStructure *s;
+ /* Set channel position if we know it */
+ if (self->keep_positions)
+ position = GST_AUDIO_INFO_POSITION (&self->audio_info, i);
- pad = gst_pad_new_from_static_template (&src_template, name);
- g_free (name);
+ gst_audio_info_init (&info);
+ gst_audio_info_set_format (&info, format, rate, 1, &position);
- /* Set channel position if we know it */
- if (self->keep_positions) {
- GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE };
+ srccaps = gst_audio_info_to_caps (&info);
- srccaps = gst_caps_copy (caps);
- s = gst_caps_get_structure (srccaps, 0);
- if (self->pos)
- gst_audio_set_channel_positions (s, &self->pos[i]);
- else
- gst_audio_set_channel_positions (s, pos);
- } else {
- srccaps = caps;
- }
+ pad = gst_pad_new_from_static_template (&src_template, name);
+ g_free (name);
- gst_pad_set_caps (pad, srccaps);
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);
+ gst_pad_set_caps (pad, srccaps);
gst_element_add_pad (GST_ELEMENT (self), pad);
self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
- if (self->keep_positions)
- gst_caps_unref (srccaps);
+ gst_caps_unref (srccaps);
}
gst_element_no_more_pads (GST_ELEMENT (self));
gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps)
{
GList *l;
-
- GstStructure *s;
-
gint i;
for (l = self->srcpads, i = 0; l; l = l->next, i++) {
GstPad *pad = GST_PAD (l->data);
GstCaps *srccaps;
+ GstAudioInfo info;
+ gst_audio_info_from_caps (&info, caps);
+ if (self->keep_positions)
+ GST_AUDIO_INFO_POSITION (&info, i) =
+ GST_AUDIO_INFO_POSITION (&self->audio_info, i);
- /* Set channel position if we know it */
- if (self->keep_positions) {
- GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE };
-
- srccaps = gst_caps_copy (caps);
- s = gst_caps_get_structure (srccaps, 0);
- if (self->pos)
- gst_audio_set_channel_positions (s, &self->pos[i]);
- else
- gst_audio_set_channel_positions (s, pos);
- } else {
- srccaps = caps;
- }
+ srccaps = gst_audio_info_to_caps (&info);
gst_pad_set_caps (pad, srccaps);
-
- if (self->keep_positions)
- gst_caps_unref (srccaps);
+ gst_caps_unref (srccaps);
}
}
g_list_free (self->srcpads);
self->srcpads = NULL;
- gst_pad_set_caps (self->sink, NULL);
gst_caps_replace (&self->sinkcaps, NULL);
}
static gboolean
-gst_deinterleave_set_process_function (GstDeinterleave * self, GstCaps * caps)
+gst_deinterleave_set_process_function (GstDeinterleave * self)
{
- GstStructure *s;
-
- s = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (s, "width", &self->width))
- return FALSE;
-
- switch (self->width) {
+ switch (GST_AUDIO_INFO_WIDTH (&self->audio_info)) {
case 8:
self->func = (GstDeinterleaveFunc) deinterleave_8;
break;
}
static gboolean
-gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_deinterleave_sink_setcaps (GstDeinterleave * self, GstCaps * caps)
{
- GstDeinterleave *self;
-
GstCaps *srccaps;
-
GstStructure *s;
- self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps);
- if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) {
- gint new_channels, i;
+ if (!gst_audio_info_from_caps (&self->audio_info, caps))
+ goto invalid_caps;
- GstAudioChannelPosition *pos;
+ if (!gst_deinterleave_set_process_function (self))
+ goto unsupported_caps;
+ if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) {
+ gint i;
gboolean same_layout = TRUE;
-
- s = gst_caps_get_structure (caps, 0);
+ gboolean was_unpositioned;
+ gboolean is_unpositioned =
+ GST_AUDIO_INFO_IS_UNPOSITIONED (&self->audio_info);
+ gint new_channels = GST_AUDIO_INFO_CHANNELS (&self->audio_info);
+ gint old_channels;
+ GstAudioInfo old_info;
+
+ gst_audio_info_init (&old_info);
+ gst_audio_info_from_caps (&old_info, self->sinkcaps);
+ was_unpositioned = GST_AUDIO_INFO_IS_UNPOSITIONED (&old_info);
+ old_channels = GST_AUDIO_INFO_CHANNELS (&old_info);
/* 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 (!gst_structure_get_int (s, "channels", &new_channels) ||
- new_channels != self->channels ||
- !gst_deinterleave_set_process_function (self, caps))
+ if (new_channels != old_channels ||
+ !gst_deinterleave_set_process_function (self))
goto cannot_change_caps;
/* Now check the channel positions. If we had no channel positions
* If we had channel positions and get different ones things have
* changed too of course
*/
- pos = gst_audio_get_channel_positions (s);
- if ((pos && !self->pos) || (!pos && self->pos))
+ if ((!was_unpositioned && is_unpositioned) || (was_unpositioned
+ && !is_unpositioned))
goto cannot_change_caps;
- if (pos) {
- for (i = 0; i < self->channels; i++) {
- if (self->pos[i] != pos[i]) {
+ if (!is_unpositioned) {
+ if (GST_AUDIO_INFO_CHANNELS (&old_info) !=
+ GST_AUDIO_INFO_CHANNELS (&self->audio_info))
+ goto cannot_change_caps;
+ for (i = 0; i < GST_AUDIO_INFO_CHANNELS (&old_info); i++) {
+ if (self->audio_info.position[i] != old_info.position[i]) {
same_layout = FALSE;
break;
}
}
- g_free (pos);
if (!same_layout)
goto cannot_change_caps;
}
- } else {
- s = gst_caps_get_structure (caps, 0);
-
- if (!gst_structure_get_int (s, "channels", &self->channels))
- goto no_channels;
-
- if (!gst_deinterleave_set_process_function (self, caps))
- goto unsupported_caps;
-
- self->pos = gst_audio_get_channel_positions (s);
}
gst_caps_replace (&self->sinkcaps, 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-positions");
+ gst_structure_remove_field (s, "channel-mask");
/* If we already have pads, update the caps otherwise
* add new pads */
}
gst_caps_unref (srccaps);
- gst_object_unref (self);
return TRUE;
cannot_change_caps:
{
GST_ERROR_OBJECT (self, "can't set new caps: %" GST_PTR_FORMAT, caps);
- gst_object_unref (self);
return FALSE;
}
unsupported_caps:
{
GST_ERROR_OBJECT (self, "caps not supported: %" GST_PTR_FORMAT, caps);
- gst_object_unref (self);
return FALSE;
}
-no_channels:
+invalid_caps:
{
GST_ERROR_OBJECT (self, "invalid caps");
- gst_object_unref (self);
return FALSE;
}
}
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-positions");
+ gst_structure_remove_field (s, "channel-mask");
gst_structure_remove_field (s, "channels");
}
}
}
static GstCaps *
-gst_deinterleave_sink_getcaps (GstPad * pad)
+gst_deinterleave_sink_getcaps (GstPad * pad, GstObject * parent,
+ GstCaps * filter)
{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
+ GstDeinterleave *self = GST_DEINTERLEAVE (parent);
GstCaps *ret;
if (pad == ourpad) {
if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK)
- __set_channels (ourcaps, self->channels);
+ __set_channels (ourcaps, GST_AUDIO_INFO_CHANNELS (&self->audio_info));
else
__set_channels (ourcaps, 1);
} else {
* as otherwise gst_pad_peer_get_caps() might call
* back into this function and deadlock
*/
- peercaps = gst_pad_peer_get_caps (ourpad);
+ 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,
}
GST_OBJECT_UNLOCK (self);
- gst_object_unref (self);
-
GST_DEBUG_OBJECT (pad, "Intersected caps to %" GST_PTR_FORMAT, ret);
return ret;
}
static gboolean
-gst_deinterleave_sink_event (GstPad * pad, GstEvent * event)
+gst_deinterleave_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
+ GstDeinterleave *self = GST_DEINTERLEAVE (parent);
gboolean ret;
case GST_EVENT_FLUSH_STOP:
case GST_EVENT_FLUSH_START:
case GST_EVENT_EOS:
- ret = gst_pad_event_default (pad, event);
+ 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) {
- ret = gst_pad_event_default (pad, event);
+ ret = gst_pad_event_default (pad, parent, event);
} else {
GST_OBJECT_LOCK (self);
self->pending_events = g_list_append (self->pending_events, event);
break;
}
- gst_object_unref (self);
-
return ret;
}
static gboolean
-gst_deinterleave_src_query (GstPad * pad, GstQuery * query)
+gst_deinterleave_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
+ GstDeinterleave *self = GST_DEINTERLEAVE (parent);
gboolean res;
- res = gst_pad_query_default (pad, query);
+ res = gst_pad_query_default (pad, parent, query);
if (res && GST_QUERY_TYPE (query) == GST_QUERY_DURATION) {
GstFormat 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 / self->channels);
+ 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;
* 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 / self->channels);
+ 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_sink_getcaps (pad, parent, filter);
+ gst_query_set_caps_result (query, caps);
+ gst_caps_unref (caps);
}
- gst_object_unref (self);
return res;
}
{
GstFlowReturn ret = GST_FLOW_OK;
- guint channels = self->channels;
+ guint channels = GST_AUDIO_INFO_CHANNELS (&self->audio_info);
guint pads_pushed = 0, buffers_allocated = 0;
- guint nframes = GST_BUFFER_SIZE (buf) / channels / (self->width / 8);
+ guint nframes =
+ gst_buffer_get_size (buf) / channels /
+ (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
- guint bufsize = nframes * (self->width / 8);
+ guint bufsize = nframes * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
guint i;
guint8 *in, *out;
+ GstMapInfo read_info;
+ gst_buffer_map (buf, &read_info, GST_MAP_READ);
+
/* Send any pending events to all src pads */
GST_OBJECT_LOCK (self);
if (self->pending_events) {
/* Allocate buffers */
for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
- GstPad *pad = (GstPad *) srcs->data;
-
- buffers_out[i] = NULL;
- ret =
- gst_pad_alloc_buffer (pad, GST_BUFFER_OFFSET_NONE, bufsize,
- GST_PAD_CAPS (pad), &buffers_out[i]);
+ buffers_out[i] = gst_buffer_new_allocate (NULL, bufsize, 0);
/* Make sure we got a correct buffer. The only other case we allow
* here is an unliked pad */
- if (ret != GST_FLOW_OK && ret != GST_FLOW_NOT_LINKED)
+ if (!buffers_out[i])
goto alloc_buffer_failed;
- else if (buffers_out[i] && GST_BUFFER_SIZE (buffers_out[i]) != bufsize)
+ else if (buffers_out[i] && gst_buffer_get_size (buffers_out[i]) != bufsize)
goto alloc_buffer_bad_size;
- else if (buffers_out[i] &&
- !gst_caps_is_equal (GST_BUFFER_CAPS (buffers_out[i]),
- GST_PAD_CAPS (pad)))
- goto invalid_caps;
if (buffers_out[i]) {
- gst_buffer_copy_metadata (buffers_out[i], buf,
- GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS);
+ gst_buffer_copy_into (buffers_out[i], buf, GST_BUFFER_COPY_METADATA, 0,
+ -1);
buffers_allocated++;
}
}
/* deinterleave */
for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
GstPad *pad = (GstPad *) srcs->data;
+ GstMapInfo write_info;
+
- in = (guint8 *) GST_BUFFER_DATA (buf);
- in += i * (self->width / 8);
+ in = (guint8 *) read_info.data;
+ in += i * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
if (buffers_out[i]) {
- out = (guint8 *) GST_BUFFER_DATA (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)
ret = GST_FLOW_NOT_LINKED;
done:
+ gst_buffer_unmap (buf, &read_info);
gst_buffer_unref (buf);
g_free (buffers_out);
return ret;
ret = GST_FLOW_NOT_NEGOTIATED;
goto clean_buffers;
}
-invalid_caps:
- {
- GST_WARNING ("called alloc_buffer(), but didn't get requested caps");
- ret = GST_FLOW_NOT_NEGOTIATED;
- goto clean_buffers;
- }
push_failed:
{
GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret));
}
clean_buffers:
{
+ gst_buffer_unmap (buf, &read_info);
for (i = 0; i < channels; i++) {
if (buffers_out[i])
gst_buffer_unref (buffers_out[i]);
}
static GstFlowReturn
-gst_deinterleave_chain (GstPad * pad, GstBuffer * buffer)
+gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
- GstDeinterleave *self = GST_DEINTERLEAVE (GST_PAD_PARENT (pad));
+ GstDeinterleave *self = GST_DEINTERLEAVE (parent);
GstFlowReturn ret;
g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->channels > 0, 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);
return ret;
}
-static gboolean
-gst_deinterleave_sink_activate_push (GstPad * pad, gboolean active)
+static GstStateChangeReturn
+gst_deinterleave_change_state (GstElement * element, GstStateChange transition)
{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
- /* Reset everything when the pad is deactivated */
- if (!active) {
- gst_deinterleave_remove_pads (self);
- if (self->pos) {
- g_free (self->pos);
- self->pos = NULL;
- }
- self->channels = 0;
- self->width = 0;
- 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;
- }
+ 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;
}
- gst_object_unref (self);
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
- return TRUE;
+ 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;
}
#endif
#include <stdio.h>
+#include <gst/audio/audio.h>
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
- "channels = (int) 1, " "rate = (int) {32000, 48000}"));
+ "channels = (int) 1, layout = (string) {interleaved, non-interleaved}, rate = (int) {32000, 48000}"));
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
- "channels = (int) { 2, 3 }, " "rate = (int) {32000, 48000}"));
+ "channels = (int) { 2, 3 }, layout = (string) interleaved, rate = (int) {32000, 48000}"));
#define CAPS_32khz \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE (F32) ", " \
- "channels = (int) 2, " \
+ "channels = (int) 2, layout = (string) interleaved, " \
"rate = (int) 32000"
#define CAPS_48khz \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE (F32) ", " \
- "channels = (int) 2, " \
+ "channels = (int) 2, layout = (string) interleaved, " \
"rate = (int) 48000"
#define CAPS_48khz_3CH \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE (F32) ", " \
- "channels = (int) 3, " \
+ "channels = (int) 3, layout = (string) interleaved, " \
"rate = (int) 48000"
static GstFlowReturn
GstCaps *caps;
gfloat *indata;
GstMapInfo map;
+ guint64 channel_mask = 0;
mysinkpads = g_new0 (GstPad *, 2);
nsinkpads = 0;
mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
fail_unless (mysrcpad != NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
caps = gst_caps_from_string (CAPS_48khz);
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
+ NULL);
+
fail_unless (gst_pad_set_caps (mysrcpad, caps));
gst_pad_use_fixed_caps (mysrcpad);
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
+ inbuf = gst_buffer_make_writable (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
indata = (gfloat *) map.data;
for (i = 0; i < 2 * 48000; i += 2) {
GstCaps *caps;
gfloat *indata;
GstMapInfo map;
+ guint64 channel_mask = 0;
nsinkpads = 0;
mysinkpads = g_new0 (GstPad *, 2);
mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
fail_unless (mysrcpad != NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
caps = gst_caps_from_string (CAPS_48khz);
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
+ NULL);
+
fail_unless (gst_pad_set_caps (mysrcpad, caps));
gst_pad_use_fixed_caps (mysrcpad);
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
+ inbuf = gst_buffer_make_writable (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
indata = (gfloat *) map.data;
for (i = 0; i < 2 * 48000; i += 2) {
GstBuffer *inbuf;
gfloat *indata;
GstMapInfo map;
+ guint64 channel_mask;
nsinkpads = 0;
mysinkpads = g_new0 (GstPad *, 2);
mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
fail_unless (mysrcpad != NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
+
caps = gst_caps_from_string (CAPS_48khz);
+ channel_mask = 0;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
+ NULL);
fail_unless (gst_pad_set_caps (mysrcpad, caps));
gst_pad_use_fixed_caps (mysrcpad);
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
+ inbuf = gst_buffer_make_writable (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
indata = (gfloat *) map.data;
for (i = 0; i < 2 * 48000; i += 2) {
fail_unless (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
caps2 = gst_caps_from_string (CAPS_32khz);
+ channel_mask = 0;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ gst_caps_set_simple (caps2, "channel-mask", GST_TYPE_BITMASK, channel_mask,
+ NULL);
gst_pad_set_caps (mysrcpad, caps2);
inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
+ inbuf = gst_buffer_make_writable (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
indata = (gfloat *) map.data;
for (i = 0; i < 2 * 48000; i += 2) {
gst_caps_unref (caps2);
caps2 = gst_caps_from_string (CAPS_48khz_3CH);
+ channel_mask = 0;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
+ channel_mask |=
+ G_GUINT64_CONSTANT (1) << GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
+ gst_caps_set_simple (caps2, "channel-mask", GST_TYPE_BITMASK, channel_mask,
+ NULL);
gst_pad_set_caps (mysrcpad, caps2);
inbuf = gst_buffer_new_and_alloc (3 * 48000 * sizeof (gfloat));
+ inbuf = gst_buffer_make_writable (inbuf);
gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
indata = (gfloat *) map.data;
for (i = 0; i < 3 * 48000; i += 3) {
set_channel_positions (GstCaps * caps, int channels,
GstAudioChannelPosition * channelpositions)
{
-#if 0
- GValue chanpos = { 0 };
- GValue pos = { 0 };
- GstStructure *structure = gst_caps_get_structure (caps, 0);
int c;
+ guint64 channel_mask = 0;
- g_value_init (&chanpos, GST_TYPE_ARRAY);
- g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
-
- for (c = 0; c < channels; c++) {
- g_value_set_enum (&pos, channelpositions[c]);
- gst_value_array_append_value (&chanpos, &pos);
- }
- g_value_unset (&pos);
+ for (c = 0; c < channels; c++)
+ channel_mask |= G_GUINT64_CONSTANT (1) << channelpositions[c];
- gst_structure_set_value (structure, "channel-positions", &chanpos);
- g_value_unset (&chanpos);
-#endif
+ gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask,
+ NULL);
}
static void
src_handoff_float32_8ch (GstElement * src, GstBuffer * buf, GstPad * pad,
gpointer user_data)
{
- GstAudioChannelPosition layout[NUM_CHANNELS];
- GstCaps *caps;
gfloat *data, *p;
guint size, i, c;
+ GstAudioChannelPosition layout[NUM_CHANNELS];
+ GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (F32),
"channels", G_TYPE_INT, NUM_CHANNELS,
+ "layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, SAMPLE_RATE, NULL);
for (i = 0; i < NUM_CHANNELS; ++i)
- layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+ layout[i] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT + i;
set_channel_positions (caps, NUM_CHANNELS, layout);
+ gst_pad_set_caps (pad, caps);
+ gst_caps_unref (caps);
size = sizeof (gfloat) * SAMPLES_PER_BUFFER * NUM_CHANNELS;
data = p = (gfloat *) g_malloc (size);
++p;
}
}
+
+ buf = gst_buffer_new ();
gst_buffer_take_memory (buf, -1, gst_memory_new_wrapped (0, data, g_free,
size, 0, size));
-
GST_BUFFER_OFFSET (buf) = 0;
GST_BUFFER_TIMESTAMP (buf) = 0;
-
- /* FIXME, caps */
}
static GstPadProbeReturn
guint num, i;
GstCaps *caps;
GstStructure *s;
-#if 0
GstAudioChannelPosition *pos;
-#endif
gint channels;
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
+ GstAudioInfo audio_info;
+ gint pad_id = (gint) userdata;
- fail_unless_equals_int (sscanf (GST_PAD_NAME (pad), "src%u", &padnum), 1);
+ fail_unless_equals_int (sscanf (GST_PAD_NAME (pad), "src_%u", &padnum), 1);
numpads = pads_created;
s = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_get_int (s, "channels", &channels));
fail_unless_equals_int (channels, 1);
- fail_unless (gst_structure_has_field (s, "channel-positions"));
-#if 0
- pos = gst_audio_get_channel_positions (s);
- fail_unless (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE);
- g_free (pos);
-#endif
+
+ gst_audio_info_init (&audio_info);
+ fail_unless (gst_audio_info_from_caps (&audio_info, caps));
+
+ pos = audio_info.position;
+ fail_unless (pos != NULL
+ && pos[0] == GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT + pad_id);
gst_caps_unref (caps);
gst_buffer_map (buffer, &map, GST_MAP_READ);
fail_unless (sink != NULL);
gst_bin_add_many (GST_BIN (pipeline), queue, sink, NULL);
- fail_unless (gst_element_link_many (queue, sink, NULL));
+ gst_element_link_pads_full (queue, "src", sink, "sink",
+ GST_PAD_LINK_CHECK_NOTHING);
sinkpad = gst_element_get_static_pad (queue, "sink");
+
fail_unless_equals_int (gst_pad_link (pad, sinkpad), GST_PAD_LINK_OK);
gst_object_unref (sinkpad);
+
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_BUFFER, float_buffer_check_probe,
- NULL, NULL);
+ (gpointer) pads_created, NULL);
gst_element_set_state (sink, GST_STATE_PLAYING);
gst_element_set_state (queue, GST_STATE_PLAYING);
projects / platform / upstream / gst-plugins-good.git / commitdiff