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authorAndy Wingo <wingo@pobox.com>2003-07-16 16:08:13 +0000
committerAndy Wingo <wingo@pobox.com>2003-07-16 16:08:13 +0000
commit2ff63e563b21cac87489a0d989c3aa957d5f2fb9 (patch)
tree614234db5c4422d29fd25866d20b9c4cdc8729f0 /ext/sndfile
parent0e04196b7136fee8628669a769838ecf78416dea (diff)
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actually recurse into sndfile if we are able big ladspa cleanups, mainly to comply with the buffer-frames caps proper...
Original commit message from CVS: * actually recurse into sndfile if we are able * big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general cleanups - the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins) you need to use a filtered connection, just like with buffer-frames * big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit simpler * make the ossclock general, add it to gstaudio, and use it in sndfile as well i need to update mimetypes, but that's coming soon. there are some other plugins that don't support buffer-frames, i guess i need to get around to fixing them as well.
Diffstat (limited to 'ext/sndfile')
-rw-r--r--ext/sndfile/gstsf.c419
-rw-r--r--ext/sndfile/gstsf.h14
2 files changed, 316 insertions, 117 deletions
diff --git a/ext/sndfile/gstsf.c b/ext/sndfile/gstsf.c
index 998bf9e7..4570c108 100644
--- a/ext/sndfile/gstsf.c
+++ b/ext/sndfile/gstsf.c
@@ -1,9 +1,5 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2000 Wim Taymans <wtay@chello.be>
- * 2003 Andy Wingo <wingo at pobox dot com>
- *
- * gstsf.c: libsndfile plugin for GStreamer
+/* GStreamer libsndfile plugin
+ * Copyright (C) 2003 Andy Wingo <wingo at pobox dot com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@@ -22,13 +18,15 @@
*/
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-#include <gst/gst.h>
#include <string.h>
+#include <gst/gst.h>
+
+#include <config.h>
+#include <gst/audio/audio.h>
+
#include "gstsf.h"
+
static GstElementDetails sfsrc_details = {
"Sndfile Source",
"Source/Audio",
@@ -58,9 +56,6 @@ enum {
ARG_CREATE_PADS
};
-#define GST_SF_BUF_BYTES 2048
-#define GST_SF_BUF_FRAMES (GST_SF_BUF_BYTES / sizeof(float))
-
GST_PAD_TEMPLATE_FACTORY (sf_src_factory,
"src%d",
GST_PAD_SRC,
@@ -68,12 +63,11 @@ GST_PAD_TEMPLATE_FACTORY (sf_src_factory,
GST_CAPS_NEW (
"sf_src",
"audio/x-raw-float",
- "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),
- "intercept", GST_PROPS_FLOAT(0.0),
- "slope", GST_PROPS_FLOAT(1.0),
- "channels", GST_PROPS_INT (1),
- "width", GST_PROPS_INT (32),
- "endianness", GST_PROPS_INT (G_BYTE_ORDER)
+ "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),
+ "width", GST_PROPS_INT (32),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),
+ "channels", GST_PROPS_INT (1)
)
);
@@ -84,12 +78,11 @@ GST_PAD_TEMPLATE_FACTORY (sf_sink_factory,
GST_CAPS_NEW (
"sf_sink",
"audio/x-raw-float",
- "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),
- "intercept", GST_PROPS_FLOAT(0.0),
- "slope", GST_PROPS_FLOAT(1.0),
- "channels", GST_PROPS_INT (1),
- "width", GST_PROPS_INT (32),
- "endianness", GST_PROPS_INT (G_BYTE_ORDER)
+ "rate", GST_PROPS_INT_RANGE (1, G_MAXINT),
+ "width", GST_PROPS_INT (32),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "buffer-frames", GST_PROPS_INT_RANGE (1, G_MAXINT),
+ "channels", GST_PROPS_INT (1)
)
);
@@ -163,27 +156,38 @@ gst_sf_minor_types_get_type (void)
return sf_minor_types_type;
}
-static void gst_sf_class_init (GstSFClass *klass);
-static void gst_sf_init (GstSF *this);
-
-static gboolean gst_sf_open_file (GstSF *this);
-static void gst_sf_close_file (GstSF *this);
+static void gst_sf_class_init (GstSFClass *klass);
+static void gst_sf_init (GstSF *this);
+static void gst_sf_dispose (GObject *object);
+static void gst_sf_set_property (GObject *object, guint prop_id,
+ const GValue *value, GParamSpec *pspec);
+static void gst_sf_get_property (GObject *object, guint prop_id,
+ GValue *value, GParamSpec *pspec);
+
+static GstClock* gst_sf_get_clock (GstElement *element);
+static void gst_sf_set_clock (GstElement *element, GstClock *clock);
+static GstPad* gst_sf_request_new_pad (GstElement *element, GstPadTemplate *templ,
+ const gchar *unused);
+static void gst_sf_release_request_pad (GstElement *element, GstPad *pad);
+static GstElementStateReturn gst_sf_change_state (GstElement *element);
-static void gst_sf_loop (GstElement *element);
+static GstPadLinkReturn gst_sf_link (GstPad *pad, GstCaps *caps);
-static void gst_sf_set_property (GObject *object, guint prop_id, const GValue *value,
- GParamSpec *pspec);
-static void gst_sf_get_property (GObject *object, guint prop_id, GValue *value,
- GParamSpec *pspec);
+static void gst_sf_loop (GstElement *element);
-static GstPad* gst_sf_request_new_pad (GstElement *element, GstPadTemplate *templ,
- const gchar *unused);
+static GstClockTime gst_sf_get_time (GstClock *clock, gpointer data);
-static GstElementStateReturn gst_sf_change_state (GstElement *element);
-static GstPadLinkReturn gst_sf_link (GstPad *pad, GstCaps *caps);
+static gboolean gst_sf_open_file (GstSF *this);
+static void gst_sf_close_file (GstSF *this);
static GstElementClass *parent_class = NULL;
+GST_DEBUG_CATEGORY_STATIC (gstsf_debug);
+#define INFO(...) \
+ GST_CAT_LEVEL_LOG (gstsf_debug, GST_LEVEL_INFO, NULL, __VA_ARGS__)
+#define INFO_OBJ(obj,...) \
+ GST_CAT_LEVEL_LOG (gstsf_debug, GST_LEVEL_INFO, obj, __VA_ARGS__)
+
GType
gst_sf_get_type (void)
{
@@ -281,64 +285,33 @@ gst_sf_class_init (GstSFClass *klass)
g_object_class_install_property (gobject_class, ARG_CREATE_PADS, pspec);
}
+ gobject_class->dispose = gst_sf_dispose;
gobject_class->set_property = gst_sf_set_property;
gobject_class->get_property = gst_sf_get_property;
+ gstelement_class->get_clock = gst_sf_get_clock;
+ gstelement_class->set_clock = gst_sf_set_clock;
gstelement_class->change_state = gst_sf_change_state;
gstelement_class->request_new_pad = gst_sf_request_new_pad;
+ gstelement_class->release_pad = gst_sf_release_request_pad;
}
static void
gst_sf_init (GstSF *this)
{
gst_element_set_loop_function (GST_ELEMENT (this), gst_sf_loop);
+ this->provided_clock = gst_audio_clock_new ("sfclock", gst_sf_get_time, this);
+ gst_object_set_parent (GST_OBJECT (this->provided_clock), GST_OBJECT (this));
}
-static GstPad*
-gst_sf_request_new_pad (GstElement *element, GstPadTemplate *templ,
- const gchar *unused)
+static void
+gst_sf_dispose (GObject *object)
{
- gchar *name;
- GstSF *this;
- GstSFChannel *channel;
+ GstSF *this = (GstSF*)object;
- this = GST_SF (element);
- channel = g_new0 (GstSFChannel, 1);
-
- if (templ->direction == GST_PAD_SINK) {
- /* we have an SFSink */
- name = g_strdup_printf ("sink%d", this->channelcount);
- this->numchannels++;
- if (this->file) {
- gst_sf_close_file (this);
- gst_sf_open_file (this);
- }
- } else {
- /* we have an SFSrc */
- name = g_strdup_printf ("src%d", this->channelcount);
- }
-
- channel->pad = gst_pad_new_from_template (templ, name);
- gst_element_add_pad (GST_ELEMENT (this), channel->pad);
- gst_pad_set_link_function (channel->pad, gst_sf_link);
-
- this->channels = g_list_append (this->channels, channel);
- this->channelcount++;
-
- GST_DEBUG ("sf added pad %s\n", name);
-
- g_free (name);
- return channel->pad;
-}
-
-static GstPadLinkReturn
-gst_sf_link (GstPad *pad, GstCaps *caps)
-{
- GstSF *this = (GstSF*)GST_OBJECT_PARENT (pad);
-
- gst_caps_get_int (caps, "rate", &this->rate);
+ gst_object_unparent (GST_OBJECT (this->provided_clock));
- return GST_PAD_LINK_OK;
+ G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
@@ -420,6 +393,156 @@ gst_sf_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *
}
}
+static GstClock*
+gst_sf_get_clock (GstElement *element)
+{
+ GstSF *this = GST_SF (element);
+
+ return this->provided_clock;
+}
+
+static void
+gst_sf_set_clock (GstElement *element, GstClock *clock)
+{
+ GstSF *this = GST_SF (element);
+
+ this->clock = clock;
+}
+
+static GstClockTime
+gst_sf_get_time (GstClock *clock, gpointer data)
+{
+ GstSF *this = GST_SF (data);
+
+ return this->time;
+}
+
+static GstElementStateReturn
+gst_sf_change_state (GstElement *element)
+{
+ GstSF *this = GST_SF (element);
+
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:
+ break;
+ case GST_STATE_READY_TO_PAUSED:
+ break;
+ case GST_STATE_PAUSED_TO_PLAYING:
+ gst_audio_clock_set_active (GST_AUDIO_CLOCK (this->provided_clock), TRUE);
+ break;
+ case GST_STATE_PLAYING_TO_PAUSED:
+ gst_audio_clock_set_active (GST_AUDIO_CLOCK (this->provided_clock), FALSE);
+ break;
+ case GST_STATE_PAUSED_TO_READY:
+ break;
+ case GST_STATE_READY_TO_NULL:
+ if (GST_FLAG_IS_SET (this, GST_SF_OPEN))
+ gst_sf_close_file (this);
+ break;
+ }
+
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+static GstPad*
+gst_sf_request_new_pad (GstElement *element, GstPadTemplate *templ,
+ const gchar *unused)
+{
+ gchar *name;
+ GstSF *this;
+ GstSFChannel *channel;
+
+ this = GST_SF (element);
+ channel = g_new0 (GstSFChannel, 1);
+
+ if (templ->direction == GST_PAD_SINK) {
+ /* we have an SFSink */
+ name = g_strdup_printf ("sink%d", this->channelcount);
+ this->numchannels++;
+ if (this->file) {
+ gst_sf_close_file (this);
+ gst_sf_open_file (this);
+ }
+ } else {
+ /* we have an SFSrc */
+ name = g_strdup_printf ("src%d", this->channelcount);
+ }
+
+ channel->pad = gst_pad_new_from_template (templ, name);
+ gst_element_add_pad (GST_ELEMENT (this), channel->pad);
+ gst_pad_set_link_function (channel->pad, gst_sf_link);
+
+ this->channels = g_list_append (this->channels, channel);
+ this->channelcount++;
+
+ INFO_OBJ (element, "added pad %s\n", name);
+
+ g_free (name);
+ return channel->pad;
+}
+
+static void
+gst_sf_release_request_pad (GstElement *element, GstPad *pad)
+{
+ GstSF *this;
+ GstSFChannel *channel = NULL;
+ GList *l;
+
+ this = GST_SF (element);
+
+ if (GST_STATE (element) == GST_STATE_PLAYING) {
+ g_warning ("You can't release a request pad if the element is PLAYING, sorry.");
+ return;
+ }
+
+ for (l=this->channels; l; l=l->next) {
+ if (GST_SF_CHANNEL (l)->pad == pad) {
+ channel = GST_SF_CHANNEL (l);
+ break;
+ }
+ }
+
+ g_return_if_fail (channel != NULL);
+
+ INFO_OBJ (element, "Releasing request pad %s", GST_PAD_NAME (channel->pad));
+
+ if (GST_FLAG_IS_SET (element, GST_SF_OPEN))
+ gst_sf_close_file (this);
+
+ gst_element_remove_pad (element, channel->pad);
+ this->channels = g_list_remove (this->channels, channel);
+ this->numchannels--;
+ g_free (channel);
+}
+
+static GstPadLinkReturn
+gst_sf_link (GstPad *pad, GstCaps *caps)
+{
+ GstSF *this = (GstSF*)GST_OBJECT_PARENT (pad);
+
+ if (GST_CAPS_IS_FIXED (caps)) {
+ gst_caps_get_int (caps, "rate", &this->rate);
+ gst_caps_get_int (caps, "buffer-frames", &this->buffer_frames);
+
+ INFO_OBJ (this, "linked pad %s:%s with fixed caps, frames=%d, rate=%d",
+ GST_DEBUG_PAD_NAME (pad), this->rate, this->buffer_frames);
+
+ if (this->numchannels) {
+ /* we can go ahead and allocate our buffer */
+ if (this->buffer)
+ g_free (this->buffer);
+ this->buffer = g_malloc (this->numchannels * this->buffer_frames * sizeof (float));
+ memset (this->buffer, 0, this->numchannels * this->buffer_frames * sizeof (float));
+ }
+ return GST_PAD_LINK_OK;
+ }
+
+ return GST_PAD_LINK_DELAYED;
+}
+
static gboolean
gst_sf_open_file (GstSF *this)
{
@@ -428,8 +551,10 @@ gst_sf_open_file (GstSF *this)
g_return_val_if_fail (!GST_FLAG_IS_SET (this, GST_SF_OPEN), FALSE);
+ this->time = 0;
+
if (!this->filename) {
- gst_element_error (GST_ELEMENT (this), "sndfile::location was not set");
+ gst_element_error (GST_ELEMENT (this), "sndfile: 'location' was not set");
return FALSE;
}
@@ -437,15 +562,26 @@ gst_sf_open_file (GstSF *this)
mode = SFM_READ;
info.format = 0;
} else {
+ if (!this->rate) {
+ INFO_OBJ (this, "Not opening %s yet because caps are not set", this->filename);
+ return FALSE;
+ } else if (!this->numchannels) {
+ INFO_OBJ (this, "Not opening %s yet because we have no input channels", this->filename);
+ return FALSE;
+ }
+
mode = SFM_WRITE;
this->format = this->format_major | this->format_subtype;
info.samplerate = this->rate;
info.channels = this->numchannels;
info.format = this->format;
+ INFO_OBJ (this, "Opening %s with rate %d, %d channels, format 0x%x",
+ this->filename, info.samplerate, info.channels, info.format);
+
if (!sf_format_check (&info)) {
gst_element_error (GST_ELEMENT (this),
- g_strdup_printf ("Input parameters (rate:%d, channels:%d, format:%x) invalid",
+ g_strdup_printf ("Input parameters (rate:%d, channels:%d, format:0x%x) invalid",
info.samplerate, info.channels, info.format));
return FALSE;
}
@@ -478,7 +614,6 @@ gst_sf_open_file (GstSF *this)
GST_SF_CHANNEL (l)->caps_set = FALSE;
}
- this->buffer = g_malloc (this->numchannels * GST_SF_BUF_BYTES);
GST_FLAG_SET (this, GST_SF_OPEN);
return TRUE;
@@ -491,6 +626,8 @@ gst_sf_close_file (GstSF *this)
g_return_if_fail (GST_FLAG_IS_SET (this, GST_SF_OPEN));
+ INFO_OBJ (this, "Closing file %s", this->filename);
+
if ((err = sf_close (this->file)))
gst_element_error (GST_ELEMENT (this),
g_strdup_printf ("sndfile: could not close file \"%s\": %s",
@@ -513,25 +650,36 @@ gst_sf_loop (GstElement *element)
this = (GstSF*)element;
if (this->channels == NULL) {
- gst_element_error (element, "You must connect at least one pad to soundfile elements.");
+ gst_element_error (element, "You must connect at least one pad to sndfile elements.");
return;
}
- if (!GST_FLAG_IS_SET (this, GST_SF_OPEN))
- if (!gst_sf_open_file (this))
- return; /* we've already set gst_element_error */
if (GST_IS_SFSRC (this)) {
sf_count_t read;
gint i, j;
int eos = 0;
+ int buffer_frames = this->buffer_frames;
int nchannels = this->numchannels;
GstSFChannel *channel = NULL;
gfloat *data;
gfloat *buf = this->buffer;
GstBuffer *out;
- read = sf_readf_float (this->file, buf, GST_SF_BUF_FRAMES);
- if (read < GST_SF_BUF_FRAMES)
+ if (!GST_FLAG_IS_SET (this, GST_SF_OPEN))
+ if (!gst_sf_open_file (this))
+ return; /* we've already set gst_element_error */
+
+ if (buffer_frames == 0) {
+ /* we have to set the caps later */
+ buffer_frames = this->buffer_frames = 1024;
+ }
+ if (buf == NULL) {
+ buf = this->buffer = g_malloc (this->numchannels * this->buffer_frames * sizeof (float));
+ memset (this->buffer, 0, this->numchannels * this->buffer_frames * sizeof (float));
+ }
+
+ read = sf_readf_float (this->file, buf, buffer_frames);
+ if (read < buffer_frames)
eos = 1;
if (read)
@@ -548,16 +696,12 @@ gst_sf_loop (GstElement *element)
caps = gst_caps_copy
(GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (GST_SF_CHANNEL (l)->pad)));
gst_caps_set (caps, "rate", GST_PROPS_INT (this->rate), NULL);
- /* we know it's fixed, yo. */
- GST_CAPS_FLAG_SET (caps, GST_CAPS_FIXED);
+ gst_caps_set (caps, "buffer-frames", GST_PROPS_INT (this->buffer_frames), NULL);
if (!gst_pad_try_set_caps (GST_SF_CHANNEL (l)->pad, caps)) {
gst_element_error (GST_ELEMENT (this),
g_strdup_printf ("Opened file with sample rate %d, but could not set caps",
this->rate));
- sf_close (this->file);
- this->file = NULL;
- g_free (this->buffer);
- this->buffer = NULL;
+ gst_sf_close_file (this->file);
return;
}
channel->caps_set = TRUE;
@@ -570,35 +714,81 @@ gst_sf_loop (GstElement *element)
gst_pad_push (channel->pad, out);
}
+ this->time += read * (GST_SECOND / this->rate);
+ gst_audio_clock_update_time ((GstAudioClock*)this->provided_clock, this->time);
+
if (eos) {
if (this->loop) {
sf_seek (this->file, (sf_count_t)0, SEEK_SET);
eos = 0;
} else {
for (l=this->channels; l; l=l->next)
- gst_pad_push (GST_SF_CHANNEL (l)->pad, gst_event_new (GST_EVENT_EOS));
+ gst_pad_push (GST_SF_CHANNEL (l)->pad, (GstBuffer*)gst_event_new (GST_EVENT_EOS));
gst_element_set_eos (element);
}
}
} else {
- /* unimplemented */
- }
-}
+ sf_count_t written, num_to_write;
+ gint i, j;
+ int buffer_frames = this->buffer_frames;
+ int nchannels = this->numchannels;
+ GstSFChannel *channel = NULL;
+ gfloat *data;
+ gfloat *buf = this->buffer;
+ GstBuffer *in;
-static GstElementStateReturn
-gst_sf_change_state (GstElement *element)
-{
- g_return_val_if_fail (GST_IS_SF (element), GST_STATE_FAILURE);
+ /* the problem: we can't allocate a buffer for pulled data before caps is
+ * set, and we can't open the file without the sample rate from the
+ * caps... */
- /* if going to NULL then close the file */
- if (GST_STATE_PENDING (element) == GST_STATE_NULL)
- if (GST_FLAG_IS_SET (element, GST_SF_OPEN))
- gst_sf_close_file (GST_SF (element));
+ num_to_write = buffer_frames;
- if (GST_ELEMENT_CLASS (parent_class)->change_state)
- return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+ INFO_OBJ (this, "looping, buffer_frames=%d, nchannels=%d", buffer_frames, nchannels);
- return GST_STATE_SUCCESS;
+ for (i=0,l=this->channels; l; l=l->next,i++) {
+ channel = GST_SF_CHANNEL (l);
+
+ in = gst_pad_pull (channel->pad);
+
+ if (buffer_frames == 0) {
+ /* pulling a buffer from the pad should have caused capsnego to occur,
+ which then would set this->buffer_frames to a new value */
+ buffer_frames = this->buffer_frames;
+ if (buffer_frames == 0) {
+ gst_element_error (element, "Caps were never set, bailing...");
+ return;
+ }
+ buf = this->buffer;
+ num_to_write = buffer_frames;
+ }
+
+ if (!GST_FLAG_IS_SET (this, GST_SF_OPEN))
+ if (!gst_sf_open_file (this))
+ return; /* we've already set gst_element_error */
+
+ if (GST_IS_EVENT (in)) {
+ num_to_write = 0;
+ } else {
+ data = (gfloat*)GST_BUFFER_DATA (in);
+ num_to_write = MIN (num_to_write, GST_BUFFER_SIZE (in) / sizeof (gfloat));
+ for (j=0; j<num_to_write; j++)
+ buf[j * nchannels + i % nchannels] = data[j];
+ }
+ gst_data_unref ((GstData*)in);
+ }
+
+ if (num_to_write) {
+ written = sf_writef_float (this->file, buf, num_to_write);
+ if (written != num_to_write)
+ gst_element_error (element, "Error writing file: %s", sf_strerror (this->file));
+ }
+
+ this->time += num_to_write * (GST_SECOND / this->rate);
+ gst_audio_clock_update_time ((GstAudioClock*)this->provided_clock, this->time);
+
+ if (num_to_write != buffer_frames)
+ gst_element_set_eos (element);
+ }
}
static gboolean
@@ -606,6 +796,13 @@ plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
+ if (!gst_library_load ("gstaudio"))
+ return FALSE;
+
+ GST_DEBUG_CATEGORY_INIT (gstsf_debug, "sf",
+ GST_DEBUG_FG_WHITE | GST_DEBUG_BG_GREEN | GST_DEBUG_BOLD,
+ "libsndfile plugin");
+
factory = gst_element_factory_new ("sfsrc", GST_TYPE_SFSRC,
&sfsrc_details);
g_return_val_if_fail (factory != NULL, FALSE);
diff --git a/ext/sndfile/gstsf.h b/ext/sndfile/gstsf.h
index 1a8e6eaa..3197effd 100644
--- a/ext/sndfile/gstsf.h
+++ b/ext/sndfile/gstsf.h
@@ -1,9 +1,5 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2000 Wim Taymans <wtay@chello.be>
- * 2003 Andy Wingo <wingo at pobox dot com>
- *
- * gstsf.c: libsndfile plugin for GStreamer
+/* GStreamer libsndfile plugin
+ * Copyright (C) 2003 Andy Wingo <wingo at pobox dot com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@@ -87,6 +83,8 @@ struct _GstSF {
GstElement element;
GList *channels;
+ GstClock *clock, *provided_clock;
+
gchar *filename;
SNDFILE *file;
void *buffer;
@@ -98,7 +96,11 @@ struct _GstSF {
gint format_major;
gint format_subtype;
gint format;
+
gint rate;
+ gint buffer_frames;
+
+ guint64 time;
};
struct _GstSFClass {