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actually recurse into sndfile if we are able big ladspa cleanups, mainly to comply with the buffer-frames caps proper...

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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.
This commit is contained in:
Andy Wingo 2003-07-16 16:08:13 +00:00
parent 0e04196b71
commit 2ff63e563b
12 changed files with 945 additions and 641 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
common

@ -1 +1 @@
Subproject commit 74856703c922315f64b9314547966bd4963db968
Subproject commit 063ebfd201ca87f316bf478d19440a585ce0af2c

2
ext/Makefile.am
View file

@ -273,7 +273,7 @@ SUBDIRS=$(A52DEC_DIR) $(AALIB_DIR) $(ALSA_DIR) \
$(MAD_DIR) $(MATROSKA_DIR) $(MIKMOD_DIR) \
$(MPEG2DEC_DIR) $(PANGO_DIR) $(RAW1394_DIR) \
$(SDL_DIR) $(SHOUT_DIR) $(SIDPLAY_DIR) \
$(SMOOTHWAVE_DIR) $(SWFDEC_DIR) $(TARKIN_DIR) \
$(SMOOTHWAVE_DIR) $(SNDFILE_DIR) $(SWFDEC_DIR) $(TARKIN_DIR) \
$(VORBIS_DIR) $(XVID_DIR) $(SNAPSHOT_DIR)
DIST_SUBDIRS=\

847
ext/ladspa/gstladspa.c
View file

File diff suppressed because it is too large Load diff

5
ext/ladspa/gstladspa.h
View file

@ -60,12 +60,11 @@ struct _GstLADSPA {
GstPad **sinkpads,
**srcpads;
GstBufferPool *bufpool;
gboolean newcaps, activated;
gboolean activated;
gint samplerate, buffersize, numbuffers;
gint samplerate, buffer_frames;
gint64 timestamp;
gboolean inplace_broken;
};

429
ext/sndfile/gstsf.c
View file

@ -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_loop (GstElement *element);
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 GstPad* gst_sf_request_new_pad (GstElement *element, GstPadTemplate *templ,
const gchar *unused);
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 GstPadLinkReturn gst_sf_link (GstPad *pad, GstCaps *caps);
static GstPadLinkReturn gst_sf_link (GstPad *pad, GstCaps *caps);
static void gst_sf_loop (GstElement *element);
static GstClockTime gst_sf_get_time (GstClock *clock, gpointer data);
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);
gst_object_unparent (GST_OBJECT (this->provided_clock));
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);
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;
@ -570,42 +714,95 @@ 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;
/* 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... */
num_to_write = buffer_frames;
INFO_OBJ (this, "looping, buffer_frames=%d, nchannels=%d", buffer_frames, nchannels);
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 GstElementStateReturn
gst_sf_change_state (GstElement *element)
{
g_return_val_if_fail (GST_IS_SF (element), GST_STATE_FAILURE);
/* 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));
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static gboolean
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);

14
ext/sndfile/gstsf.h
View file

@ -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 {

2
gst-libs/ext/ffmpeg/Tag
View file

@ -1 +1 @@
2003-06-09 22:00 GMT
2003-07-05 22:00 GMT

4
gst-libs/gst/audio/Makefile.am
View file

@ -2,10 +2,10 @@ librarydir = $(libdir)/gstreamer-@GST_MAJORMINOR@
library_LTLIBRARIES = libgstaudio.la
libgstaudio_la_SOURCES = audio.c
libgstaudio_la_SOURCES = audio.c audioclock.c
libgstaudioincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/audio
libgstaudioinclude_HEADERS = audio.h
libgstaudioinclude_HEADERS = audio.h audioclock.h
libgstaudio_la_LIBADD =
libgstaudio_la_CFLAGS = $(GST_CFLAGS) $(GST_OPT_CFLAGS)

2
gst-libs/gst/audio/audio.c
View file

@ -177,7 +177,7 @@ gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
gst_plugin_set_longname (plugin, "Convenience routines for audio plugins");
gst_plugin_set_longname (plugin, "Support services for audio plugins");
return TRUE;
}

2
gst-libs/gst/audio/audio.h
View file

@ -20,6 +20,8 @@
#include <gst/gst.h>
#include <gst/audio/audioclock.h>
/* For people that are looking at this source: the purpose of these defines is
* to make GstCaps a bit easier, in that you don't have to know all of the
* properties that need to be defined. you can just use these macros. currently

194
gst-libs/gst/audio/audioclock.c Normal file
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@ -0,0 +1,194 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* audioclock.c: Clock for use by audio plugins
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include "audioclock.h"
static void gst_audio_clock_class_init (GstAudioClockClass *klass);
static void gst_audio_clock_init (GstAudioClock *clock);
static GstClockTime gst_audio_clock_get_internal_time (GstClock *clock);
static GstClockReturn gst_audio_clock_id_wait_async (GstClock *clock,
GstClockEntry *entry);
static void gst_audio_clock_id_unschedule (GstClock *clock,
GstClockEntry *entry);
static GstSystemClockClass *parent_class = NULL;
/* static guint gst_audio_clock_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_audio_clock_get_type (void)
{
static GType clock_type = 0;
if (!clock_type) {
static const GTypeInfo clock_info = {
sizeof (GstAudioClockClass),
NULL,
NULL,
(GClassInitFunc) gst_audio_clock_class_init,
NULL,
NULL,
sizeof (GstAudioClock),
4,
(GInstanceInitFunc) gst_audio_clock_init,
NULL
};
clock_type = g_type_register_static (GST_TYPE_SYSTEM_CLOCK, "GstAudioClock",
&clock_info, 0);
}
return clock_type;
}
static void
gst_audio_clock_class_init (GstAudioClockClass *klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstClockClass *gstclock_class;
gobject_class = (GObjectClass*) klass;
gstobject_class = (GstObjectClass*) klass;
gstclock_class = (GstClockClass*) klass;
parent_class = g_type_class_ref (GST_TYPE_SYSTEM_CLOCK);
gstclock_class->get_internal_time = gst_audio_clock_get_internal_time;
gstclock_class->wait_async = gst_audio_clock_id_wait_async;
gstclock_class->unschedule = gst_audio_clock_id_unschedule;
}
static void
gst_audio_clock_init (GstAudioClock *clock)
{
gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
clock->prev1 = 0;
clock->prev2 = 0;
}
GstClock*
gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func, gpointer user_data)
{
GstAudioClock *aclock = GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
aclock->func = func;
aclock->user_data = user_data;
aclock->adjust = 0;
return (GstClock*)aclock;
}
void
gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
{
GTimeVal timeval;
GstClockTime time;
GstClockTime audiotime;
g_get_current_time (&timeval);
time = GST_TIMEVAL_TO_TIME (timeval);
audiotime = aclock->func ((GstClock*)aclock, aclock->user_data);
if (active) {
aclock->adjust = time - audiotime;
}
else {
aclock->adjust = audiotime - time;
}
aclock->active = active;
}
static GstClockTime
gst_audio_clock_get_internal_time (GstClock *clock)
{
GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
if (aclock->active) {
GstClockTime audiotime;
audiotime = aclock->func (clock, aclock->user_data) + aclock->adjust;
return audiotime;
}
else {
GstClockTime time;
GTimeVal timeval;
g_get_current_time (&timeval);
time = GST_TIMEVAL_TO_TIME (timeval);
return time;
}
}
void
gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time)
{
/* I don't know of a purpose in updating these; perhaps they can be removed */
aclock->prev2 = aclock->prev1;
aclock->prev1 = time;
/* FIXME: the wait_async subsystem should be made threadsafe, but I don't want
* to lock and unlock a mutex on every iteration... */
while (aclock->async_entries) {
GstClockEntry *entry = (GstClockEntry*)aclock->async_entries->data;
if (entry->time > time)
break;
entry->func ((GstClock*)aclock, time, entry, entry->user_data);
aclock->async_entries = g_slist_delete_link (aclock->async_entries,
aclock->async_entries);
/* do I need to free the entry? */
}
}
static gint
compare_clock_entries (GstClockEntry *entry1, GstClockEntry *entry2)
{
return entry1->time - entry2->time;
}
static GstClockReturn
gst_audio_clock_id_wait_async (GstClock *clock, GstClockEntry *entry)
{
GstAudioClock *aclock = (GstAudioClock*)clock;
aclock->async_entries = g_slist_insert_sorted (aclock->async_entries,
entry,
(GCompareFunc)compare_clock_entries);
/* is this the proper return val? */
return GST_CLOCK_EARLY;
}
static void
gst_audio_clock_id_unschedule (GstClock *clock, GstClockEntry *entry)
{
GstAudioClock *aclock = (GstAudioClock*)clock;
aclock->async_entries = g_slist_remove (aclock->async_entries,
entry);
}

83
gst-libs/gst/audio/audioclock.h Normal file
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@ -0,0 +1,83 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* audioclock.h: Clock for use by audio plugins
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_AUDIO_CLOCK_H__
#define __GST_AUDIO_CLOCK_H__
#include <gst/gstsystemclock.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#define GST_TYPE_AUDIO_CLOCK \
(gst_audio_clock_get_type())
#define GST_AUDIO_CLOCK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CLOCK,GstAudioClock))
#define GST_AUDIO_CLOCK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CLOCK,GstAudioClockClass))
#define GST_IS_AUDIO_CLOCK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CLOCK))
#define GST_IS_AUDIO_CLOCK_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CLOCK))
typedef struct _GstAudioClock GstAudioClock;
typedef struct _GstAudioClockClass GstAudioClockClass;
typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock *clock, gpointer user_data);
struct _GstAudioClock {
GstSystemClock clock;
GstClockTime prev1, prev2;
/* --- protected --- */
GstAudioClockGetTimeFunc func;
gpointer user_data;
GstClockTimeDiff adjust;
GSList *async_entries;
gboolean active;
};
struct _GstAudioClockClass {
GstSystemClockClass parent_class;
};
GType gst_audio_clock_get_type (void);
GstClock* gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func,
gpointer user_data);
void gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active);
void gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_AUDIO_CLOCK_H__ */