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From: <ensonic@fr...> - 2005-08-23 21:32:46
|
CVS Root: /cvs/gstreamer
Module: gstreamer
Changes by: ensonic
Date: Tue Aug 23 2005 14:32:43 PDT
Log message:
* check/gst-libs/controller.c: (GST_START_TEST),
(gst_controller_suite):
* docs/gst/tmpl/gstcaps.sgml:
* docs/gst/tmpl/gstghostpad.sgml:
* docs/gst/tmpl/gstquery.sgml:
* docs/gst/tmpl/gstutils.sgml:
* libs/gst/controller/gst-helper.c: (gst_object_set_controller),
(gst_object_sink_values), (gst_object_get_value_arrays),
(gst_object_get_value_array):
gracefully handle helper method calls to objects that are not beeing
controlled, added test case for that
Modified files:
. : ChangeLog
check/gst-libs : controller.c
docs/gst/tmpl : gstcaps.sgml gstghostpad.sgml gstquery.sgml
gstutils.sgml
libs/gst/controller: gst-helper.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/ChangeLog.diff?r1=1.1386&r2=1.1387
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/check/gst-libs/controller.c.diff?r1=1.5&r2=1.6
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/gstcaps.sgml.diff?r1=1.50&r2=1.51
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/gstghostpad.sgml.diff?r1=1.8&r2=1.9
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/gstquery.sgml.diff?r1=1.13&r2=1.14
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/gstutils.sgml.diff?r1=1.32&r2=1.33
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/libs/gst/controller/gst-helper.c.diff?r1=1.5&r2=1.6
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gstreamer/ChangeLog,v
retrieving revision 1.1386
retrieving revision 1.1387
diff -u -d -r1.1386 -r1.1387
--- ChangeLog 23 Aug 2005 18:17:01 -0000 1.1386
+++ ChangeLog 23 Aug 2005 21:32:31 -0000 1.1387
@@ -1,3 +1,17 @@
+2005-08-24 Stefan Kost <ensonic@...>
+
+ * check/gst-libs/controller.c: (GST_START_TEST),
+ (gst_controller_suite):
+ * docs/gst/tmpl/gstcaps.sgml:
+ * docs/gst/tmpl/gstghostpad.sgml:
+ * docs/gst/tmpl/gstquery.sgml:
+ * docs/gst/tmpl/gstutils.sgml:
+ * libs/gst/controller/gst-helper.c: (gst_object_set_controller),
+ (gst_object_sink_values), (gst_object_get_value_arrays),
+ (gst_object_get_value_array):
+ gracefully handle helper method calls to objects that are not beeing
+ controlled, added test case for that
2005-08-23 Wim Taymans <wim@...>
* gst/gstevent.c: (_gst_event_copy), (gst_event_new_custom),
Index: controller.c
RCS file: /cvs/gstreamer/gstreamer/check/gst-libs/controller.c,v
retrieving revision 1.5
retrieving revision 1.6
diff -u -d -r1.5 -r1.6
--- controller.c 20 Aug 2005 12:14:27 -0000 1.5
+++ controller.c 23 Aug 2005 21:32:31 -0000 1.6
@@ -330,8 +330,24 @@
GST_END_TEST;
-/* @TODO write more tests (using an internal element that has controlable params)
- */
+/* tests if we can run helper methods against any GObject */
+GST_START_TEST (controller_helper_any_gobject)
+{
+ GstElement *elem;
+ gboolean res;
+ elem = gst_element_factory_make ("bin", "test_elem");
+ /* that element is not controllable */
+ res = gst_object_sink_values (G_OBJECT (elem), 0LL);
+ fail_unless (res == FALSE, NULL);
+ g_object_unref (elem);
+}
+GST_END_TEST;
Suite *
gst_controller_suite (void)
{
@@ -346,6 +362,7 @@
tcase_add_test (tc, controller_param_twice);
tcase_add_test (tc, controller_finalize);
tcase_add_test (tc, controller_interpolate_none);
+ tcase_add_test (tc, controller_helper_any_gobject);
return s;
}
Index: gstcaps.sgml
RCS file: /cvs/gstreamer/gstreamer/docs/gst/tmpl/gstcaps.sgml,v
retrieving revision 1.50
retrieving revision 1.51
diff -u -d -r1.50 -r1.51
--- gstcaps.sgml 5 Aug 2005 10:02:43 -0000 1.50
+++ gstcaps.sgml 23 Aug 2005 21:32:31 -0000 1.51
@@ -160,6 +160,16 @@
@Returns:
+<!-- ##### FUNCTION gst_caps_copy_nth ##### -->
+<para>
+</para>
+@caps:
+@nth:
+@Returns:
<!-- ##### FUNCTION gst_static_caps_get ##### -->
<para>
Index: gstghostpad.sgml
RCS file: /cvs/gstreamer/gstreamer/docs/gst/tmpl/gstghostpad.sgml,v
retrieving revision 1.8
retrieving revision 1.9
diff -u -d -r1.8 -r1.9
--- gstghostpad.sgml 5 Aug 2005 10:02:43 -0000 1.8
+++ gstghostpad.sgml 23 Aug 2005 21:32:31 -0000 1.9
@@ -35,3 +35,13 @@
@pad:
+<!-- ##### FUNCTION gst_ghost_pad_set_target ##### -->
+@gpad:
+@newtarget:
Index: gstquery.sgml
RCS file: /cvs/gstreamer/gstreamer/docs/gst/tmpl/gstquery.sgml,v
retrieving revision 1.13
retrieving revision 1.14
diff -u -d -r1.13 -r1.14
--- gstquery.sgml 5 Aug 2005 10:02:43 -0000 1.13
+++ gstquery.sgml 23 Aug 2005 21:32:31 -0000 1.14
@@ -18,6 +18,15 @@
<!-- ##### SECTION Stability_Level ##### -->
+<!-- ##### STRUCT GstQuery ##### -->
+@mini_object:
+@type:
+@structure:
<!-- ##### ENUM GstQueryType ##### -->
Standard predefined Query types
Index: gstutils.sgml
RCS file: /cvs/gstreamer/gstreamer/docs/gst/tmpl/gstutils.sgml,v
retrieving revision 1.32
retrieving revision 1.33
diff -u -d -r1.32 -r1.33
--- gstutils.sgml 15 Aug 2005 16:57:33 -0000 1.32
+++ gstutils.sgml 23 Aug 2005 21:32:31 -0000 1.33
@@ -65,6 +65,50 @@
@element:
+<!-- ##### FUNCTION gst_bin_watch_for_state_change ##### -->
+@bin:
+<!-- ##### MACRO GST_BOILERPLATE_WITH_INTERFACE ##### -->
+@type_as_function:
+@parent_type:
+@\
+ parent_type_as_macro:
+ parent_type_as_macro:
+@interface_type:
+@interface_type_as_macro:
+ interface_as_function:
+ interface_as_function:
<!-- ##### MACRO GST_BOILERPLATE_FULL ##### -->
Index: gst-helper.c
RCS file: /cvs/gstreamer/gstreamer/libs/gst/controller/gst-helper.c,v
--- gst-helper.c 11 Aug 2005 17:39:48 -0000 1.5
+++ gst-helper.c 23 Aug 2005 21:32:31 -0000 1.6
@@ -131,10 +131,11 @@
g_return_val_if_fail (G_IS_OBJECT (object), FALSE);
g_return_val_if_fail (controller, FALSE);
- ctrl = g_object_get_qdata (object, controller_key);
- g_return_val_if_fail (!ctrl, FALSE);
- g_object_set_qdata (object, controller_key, controller);
- return (TRUE);
+ if (!(ctrl = g_object_get_qdata (object, controller_key))) {
+ g_object_set_qdata (object, controller_key, controller);
+ return (TRUE);
+ }
+ return (FALSE);
/**
@@ -155,9 +156,10 @@
g_return_val_if_fail (GST_CLOCK_TIME_IS_VALID (timestamp), FALSE);
- g_return_val_if_fail (ctrl, FALSE);
- return gst_controller_sink_values (ctrl, timestamp);
+ if ((ctrl = g_object_get_qdata (object, controller_key))) {
+ return gst_controller_sink_values (ctrl, timestamp);
@@ -187,9 +189,10 @@
- return gst_controller_get_value_arrays (ctrl, timestamp, value_arrays);
+ return gst_controller_get_value_arrays (ctrl, timestamp, value_arrays);
@@ -217,8 +220,8 @@
-
- return gst_controller_get_value_array (ctrl, timestamp, value_array);
+ return gst_controller_get_value_array (ctrl, timestamp, value_array);
|
|
From: <ensonic@fr...> - 2005-08-23 19:40:27
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: ensonic
Date: Tue Aug 23 2005 12:40:23 PDT
Log message:
* ext/speex/gstspeexenc.h:
Fixed include path of adapter
Modified files:
. : ChangeLog
ext/speex : gstspeexenc.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1809&r2=1.1810
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/speex/gstspeexenc.h.diff?r1=1.8&r2=1.9
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1809
retrieving revision 1.1810
diff -u -d -r1.1809 -r1.1810
--- ChangeLog 23 Aug 2005 18:24:36 -0000 1.1809
+++ ChangeLog 23 Aug 2005 19:40:11 -0000 1.1810
@@ -1,3 +1,8 @@
+2005-08-23 Stefan Kost <ensonic@...>
+
+ * ext/speex/gstspeexenc.h:
+ Fixed include path of adapter
2005-08-23 Wim Taymans <wim@...>
* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init):
Index: gstspeexenc.h
RCS file: /cvs/gstreamer/gst-plugins/ext/speex/gstspeexenc.h,v
retrieving revision 1.8
retrieving revision 1.9
diff -u -d -r1.8 -r1.9
--- gstspeexenc.h 23 Dec 2004 14:45:25 -0000 1.8
+++ gstspeexenc.h 23 Aug 2005 19:40:11 -0000 1.9
@@ -23,7 +23,7 @@
#include <gst/gst.h>
-#include <gst/bytestream/adapter.h>
+#include <gst/base/gstadapter.h>
#include <speex/speex.h>
#include <speex/speex_header.h>
|
|
From: <ds@fr...> - 2005-08-23 19:29:57
|
CVS Root: /cvs/gstreamer
Module: gst-plugins-base
Changes by: ds
Date: Tue Aug 23 2005 12:29:50 PDT
Log message:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
Modified files:
. : ChangeLog
Added files:
gst/audioresample: Makefile.am buffer.c buffer.h debug.c debug.h
functable.c functable.h gstaudioresample.c
gstaudioresample.h resample.c resample.h
resample_chunk.c resample_functable.c
resample_ref.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/ChangeLog.diff?r1=1.1846&r2=1.1847
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/Makefile.am?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/buffer.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/buffer.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/debug.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/debug.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/functable.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/functable.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/gstaudioresample.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/gstaudioresample.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/resample.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/resample.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/resample_chunk.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/resample_functable.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/audioresample/resample_ref.c?rev=1.1&content-type=text/vnd.viewcvs-markup
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/ChangeLog,v
retrieving revision 1.1846
retrieving revision 1.1847
diff -u -d -r1.1846 -r1.1847
--- ChangeLog 23 Aug 2005 18:30:06 -0000 1.1846
+++ ChangeLog 23 Aug 2005 19:29:38 -0000 1.1847
@@ -1,3 +1,20 @@
+2005-08-23 David Schleef <ds@...>
+
+ * gst/audioresample/Makefile.am: Leet audioresampling code
+ * gst/audioresample/buffer.c:
+ * gst/audioresample/buffer.h:
+ * gst/audioresample/debug.c:
+ * gst/audioresample/debug.h:
+ * gst/audioresample/functable.c:
+ * gst/audioresample/functable.h:
+ * gst/audioresample/gstaudioresample.c:
+ * gst/audioresample/gstaudioresample.h:
+ * gst/audioresample/resample.c:
+ * gst/audioresample/resample.h:
+ * gst/audioresample/resample_chunk.c:
+ * gst/audioresample/resample_functable.c:
+ * gst/audioresample/resample_ref.c:
2005-08-23 Wim Taymans <wim@...>
* examples/seeking/seek.c: (make_vorbis_pipeline),
--- NEW FILE: Makefile.am ---
plugin_LTLIBRARIES = libgstaudioresample.la
resample_SOURCES = \
functable.c \
functable.h \
resample.c \
resample_functable.c \
resample_ref.c \
resample_chunk.c \
resample.h \
debug.c \
debug.h \
buffer.c \
buffer.h
libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = $(LIBOIL_LIBS)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
--- NEW FILE: buffer.c ---
#ifndef HAVE_CONFIG_H
#include "config.h"
#endif
#include <audioresample/buffer.h>
#include <glib.h>
#include <string.h>
#include <audioresample/debug.h>
static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer,
void *);
static void audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer,
void *priv);
AudioresampleBuffer *
audioresample_buffer_new (void)
{
AudioresampleBuffer *buffer;
buffer = g_new0 (AudioresampleBuffer, 1);
buffer->ref_count = 1;
return buffer;
}
audioresample_buffer_new_and_alloc (int size)
AudioresampleBuffer *buffer = audioresample_buffer_new ();
buffer->data = g_malloc (size);
buffer->length = size;
buffer->free = audioresample_buffer_free_mem;
audioresample_buffer_new_with_data (void *data, int size)
buffer->data = data;
audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
int length)
AudioresampleBuffer *subbuffer = audioresample_buffer_new ();
if (buffer->parent) {
audioresample_buffer_ref (buffer->parent);
subbuffer->parent = buffer->parent;
} else {
audioresample_buffer_ref (buffer);
subbuffer->parent = buffer;
}
subbuffer->data = buffer->data + offset;
subbuffer->length = length;
subbuffer->free = audioresample_buffer_free_subbuffer;
return subbuffer;
void
audioresample_buffer_ref (AudioresampleBuffer * buffer)
buffer->ref_count++;
audioresample_buffer_unref (AudioresampleBuffer * buffer)
buffer->ref_count--;
if (buffer->ref_count == 0) {
if (buffer->free)
buffer->free (buffer, buffer->priv);
g_free (buffer);
static void
audioresample_buffer_free_mem (AudioresampleBuffer * buffer, void *priv)
g_free (buffer->data);
audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer, void *priv)
audioresample_buffer_unref (buffer->parent);
AudioresampleBufferQueue *
audioresample_buffer_queue_new (void)
return g_new0 (AudioresampleBufferQueue, 1);
int
audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue)
return queue->depth;
audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue)
return queue->offset;
audioresample_buffer_queue_free (AudioresampleBufferQueue * queue)
GList *g;
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
g_list_free (queue->buffers);
g_free (queue);
audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer)
queue->buffers = g_list_append (queue->buffers, buffer);
queue->depth += buffer->length;
audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int length)
AudioresampleBuffer *newbuffer;
AudioresampleBuffer *subbuffer;
g_return_val_if_fail (length > 0, NULL);
if (queue->depth < length) {
return NULL;
RESAMPLE_LOG ("pulling %d, %d available", length, queue->depth);
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length) {
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
subbuffer = audioresample_buffer_new_subbuffer (buffer, length,
buffer->length - length);
g->data = subbuffer;
audioresample_buffer_unref (buffer);
int offset = 0;
newbuffer = audioresample_buffer_new_and_alloc (length);
while (offset < length) {
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length - offset) {
int n = length - offset;
memcpy (newbuffer->data + offset, buffer->data, n);
subbuffer =
audioresample_buffer_new_subbuffer (buffer, n, buffer->length - n);
g->data = subbuffer;
audioresample_buffer_unref (buffer);
offset += n;
} else {
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
queue->buffers = g_list_delete_link (queue->buffers, g);
offset += buffer->length;
}
}
queue->depth -= length;
queue->offset += length;
return newbuffer;
audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int length)
int offset = 0;
RESAMPLE_LOG ("peeking %d, %d available", length, queue->depth);
g = g_list_next (g);
--- NEW FILE: buffer.h ---
#ifndef __AUDIORESAMPLE_BUFFER_H__
#define __AUDIORESAMPLE_BUFFER_H__
typedef struct _AudioresampleBuffer AudioresampleBuffer;
typedef struct _AudioresampleBufferQueue AudioresampleBufferQueue;
struct _AudioresampleBuffer
unsigned char *data;
int length;
int ref_count;
AudioresampleBuffer *parent;
void (*free) (AudioresampleBuffer *, void *);
void *priv;
void *priv2;
};
struct _AudioresampleBufferQueue
GList *buffers;
int depth;
int offset;
AudioresampleBuffer *audioresample_buffer_new (void);
AudioresampleBuffer *audioresample_buffer_new_and_alloc (int size);
AudioresampleBuffer *audioresample_buffer_new_with_data (void *data, int size);
AudioresampleBuffer *audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
int length);
void audioresample_buffer_ref (AudioresampleBuffer * buffer);
void audioresample_buffer_unref (AudioresampleBuffer * buffer);
AudioresampleBufferQueue *audioresample_buffer_queue_new (void);
void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer);
AudioresampleBuffer *audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
AudioresampleBuffer *audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
--- NEW FILE: debug.c ---
#ifdef HAVE_CONFIG_H
#include <stdio.h>
#include <debug.h>
static const char *resample_debug_level_names[] = {
"NONE",
"ERROR",
"WARNING",
"INFO",
"DEBUG",
"LOG"
static int resample_debug_level = RESAMPLE_LEVEL_ERROR;
resample_debug_log (int level, const char *file, const char *function,
int line, const char *format, ...)
#ifndef GLIB_COMPAT
va_list varargs;
char *s;
if (level > resample_debug_level)
return;
va_start (varargs, format);
s = g_strdup_vprintf (format, varargs);
va_end (varargs);
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
resample_debug_level_names[level], file, line, function, s);
g_free (s);
#else
char s[1000];
vsnprintf (s, 999, format, varargs);
resample_debug_set_level (int level)
resample_debug_level = level;
resample_debug_get_level (void)
return resample_debug_level;
--- NEW FILE: debug.h ---
#ifndef __RESAMPLE_DEBUG_H__
#define __RESAMPLE_DEBUG_H__
enum
RESAMPLE_LEVEL_NONE = 0,
RESAMPLE_LEVEL_ERROR,
RESAMPLE_LEVEL_WARNING,
RESAMPLE_LEVEL_INFO,
RESAMPLE_LEVEL_DEBUG,
RESAMPLE_LEVEL_LOG
#define RESAMPLE_ERROR(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_ERROR, __VA_ARGS__)
#define RESAMPLE_WARNING(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_WARNING, __VA_ARGS__)
#define RESAMPLE_INFO(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_INFO, __VA_ARGS__)
#define RESAMPLE_DEBUG(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_DEBUG, __VA_ARGS__)
#define RESAMPLE_LOG(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_LOG, __VA_ARGS__)
#define RESAMPLE_DEBUG_LEVEL(level,...) \
resample_debug_log ((level), __FILE__, __FUNCTION__, __LINE__, __VA_ARGS__)
void resample_debug_log (int level, const char *file, const char *function,
int line, const char *format, ...);
void resample_debug_set_level (int level);
int resample_debug_get_level (void);
--- NEW FILE: functable.c ---
/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@...>
*
* 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 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 <config.h>
#include <math.h>
#include <stdlib.h>
#include <audioresample/functable.h>
functable_func_sinc (double *fx, double *dfx, double x, void *closure)
if (x == 0) {
*fx = 1;
*dfx = 0;
*fx = sin (x) / x;
*dfx = (cos (x) - sin (x) / x) / x;
functable_func_boxcar (double *fx, double *dfx, double x, void *closure)
double width = *(double *) closure;
if (x < width && x > -width) {
*fx = 0;
*dfx = 0;
functable_func_hanning (double *fx, double *dfx, double x, void *closure)
x /= width;
*fx = (1 - x * x) * (1 - x * x);
*dfx = -2 * 2 * x / width * (1 - x * x);
Functable *
functable_new (void)
Functable *ft;
ft = malloc (sizeof (Functable));
memset (ft, 0, sizeof (Functable));
return ft;
functable_free (Functable * ft)
free (ft);
functable_set_length (Functable * t, int length)
t->length = length;
functable_set_offset (Functable * t, double offset)
t->offset = offset;
functable_set_multiplier (Functable * t, double multiplier)
t->multiplier = multiplier;
functable_calculate (Functable * t, FunctableFunc func, void *closure)
int i;
double x;
if (t->fx)
free (t->fx);
if (t->dfx)
free (t->dfx);
t->fx = malloc (sizeof (double) * (t->length + 1));
t->dfx = malloc (sizeof (double) * (t->length + 1));
t->inv_multiplier = 1.0 / t->multiplier;
for (i = 0; i < t->length + 1; i++) {
x = t->offset + t->multiplier * i;
func (&t->fx[i], &t->dfx[i], x, closure);
functable_calculate_multiply (Functable * t, FunctableFunc func, void *closure)
double afx, adfx, bfx, bdfx;
afx = t->fx[i];
adfx = t->dfx[i];
func (&bfx, &bdfx, x, closure);
t->fx[i] = afx * bfx;
t->dfx[i] = afx * bdfx + adfx * bfx;
double
functable_evaluate (Functable * t, double x)
double f0, f1, w0, w1;
double x2, x3;
double w;
if (x < t->offset || x > (t->offset + t->length * t->multiplier)) {
RESAMPLE_DEBUG ("x out of range %g", x);
x -= t->offset;
x *= t->inv_multiplier;
i = floor (x);
x -= i;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
/*w = t->fx[i] * (1-x) + t->fx[i+1] * x; */
return w;
functable_fir (Functable * t, double x, int n, double *data, int len)
int i, j;
double sum;
x /= t->multiplier;
sum = 0;
for (j = 0; j < len; j++) {
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
sum += data[j * 2] * w;
i += n;
return sum;
functable_fir2 (Functable * t, double *r0, double *r1, double x,
int n, double *data, int len)
double sum0, sum1;
double floor_x;
floor_x = floor (x);
i = floor_x;
x -= floor_x;
sum0 = 0;
sum1 = 0;
sum0 += data[j * 2] * w;
sum1 += data[j * 2 + 1] * w;
*r0 = sum0;
*r1 = sum1;
--- NEW FILE: functable.h ---
* Copyright (C) <2001> David Schleef <ds@...>
#ifndef __FUNCTABLE_H__
#define __FUNCTABLE_H__
typedef void FunctableFunc (double *fx, double *dfx, double x, void *closure);
typedef struct _Functable Functable;
struct _Functable {
double offset;
double multiplier;
double inv_multiplier;
double *fx;
double *dfx;
Functable *functable_new (void);
void functable_setup (Functable *t);
void functable_free (Functable *t);
void functable_set_length (Functable *t, int length);
void functable_set_offset (Functable *t, double offset);
void functable_set_multiplier (Functable *t, double multiplier);
void functable_calculate (Functable *t, FunctableFunc func, void *closure);
void functable_calculate_multiply (Functable *t, FunctableFunc func, void *closure);
double functable_evaluate (Functable *t,double x);
double functable_fir(Functable *t,double x0,int n,double *data,int len);
void functable_fir2(Functable *t,double *r0, double *r1, double x0,
int n,double *data,int len);
void functable_func_sinc(double *fx, double *dfx, double x, void *closure);
void functable_func_boxcar(double *fx, double *dfx, double x, void *closure);
void functable_func_hanning(double *fx, double *dfx, double x, void *closure);
#endif /* __PRIVATE_H__ */
--- NEW FILE: gstaudioresample.c ---
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@...>
* Copyright (C) 2003,2004 David A. Schleef <ds@...>
* version 2 of the License, or (at your option) any later version.
/* Element-Checklist-Version: 5 */
/*#define DEBUG_ENABLED */
#include "gstaudioresample.h"
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
#define GST_CAT_DEFAULT audioresample_debug
/* elementfactory information */
static GstElementDetails gst_audioresample_details =
GST_ELEMENT_DETAILS ("Audio scaler",
"Filter/Converter/Audio",
"Resample audio",
"David Schleef <ds@...>");
/* Audioresample signals and args */
/* FILL ME */
LAST_SIGNAL
ARG_0,
ARG_FILTERLEN
#define SUPPORTED_CAPS \
GST_STATIC_CAPS (\
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) true")
#if 0
/* disabled because it segfaults */
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) 32")
static GstStaticPadTemplate gst_audioresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_audioresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_base_init (gpointer g_class);
static void gst_audioresample_class_init (AudioresampleClass * klass);
static void gst_audioresample_init (Audioresample * audioresample);
static void gst_audioresample_dispose (GObject * object);
static void gst_audioresample_chain (GstPad * pad, GstData * _data);
static void gst_audioresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audioresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
GType audioresample_get_type (void)
{
static GType audioresample_type = 0;
if (!audioresample_type)
{
static const GTypeInfo audioresample_info = {
sizeof (AudioresampleClass),
gst_audioresample_base_init,
NULL,
(GClassInitFunc) gst_audioresample_class_init,
sizeof (Audioresample), 0,
(GInstanceInitFunc) gst_audioresample_init,};
audioresample_type =
g_type_register_static (GST_TYPE_ELEMENT, "Audioresample",
&audioresample_info, 0);
}
return audioresample_type;
}
static void gst_audioresample_base_init (gpointer g_class)
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_src_template));
gst_static_pad_template_get (&gst_audioresample_sink_template));
gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
static void gst_audioresample_class_init (AudioresampleClass * klass)
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
gobject_class->dispose = gst_audioresample_dispose;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0,
"audioresample element");
static void gst_audioresample_expand_caps (GstCaps * caps)
gint i;
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return;
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0);
static GstCaps *gst_audioresample_getcaps (GstPad * pad)
Audioresample *audioresample;
GstCaps *caps;
GstPad *otherpad;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
caps = gst_pad_get_allowed_caps (otherpad);
gst_audioresample_expand_caps (caps);
return caps;
static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps)
int rate;
GstCaps *copy;
GstStructure *structure;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
if (pad == audioresample->srcpad) {
otherpad = audioresample->sinkpad;
rate = audioresample->i_rate;
} else
{
otherpad = audioresample->srcpad;
rate = audioresample->o_rate;
if (!GST_PAD_IS_NEGOTIATING (otherpad))
if (gst_caps_get_size (caps) > 1)
copy = gst_caps_copy (caps);
structure = gst_caps_get_structure (copy, 0);
if (rate) {
if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate)) {
return copy;
gst_caps_free (copy);
return NULL;
static GstPadLinkReturn gst_audioresample_link (GstPad * pad,
const GstCaps * caps)
int channels;
gboolean ret;
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rate);
ret &= gst_structure_get_int (structure, "channels", &channels);
if (!ret)
return GST_PAD_LINK_REFUSED;
if (gst_pad_is_negotiated (otherpad))
GstCaps *othercaps = gst_caps_copy (caps);
int otherrate;
GstPadLinkReturn linkret;
if (pad == audioresample->srcpad) {
otherrate = audioresample->i_rate;
} else {
otherrate = audioresample->o_rate;
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL);
linkret = gst_pad_try_set_caps (otherpad, othercaps);
if (GST_PAD_LINK_FAILED (linkret)) {
return GST_PAD_LINK_REFUSED;
audioresample->channels = channels;
resample_set_n_channels (audioresample->resample, audioresample->channels);
audioresample->o_rate = rate;
resample_set_output_rate (audioresample->resample, audioresample->o_rate);
GST_DEBUG ("set o_rate to %d", rate);
audioresample->i_rate = rate;
resample_set_input_rate (audioresample->resample, audioresample->i_rate);
GST_DEBUG ("set i_rate to %d", rate);
return GST_PAD_LINK_OK;
static void gst_audioresample_init (Audioresample * audioresample)
ResampleState *r;
audioresample->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioresample_sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad);
gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain);
gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->sinkpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->sinkpad,
gst_audioresample_fixate);
audioresample->srcpad =
(&gst_audioresample_src_template), "src");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad);
gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->srcpad,
gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate);
r = resample_new ();
audioresample->resample = r;
resample_set_filter_length (r, 64);
resample_set_format (r, RESAMPLE_FORMAT_S16);
static void gst_audioresample_dispose (GObject * object)
Audioresample *audioresample = GST_AUDIORESAMPLE (object);
if (audioresample->resample) {
resample_free (audioresample->resample);
G_OBJECT_CLASS (parent_class)->dispose (object);
static void gst_audioresample_chain (GstPad * pad, GstData * _data)
GstBuffer *buf = GST_BUFFER (_data);
guchar *data;
gulong size;
int outsize;
GstBuffer *outbuf;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
if (!GST_IS_BUFFER (_data)) {
gst_pad_push (audioresample->srcpad, _data);
if (audioresample->passthru) {
gst_pad_push (audioresample->srcpad, GST_DATA (buf));
r = audioresample->resample;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG ("got buffer of %ld bytes", size);
resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref,
buf);
outsize = resample_get_output_size (r);
/* FIXME this is audioresample being dumb. dunno why */
if (outsize == 0) {
GST_ERROR ("overriding outbuf size");
outsize = size;
outbuf = gst_buffer_new_and_alloc (outsize);
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
gst_pad_push (audioresample->srcpad, GST_DATA (outbuf));
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d\n",
audioresample->filter_length);
resample_set_filter_length (audioresample->resample,
break;
default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
g_value_set_int (value, audioresample->filter_length);
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
static gboolean plugin_init (GstPlugin * plugin)
resample_init ();
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIORESAMPLE)) {
return FALSE;
return TRUE;
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioresample",
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
--- NEW FILE: gstaudioresample.h ---
#ifndef __AUDIORESAMPLE_H__
#define __AUDIORESAMPLE_H__
#include <gst/gst.h>
#include <audioresample/resample.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIORESAMPLE \
(audioresample_get_type())
#define GST_AUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,Audioresample))
#define GST_AUDIORESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,Audioresample))
#define GST_IS_AUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
#define GST_IS_AUDIORESAMPLE_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
typedef struct _Audioresample Audioresample;
typedef struct _AudioresampleClass AudioresampleClass;
struct _Audioresample {
GstElement element;
GstPad *sinkpad,*srcpad;
gboolean passthru;
gint64 offset;
int i_rate;
int o_rate;
int filter_length;
ResampleState * resample;
struct _AudioresampleClass {
GstElementClass parent_class;
GType gst_audioresample_get_type(void);
G_END_DECLS
#endif /* __AUDIORESAMPLE_H__ */
--- NEW FILE: resample.c ---
#include <limits.h>
#include <liboil/liboil.h>
void resample_scale_ref (ResampleState * r);
void resample_scale_functable (ResampleState * r);
resample_init (void)
static int inited = 0;
if (!inited) {
oil_init ();
inited = 1;
ResampleState *
resample_new (void)
r = malloc (sizeof (ResampleState));
memset (r, 0, sizeof (ResampleState));
r->filter_length = 16;
r->i_start = 0;
if (r->filter_length & 1) {
r->o_start = 0;
r->o_start = r->o_inc * 0.5;
r->queue = audioresample_buffer_queue_new ();
r->out_tmp = malloc (10000 * sizeof (double));
r->need_reinit = 1;
return r;
resample_free (ResampleState * r)
if (r->buffer) {
free (r->buffer);
if (r->ft) {
functable_free (r->ft);
if (r->queue) {
audioresample_buffer_queue_free (r->queue);
if (r->out_tmp) {
free (r->out_tmp);
free (r);
resample_buffer_free (AudioresampleBuffer * buffer, void *priv)
if (buffer->priv2) {
((void (*)(void *)) buffer->priv2) (buffer->priv);
resample_add_input_data (ResampleState * r, void *data, int size,
void (*free_func) (void *), void *closure)
RESAMPLE_DEBUG ("data %p size %d", data, size);
buffer = audioresample_buffer_new_with_data (data, size);
buffer->free = resample_buffer_free;
buffer->priv2 = free_func;
buffer->priv = closure;
audioresample_buffer_queue_push (r->queue, buffer);
resample_input_eos (ResampleState * r)
int sample_size;
sample_size = r->n_channels * resample_format_size (r->format);
buffer = audioresample_buffer_new_and_alloc (sample_size *
(r->filter_length / 2));
memset (buffer->data, 0, buffer->length);
r->eos = 1;
resample_get_output_size (ResampleState * r)
return floor (audioresample_buffer_queue_get_depth (r->queue) * r->o_rate /
r->i_rate);
resample_get_output_data (ResampleState * r, void *data, int size)
r->o_buf = data;
r->o_size = size;
switch (r->method) {
case 0:
resample_scale_ref (r);
case 1:
resample_scale_functable (r);
return size - r->o_size;
resample_set_filter_length (ResampleState * r, int length)
r->filter_length = length;
resample_set_input_rate (ResampleState * r, double rate)
r->i_rate = rate;
resample_set_output_rate (ResampleState * r, double rate)
r->o_rate = rate;
resample_set_n_channels (ResampleState * r, int n_channels)
r->n_channels = n_channels;
resample_set_format (ResampleState * r, ResampleFormat format)
r->format = format;
resample_set_method (ResampleState * r, int method)
r->method = method;
resample_format_size (ResampleFormat format)
switch (format) {
case RESAMPLE_FORMAT_S16:
return 2;
case RESAMPLE_FORMAT_S32:
case RESAMPLE_FORMAT_F32:
return 4;
case RESAMPLE_FORMAT_F64:
return 8;
return 0;
--- NEW FILE: resample.h ---
#ifndef __RESAMPLE_H__
#define __RESAMPLE_H__
typedef enum {
RESAMPLE_FORMAT_S16 = 0,
RESAMPLE_FORMAT_S32,
RESAMPLE_FORMAT_F32,
RESAMPLE_FORMAT_F64
} ResampleFormat;
typedef void (*ResampleCallback) (void *);
typedef struct _ResampleState ResampleState;
struct _ResampleState {
/* parameters */
int n_channels;
ResampleFormat format;
int filter_length;
double i_rate;
double o_rate;
int method;
/* internal parameters */
int need_reinit;
double halftaps;
/* filter state */
void *o_buf;
int o_size;
AudioresampleBufferQueue *queue;
int eos;
int started;
int sample_size;
void *buffer;
int buffer_len;
double i_start;
double o_start;
double i_inc;
double o_inc;
double sinc_scale;
double i_end;
double o_end;
int i_samples;
int o_samples;
//void *i_buf;
Functable *ft;
double *out_tmp;
void resample_init(void);
void resample_cleanup(void);
ResampleState *resample_new (void);
void resample_free (ResampleState *state);
void resample_add_input_data (ResampleState * r, void *data, int size,
ResampleCallback free_func, void *closure);
void resample_input_eos (ResampleState *r);
int resample_get_output_size (ResampleState *r);
int resample_get_output_data (ResampleState *r, void *data, int size);
void resample_set_filter_length (ResampleState *r, int length);
void resample_set_input_rate (ResampleState *r, double rate);
void resample_set_output_rate (ResampleState *r, double rate);
void resample_set_n_channels (ResampleState *r, int n_channels);
void resample_set_format (ResampleState *r, ResampleFormat format);
void resample_set_method (ResampleState *r, int method);
int resample_format_size (ResampleFormat format);
#endif /* __RESAMPLE_H__ */
--- NEW FILE: resample_chunk.c ---
static double
resample_sinc_window (double x, double halfwidth, double scale)
double y;
if (x == 0)
return 1.0;
if (x < -halfwidth || x > halfwidth)
return 0.0;
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
x /= halfwidth;
y *= (1 - x * x) * (1 - x * x);
return y;
resample_scale_chunk (ResampleState * r)
if (r->need_reinit) {
r->sample_size = r->n_channels * resample_format_size (r->format);
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
if (r->buffer)
free (r->buffer);
r->buffer_len = r->sample_size * 1000;
r->buffer = malloc (r->buffer_len);
memset (r->buffer, 0, r->buffer_len);
r->i_inc = r->o_rate / r->i_rate;
r->o_inc = r->i_rate / r->o_rate;
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
r->i_start = -r->i_inc * r->filter_length;
r->need_reinit = 0;
if (r->i_inc < 1.0) {
r->sinc_scale = r->i_inc;
if (r->sinc_scale == 0.5) {
/* strange things happen at integer multiples */
r->sinc_scale = 1.0;
r->sinc_scale = 1.0;
r->sinc_scale = 1.0;
while (r->o_size > 0) {
double midpoint;
int i;
int j;
RESAMPLE_DEBUG ("i_start %g", r->i_start);
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
if (midpoint > 0.5 * r->i_inc) {
RESAMPLE_ERROR ("inconsistent state");
while (midpoint < -0.5 * r->i_inc) {
AudioresampleBuffer *buffer;
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
if (buffer == NULL) {
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
return;
r->i_start += r->i_inc;
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
midpoint += r->i_inc;
memmove (r->buffer, r->buffer + r->sample_size,
r->buffer_len - r->sample_size);
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
r->sample_size);
audioresample_buffer_unref (buffer);
switch (r->format) {
case RESAMPLE_FORMAT_S16:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -32768.0)
acc = -32768.0;
if (acc > 32767.0)
acc = 32767.0;
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_S32:
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
if (acc < -2147483648.0)
acc = -2147483648.0;
if (acc > 2147483647.0)
acc = 2147483647.0;
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
case RESAMPLE_FORMAT_F32:
x = *(float *) (r->buffer + i * sizeof (float) +
*(float *) (r->o_buf + i * sizeof (float)) = acc;
case RESAMPLE_FORMAT_F64:
x = *(double *) (r->buffer + i * sizeof (double) +
*(double *) (r->o_buf + i * sizeof (double)) = acc;
r->i_start -= 1.0;
r->o_buf += r->sample_size;
r->o_size -= r->sample_size;
--- NEW FILE: resample_functable.c ---
func_sinc (double *fx, double *dfx, double x, void *closure)
//double scale = *(double *)closure;
double scale = M_PI;
x *= scale;
*dfx = scale * (cos (x) - sin (x) / x) / x;
func_hanning (double *fx, double *dfx, double x, void *closure)
functable_test (Functable * ft, double halfwidth)
for (i = 0; i < 100; i++) {
x = i * 0.1;
printf ("%d %g %g\n", i, resample_sinc_window (x, halfwidth, 1.0),
functable_evaluate (ft, x));
exit (0);
resample_scale_functable (ResampleState * r)
double hanning_width;
r->buffer_len = r->sample_size * r->filter_length;
if (r->ft) {
functable_free (r->ft);
r->ft = functable_new ();
functable_set_length (r->ft, r->filter_length * 16);
functable_set_offset (r->ft, -r->filter_length / 2);
functable_set_multiplier (r->ft, 1 / 16.0);
hanning_width = r->filter_length / 2;
functable_calculate (r->ft, func_sinc, NULL);
functable_calculate_multiply (r->ft, func_hanning, &hanning_width);
//functable_test(r->ft, 0.5 * r->filter_length);
acc += functable_evaluate (r->ft, offset) * x;
//acc += resample_sinc_window (offset, r->filter_length * 0.5, r->sinc_scale) * x;
--- NEW FILE: resample_ref.c ---
resample_scale_ref (ResampleState * r)
|
|
From: <wtay@fr...> - 2005-08-23 18:30:21
|
CVS Root: /cvs/gstreamer
Module: gst-plugins-base
Changes by: wtay
Date: Tue Aug 23 2005 11:30:19 PDT
Log message:
* examples/seeking/seek.c: (make_vorbis_pipeline),
(make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
Small seek updates.
Modified files:
. : ChangeLog
examples/seeking: seek.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/ChangeLog.diff?r1=1.1845&r2=1.1846
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/examples/seeking/seek.c.diff?r1=1.40&r2=1.41
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/ChangeLog,v
retrieving revision 1.1845
retrieving revision 1.1846
diff -u -d -r1.1845 -r1.1846
--- ChangeLog 23 Aug 2005 13:29:16 -0000 1.1845
+++ ChangeLog 23 Aug 2005 18:30:06 -0000 1.1846
@@ -1,3 +1,9 @@
+2005-08-23 Wim Taymans <wim@...>
+
+ * examples/seeking/seek.c: (make_vorbis_pipeline),
+ (make_theora_pipeline), (make_vorbis_theora_pipeline), (do_seek):
+ Small seek updates.
2005-08-23 Andy Wingo <wingo@...>
* gst-libs/gst/audio/gstbaseaudiosrc.c
Index: seek.c
RCS file: /cvs/gstreamer/gst-plugins-base/examples/seeking/seek.c,v
retrieving revision 1.40
retrieving revision 1.41
diff -u -d -r1.40 -r1.41
--- seek.c 16 Aug 2005 14:35:52 -0000 1.40
+++ seek.c 23 Aug 2005 18:30:07 -0000 1.41
@@ -296,7 +296,7 @@
{
GstElement *pipeline, *audio_bin;
GstElement *src, *demux, *decoder, *convert, *audiosink;
- GstPad *seekable;
+ GstPad *pad, *seekable;
pipeline = gst_pipeline_new ("app");
@@ -322,7 +322,12 @@
gst_element_link (decoder, convert);
gst_element_link (convert, audiosink);
- setup_dynamic_link (demux, NULL, gst_element_get_pad (decoder, "sink"), NULL);
+ pad = gst_element_get_pad (decoder, "sink");
+ gst_element_add_pad (audio_bin, gst_ghost_pad_new ("sink", pad));
+ gst_object_unref (pad);
+ setup_dynamic_link (demux, NULL, gst_element_get_pad (audio_bin, "sink"),
+ NULL);
seekable = gst_element_get_pad (decoder, "src");
seekable_pads = g_list_prepend (seekable_pads, seekable);
@@ -337,14 +342,12 @@
GstElement *pipeline, *video_bin;
GstElement *src, *demux, *decoder, *convert, *videosink;
- GstElement *queue;
src = gst_element_factory_make_or_warn (SOURCE, "src");
demux = gst_element_factory_make_or_warn ("oggdemux", "demux");
- queue = gst_element_factory_make_or_warn ("queue", "queue");
decoder = gst_element_factory_make_or_warn ("theoradec", "decoder");
convert = gst_element_factory_make_or_warn ("ffmpegcolorspace", "convert");
videosink = gst_element_factory_make_or_warn (VSINK, "sink");
@@ -355,18 +358,21 @@
gst_bin_add (GST_BIN (pipeline), src);
gst_bin_add (GST_BIN (pipeline), demux);
- gst_bin_add (GST_BIN (pipeline), queue);
gst_bin_add (GST_BIN (video_bin), decoder);
gst_bin_add (GST_BIN (video_bin), convert);
gst_bin_add (GST_BIN (video_bin), videosink);
gst_bin_add (GST_BIN (pipeline), video_bin);
gst_element_link (src, demux);
- gst_element_link (queue, decoder);
gst_element_link (convert, videosink);
- setup_dynamic_link (demux, NULL, gst_element_get_pad (queue, "sink"), NULL);
+ gst_element_add_pad (video_bin, gst_ghost_pad_new ("sink", pad));
+ setup_dynamic_link (demux, NULL, gst_element_get_pad (video_bin, "sink"),
@@ -416,7 +422,7 @@
pad = gst_element_get_pad (a_queue, "sink");
gst_element_add_pad (audio_bin, gst_ghost_pad_new ("sink", pad));
- gst_object_unref (GST_OBJECT_CAST (pad));
setup_dynamic_link (demux, NULL, gst_element_get_pad (audio_bin, "sink"),
NULL);
@@ -439,7 +445,7 @@
pad = gst_element_get_pad (v_queue, "sink");
gst_element_add_pad (video_bin, gst_ghost_pad_new ("sink", pad));
setup_dynamic_link (demux, NULL, gst_element_get_pad (video_bin, "sink"),
@@ -975,10 +981,9 @@
while (walk) {
GstPad *seekable = GST_PAD (walk->data);
- g_print ("seek to %" GST_TIME_FORMAT " on pad %s:%s\n",
+ GST_DEBUG ("seek to %" GST_TIME_FORMAT " on pad %s:%s",
GST_TIME_ARGS (real), GST_DEBUG_PAD_NAME (seekable));
-
s_event = gst_event_new_seek (1.0,
GST_FORMAT_TIME,
GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, real, GST_SEEK_TYPE_NONE, 0);
@@ -993,7 +998,7 @@
GstElement *seekable = GST_ELEMENT (walk->data);
- g_print ("seek to %" GST_TIME_FORMAT " on element %s\n",
+ GST_DEBUG ("seek to %" GST_TIME_FORMAT " on element %s",
GST_TIME_ARGS (real), GST_ELEMENT_NAME (seekable));
|
|
From: <wtay@fr...> - 2005-08-23 18:25:36
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wtay
Date: Tue Aug 23 2005 11:24:48 PDT
Log message:
* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init):
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
Fix property warning.
Modified files:
. : ChangeLog
ext/speex : gstspeexdec.c gstspeexenc.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1808&r2=1.1809
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/speex/gstspeexdec.c.diff?r1=1.23&r2=1.24
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/speex/gstspeexenc.c.diff?r1=1.27&r2=1.28
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1808
retrieving revision 1.1809
diff -u -d -r1.1808 -r1.1809
--- ChangeLog 23 Aug 2005 16:54:38 -0000 1.1808
+++ ChangeLog 23 Aug 2005 18:24:36 -0000 1.1809
@@ -1,5 +1,11 @@
2005-08-23 Wim Taymans <wim@...>
+ * ext/speex/gstspeexdec.c: (gst_speex_dec_class_init):
+ * ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
+ Fix property warning.
+
+2005-08-23 Wim Taymans <wim@...>
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain):
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
Index: gstspeexdec.c
RCS file: /cvs/gstreamer/gst-plugins/ext/speex/gstspeexdec.c,v
retrieving revision 1.23
retrieving revision 1.24
diff -u -d -r1.23 -r1.24
--- gstspeexdec.c 23 Aug 2005 15:14:01 -0000 1.23
+++ gstspeexdec.c 23 Aug 2005 18:24:36 -0000 1.24
@@ -99,15 +99,15 @@
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ gobject_class->set_property = gst_speexdec_set_property;
+ gobject_class->get_property = gst_speexdec_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_ENH,
g_param_spec_boolean ("enh", "Enh", "Enable perceptual enhancement",
DEFAULT_ENH, G_PARAM_READWRITE));
gstelement_class->change_state = speex_dec_change_state;
- gobject_class->set_property = gst_speexdec_set_property;
- gobject_class->get_property = gst_speexdec_get_property;
-
GST_DEBUG_CATEGORY_INIT (speexdec_debug, "speexdec", 0,
"speex decoding element");
}
Index: gstspeexenc.c
RCS file: /cvs/gstreamer/gst-plugins/ext/speex/gstspeexenc.c,v
retrieving revision 1.27
retrieving revision 1.28
diff -u -d -r1.27 -r1.28
--- gstspeexenc.c 5 Jul 2005 10:51:37 -0000 1.27
+++ gstspeexenc.c 23 Aug 2005 18:24:36 -0000 1.28
@@ -191,6 +191,9 @@
+ gobject_class->set_property = gst_speexenc_set_property;
+ gobject_class->get_property = gst_speexenc_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_QUALITY,
g_param_spec_float ("quality", "Quality", "Encoding quality",
0.0, 10.0, DEFAULT_QUALITY, G_PARAM_READWRITE));
@@ -225,9 +228,6 @@
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
- gobject_class->set_property = gst_speexenc_set_property;
- gobject_class->get_property = gst_speexenc_get_property;
gstelement_class->change_state = gst_speexenc_change_state;
|
|
From: <thomasvs@fr...> - 2005-08-23 18:19:25
|
CVS Root: /cvs/gstreamer
Module: gst-plugins-base
Changes by: thomasvs
Date: Tue Aug 23 2005 11:19:22 PDT
Log message:
style fixes
Modified files:
gst/volume : gstvolume.c gstvolume.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/volume/gstvolume.c.diff?r1=1.57&r2=1.58
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst/volume/gstvolume.h.diff?r1=1.13&r2=1.14
====Begin Diffs====
Index: gstvolume.c
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/gst/volume/gstvolume.c,v
retrieving revision 1.57
retrieving revision 1.58
diff -u -d -r1.57 -r1.58
--- gstvolume.c 22 Aug 2005 09:35:57 -0000 1.57
+++ gstvolume.c 23 Aug 2005 18:19:10 -0000 1.58
@@ -67,7 +67,7 @@
PROP_VOLUME
};
-static GstStaticPadTemplate volume_sink_factory =
+static GstStaticPadTemplate volume_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
@@ -84,7 +84,8 @@
"width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")
);
-static GstStaticPadTemplate volume_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+static GstStaticPadTemplate volume_src_template = GST_STATIC_PAD_TEMPLATE
+ ("src",
GST_PAD_SRC,
GST_STATIC_CAPS ("audio/x-raw-float, "
@@ -248,9 +249,9 @@
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&volume_src_factory));
+ gst_static_pad_template_get (&volume_src_template));
- gst_static_pad_template_get (&volume_sink_factory));
+ gst_static_pad_template_get (&volume_sink_template));
gst_element_class_set_details (element_class, &volume_details);
}
Index: gstvolume.h
RCS file: /cvs/gstreamer/gst-plugins-base/gst/volume/gstvolume.h,v
retrieving revision 1.13
retrieving revision 1.14
diff -u -d -r1.13 -r1.14
--- gstvolume.h 22 Aug 2005 09:35:57 -0000 1.13
+++ gstvolume.h 23 Aug 2005 18:19:10 -0000 1.14
@@ -24,6 +24,7 @@
#define __GST_VOLUME_H__
#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
G_BEGIN_DECLS
|
|
From: <wtay@fr...> - 2005-08-23 18:17:17
|
CVS Root: /cvs/gstreamer
Module: gstreamer
Changes by: wtay
Date: Tue Aug 23 2005 11:17:13 PDT
Log message:
* gst/gstevent.c: (_gst_event_copy), (gst_event_new_custom),
(gst_event_new_newsegment), (gst_event_parse_newsegment),
(gst_event_new_tag), (gst_event_parse_tag), (gst_event_new_qos),
(gst_event_parse_qos), (gst_event_new_seek),
(gst_event_parse_seek):
* gst/gstevent.h:
Some more debugging output and doc cleanups.
* gst/gstqueue.c: (gst_queue_handle_sink_event):
Fix possible deadlock.
Modified files:
. : ChangeLog
gst : gstevent.c gstevent.h gstqueue.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/ChangeLog.diff?r1=1.1385&r2=1.1386
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstevent.c.diff?r1=1.60&r2=1.61
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstevent.h.diff?r1=1.63&r2=1.64
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstqueue.c.diff?r1=1.144&r2=1.145
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gstreamer/ChangeLog,v
retrieving revision 1.1385
retrieving revision 1.1386
diff -u -d -r1.1385 -r1.1386
--- ChangeLog 23 Aug 2005 14:25:55 -0000 1.1385
+++ ChangeLog 23 Aug 2005 18:17:01 -0000 1.1386
@@ -1,3 +1,16 @@
+2005-08-23 Wim Taymans <wim@...>
+
+ * gst/gstevent.c: (_gst_event_copy), (gst_event_new_custom),
+ (gst_event_new_newsegment), (gst_event_parse_newsegment),
+ (gst_event_new_tag), (gst_event_parse_tag), (gst_event_new_qos),
+ (gst_event_parse_qos), (gst_event_new_seek),
+ (gst_event_parse_seek):
+ * gst/gstevent.h:
+ Some more debugging output and doc cleanups.
+ * gst/gstqueue.c: (gst_queue_handle_sink_event):
+ Fix possible deadlock.
2005-08-23 Stefan Kost <ensonic@...>
* docs/gst/gstreamer-docs.sgml:
Index: gstevent.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstevent.c,v
retrieving revision 1.60
retrieving revision 1.61
diff -u -d -r1.60 -r1.61
--- gstevent.c 22 Aug 2005 10:37:02 -0000 1.60
+++ gstevent.c 23 Aug 2005 18:17:01 -0000 1.61
@@ -301,6 +301,23 @@
gst_event_new_newsegment (gdouble rate, GstFormat format,
gint64 start_val, gint64 stop_val, gint64 base)
{
+ if (format == GST_FORMAT_TIME) {
+ GST_CAT_INFO (GST_CAT_EVENT,
+ "creating newsegment rate %lf, format GST_FORMAT_TIME, "
+ "start %" GST_TIME_FORMAT ", stop %" GST_TIME_FORMAT
+ ", base %" GST_TIME_FORMAT,
+ rate, GST_TIME_ARGS (start_val),
+ GST_TIME_ARGS (stop_val), GST_TIME_ARGS (base));
+ } else {
+ "creating newsegment rate %lf, format %d, "
+ "start %lld, stop %lld, base %lld",
+ rate, format, start_val, stop_val, base);
+ }
+ if (start_val != -1 && stop_val != -1)
+ g_return_val_if_fail (start_val < stop_val, NULL);
return gst_event_new_custom (GST_EVENT_NEWSEGMENT,
gst_structure_new ("GstEventNewsegment", "rate", G_TYPE_DOUBLE, rate,
"format", GST_TYPE_FORMAT, format,
@@ -345,7 +362,6 @@
*base = g_value_get_int64 (gst_structure_get_value (structure, "base"));
}
-/* tag event */
/**
* gst_event_new_tag:
* @taglist: metadata list
@@ -420,6 +436,11 @@
gst_event_new_qos (gdouble proportion, GstClockTimeDiff diff,
GstClockTime timestamp)
+ GST_CAT_INFO (GST_CAT_EVENT,
+ "creating qos proportion %lf, diff %" GST_TIME_FORMAT
+ ", timestamp %" GST_TIME_FORMAT, proportion,
+ GST_TIME_ARGS (diff), GST_TIME_ARGS (timestamp));
return gst_event_new_custom (GST_EVENT_QOS,
gst_structure_new ("GstEventQOS",
"proportion", G_TYPE_DOUBLE, proportion,
@@ -486,6 +507,21 @@
gst_event_new_seek (gdouble rate, GstFormat format, GstSeekFlags flags,
GstSeekType cur_type, gint64 cur, GstSeekType stop_type, gint64 stop)
+ "creating seek rate %lf, format TIME, flags %d, "
+ "cur_type %d, cur %" GST_TIME_FORMAT ", "
+ "stop_type %d, stop %" GST_TIME_FORMAT,
+ rate, flags, cur_type, GST_TIME_ARGS (cur),
+ stop_type, GST_TIME_ARGS (stop));
+ "creating seek rate %lf, format %d, flags %d, "
+ "cur_type %d, cur %" G_GINT64_FORMAT ", "
+ "stop_type %d, stop %" G_GINT64_FORMAT,
+ rate, format, flags, cur_type, cur, stop_type, stop);
return gst_event_new_custom (GST_EVENT_SEEK,
gst_structure_new ("GstEventSeek", "rate", G_TYPE_DOUBLE, rate,
@@ -539,7 +575,6 @@
*stop = g_value_get_int64 (gst_structure_get_value (structure, "stop"));
-/* navigation event */
* gst_event_new_navigation:
* @structure: description of the event
Index: gstevent.h
RCS file: /cvs/gstreamer/gstreamer/gst/gstevent.h,v
retrieving revision 1.63
retrieving revision 1.64
diff -u -d -r1.63 -r1.64
--- gstevent.h 23 Aug 2005 14:25:55 -0000 1.63
+++ gstevent.h 23 Aug 2005 18:17:01 -0000 1.64
@@ -197,7 +197,6 @@
* @GST_SEEK_FLAG_SEGMENT: perform a segment seek. After the playback
* of the segment completes, no EOS will be emmited but a
* SEGMENT_DONE message will be posted on the bus.
- *
*/
typedef enum {
GST_SEEK_FLAG_NONE = 0,
Index: gstqueue.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstqueue.c,v
retrieving revision 1.144
retrieving revision 1.145
diff -u -d -r1.144 -r1.145
--- gstqueue.c 22 Aug 2005 18:48:48 -0000 1.144
+++ gstqueue.c 23 Aug 2005 18:17:01 -0000 1.145
@@ -526,8 +526,9 @@
/* now unblock the chain function */
GST_QUEUE_MUTEX_LOCK (queue);
queue->srcresult = GST_FLOW_WRONG_STATE;
- /* unblock the loop function */
+ /* unblock the loop and chain functions */
g_cond_signal (queue->item_add);
+ g_cond_signal (queue->item_del);
GST_QUEUE_MUTEX_UNLOCK (queue);
/* make sure it pauses */
|
|
From: <wtay@fr...> - 2005-08-23 16:55:25
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wtay
Date: Tue Aug 23 2005 09:54:50 PDT
Log message:
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain):
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
(gst_rtpamrenc_init), (gst_rtpamrenc_chain):
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
(gst_rtph263penc_flush), (gst_rtph263penc_chain):
Small updates, RFC reference to payload encoders.
Modified files:
. : ChangeLog
gst/rtp : gstrtpamrdec.c gstrtpamrenc.c gstrtph263penc.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1807&r2=1.1808
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/gst/rtp/gstrtpamrdec.c.diff?r1=1.3&r2=1.4
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/gst/rtp/gstrtpamrenc.c.diff?r1=1.3&r2=1.4
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/gst/rtp/gstrtph263penc.c.diff?r1=1.3&r2=1.4
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1807
retrieving revision 1.1808
diff -u -d -r1.1807 -r1.1808
--- ChangeLog 23 Aug 2005 15:14:01 -0000 1.1807
+++ ChangeLog 23 Aug 2005 16:54:38 -0000 1.1808
@@ -1,3 +1,13 @@
+2005-08-23 Wim Taymans <wim@...>
+
+ * gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
+ (gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain):
+ * gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_class_init),
+ (gst_rtpamrenc_init), (gst_rtpamrenc_chain):
+ * gst/rtp/gstrtph263penc.c: (gst_rtph263penc_class_init),
+ (gst_rtph263penc_flush), (gst_rtph263penc_chain):
+ Small updates, RFC reference to payload encoders.
2005-08-23 Ronald S. Bultje <rbultje@...>
* configure.ac:
Index: gstrtpamrdec.c
RCS file: /cvs/gstreamer/gst-plugins/gst/rtp/gstrtpamrdec.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -u -d -r1.3 -r1.4
--- gstrtpamrdec.c 19 Aug 2005 12:44:35 -0000 1.3
+++ gstrtpamrdec.c 23 Aug 2005 16:54:38 -0000 1.4
@@ -21,6 +21,13 @@
#include <string.h>
#include "gstrtpamrdec.h"
+/* references:
+ *
+ * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File Storage Format
+ * for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio
+ * Codecs.
+ */
/* elementfactory information */
static GstElementDetails gst_rtp_amrdec_details = {
"RTP packet parser",
Index: gstrtpamrenc.c
RCS file: /cvs/gstreamer/gst-plugins/gst/rtp/gstrtpamrenc.c,v
--- gstrtpamrenc.c 19 Aug 2005 12:44:35 -0000 1.3
+++ gstrtpamrenc.c 23 Aug 2005 16:54:38 -0000 1.4
@@ -22,6 +22,13 @@
#include "gstrtpamrenc.h"
+ * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
+ * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive
+ * Multi-Rate Wideband (AMR-WB) Audio Codecs.
static GstElementDetails gst_rtp_amrenc_details = {
|
|
From: <rbultje@fr...> - 2005-08-23 15:14:19
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: rbultje
Date: Tue Aug 23 2005 08:14:13 PDT
Log message:
* configure.ac:
* ext/Makefile.am:
* ext/speex/Makefile.am:
* ext/speex/gstspeex.c: (plugin_init):
* ext/speex/gstspeexdec.c: (speex_get_query_types),
(gst_speex_dec_init), (speex_dec_src_query), (speex_dec_src_event),
(speex_dec_event), (speex_dec_chain):
Port speexdec. Leads to some unfamiliar warnings on console,
but works otherwise.
Modified files:
. : ChangeLog configure.ac
ext : Makefile.am
ext/speex : Makefile.am gstspeex.c gstspeexdec.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1806&r2=1.1807
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/configure.ac.diff?r1=1.576&r2=1.577
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/Makefile.am.diff?r1=1.141&r2=1.142
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/speex/Makefile.am.diff?r1=1.4&r2=1.5
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/speex/gstspeex.c.diff?r1=1.8&r2=1.9
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ext/speex/gstspeexdec.c.diff?r1=1.22&r2=1.23
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1806
retrieving revision 1.1807
diff -u -d -r1.1806 -r1.1807
--- ChangeLog 23 Aug 2005 14:10:04 -0000 1.1806
+++ ChangeLog 23 Aug 2005 15:14:01 -0000 1.1807
@@ -1,3 +1,15 @@
+2005-08-23 Ronald S. Bultje <rbultje@...>
+
+ * configure.ac:
+ * ext/Makefile.am:
+ * ext/speex/Makefile.am:
+ * ext/speex/gstspeex.c: (plugin_init):
+ * ext/speex/gstspeexdec.c: (speex_get_query_types),
+ (gst_speex_dec_init), (speex_dec_src_query), (speex_dec_src_event),
+ (speex_dec_event), (speex_dec_chain):
+ Port speexdec. Leads to some unfamiliar warnings on console,
+ but works otherwise.
2005-08-23 Andy Wingo <wingo@...>
* sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name
Index: configure.ac
RCS file: /cvs/gstreamer/gst-plugins/configure.ac,v
retrieving revision 1.576
retrieving revision 1.577
diff -u -d -r1.576 -r1.577
--- configure.ac 22 Aug 2005 11:20:18 -0000 1.576
+++ configure.ac 23 Aug 2005 15:14:01 -0000 1.577
@@ -583,7 +583,50 @@
AC_SUBST(LIBPNG_LIBS)
])
+dnl *** speex >= 1.0.4 or >= 1.1.5 ***
+dnl 1.1.4 and earlier were not API/ABI compatible with 1.0
+dnl 1.1.6 is the first to use a .pc/pkg-config file ***
+dnl speex_jitter.h is 1.1.x only
+translit(dnm, m, l) AM_CONDITIONAL(USE_SPEEX, true)
+GST_CHECK_FEATURE(SPEEX, [speex plug-in], speex, [
+ PKG_CHECK_MODULES(SPEEX, speex >= 1.1.6, [
+ HAVE_SPEEX="yes"
+ AC_SUBST(SPEEX_CFLAGS)
+ AC_SUBST(SPEEX_LIBS)
+ ], [
+ GST_CHECK_LIBHEADER(SPEEX, speex, speex_bits_init, , speex/speex.h, [
+ AC_CHECK_HEADER(speex/speex_jitter.h, [
+ dnl speex 1.1.x :
+ GST_CHECK_LIBHEADER(SPEEX, speex, speex_encode_int, , speex/speex.h, [
+ dnl speex 1.1.5 or + :
+ HAVE_SPEEX="yes"
+ SPEEX_LIBS="-lspeex"
+ AC_SUBST(SPEEX_CFLAGS)
+ AC_SUBST(SPEEX_LIBS)
+ ],[
+ HAVE_SPEEX="no"
+ ])
+ ],[
+ dnl speex 1.0.x :
+ AC_CHECK_DECL(SPEEX_GET_LOOKAHEAD, [
+ dnl speex 1.0.4
+ HAVE_SPEEX="yes"
+ SPEEX_LIBS="-lspeex"
+ AC_SUBST(SPEEX_CFLAGS)
+ AC_SUBST(SPEEX_LIBS)
+ AC_DEFINE_UNQUOTED(SPEEX_1_0, 1,
+ [defined if speex 1.0.x API detected])
+ HAVE_SPEEX="no"
+ AC_MSG_NOTICE(You need at least 1.0.4 to compile the speex plugin)
+ ], [
+#include <speex/speex.h>
+ ])
+ ])
+ ])
+])
dnl also add builddir include for enumtypes and marshal
GST_CFLAGS="-I\$(top_srcdir)/gst-libs -I\$(top_builddir)/gst-libs $GST_CFLAGS $GST_ERROR"
@@ -676,6 +719,7 @@
ext/raw1394/Makefile
ext/shout2/Makefile
ext/sidplay/Makefile
+ext/speex/Makefile
ext/esd/Makefile
ext/mpeg2dec/Makefile
ext/faad/Makefile
Index: Makefile.am
RCS file: /cvs/gstreamer/gst-plugins/ext/Makefile.am,v
retrieving revision 1.141
retrieving revision 1.142
diff -u -d -r1.141 -r1.142
--- Makefile.am 22 Aug 2005 11:20:18 -0000 1.141
+++ Makefile.am 23 Aug 2005 15:14:01 -0000 1.142
@@ -364,11 +364,11 @@
SNAPSHOT_DIR=
# endif
-# if USE_SPEEX
-# SPEEX_DIR=speex
-# else
+if USE_SPEEX
+SPEEX_DIR=speex
+else
SPEEX_DIR=
-# endif
+endif
# if USE_XINE
# XINE_DIR=xine
@@ -455,4 +455,5 @@
mpeg2dec \
jpeg \
libpng \
- lame
+ lame \
+ speex
RCS file: /cvs/gstreamer/gst-plugins/ext/speex/Makefile.am,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -d -r1.4 -r1.5
--- Makefile.am 9 Aug 2003 23:56:39 -0000 1.4
+++ Makefile.am 23 Aug 2005 15:14:01 -0000 1.5
@@ -1,7 +1,8 @@
plugin_LTLIBRARIES = libgstspeex.la
-libgstspeex_la_SOURCES = gstspeex.c gstspeexdec.c gstspeexenc.c
+libgstspeex_la_SOURCES = gstspeex.c gstspeexdec.c
+#gstspeexenc.c
libgstspeex_la_CFLAGS = $(GST_CFLAGS) $(SPEEX_CFLAGS)
libgstspeex_la_LIBADD = $(SPEEX_LIBS)
libgstspeex_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
Index: gstspeex.c
RCS file: /cvs/gstreamer/gst-plugins/ext/speex/gstspeex.c,v
retrieving revision 1.8
retrieving revision 1.9
diff -u -d -r1.8 -r1.9
--- gstspeex.c 28 Sep 2004 16:44:12 -0000 1.8
+++ gstspeex.c 23 Aug 2005 15:14:01 -0000 1.9
@@ -26,16 +26,11 @@
static gboolean
plugin_init (GstPlugin * plugin)
{
- if (!gst_library_load ("gstbytestream"))
- return FALSE;
-
- if (!gst_library_load ("gsttags"))
+#if 0
if (!gst_element_register (plugin, "speexenc", GST_RANK_NONE,
GST_TYPE_SPEEXENC))
return FALSE;
+#endif
if (!gst_element_register (plugin, "speexdec", GST_RANK_PRIMARY,
GST_TYPE_SPEEXDEC))
Index: gstspeexdec.c
RCS file: /cvs/gstreamer/gst-plugins/ext/speex/gstspeexdec.c,v
retrieving revision 1.22
retrieving revision 1.23
diff -u -d -r1.22 -r1.23
--- gstspeexdec.c 5 Jul 2005 10:51:37 -0000 1.22
+++ gstspeexdec.c 23 Aug 2005 15:14:01 -0000 1.23
@@ -23,7 +23,7 @@
#include "gstspeexdec.h"
#include <string.h>
-#include <gst/tag/tag.h>
+//#include <gst/tag/tag.h>
GST_DEBUG_CATEGORY (speexdec_debug);
#define GST_CAT_DEFAULT speexdec_debug
@@ -35,13 +35,6 @@
"Wim Taymans <wim@...>",
};
-/* Filter signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
#define DEFAULT_ENH TRUE
enum
@@ -70,13 +63,12 @@
GST_BOILERPLATE (GstSpeexDec, gst_speex_dec, GstElement, GST_TYPE_ELEMENT);
-static void speex_dec_chain (GstPad * pad, GstData * data);
+static gboolean speex_dec_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn speex_dec_chain (GstPad * pad, GstBuffer * buf);
static GstElementStateReturn speex_dec_change_state (GstElement * element);
-static const GstFormat *speex_dec_get_formats (GstPad * pad);
static gboolean speex_dec_src_event (GstPad * pad, GstEvent * event);
-static gboolean speex_dec_src_query (GstPad * pad,
- GstQueryType query, GstFormat * format, gint64 * value);
+static gboolean speex_dec_src_query (GstPad * pad, GstQuery * query);
static gboolean speex_dec_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value);
@@ -120,41 +112,10 @@
"speex decoding element");
}
-static const GstFormat *
-speex_dec_get_formats (GstPad * pad)
- static GstFormat src_formats[] = {
- GST_FORMAT_BYTES,
- GST_FORMAT_DEFAULT, /* samples in the audio case */
- GST_FORMAT_TIME,
- 0
- };
- static GstFormat sink_formats[] = {
- GST_FORMAT_DEFAULT, /* samples */
- return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats);
-}
-static const GstEventMask *
-speex_get_event_masks (GstPad * pad)
- static const GstEventMask speex_dec_src_event_masks[] = {
- {GST_EVENT_SEEK, GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH},
- {0,}
- return speex_dec_src_event_masks;
static const GstQueryType *
speex_get_query_types (GstPad * pad)
static const GstQueryType speex_dec_src_query_types[] = {
- GST_QUERY_TOTAL,
GST_QUERY_POSITION,
0
};
@@ -169,25 +130,19 @@
gst_pad_new_from_template (gst_static_pad_template_get
(&speex_dec_sink_factory), "sink");
gst_pad_set_chain_function (dec->sinkpad, speex_dec_chain);
- gst_pad_set_formats_function (dec->sinkpad, speex_dec_get_formats);
- gst_pad_set_convert_function (dec->sinkpad, speex_dec_convert);
+ gst_pad_set_event_function (dec->sinkpad, speex_dec_event);
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
dec->srcpad =
(&speex_dec_src_factory), "src");
- gst_pad_use_explicit_caps (dec->srcpad);
- gst_pad_set_event_mask_function (dec->srcpad, speex_get_event_masks);
+ gst_pad_use_fixed_caps (dec->srcpad);
gst_pad_set_event_function (dec->srcpad, speex_dec_src_event);
gst_pad_set_query_type_function (dec->srcpad, speex_get_query_types);
gst_pad_set_query_function (dec->srcpad, speex_dec_src_query);
- gst_pad_set_formats_function (dec->srcpad, speex_dec_get_formats);
- gst_pad_set_convert_function (dec->srcpad, speex_dec_convert);
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->enh = DEFAULT_ENH;
- GST_FLAG_SET (dec, GST_ELEMENT_EVENT_AWARE);
@@ -253,29 +208,26 @@
-speex_dec_src_query (GstPad * pad, GstQueryType query, GstFormat * format,
- gint64 * value)
+speex_dec_src_query (GstPad * pad, GstQuery * query)
- gint64 samples_out = 0;
- GstSpeexDec *dec = GST_SPEEXDEC (gst_pad_get_parent (pad));
- GstFormat my_format = GST_FORMAT_DEFAULT;
- if (query == GST_QUERY_POSITION) {
- samples_out = dec->samples_out;
- } else {
- /* query peer in default format */
- if (!gst_pad_query (GST_PAD_PEER (dec->sinkpad), query, &my_format,
- &samples_out))
- return FALSE;
- }
+ gint64 samples_out = 0, total_samples;
+ GstSpeexDec *dec = GST_SPEEXDEC (GST_OBJECT_PARENT (pad));
+ GstFormat my_format = GST_FORMAT_TIME;
+ GstPad *peer;
- /* and convert to the final format */
- if (!gst_pad_convert (pad, GST_FORMAT_DEFAULT, samples_out, format, value))
+ if (GST_QUERY_TYPE (query) != GST_QUERY_POSITION)
+ return FALSE;
+ if (!(peer = gst_pad_get_peer (dec->sinkpad)))
+ gst_pad_query_position (peer, &my_format, NULL, &total_samples);
+ gst_object_unref (peer);
+ samples_out = dec->samples_out;
+ speex_dec_convert (dec->srcpad, GST_FORMAT_DEFAULT, samples_out,
+ &my_format, &samples_out);
+ speex_dec_convert (dec->srcpad, GST_FORMAT_DEFAULT, total_samples,
+ &my_format, &total_samples);
+ gst_query_set_position (query, GST_FORMAT_TIME, samples_out, total_samples);
- GST_LOG_OBJECT (dec,
- "query %u: peer returned samples_out: %llu - we return %llu (format %u)\n",
- query, samples_out, *value, *format);
return TRUE;
@@ -287,20 +239,24 @@
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
- guint64 value;
- GstFormat my_format = GST_FORMAT_DEFAULT;
+ gint64 cur, stop;
+ GstFormat format, my_format = GST_FORMAT_DEFAULT;
+ GstSeekType cur_type, stop_type;
+ GstSeekFlags flags;
+ gst_event_parse_seek (event, NULL, &format, &flags, &cur_type, &cur,
+ &stop_type, &stop);
/* convert to samples_out */
- res = speex_dec_convert (pad, GST_EVENT_SEEK_FORMAT (event),
- GST_EVENT_SEEK_OFFSET (event), &my_format, &value);
- if (res) {
- GstEvent *real_seek = gst_event_new_seek (
- (GST_EVENT_SEEK_TYPE (event) & ~GST_SEEK_FORMAT_MASK) |
- GST_FORMAT_DEFAULT,
- value);
+ if (speex_dec_convert (pad, format, cur, &my_format, &cur) &&
+ (stop == -1 ||
+ speex_dec_convert (pad, format, stop, &my_format, &stop))) {
+ GstEvent *real_seek = gst_event_new_seek (1.0, GST_FORMAT_DEFAULT,
+ flags, cur_type, cur, stop_type, stop);
res = gst_pad_send_event (GST_PAD_PEER (dec->sinkpad), real_seek);
- }
+ } else
+ res = FALSE;
gst_event_unref (event);
break;
}
@@ -312,16 +268,18 @@
return res;
-static void
-speex_dec_event (GstSpeexDec * dec, GstEvent * event)
+static gboolean
+speex_dec_event (GstPad * pad, GstEvent * event)
- guint64 value, time, bytes;
+ gint64 value, time;
+ GstFormat fmt;
GST_LOG_OBJECT (dec, "handling event");
- case GST_EVENT_DISCONTINUOUS:
- if (gst_event_discont_get_value (event, GST_FORMAT_DEFAULT,
- (gint64 *) & value)) {
+ case GST_EVENT_NEWSEGMENT:
+ gst_event_parse_newsegment (event, NULL, &fmt, &value, NULL, NULL);
+ if (fmt == GST_FORMAT_DEFAULT) {
dec->samples_out = value;
GST_DEBUG_OBJECT (dec,
"setting samples_out to %" G_GUINT64_FORMAT " after discont",
@@ -329,77 +287,66 @@
} else {
GST_WARNING_OBJECT (dec,
"discont event didn't include offset, we might set it wrong now");
+ value = 0;
}
if (dec->packetno < 2) {
if (dec->samples_out != 0)
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
("can't handle discont before parsing first 2 packets"));
dec->packetno = 0;
- gst_pad_push (dec->srcpad, GST_DATA (gst_event_new_discontinuous (FALSE,
- GST_FORMAT_TIME, (guint64) 0, GST_FORMAT_DEFAULT,
- (guint64) 0, GST_FORMAT_BYTES, (guint64) 0, 0)));
+ gst_pad_push_event (dec->srcpad,
+ gst_event_new_newsegment (1.0, GST_FORMAT_TIME,
+ 0, GST_CLOCK_TIME_NONE, 0));
- GstFormat time_format, default_format, bytes_format;
- time_format = GST_FORMAT_TIME;
- default_format = GST_FORMAT_DEFAULT;
- bytes_format = GST_FORMAT_BYTES;
+ GstFormat time_format = GST_FORMAT_TIME;
dec->packetno = 2;
/* if one of them works, all of them work */
if (speex_dec_convert (dec->srcpad, GST_FORMAT_DEFAULT,
- dec->samples_out, &time_format, &time)
- && speex_dec_convert (dec->srcpad, GST_FORMAT_DEFAULT,
- dec->samples_out, &bytes_format, &bytes)) {
- gst_pad_push (dec->srcpad,
- GST_DATA (gst_event_new_discontinuous (FALSE, GST_FORMAT_TIME,
- time, GST_FORMAT_DEFAULT, dec->samples_out,
- GST_FORMAT_BYTES, bytes, 0)));
+ dec->samples_out, &time_format, &time)) {
+ gst_pad_push_event (dec->srcpad,
+ gst_event_new_newsegment (1.0, GST_FORMAT_TIME,
+ time, GST_CLOCK_TIME_NONE, 0));
} else {
GST_ERROR_OBJECT (dec,
"failed to parse data for DISCONT event, not sending any");
}
- gst_data_unref (GST_DATA (event));
+ gst_event_unref (event);
default:
- gst_pad_event_default (dec->sinkpad, event);
- break;
+ return gst_pad_event_default (dec->sinkpad, event);
}
+ return TRUE;
-speex_dec_chain (GstPad * pad, GstData * data)
+static GstFlowReturn
+speex_dec_chain (GstPad * pad, GstBuffer * buf)
- GstBuffer *buf;
+ GstFlowReturn res;
GstSpeexDec *dec;
dec = GST_SPEEXDEC (gst_pad_get_parent (pad));
- if (GST_IS_EVENT (data)) {
- speex_dec_event (dec, GST_EVENT (data));
- return;
- buf = GST_BUFFER (data);
if (dec->packetno == 0) {
GstCaps *caps;
/* get the header */
- dec->header =
- speex_packet_to_header (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
- gst_data_unref (data);
+ dec->header = speex_packet_to_header ((char *) GST_BUFFER_DATA (buf),
+ GST_BUFFER_SIZE (buf));
+ gst_buffer_unref (buf);
if (!dec->header) {
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't read header"));
- return;
+ return GST_FLOW_ERROR;
if (dec->header->mode >= SPEEX_NB_MODES) {
(NULL),
("Mode number %d does not (yet/any longer) exist in this version",
dec->header->mode));
dec->mode = (SpeexMode *) speex_mode_list[dec->header->mode];
@@ -409,8 +356,8 @@
if (!dec->state) {
(NULL), ("couldn't initialize decoder"));
- gst_data_unref (data);
+ gst_buffer_unref (buf);
speex_decoder_ctl (dec->state, SPEEX_SET_ENH, &dec->enh);
@@ -435,19 +382,23 @@
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
- if (!gst_pad_set_explicit_caps (dec->srcpad, caps)) {
- gst_caps_free (caps);
+ if (!gst_pad_set_caps (dec->srcpad, caps)) {
+ gst_caps_unref (caps);
+ return GST_FLOW_NOT_NEGOTIATED;
- gst_caps_free (caps);
+ gst_caps_unref (caps);
+ gst_pad_push_event (dec->srcpad,
+ gst_event_new_newsegment (1.0, GST_FORMAT_TIME,
+ 0, GST_CLOCK_TIME_NONE, 0));
} else if (dec->packetno == 1) {
gchar *encoder = NULL;
/* FIXME parse comments */
- GstTagList *list = gst_tag_list_from_vorbiscomment_buffer (buf, "", 1,
- &encoder);
+ GstTagList *list = gst_tag_list_new (); //gst_tag_list_from_vorbiscomment_buffer (buf, "", 1,
+// &encoder);
if (!list) {
GST_WARNING_OBJECT (dec, "couldn't decode comments");
@@ -474,14 +425,14 @@
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL);
*/
- gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, 0, list);
+ gst_element_found_tags_for_pad (GST_ELEMENT (dec), dec->srcpad, list);
} else {
gint i;
/* send data to the bitstream */
- speex_bits_read_from (&dec->bits, GST_BUFFER_DATA (buf),
+ speex_bits_read_from (&dec->bits, (char *) GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf));
/* now decode each frame */
for (i = 0; i < dec->header->frames_per_packet; i++) {
@@ -505,8 +456,10 @@
if (dec->header->nb_channels == 2)
speex_decode_stereo (dec->output, dec->frame_size, &dec->stereo);
- outbuf = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET_NONE,
- dec->frame_size * dec->header->nb_channels * 2);
+ if ((res = gst_pad_alloc_buffer (dec->srcpad, GST_BUFFER_OFFSET_NONE,
+ dec->frame_size * dec->header->nb_channels * 2,
+ GST_PAD_CAPS (dec->srcpad), &outbuf)) != GST_FLOW_OK)
+ return res;
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
/*PCM saturation (just in case) */
@@ -525,11 +478,15 @@
dec->samples_out * GST_SECOND / dec->header->rate;
GST_BUFFER_DURATION (outbuf) =
dec->frame_size * GST_SECOND / dec->header->rate;
- gst_pad_push (dec->srcpad, GST_DATA (outbuf));
+ res = gst_pad_push (dec->srcpad, outbuf);
+ if (res != GST_FLOW_OK && res != GST_FLOW_NOT_LINKED)
dec->samples_out += dec->frame_size;
dec->packetno++;
+ return GST_FLOW_OK;
static void
|
|
From: <ensonic@fr...> - 2005-08-23 14:26:14
|
CVS Root: /cvs/gstreamer
Module: gstreamer
Changes by: ensonic
Date: Tue Aug 23 2005 07:26:08 PDT
Log message:
* docs/gst/gstreamer-docs.sgml:
* docs/gst/gstreamer-sections.txt:
* docs/gst/gstreamer.types:
* docs/gst/tmpl/.cvsignore:
* gst/gstbin.h:
* gst/gstbus.c:
* gst/gstelement.c:
* gst/gstevent.h:
added about 100 symbols from gstreamer-unused.txt to the right sections
fixed more broken comments
added GstBus to docs
Modified files:
. : ChangeLog
docs/gst : gstreamer-docs.sgml gstreamer-sections.txt
gstreamer.types
docs/gst/tmpl : .cvsignore
gst : gstbin.h gstbus.c gstelement.c gstevent.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/ChangeLog.diff?r1=1.1384&r2=1.1385
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/gstreamer-docs.sgml.diff?r1=1.73&r2=1.74
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/gstreamer-sections.txt.diff?r1=1.155&r2=1.156
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/gstreamer.types.diff?r1=1.20&r2=1.21
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/.cvsignore.diff?r1=1.4&r2=1.5
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstbin.h.diff?r1=1.75&r2=1.76
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstbus.c.diff?r1=1.18&r2=1.19
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstelement.c.diff?r1=1.355&r2=1.356
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstevent.h.diff?r1=1.62&r2=1.63
====Begin Diffs====
Index: gstreamer-docs.sgml
===================================================================
RCS file: /cvs/gstreamer/gstreamer/docs/gst/gstreamer-docs.sgml,v
retrieving revision 1.73
retrieving revision 1.74
diff -u -d -r1.73 -r1.74
--- gstreamer-docs.sgml 22 Aug 2005 21:03:33 -0000 1.73
+++ gstreamer-docs.sgml 23 Aug 2005 14:25:55 -0000 1.74
@@ -6,6 +6,7 @@
<!ENTITY Gst SYSTEM "xml/gst.xml">
<!ENTITY GstBin SYSTEM "xml/gstbin.xml">
<!ENTITY GstBuffer SYSTEM "xml/gstbuffer.xml">
+<!ENTITY GstBus SYSTEM "xml/gstbus.xml">
<!ENTITY GstCaps SYSTEM "xml/gstcaps.xml">
<!ENTITY GstChildProxy SYSTEM "xml/gstchildproxy.xml">
<!ENTITY GstClock SYSTEM "xml/gstclock.xml">
@@ -102,6 +103,7 @@
&Gst;
&GstBin;
&GstBuffer;
+ &GstBus;
&GstCaps;
&GstChildProxy;
&GstClock;
Index: gstreamer-sections.txt
RCS file: /cvs/gstreamer/gstreamer/docs/gst/gstreamer-sections.txt,v
retrieving revision 1.155
retrieving revision 1.156
diff -u -d -r1.155 -r1.156
--- gstreamer-sections.txt 23 Aug 2005 11:53:58 -0000 1.155
+++ gstreamer-sections.txt 23 Aug 2005 14:25:55 -0000 1.156
@@ -22,6 +22,7 @@
gst_init_with_popt_table
gst_init_check_with_popt_table
gst_init_get_popt_table
+gst_deinit
<SUBSECTION Private>
#GstPoptOption
</SECTION>
@@ -46,6 +47,9 @@
gst_bin_iterate_sinks
gst_bin_iterate_all_by_interface
+GST_BIN_CHILDREN
+GST_BIN_CHILDREN_COOKIE
+GST_BIN_NUMCHILDREN
<SUBSECTION>
gst_bin_add_many
gst_bin_remove_many
@@ -59,7 +63,7 @@
GST_IS_BIN_CLASS
GST_BIN_GET_CLASS
GST_TYPE_BIN_FLAGS
-
+GST_BIN_CAST
gst_bin_get_type
gst_bin_flags_get_type
@@ -67,6 +71,41 @@
<SECTION>
+<FILE>gstbus</FILE>
+<TITLE>GstBus</TITLE>
+GstBus
+GstBusFlags
+GstBusSyncReply
+GstBusHandler
+GstBusSyncHandler
+gst_bus_new
+gst_bus_post
+gst_bus_have_pending
+gst_bus_peek
+gst_bus_pop
+gst_bus_set_flushing
+gst_bus_set_sync_handler
+gst_bus_create_watch
+gst_bus_add_watch_full
+gst_bus_add_watch
+gst_bus_poll
+<SUBSECTION Standard>
+GstBusClass
+GST_BUS
+GST_IS_BUS
+GST_TYPE_BUS
+GST_BUS_CLASS
+GST_IS_BUS_CLASS
+GST_BUS_GET_CLASS
+GST_TYPE_BUS_FLAGS
+GST_BUS_CAST
+<SUBSECTION Private>
+gst_bus_get_type
+gst_bus_flags_get_type
+gst_bus_sync_reply_get_type
+</SECTION>
+
+<SECTION>
<FILE>gstbuffer</FILE>
<TITLE>GstBuffer</TITLE>
GstBuffer
@@ -117,8 +156,10 @@
gst_buffer_merge
<SUBSECTION Standard>
+GstBufferClass
GST_BUFFER
GST_BUFFER_CLASS
+GST_BUFFER_GET_CLASS
GST_IS_BUFFER
GST_IS_BUFFER_CLASS
GST_TYPE_BUFFER
@@ -149,6 +190,7 @@
gst_caps_new_full
gst_caps_new_full_valist
gst_caps_copy
+gst_caps_copy_nth
gst_static_caps_get
gst_caps_append
gst_caps_append_structure
@@ -354,6 +396,7 @@
gst_element_get_manager
gst_element_no_more_pads
+GstEventClass
GST_ELEMENT
GST_IS_ELEMENT
GST_ELEMENT_CLASS
@@ -444,6 +487,9 @@
GST_EVENT_TIMESTAMP
GST_EVENT_SRC
GST_EVENT_IS_INTERRUPT
+GST_EVENT_IS_DOWNSTREAM
+GST_EVENT_IS_SERIALIZED
+GST_EVENT_IS_UPSTREAM
GstEventFlag
GST_EVENT_MASK_FUNCTION
GstSeekType
@@ -460,6 +506,7 @@
GST_EVENT_SIZE_FORMAT
GST_EVENT_SIZE_VALUE
GST_EVENT_RATE_VALUE
+GST_EVENT_MAKE_TYPE
gst_event_ref
gst_event_unref
gst_event_copy
@@ -481,6 +528,9 @@
GST_EVENT
GST_IS_EVENT
+GST_IS_EVENT_CLASS
+GST_EVENT_CLASS
+GST_EVENT_GET_CLASS
GST_TYPE_EVENT
GST_TYPE_EVENT_FLAG
GST_TYPE_EVENT_TYPE
@@ -524,6 +574,7 @@
<TITLE>GstGhostPad</TITLE>
GstGhostPad
gst_ghost_pad_new
+gst_ghost_pad_set_target
GstGhostPadClass
GST_GHOST_PAD
@@ -1185,6 +1236,7 @@
<FILE>gstquery</FILE>
<TITLE>GstQuery</TITLE>
+GstQuery
GstQueryType
GST_QUERY_TYPE_RATE_DEN
GstQueryTypeDefinition
@@ -1194,8 +1246,15 @@
gst_query_types_contains
gst_query_type_get_details
+GstQueryClass
+GST_QUERY
+GST_IS_QUERY
+GST_QUERY_CLASS
+GST_IS_QUERY_CLASS
+GST_TYPE_QUERY
GST_TYPE_QUERY_TYPE
+gst_query_get_type
gst_query_type_get_type
@@ -1621,6 +1680,8 @@
gst_util_dump_mem
gst_print_pad_caps
gst_print_element_args
+gst_bin_watch_for_state_change
+GST_BOILERPLATE_WITH_INTERFACE
GST_BOILERPLATE_FULL
GST_BOILERPLATE
GST_CALL_PARENT
@@ -1772,15 +1833,16 @@
gst_adapter_flush
gst_adapter_available
gst_adapter_available_fast
+gst_adapter_take
-GST_ADAPTER_SRC
-GST_IS_ADAPTER_SRC
-GST_TYPE_ADAPTER_SRC
-GST_ADAPTER_SRC_CLASS
-GST_IS_ADAPTER_SRC_CLASS
-GST_ADAPTER_SRC_GET_CLASS
+GST_ADAPTER
+GST_IS_ADAPTER
+GST_TYPE_ADAPTER
+GST_ADAPTER_CLASS
+GST_IS_ADAPTER_CLASS
+GST_ADAPTER_GET_CLASS
-gst_adapter_src_get_type
+gst_adapter_get_type
@@ -1837,6 +1899,13 @@
<INCLUDE>gst/base/gstbasetransform.h</INCLUDE>
GstBaseTransform
GstBaseTransformClass
+gst_base_transform_is_passthrough
+gst_base_transform_set_passthrough
+GST_BASE_TRANSFORM_SINK_NAME
+GST_BASE_TRANSFORM_SRC_NAME
GST_BASE_TRANSFORM
GST_IS_BASE_TRANSFORM
@@ -1856,6 +1925,7 @@
GstCollectPads
GstCollectPadsClass
GstCollectData
+GstCollectPadsFunction
gst_collectpads_new
gst_collectpads_set_function
gst_collectpads_add_pad
@@ -1870,6 +1940,10 @@
gst_collectpads_available
gst_collectpads_read
gst_collectpads_flush
+GST_COLLECTPADS_BROADCAST
+GST_COLLECTPADS_GET_COND
+GST_COLLECTPADS_SIGNAL
+GST_COLLECTPADS_WAIT
GST_COLLECTPADS
GST_IS_COLLECTPADS
@@ -1907,13 +1981,17 @@
<TITLE>GstFakeSrc</TITLE>
<INCLUDE>gst/elements/gstfakesrc.h</INCLUDE>
GstFakeSrc
+GstFakeSrcDataType
+GstFakeSrcFillType
+GstFakeSrcOutputType
+GstFakeSrcSizeType
GstFakeSrcClass
-GST_FAKESRC
-GST_IS_FAKESRC
-GST_TYPE_FAKESRC
-GST_FAKESRC_CLASS
-GST_IS_FAKESRC_CLASS
+GST_FAKE_SRC
+GST_IS_FAKE_SRC
+GST_TYPE_FAKE_SRC
+GST_FAKE_SRC_CLASS
+GST_IS_FAKE_SRC_CLASS
gst_fake_src_get_type
@@ -1923,13 +2001,14 @@
<TITLE>GstFakeSink</TITLE>
<INCLUDE>gst/elements/gstfakesink.h</INCLUDE>
GstFakeSink
+GstFakeSinkStateError
GstFakeSinkClass
-GST_FAKESINK
-GST_IS_FAKESINK
-GST_TYPE_FAKESINK
-GST_FAKESINK_CLASS
-GST_IS_FAKESINK_CLASS
+GST_FAKE_SINK
+GST_IS_FAKE_SINK
+GST_TYPE_FAKE_SINK
+GST_FAKE_SINK_CLASS
+GST_IS_FAKE_SINK_CLASS
gst_fake_sink_get_type
@@ -1942,10 +2021,10 @@
GstFileSrcClass
GST_FILE_SRC
-GST_IS_FILESRC
-GST_TYPE_FILESRC
+GST_IS_FILE_SRC
+GST_TYPE_FILE_SRC
GST_FILE_SRC_CLASS
-GST_IS_FILESRC_CLASS
+GST_IS_FILE_SRC_CLASS
gst_file_src_get_type
@@ -1958,10 +2037,10 @@
GstFileSinkClass
GST_FILE_SINK
-GST_IS_FILESINK
-GST_TYPE_FILESINK
+GST_IS_FILE_SINK
+GST_TYPE_FILE_SINK
GST_FILE_SINK_CLASS
-GST_IS_FILESINK_CLASS
+GST_IS_FILE_SINK_CLASS
gst_file_sink_get_type
Index: gstreamer.types
RCS file: /cvs/gstreamer/gstreamer/docs/gst/gstreamer.types,v
retrieving revision 1.20
retrieving revision 1.21
diff -u -d -r1.20 -r1.21
--- gstreamer.types 5 Aug 2005 12:59:45 -0000 1.20
+++ gstreamer.types 23 Aug 2005 14:25:55 -0000 1.21
@@ -10,6 +10,7 @@
#include <gst/gstqueue.h>
gst_clock_get_type
gst_element_factory_get_type
gst_element_get_type
Index: .cvsignore
RCS file: /cvs/gstreamer/gstreamer/docs/gst/tmpl/.cvsignore,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -d -r1.4 -r1.5
--- .cvsignore 23 Aug 2005 11:53:58 -0000 1.4
+++ .cvsignore 23 Aug 2005 14:25:55 -0000 1.5
@@ -7,6 +7,7 @@
gstbasetransform.sgml
gstbin.sgml
gstbuffer.sgml
+gstbus.sgml
gstcollectpads.sgml
gstevent.sgml
gstfakesrc.sgml
Index: ChangeLog
RCS file: /cvs/gstreamer/gstreamer/ChangeLog,v
retrieving revision 1.1384
retrieving revision 1.1385
diff -u -d -r1.1384 -r1.1385
--- ChangeLog 23 Aug 2005 11:53:57 -0000 1.1384
+++ ChangeLog 23 Aug 2005 14:25:55 -0000 1.1385
@@ -1,5 +1,19 @@
2005-08-23 Stefan Kost <ensonic@...>
+ * docs/gst/gstreamer-docs.sgml:
+ * docs/gst/gstreamer-sections.txt:
+ * docs/gst/gstreamer.types:
+ * docs/gst/tmpl/.cvsignore:
+ * gst/gstbin.h:
+ * gst/gstbus.c:
+ * gst/gstelement.c:
+ * gst/gstevent.h:
+ added about 100 symbols from gstreamer-unused.txt to the right sections
+ fixed more broken comments
+ added GstBus to docs
+2005-08-23 Stefan Kost <ensonic@...>
* docs/gst/gstreamer-sections.txt:
* docs/gst/tmpl/.cvsignore:
* docs/gst/tmpl/gstbin.sgml:
Index: gstbin.h
RCS file: /cvs/gstreamer/gstreamer/gst/gstbin.h,v
retrieving revision 1.75
retrieving revision 1.76
diff -u -d -r1.75 -r1.76
--- gstbin.h 18 Jul 2005 12:49:52 -0000 1.75
+++ gstbin.h 23 Aug 2005 14:25:55 -0000 1.76
@@ -57,8 +57,32 @@
typedef struct _GstBin GstBin;
typedef struct _GstBinClass GstBinClass;
+/**
+ * GST_BIN_NUMCHILDREN:
+ * @bin: the bin to get the number of children from
+ *
+ * Gets the number of children a bin manages.
+ * Returns: the number of child objects
+ */
#define GST_BIN_NUMCHILDREN(bin) (GST_BIN_CAST(bin)->numchildren);
+ * GST_BIN_CHILDREN:
+ * @bin: the bin to get the list with children from
+ * Gets the list with children a bin manages.
+ * Returns: the children list
#define GST_BIN_CHILDREN(bin) (GST_BIN_CAST(bin)->children);
+ * GST_BIN_CHILDREN_COOKIE:
+ * @bin: the bin to get the children cookie from
+ * Gets the children cookie that watches the children list.
+ * Returns: the children cookie
#define GST_BIN_CHILDREN_COOKIE(bin) (GST_BIN_CAST(bin)->children_cookie);
struct _GstBin {
Index: gstbus.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstbus.c,v
retrieving revision 1.18
retrieving revision 1.19
diff -u -d -r1.18 -r1.19
--- gstbus.c 18 Aug 2005 15:31:28 -0000 1.18
+++ gstbus.c 23 Aug 2005 14:25:55 -0000 1.19
@@ -18,7 +18,10 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
+ * SECTION:gstbus
+ * @short_description: Asynchronous message bus subsystem
#include <errno.h>
#include <unistd.h>
@@ -151,6 +154,13 @@
}
}
+ * gst_bus_new:
+ * Creates a new #GstBuus instance.
+ * Returns: a new #GstBus instance
GstBus *
gst_bus_new (void)
{
Index: gstelement.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstelement.c,v
retrieving revision 1.355
retrieving revision 1.356
diff -u -d -r1.355 -r1.356
--- gstelement.c 22 Aug 2005 14:35:42 -0000 1.355
+++ gstelement.c 23 Aug 2005 14:25:55 -0000 1.356
@@ -1140,10 +1140,13 @@
/**
* gst_element_seek:
* @element: a #GstElement to send the event to.
- * @seek_method: the method to use for seeking (GST_SEEK_METHOD_*).
- * @seek_format: the #GstFormat to use for seeking (GST_FORMAT_*).
- * @seek_flags: the flags to use for seeking (GST_SEEK_FLAG_*).
- * @offset: the offset to seek to (in the given seek_format).
+ * @rate: the rate of the seek
+ * @format: the #GstFormat to use for seeking (GST_FORMAT_*).
+ * @flags: the flags to use for seeking (GST_SEEK_FLAG_*).
+ * @cur_type: the #GstSeekType for the start offset
+ * @cur: the start offset to seek to (in the given seek_format).
+ * @stop_type: the #GstSeekType for the end offset
+ * @stop: the end offset to seek to (in the given seek_format).
*
* Sends a seek event to an element.
Index: gstevent.h
RCS file: /cvs/gstreamer/gstreamer/gst/gstevent.h,v
retrieving revision 1.62
retrieving revision 1.63
diff -u -d -r1.62 -r1.63
--- gstevent.h 22 Aug 2005 10:37:02 -0000 1.62
+++ gstevent.h 23 Aug 2005 14:25:55 -0000 1.63
@@ -98,6 +98,7 @@
* GST_EVENT_TRACE_NAME:
* The name used for memory allocation tracing
#define GST_EVENT_TRACE_NAME "GstEvent"
|
|
From: <wingo@fr...> - 2005-08-23 14:10:20
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wingo
Date: Tue Aug 23 2005 07:10:17 PDT
Log message:
2005-08-23 Andy Wingo <wingo@...>
* sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name
property after opening the mixer.
Modified files:
. : ChangeLog
sys/oss : gstosssrc.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1805&r2=1.1806
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.c.diff?r1=1.60&r2=1.61
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1805
retrieving revision 1.1806
diff -u -d -r1.1805 -r1.1806
--- ChangeLog 23 Aug 2005 13:59:22 -0000 1.1805
+++ ChangeLog 23 Aug 2005 14:10:04 -0000 1.1806
@@ -1,5 +1,8 @@
2005-08-23 Andy Wingo <wingo@...>
+ * sys/oss/gstosssrc.c (gst_oss_src_open): Set the device-name
+ property after opening the mixer.
+
* sys/oss/gstosssrc.c:
* sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
Index: gstosssrc.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.c,v
retrieving revision 1.60
retrieving revision 1.61
diff -u -d -r1.60 -r1.61
--- gstosssrc.c 23 Aug 2005 13:59:22 -0000 1.60
+++ gstosssrc.c 23 Aug 2005 14:10:05 -0000 1.61
@@ -306,9 +306,17 @@
return FALSE;
}
- if (!oss->mixer)
+ if (!oss->mixer) {
oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
+ if (oss->mixer) {
+ if (oss->device_name) {
+ g_free (oss->device_name);
+ }
+ oss->device_name = g_strdup (oss->mixer->cardname);
+ }
+ }
return TRUE;
}
|
|
From: <wingo@fr...> - 2005-08-23 13:59:36
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wingo
Date: Tue Aug 23 2005 06:59:34 PDT
Log message:
2005-08-23 Andy Wingo <wingo@...>
* sys/oss/gstosssrc.c:
* sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
Modified files:
. : ChangeLog
sys/oss : gstosssrc.c gstosssrc.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1804&r2=1.1805
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.c.diff?r1=1.59&r2=1.60
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.h.diff?r1=1.16&r2=1.17
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1804
retrieving revision 1.1805
diff -u -d -r1.1804 -r1.1805
--- ChangeLog 23 Aug 2005 13:53:17 -0000 1.1804
+++ ChangeLog 23 Aug 2005 13:59:22 -0000 1.1805
@@ -1,5 +1,8 @@
2005-08-23 Andy Wingo <wingo@...>
+ * sys/oss/gstosssrc.c:
+ * sys/oss/gstosssrc.h: Easy to implement a mixer, eh...
+
* sys/oss/gstossmixerelement.h:
* sys/oss/gstossmixerelement.c: Added mixer element like
alsamixer.
Index: gstosssrc.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.c,v
retrieving revision 1.59
retrieving revision 1.60
diff -u -d -r1.59 -r1.60
--- gstosssrc.c 23 Aug 2005 13:53:17 -0000 1.59
+++ gstosssrc.c 23 Aug 2005 13:59:22 -0000 1.60
@@ -47,13 +47,10 @@
PROP_DEVICE_NAME,
};
-GST_BOILERPLATE (GstOssSrc, gst_oss_src, GstAudioSrc, GST_TYPE_AUDIO_SRC);
+GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc,
+ GST_TYPE_AUDIO_SRC, GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
-/*
-GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc, GST_TYPE_AUDIO_SRC,
- GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
-*/
static void gst_oss_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
@@ -309,13 +306,26 @@
return FALSE;
}
+ if (!oss->mixer)
+ oss->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_CAPTURE);
return TRUE;
}
static gboolean
gst_oss_src_close (GstAudioSrc * asrc)
{
- close (GST_OSS_SRC (asrc)->fd);
+ GstOssSrc *oss;
+ oss = GST_OSS_SRC (asrc);
+ close (oss->fd);
+ if (oss->mixer) {
+ gst_ossmixer_free (oss->mixer);
+ oss->mixer = NULL;
+ }
Index: gstosssrc.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.h,v
retrieving revision 1.16
retrieving revision 1.17
diff -u -d -r1.16 -r1.17
--- gstosssrc.h 23 Aug 2005 13:26:21 -0000 1.16
+++ gstosssrc.h 23 Aug 2005 13:59:22 -0000 1.17
@@ -29,6 +29,7 @@
#include <gst/audio/gstaudiosrc.h>
#include "gstosshelper.h"
+#include "gstossmixer.h"
G_BEGIN_DECLS
@@ -49,6 +50,8 @@
gchar *device;
gchar *device_name;
+ GstOssMixer *mixer;
struct _GstOssSrcClass {
|
|
From: <wingo@fr...> - 2005-08-23 13:53:32
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wingo
Date: Tue Aug 23 2005 06:53:29 PDT
Log message:
2005-08-23 Andy Wingo <wingo@...>
* sys/oss/gstossmixerelement.h:
* sys/oss/gstossmixerelement.c: Added mixer element like
alsamixer.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Register the ossmixer element.
Modified files:
. : ChangeLog
sys/oss : Makefile.am gstossaudio.c gstossmixer.c
gstosssrc.c
Added files:
sys/oss : gstossmixerelement.c gstossmixerelement.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1803&r2=1.1804
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/Makefile.am.diff?r1=1.29&r2=1.30
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossaudio.c.diff?r1=1.18&r2=1.19
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixer.c.diff?r1=1.20&r2=1.21
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixerelement.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixerelement.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.c.diff?r1=1.58&r2=1.59
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1803
retrieving revision 1.1804
diff -u -d -r1.1803 -r1.1804
--- ChangeLog 23 Aug 2005 13:27:30 -0000 1.1803
+++ ChangeLog 23 Aug 2005 13:53:17 -0000 1.1804
@@ -1,5 +1,12 @@
2005-08-23 Andy Wingo <wingo@...>
+ * sys/oss/gstossmixerelement.h:
+ * sys/oss/gstossmixerelement.c: Added mixer element like
+ alsamixer.
+
+ * sys/oss/Makefile.am:
+ * sys/oss/gstossaudio.c: Register the ossmixer element.
+
* sys/oss/gstossmixer.h:
* sys/oss/gstossmixer.c: Refactored to be more like alsamixer.
Index: Makefile.am
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/Makefile.am,v
retrieving revision 1.29
retrieving revision 1.30
diff -u -d -r1.29 -r1.30
--- Makefile.am 23 Aug 2005 13:26:21 -0000 1.29
+++ Makefile.am 23 Aug 2005 13:53:17 -0000 1.30
@@ -1,10 +1,11 @@
plugin_LTLIBRARIES = libgstossaudio.la
-libgstossaudio_la_SOURCES = gstossaudio.c \
- gstosshelper.c \
- gstossmixer.c \
- gstossmixertrack.c \
- gstosssink.c \
+libgstossaudio_la_SOURCES = gstossaudio.c \
+ gstosshelper.c \
+ gstossmixer.c \
+ gstossmixerelement.c \
+ gstossmixertrack.c \
+ gstosssink.c \
gstosssrc.c
# gstossdmabuffer.c
@@ -15,11 +16,12 @@
-lgstaudio-@...@
libgstossaudio_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-noinst_HEADERS = gstosssink.h \
- gstosssrc.h \
- gstosshelper.h \
- gstossdmabuffer.h \
- gstossmixer.h \
+noinst_HEADERS = gstosssink.h \
+ gstosssrc.h \
+ gstosshelper.h \
+ gstossdmabuffer.h \
+ gstossmixer.h \
+ gstossmixerelement.h \
gstossmixertrack.h
# noinst_PROGRAMS = #oss_probe
Index: gstossaudio.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossaudio.c,v
retrieving revision 1.18
retrieving revision 1.19
diff -u -d -r1.18 -r1.19
--- gstossaudio.c 23 Aug 2005 13:26:21 -0000 1.18
+++ gstossaudio.c 23 Aug 2005 13:53:17 -0000 1.19
@@ -23,6 +23,7 @@
#include "gst/gst-i18n-plugin.h"
+#include "gstossmixerelement.h"
#include "gstosssink.h"
#include "gstosssrc.h"
@@ -33,8 +34,8 @@
static gboolean
plugin_init (GstPlugin * plugin)
{
- if ( /*!gst_element_register (plugin, "ossmixer", GST_RANK_PRIMARY,
- GST_TYPE_OSSELEMENT) || */
+ if (!gst_element_register (plugin, "ossmixer", GST_RANK_NONE,
+ GST_TYPE_OSS_MIXER_ELEMENT) ||
!gst_element_register (plugin, "osssrc", GST_RANK_PRIMARY,
GST_TYPE_OSS_SRC) ||
!gst_element_register (plugin, "osssink", GST_RANK_SECONDARY,
Index: gstossmixer.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossmixer.c,v
retrieving revision 1.20
retrieving revision 1.21
diff -u -d -r1.20 -r1.21
--- gstossmixer.c 23 Aug 2005 13:26:21 -0000 1.20
+++ gstossmixer.c 23 Aug 2005 13:53:17 -0000 1.21
@@ -143,6 +143,7 @@
ret->device = g_strdup (device);
ret->dir = dir;
+ ret->mixer_fd = -1;
if (!gst_ossmixer_open (ret))
goto error;
--- NEW FILE: gstossmixerelement.c ---
/* OSS mixer interface element.
* Copyright (C) 2005 Andrew Vander Wingo <wingo@...>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstossmixerelement.h"
static GstElementDetails gst_oss_mixer_element_details =
GST_ELEMENT_DETAILS ("OSS Mixer",
"Generic/Audio",
"Control sound input and output levels with OSS",
"Andrew Vander Wingo <wingo@...>");
GST_BOILERPLATE_WITH_INTERFACE (GstOssMixerElement, gst_oss_mixer_element,
GstElement, GST_TYPE_ELEMENT, GstMixer, GST_TYPE_MIXER,
gst_oss_mixer_element);
GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssMixerElement, gst_oss_mixer_element);
static GstElementStateReturn gst_oss_mixer_element_change_state (GstElement *
element);
static void
gst_oss_mixer_element_base_init (gpointer klass)
{
gst_element_class_set_details (GST_ELEMENT_CLASS (klass),
&gst_oss_mixer_element_details);
}
gst_oss_mixer_element_class_init (GstOssMixerElementClass * klass)
GstElementClass *element_class;
element_class = (GstElementClass *) klass;
element_class->change_state = gst_oss_mixer_element_change_state;
gst_oss_mixer_element_init (GstOssMixerElement * this)
this->mixer = NULL;
static GstElementStateReturn
gst_oss_mixer_element_change_state (GstElement * element)
GstOssMixerElement *this = GST_OSS_MIXER_ELEMENT (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
if (!this->mixer) {
this->mixer = gst_ossmixer_new ("/dev/mixer", GST_OSS_MIXER_ALL);
}
break;
case GST_STATE_READY_TO_NULL:
if (this->mixer) {
gst_ossmixer_free (this->mixer);
this->mixer = NULL;
default:
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
--- NEW FILE: gstossmixerelement.h ---
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* License along with this library; if not, write to the Free
* Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
#ifndef __GST_OSS_MIXER_ELEMENT_H__
#define __GST_OSS_MIXER_ELEMENT_H__
#include "gstossmixer.h"
G_BEGIN_DECLS
#define GST_OSS_MIXER_ELEMENT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElement))
#define GST_OSS_MIXER_ELEMENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_MIXER_ELEMENT,GstOssMixerElementClass))
#define GST_IS_OSS_MIXER_ELEMENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_MIXER_ELEMENT))
#define GST_IS_OSS_MIXER_ELEMENT_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_MIXER_ELEMENT))
#define GST_TYPE_OSS_MIXER_ELEMENT (gst_oss_mixer_element_get_type())
typedef struct _GstOssMixerElement GstOssMixerElement;
typedef struct _GstOssMixerElementClass GstOssMixerElementClass;
struct _GstOssMixerElement {
GstElement parent;
GstOssMixer *mixer;
};
struct _GstOssMixerElementClass {
GstElementClass parent;
GType gst_oss_mixer_element_get_type (void);
G_END_DECLS
#endif /* __GST_OSS_MIXER_ELEMENT_H__ */
|
|
From: <wingo@fr...> - 2005-08-23 13:29:33
|
CVS Root: /cvs/gstreamer
Module: gst-plugins-base
Changes by: wingo
Date: Tue Aug 23 2005 06:29:29 PDT
Log message:
2005-08-23 Andy Wingo <wingo@...>
* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Only fixate endianness if it is
present in the caps.
Modified files:
. : ChangeLog
gst-libs/gst/audio: gstbaseaudiosrc.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/ChangeLog.diff?r1=1.1844&r2=1.1845
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst-libs/gst/audio/gstbaseaudiosrc.c.diff?r1=1.6&r2=1.7
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/ChangeLog,v
retrieving revision 1.1844
retrieving revision 1.1845
diff -u -d -r1.1844 -r1.1845
--- ChangeLog 22 Aug 2005 16:50:59 -0000 1.1844
+++ ChangeLog 23 Aug 2005 13:29:16 -0000 1.1845
@@ -1,3 +1,9 @@
+2005-08-23 Andy Wingo <wingo@...>
+
+ * gst-libs/gst/audio/gstbaseaudiosrc.c
+ (gst_base_audio_src_fixate): Only fixate endianness if it is
+ present in the caps.
2005-08-22 Andy Wingo <wingo@...>
* ext/alsa/gstalsasink.c (gst_alsasink_get_property):
Index: gstbaseaudiosrc.c
RCS file: /cvs/gstreamer/gst-plugins-base/gst-libs/gst/audio/gstbaseaudiosrc.c,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -d -r1.6 -r1.7
--- gstbaseaudiosrc.c 16 Aug 2005 15:53:59 -0000 1.6
+++ gstbaseaudiosrc.c 23 Aug 2005 13:29:17 -0000 1.7
@@ -211,7 +211,8 @@
gst_caps_structure_fixate_field_nearest_int (s, "depth", 16);
gst_caps_structure_fixate_field_nearest_int (s, "width", 16);
gst_structure_set (s, "signed", G_TYPE_BOOLEAN, TRUE, NULL);
- gst_caps_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
+ if (gst_structure_has_field (s, "endianness"))
+ gst_caps_structure_fixate_field_nearest_int (s, "endianness", G_BYTE_ORDER);
}
static gboolean
|
|
From: <wingo@fr...> - 2005-08-23 13:27:54
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wingo
Date: Tue Aug 23 2005 06:27:42 PDT
Log message:
changelog
Modified files:
. : ChangeLog
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1802&r2=1.1803
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1802
retrieving revision 1.1803
diff -u -d -r1.1802 -r1.1803
--- ChangeLog 23 Aug 2005 09:46:29 -0000 1.1802
+++ ChangeLog 23 Aug 2005 13:27:30 -0000 1.1803
@@ -1,5 +1,29 @@
2005-08-23 Andy Wingo <wingo@...>
+ * sys/oss/gstossmixer.h:
+ * sys/oss/gstossmixer.c: Refactored to be more like alsamixer.
+
+ * sys/oss/gstossmixertrack.h:
+ * sys/oss/gstossmixertrack.c: Split out from gstossmixer.[ch],
+ like gstalsamixer.
+ * sys/oss/gstosssrc.c:
+ * sys/oss/gstosssink.c: Where before we used a gstosselement
+ object as a helper library, now just call functions from
+ gstosshelper.
+ * sys/oss/gstosshelper.h:
+ * sys/oss/gstosshelper.c: Made a real library. Removed
+ propertyprobe for now, should add it back later.
+
+ * sys/oss/gstosselement.h:
+ * sys/oss/gstosselement.c: Removed, we don't have a shared base
+ class.
+ * sys/oss/gstosshelper.c (gst_oss_helper_probe_caps): Search
+ higher-to-lower, makes 16 bit appear earlier in the caps, which
+ makes it preferred.
* sys/oss/gstosssrc.h:
* sys/oss/gstosssrc.c: Totally ported, dude.
|
|
From: <wingo@fr...> - 2005-08-23 13:26:36
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wingo
Date: Tue Aug 23 2005 06:26:33 PDT
Log message:
2005-08-23 Andy Wingo <wingo@...>
* sys/oss/gstossmixer.h:
* sys/oss/gstossmixer.c: Refactored to be more like alsamixer.
* sys/oss/gstossmixertrack.h:
* sys/oss/gstossmixertrack.c: Split out from gstossmixer.[ch],
like gstalsamixer.
* sys/oss/gstosssrc.c:
* sys/oss/gstosssink.c: Where before we used a gstosselement
object as a helper library, now just call functions from
gstosshelper.
* sys/oss/gstosshelper.h:
* sys/oss/gstosshelper.c: Made a real library. Removed
propertyprobe for now, should add it back later.
* sys/oss/gstosselement.h:
* sys/oss/gstosselement.c: Removed, we don't have a shared base
class.
* sys/oss/gstosshelper.c (gst_oss_helper_probe_caps): Search
higher-to-lower, makes 16 bit appear earlier in the caps, which
makes it preferred.
Modified files:
sys/oss : Makefile.am gstossaudio.c gstossdmabuffer.h
gstosshelper.c gstosshelper.h gstossmixer.c
gstossmixer.h gstosssink.c gstosssink.h
gstosssrc.c gstosssrc.h
Added files:
sys/oss : gstossmixertrack.c gstossmixertrack.h
Removed files:
sys/oss : gstosselement.c gstosselement.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/Makefile.am.diff?r1=1.28&r2=1.29
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossaudio.c.diff?r1=1.17&r2=1.18
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossdmabuffer.h.diff?r1=1.2&r2=1.3
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosselement.c
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosselement.h
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosshelper.c.diff?r1=1.6&r2=1.7
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosshelper.h.diff?r1=1.3&r2=1.4
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixer.c.diff?r1=1.19&r2=1.20
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixer.h.diff?r1=1.8&r2=1.9
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixertrack.c?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossmixertrack.h?rev=1.1&content-type=text/vnd.viewcvs-markup
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssink.c.diff?r1=1.104&r2=1.105
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssink.h.diff?r1=1.22&r2=1.23
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.c.diff?r1=1.57&r2=1.58
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.h.diff?r1=1.15&r2=1.16
====Begin Diffs====
Index: Makefile.am
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/Makefile.am,v
retrieving revision 1.28
retrieving revision 1.29
diff -u -d -r1.28 -r1.29
--- Makefile.am 23 Aug 2005 09:46:29 -0000 1.28
+++ Makefile.am 23 Aug 2005 13:26:21 -0000 1.29
@@ -1,9 +1,9 @@
plugin_LTLIBRARIES = libgstossaudio.la
libgstossaudio_la_SOURCES = gstossaudio.c \
- gstosselement.c \
gstosshelper.c \
gstossmixer.c \
+ gstossmixertrack.c \
gstosssink.c \
gstosssrc.c
@@ -17,10 +17,10 @@
noinst_HEADERS = gstosssink.h \
gstosssrc.h \
- gstosselement.h \
gstosshelper.h \
gstossdmabuffer.h \
- gstossmixer.h
+ gstossmixer.h \
+ gstossmixertrack.h
# noinst_PROGRAMS = #oss_probe
Index: gstossaudio.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossaudio.c,v
retrieving revision 1.17
retrieving revision 1.18
diff -u -d -r1.17 -r1.18
--- gstossaudio.c 23 Aug 2005 09:46:29 -0000 1.17
+++ gstossaudio.c 23 Aug 2005 13:26:21 -0000 1.18
@@ -23,7 +23,6 @@
#include "gst/gst-i18n-plugin.h"
-#include "gstosselement.h"
#include "gstosssink.h"
#include "gstosssrc.h"
Index: gstossdmabuffer.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossdmabuffer.h,v
retrieving revision 1.2
retrieving revision 1.3
diff -u -d -r1.2 -r1.3
--- gstossdmabuffer.h 8 Jul 2005 11:19:19 -0000 1.2
+++ gstossdmabuffer.h 23 Aug 2005 13:26:21 -0000 1.3
@@ -25,7 +25,7 @@
#include <gst/gst.h>
+#include "gstosshelper.h"
#include <gst/audio/gstringbuffer.h>
G_BEGIN_DECLS
--- gstosselement.c DELETED ---
--- gstosselement.h DELETED ---
Index: gstosshelper.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosshelper.c,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -d -r1.6 -r1.7
--- gstosshelper.c 8 Jul 2005 13:19:16 -0000 1.6
+++ gstosshelper.c 23 Aug 2005 13:26:21 -0000 1.7
@@ -2,8 +2,7 @@
* Copyright (C) 1999,2000 Erik Walthinsen <omega@...>
* 2000 Wim Taymans <wim.taymans@...>
*
- * gstosshelper.c: helper functions for easy OSS device handling.
- * See gstosshelper.h for details.
+ * gstosshelper.c: OSS helper routines
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@@ -33,6 +32,7 @@
#include <unistd.h>
[...1034 lines suppressed...]
- gst_oss_rate_add_rate (probe->rates, rate);
+ gst_oss_helper_rate_add_rate (probe->rates, rate);
return rate;
}
static void
-gst_oss_rate_add_rate (GArray * array, int rate)
+gst_oss_helper_rate_add_rate (GArray * array, int rate)
{
int i;
int val;
@@ -1212,7 +395,7 @@
static int
-gst_oss_rate_int_compare (gconstpointer a, gconstpointer b)
+gst_oss_helper_rate_int_compare (gconstpointer a, gconstpointer b)
const int *va = (const int *) a;
const int *vb = (const int *) b;
Index: gstosshelper.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosshelper.h,v
retrieving revision 1.4
diff -u -d -r1.3 -r1.4
--- gstosshelper.h 8 Jul 2005 13:19:16 -0000 1.3
+++ gstosshelper.h 23 Aug 2005 13:26:21 -0000 1.4
@@ -2,11 +2,7 @@
- * gstosshelper.h: helper functions for OSS Device handling. This
- * set of functions takes care of device setting/getting,
- * opening/closing devices, parsing caps to OSS formats/settings
- * or the other way around, device probing, supported format
- * probing and mixer integration.
+ * gstosshelper.h: OSS helper routines.
@@ -24,128 +20,24 @@
* Boston, MA 02111-1307, USA.
*/
+
#ifndef __GST_OSS_HELPER_H__
#define __GST_OSS_HELPER_H__
#include <sys/types.h>
-/* debugging category */
-GST_DEBUG_CATEGORY_EXTERN (oss_debug);
-#define GST_CAT_DEFAULT oss_debug
-
-G_BEGIN_DECLS
-enum {
- ARG_0,
- OSS_ARG_DEVICE,
- OSS_ARG_MIXER_DEVICE,
- OSS_ARG_DEVICE_NAME,
- OSS_ARG_0
-};
-typedef enum {
- GST_OSS_MODE_READ,
- GST_OSS_MODE_WRITE,
- GST_OSS_MODE_VOLUME,
- GST_OSS_MODE_MIXER
-} GstOssOpenMode;
-/*
- * Embed those two in whatever object you're creating.
- */
-typedef struct _GstOssDeviceCombination {
- gchar *dsp, *mixer;
- dev_t dev;
-} GstOssDeviceCombination;
-typedef struct _GstOssDevice {
- /* device state */
- int fd;
- int caps; /* the capabilities */
- gint format;
- gint fragment;
- guint64 fragment_time;
- gint fragment_size;
- GstOssOpenMode mode;
- GstCaps *probed_caps;
- /* stats bytes per *second* */
- guint bps;
- /* sample width in bytes */
- guint sample_width;
- /* parameters */
- gint law;
- gint endianness;
- gboolean sign;
- gint width;
- gint depth;
- gint channels;
- gint rate;
- /* mixer stuff */
- GList *tracklist;
- guint32 stereomask,
- recdevs,
- recmask,
- mixcaps;
- gint mixer_fd;
- gchar *device_name;
-} GstOssDevice;
- * class/type/interface handling for mixer/device handling.
-void gst_oss_add_mixer_type (GType type);
-void gst_oss_add_device_properties (GstElementClass * klass);
-void gst_oss_set_device_property (GstElement * element,
- GstOssDeviceCombination * c,
- GstOssDevice * dev,
- guint prop_id,
- GParamSpec * pspec,
- const GValue * value);
-void gst_oss_get_device_property (GstElement * element,
- GstOssDevice * d,
- GValue * value);
- * device open/close.
-void gst_oss_init (GObject * obj,
- GstOssDeviceCombination * c,
- GstOssDevice * dev,
- GstOssOpenMode mode);
-void gst_oss_dispose (GstOssDeviceCombination * c,
- GstOssDevice * dev);
-gboolean gst_oss_open (GstElement * element,
-void gst_oss_close (GstOssDevice * dev);
+G_BEGIN_DECLS
- * caps parsing/probing.
-gboolean gst_oss_parse_caps (GstOssDevice * dev,
- const GstCaps * caps);
-gboolean gst_oss_merge_fixed_caps (GstOssDevice * dev,
- GstCaps * caps);
-gboolean gst_oss_sync_parms (GstOssDevice * dev);
-void gst_oss_reset (GstOssDevice * dev);
+GstCaps* gst_oss_helper_probe_caps (gint fd);
-gboolean gst_oss_convert (GstOssDevice * dev,
- GstFormat src_format,
- gint64 src_value,
- GstFormat * dest_format,
- gint64 * dest_value);
-void gst_oss_probe_caps (GstOssDevice * dev);
G_END_DECLS
#endif /* __GST_OSS_HELPER_H__ */
Index: gstossmixer.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossmixer.c,v
retrieving revision 1.19
retrieving revision 1.20
diff -u -d -r1.19 -r1.20
--- gstossmixer.c 8 Jul 2005 11:19:19 -0000 1.19
+++ gstossmixer.c 23 Aug 2005 13:26:21 -0000 1.20
@@ -35,257 +35,186 @@
#include <gst/gst-i18n-plugin.h>
#include "gstossmixer.h"
+#include "gstossmixertrack.h"
#define MASK_BIT_IS_SET(mask, bit) \
(mask & (1 << bit))
-static void gst_ossmixer_track_class_init (GstOssMixerTrackClass * klass);
-static void gst_ossmixer_track_init (GstOssMixerTrack * track);
+static gboolean
+gst_ossmixer_open (GstOssMixer * mixer)
+{
+#ifdef SOUND_MIXER_INFO
+ struct mixer_info minfo;
+#endif
-static gboolean gst_ossmixer_supported (GstImplementsInterface * iface,
- GType iface_type);
-static const GList *gst_ossmixer_list_tracks (GstMixer * ossmixer);
+ g_return_val_if_fail (mixer->mixer_fd == -1, FALSE);
-static void gst_ossmixer_set_volume (GstMixer * ossmixer,
- GstMixerTrack * track, gint * volumes);
-static void gst_ossmixer_get_volume (GstMixer * ossmixer,
+ mixer->mixer_fd = open (mixer->device, O_RDWR);
+ if (mixer->mixer_fd == -1) {
+ /* this is valid. OSS devices don't need to expose a mixer */
+ GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s",
+ mixer->device, strerror (errno));
+ return FALSE;
+ }
-static void gst_ossmixer_set_record (GstMixer * ossmixer,
- GstMixerTrack * track, gboolean record);
-static void gst_ossmixer_set_mute (GstMixer * ossmixer,
- GstMixerTrack * track, gboolean mute);
+ /* get masks */
+ if (ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECMASK, &mixer->recmask) < 0
+ || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_RECSRC, &mixer->recdevs) < 0
+ || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_STEREODEVS,
+ &mixer->stereomask) < 0
+ || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_DEVMASK, &mixer->devmask) < 0
+ || ioctl (mixer->mixer_fd, SOUND_MIXER_READ_CAPS, &mixer->mixcaps) < 0) {
+ GST_DEBUG ("Failed to get device masks");
+ close (mixer->mixer_fd);
+ mixer->mixer_fd = -1;
-static const gchar **labels = NULL;
-static GstMixerTrackClass *parent_class = NULL;
+ /* get name */
+ if (ioctl (mixer->mixer_fd, SOUND_MIXER_INFO, &minfo) == 0) {
+ mixer->cardname = g_strdup (minfo.name);
+#else
+ oss->cardname = g_strdup ("Unknown");
-/* three functions: firstly, OSS has the nasty habit of inserting
- * spaces in the labels, we want to get rid of them. Secondly,
- * i18n is impossible with OSS' way of providing us with mixer
- * labels, so we make a 'given' list of i18n'ed labels. Thirdly, I
- * personally don't like the "1337" names that OSS gives to their
- * labels ("Vol", "Mic", "Rec"), I'd rather see full names. */
+ return TRUE;
+}
-fill_labels (void)
+gst_ossmixer_ensure_track_list (GstOssMixer * mixer)
- gint i, pos;
- gchar *origs[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_LABELS;
- struct
- {
- gchar *given, *wanted;
- }
- cases[] = {
- /* Note: this list is simply ripped from soundcard.h. For
- * some people, some values might be missing (3D surround,
- * etc.) - feel free to add them. That's the reason why
- * I'm doing this in such a horribly complicated way. */
- {
- "Vol ", _("Volume")}
- , {
- "Bass ", _("Bass")}
- "Trebl", _("Treble")}
- "Synth", _("Synth")}
- "Pcm ", _("PCM")}
- "Spkr ", _("Speaker")}
- "Line ", _("Line-in")}
- "Mic ", _("Microphone")}
- "CD ", _("CD")}
- "Mix ", _("Mixer")}
- "Pcm2 ", _("PCM-2")}
- "Rec ", _("Record")}
- "IGain", _("In-gain")}
- "OGain", _("Out-gain")}
- "Line1", _("Line-1")}
- "Line2", _("Line-2")}
- "Line3", _("Line-3")}
- "Digital1", _("Digital-1")}
- "Digital2", _("Digital-2")}
- "Digital3", _("Digital-3")}
- "PhoneIn", _("Phone-in")}
- "PhoneOut", _("Phone-out")}
- "Video", _("Video")}
- "Radio", _("Radio")}
- "Monitor", _("Monitor")}
- NULL, NULL}
- };
+ gint i, master = -1;
- labels = g_malloc (sizeof (gchar *) * SOUND_MIXER_NRDEVICES);
+ g_return_if_fail (mixer->mixer_fd != -1);
+ if (mixer->tracklist)
+ return;
+ /* find master volume */
+ if (mixer->devmask & SOUND_MASK_VOLUME)
+ master = SOUND_MIXER_VOLUME;
+ else if (mixer->devmask & SOUND_MASK_PCM)
+ master = SOUND_MIXER_PCM;
+ else if (mixer->devmask & SOUND_MASK_SPEAKER)
+ master = SOUND_MIXER_SPEAKER; /* doubtful... */
+ /* else: no master, so we won't set any */
+ /* build track list */
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
- for (pos = 0; cases[pos].given != NULL; pos++) {
- if (!strcmp (cases[pos].given, origs[i])) {
- labels[i] = g_strdup (cases[pos].wanted);
- break;
- }
- }
- if (cases[pos].given == NULL)
- labels[i] = g_strdup (origs[i]);
-}
+ if (mixer->devmask & (1 << i)) {
+ GstMixerTrack *track;
+ gboolean input = FALSE, stereo = FALSE, record = FALSE;
-GType
-gst_ossmixer_track_get_type (void)
-{
- static GType gst_ossmixer_track_type = 0;
+ /* track exists, make up capabilities */
+ if (MASK_BIT_IS_SET (mixer->stereomask, i))
+ stereo = TRUE;
+ if (MASK_BIT_IS_SET (mixer->recmask, i))
+ input = TRUE;
+ if (MASK_BIT_IS_SET (mixer->recdevs, i))
+ record = TRUE;
- if (!gst_ossmixer_track_type) {
- static const GTypeInfo ossmixer_track_info = {
- sizeof (GstOssMixerTrackClass),
- NULL,
- (GClassInitFunc) gst_ossmixer_track_class_init,
- sizeof (GstOssMixerTrack),
- 0,
- (GInstanceInitFunc) gst_ossmixer_track_init,
- NULL
- };
+ /* do we want this in our list? */
+ if (!((mixer->dir & GST_OSS_MIXER_CAPTURE && input == TRUE) ||
+ (mixer->dir & GST_OSS_MIXER_PLAYBACK && i != SOUND_MIXER_PCM)))
+ /* the PLAYBACK case seems hacky, but that's how 0.8 had it */
+ continue;
- gst_ossmixer_track_type =
- g_type_register_static (GST_TYPE_MIXER_TRACK,
- "GstOssMixerTrack", &ossmixer_track_info, 0);
+ /* add track to list */
+ track = gst_ossmixer_track_new (mixer->mixer_fd, i, stereo ? 2 : 1,
+ (record ? GST_MIXER_TRACK_RECORD : 0) |
+ (input ? GST_MIXER_TRACK_INPUT :
+ GST_MIXER_TRACK_OUTPUT) |
+ ((master != i) ? 0 : GST_MIXER_TRACK_MASTER));
+ mixer->tracklist = g_list_append (mixer->tracklist, track);
+ }
}
- return gst_ossmixer_track_type;
-static void
-gst_ossmixer_track_class_init (GstOssMixerTrackClass * klass)
+GstOssMixer *
+gst_ossmixer_new (const char *device, GstOssMixerDirection dir)
- parent_class = g_type_class_ref (GST_TYPE_MIXER_TRACK);
+ GstOssMixer *ret = NULL;
-gst_ossmixer_track_init (GstOssMixerTrack * track)
- track->lvol = track->rvol = 0;
- track->track_num = 0;
+ g_return_val_if_fail (device != NULL, NULL);
-GstMixerTrack *
-gst_ossmixer_track_new (GstOssDevice * oss,
- gint track_num, gint max_chans, gint flags)
- GstOssMixerTrack *osstrack;
- GstMixerTrack *track;
- gint volume;
+ ret = g_new0 (GstOssMixer, 1);
- if (!labels)
- fill_labels ();
+ ret->device = g_strdup (device);
+ ret->dir = dir;
- osstrack = g_object_new (GST_TYPE_OSSMIXER_TRACK, NULL);
- track = GST_MIXER_TRACK (osstrack);
- track->label = g_strdup (labels[track_num]);
- track->num_channels = max_chans;
- track->flags = flags;
- track->min_volume = 0;
- track->max_volume = 100;
- osstrack->track_num = track_num;
+ if (!gst_ossmixer_open (ret))
+ goto error;
- /* volume */
- if (ioctl (oss->mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
- g_warning ("Error getting device (%d) volume: %s",
- osstrack->track_num, strerror (errno));
- volume = 0;
- osstrack->lvol = (volume & 0xff);
- if (track->num_channels == 2) {
- osstrack->rvol = ((volume >> 8) & 0xff);
+ return ret;
- return track;
+error:
+ if (ret)
+ gst_ossmixer_free (ret);
-void
-gst_oss_interface_init (GstImplementsInterfaceClass * klass)
- /* default virtual functions */
- klass->supported = gst_ossmixer_supported;
+ return NULL;
void
-gst_ossmixer_interface_init (GstMixerClass * klass)
+gst_ossmixer_free (GstOssMixer * mixer)
- GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE;
+ g_return_if_fail (mixer != NULL);
- klass->list_tracks = gst_ossmixer_list_tracks;
- klass->set_volume = gst_ossmixer_set_volume;
- klass->get_volume = gst_ossmixer_get_volume;
- klass->set_mute = gst_ossmixer_set_mute;
- klass->set_record = gst_ossmixer_set_record;
+ if (mixer->device) {
+ g_free (mixer->device);
+ mixer->device = NULL;
-static gboolean
-gst_ossmixer_supported (GstImplementsInterface * iface, GType iface_type)
- GstOssDevice *oss = g_object_get_data (G_OBJECT (iface), "oss-data");
+ if (mixer->cardname) {
+ g_free (mixer->cardname);
+ mixer->cardname = NULL;
- g_return_val_if_fail (oss != NULL, FALSE);
- g_assert (iface_type == GST_TYPE_MIXER);
+ if (mixer->tracklist) {
+ g_list_foreach (mixer->tracklist, (GFunc) g_object_unref, NULL);
+ g_list_free (mixer->tracklist);
+ mixer->tracklist = NULL;
- return (oss->mixer_fd != -1);
+ if (mixer->mixer_fd != -1) {
+ g_free (mixer);
/* unused with G_DISABLE_* */
static G_GNUC_UNUSED gboolean
-gst_ossmixer_contains_track (GstOssDevice * oss, GstOssMixerTrack * osstrack)
+gst_ossmixer_contains_track (GstOssMixer * mixer, GstOssMixerTrack * osstrack)
const GList *item;
- for (item = oss->tracklist; item != NULL; item = item->next)
+ for (item = mixer->tracklist; item != NULL; item = item->next)
if (item->data == osstrack)
return TRUE;
return FALSE;
-static const GList *
-gst_ossmixer_list_tracks (GstMixer * mixer)
+const GList *
+gst_ossmixer_list_tracks (GstOssMixer * mixer)
- GstOssDevice *oss = g_object_get_data (G_OBJECT (mixer), "oss-data");
+ gst_ossmixer_ensure_track_list (mixer);
- return (const GList *) oss->tracklist;
+ return (const GList *) mixer->tracklist;
-gst_ossmixer_get_volume (GstMixer * mixer,
+void
+gst_ossmixer_get_volume (GstOssMixer * mixer,
GstMixerTrack * track, gint * volumes)
gint volume;
GstOssMixerTrack *osstrack = GST_OSSMIXER_TRACK (track);
- /* assert that we're opened and that we're using a known item */
- g_return_if_fail (oss != NULL);
- g_return_if_fail (oss->mixer_fd != -1);
- g_return_if_fail (gst_ossmixer_contains_track (oss, osstrack));
+ g_return_if_fail (gst_ossmixer_contains_track (mixer, osstrack));
if (track->flags & GST_MIXER_TRACK_MUTE) {
volumes[0] = osstrack->lvol;
@@ -294,7 +223,7 @@
}
} else {
/* get */
- if (ioctl (oss->mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
+ if (ioctl (mixer->mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
g_warning ("Error getting recording device (%d) volume: %s",
osstrack->track_num, strerror (errno));
volume = 0;
@@ -307,18 +236,15 @@
-gst_ossmixer_set_volume (GstMixer * mixer,
+gst_ossmixer_set_volume (GstOssMixer * mixer,
/* prepare the value for ioctl() */
if (!(track->flags & GST_MIXER_TRACK_MUTE)) {
@@ -328,7 +254,7 @@
/* set */
- if (ioctl (oss->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
+ if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
g_warning ("Error setting recording device (%d) volume (0x%x): %s",
osstrack->track_num, volume, strerror (errno));
return;
@@ -341,28 +267,26 @@
-gst_ossmixer_set_mute (GstMixer * mixer, GstMixerTrack * track, gboolean mute)
+gst_ossmixer_set_mute (GstOssMixer * mixer, GstMixerTrack * track,
+ gboolean mute)
int volume;
if (mute) {
volume = 0;
volume = (osstrack->lvol & 0xff);
- if (MASK_BIT_IS_SET (oss->stereomask, osstrack->track_num)) {
+ if (MASK_BIT_IS_SET (mixer->stereomask, osstrack->track_num)) {
volume |= ((osstrack->rvol & 0xff) << 8);
- if (ioctl (oss->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
+ if (ioctl (mixer->mixer_fd, MIXER_WRITE (osstrack->track_num), &volume) < 0) {
g_warning ("Error setting mixer recording device volume (0x%x): %s",
volume, strerror (errno));
return;
@@ -375,17 +299,14 @@
-gst_ossmixer_set_record (GstMixer * mixer,
+gst_ossmixer_set_record (GstOssMixer * mixer,
GstMixerTrack * track, gboolean record)
/* if there's nothing to do... */
if ((record && GST_MIXER_TRACK_HAS_FLAG (track, GST_MIXER_TRACK_RECORD)) ||
@@ -393,28 +314,28 @@
/* if we're exclusive, then we need to unset the current one(s) */
- if (oss->mixcaps & SOUND_CAP_EXCL_INPUT) {
+ if (mixer->mixcaps & SOUND_CAP_EXCL_INPUT) {
GList *track;
- for (track = oss->tracklist; track != NULL; track = track->next) {
+ for (track = mixer->tracklist; track != NULL; track = track->next) {
GstMixerTrack *turn = (GstMixerTrack *) track->data;
turn->flags &= ~GST_MIXER_TRACK_RECORD;
- oss->recdevs = 0;
+ mixer->recdevs = 0;
/* set new record bit, if needed */
if (record) {
- oss->recdevs |= (1 << osstrack->track_num);
+ mixer->recdevs |= (1 << osstrack->track_num);
- oss->recdevs &= ~(1 << osstrack->track_num);
+ mixer->recdevs &= ~(1 << osstrack->track_num);
/* set it to the device */
- if (ioctl (oss->mixer_fd, SOUND_MIXER_WRITE_RECSRC, &oss->recdevs) < 0) {
+ if (ioctl (mixer->mixer_fd, SOUND_MIXER_WRITE_RECSRC, &mixer->recdevs) < 0) {
g_warning ("Error setting mixer recording devices (0x%x): %s",
- oss->recdevs, strerror (errno));
+ mixer->recdevs, strerror (errno));
@@ -424,101 +345,3 @@
track->flags &= ~GST_MIXER_TRACK_RECORD;
-gst_ossmixer_build_list (GstOssDeviceCombination * c, GstOssDevice * oss)
- gint i, devmask, master = -1;
-#ifdef SOUND_MIXER_INFO
- struct mixer_info minfo;
-#endif
- g_return_if_fail (oss->mixer_fd == -1);
- oss->mixer_fd = open (c->mixer, O_RDWR);
- if (oss->mixer_fd == -1) {
- /* this is valid. OSS devices don't need to expose a mixer */
- GST_DEBUG ("Failed to open mixer device %s, mixing disabled: %s",
- c->mixer, strerror (errno));
- return;
- /* get masks */
- if (ioctl (oss->mixer_fd, SOUND_MIXER_READ_RECMASK, &oss->recmask) < 0 ||
- ioctl (oss->mixer_fd, SOUND_MIXER_READ_RECSRC, &oss->recdevs) < 0 ||
- ioctl (oss->mixer_fd, SOUND_MIXER_READ_STEREODEVS, &oss->stereomask) < 0
- || ioctl (oss->mixer_fd, SOUND_MIXER_READ_DEVMASK, &devmask) < 0
- || ioctl (oss->mixer_fd, SOUND_MIXER_READ_CAPS, &oss->mixcaps) < 0) {
- GST_DEBUG ("Failed to get device masks - disabling mixer");
- close (oss->mixer_fd);
- oss->mixer_fd = -1;
- /* get name */
- if (ioctl (oss->mixer_fd, SOUND_MIXER_INFO, &minfo) == 0) {
- oss->device_name = g_strdup (minfo.name);
-#else
- oss->device_name = g_strdup ("Unknown");
- /* find master volume */
- if (devmask & SOUND_MASK_VOLUME)
- master = SOUND_MIXER_VOLUME;
- else if (devmask & SOUND_MASK_PCM)
- master = SOUND_MIXER_PCM;
- else if (devmask & SOUND_MASK_SPEAKER)
- master = SOUND_MIXER_SPEAKER; /* doubtful... */
- /* else: no master, so we won't set any */
- /* build track list */
- for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
- if (devmask & (1 << i)) {
- GstMixerTrack *track;
- gboolean input = FALSE, stereo = FALSE, record = FALSE;
- /* track exists, make up capabilities */
- if (MASK_BIT_IS_SET (oss->stereomask, i))
- stereo = TRUE;
- if (MASK_BIT_IS_SET (oss->recmask, i))
- input = TRUE;
- if (MASK_BIT_IS_SET (oss->recdevs, i))
- record = TRUE;
- /* do we want this in our list? */
- if ((oss->mode == GST_OSS_MODE_READ && input == FALSE) ||
- (oss->mode == GST_OSS_MODE_WRITE && i != SOUND_MIXER_PCM))
- continue;
- /* add track to list */
- track = gst_ossmixer_track_new (oss, i, stereo ? 2 : 1,
- (record ? GST_MIXER_TRACK_RECORD : 0) |
- (input ? GST_MIXER_TRACK_INPUT :
- GST_MIXER_TRACK_OUTPUT) |
- ((master != i) ? 0 : GST_MIXER_TRACK_MASTER));
- oss->tracklist = g_list_append (oss->tracklist, track);
-gst_ossmixer_free_list (GstOssDevice * oss)
- if (oss->mixer_fd == -1)
- g_list_foreach (oss->tracklist, (GFunc) g_object_unref, NULL);
- g_list_free (oss->tracklist);
- oss->tracklist = NULL;
- if (oss->device_name) {
- g_free (oss->device_name);
- oss->device_name = NULL;
- close (oss->mixer_fd);
- oss->mixer_fd = -1;
Index: gstossmixer.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossmixer.h,v
retrieving revision 1.8
retrieving revision 1.9
diff -u -d -r1.8 -r1.9
--- gstossmixer.h 8 Jul 2005 11:19:19 -0000 1.8
+++ gstossmixer.h 23 Aug 2005 13:26:21 -0000 1.9
@@ -19,48 +19,153 @@
#ifndef __GST_OSS_MIXER_H__
#define __GST_OSS_MIXER_H__
#include <gst/interfaces/mixer.h>
#include "gstosshelper.h"
-#define GST_TYPE_OSSMIXER_TRACK \
- (gst_ossmixer_track_get_type ())
-#define GST_OSSMIXER_TRACK(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_OSSMIXER_TRACK, \
- GstOssMixerTrack))
-#define GST_OSSMIXER_TRACK_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_OSSMIXER_TRACK, \
- GstOssMixerTrackClass))
-#define GST_IS_OSSMIXER_TRACK(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_OSSMIXER_TRACK))
-#define GST_IS_OSSMIXER_TRACK_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_OSSMIXER_TRACK))
-typedef struct _GstOssMixerTrack {
- GstMixerTrack parent;
+#define GST_OSS_MIXER(obj) ((GstOssMixer*)(obj))
- gint lvol, rvol;
- gint track_num;
-} GstOssMixerTrack;
-typedef struct _GstOssMixerTrackClass {
- GstMixerTrackClass parent;
-} GstOssMixerTrackClass;
+typedef enum {
+ GST_OSS_MIXER_CAPTURE = 1<<0,
+ GST_OSS_MIXER_PLAYBACK = 1<<1,
+ GST_OSS_MIXER_ALL = GST_OSS_MIXER_CAPTURE | GST_OSS_MIXER_PLAYBACK
+} GstOssMixerDirection;
-GType gst_ossmixer_track_get_type (void);
-void gst_ossmixer_interface_init (GstMixerClass *klass);
-void gst_oss_interface_init (GstImplementsInterfaceClass *klass);
-void gst_ossmixer_build_list (GstOssDeviceCombination * c,
- GstOssDevice *oss);
-void gst_ossmixer_free_list (GstOssDevice *oss);
+typedef struct _GstOssMixer GstOssMixer;
+struct _GstOssMixer {
+ GList * tracklist; /* list of available tracks */
+ gint mixer_fd;
+ gchar * device;
+ gchar * cardname;
+ gint recmask;
+ gint recdevs;
+ gint stereomask;
+ gint devmask;
+ gint mixcaps;
+ GstOssMixerDirection dir;
+};
+GstOssMixer* gst_ossmixer_new (const gchar *device,
+ GstOssMixerDirection dir);
+void gst_ossmixer_free (GstOssMixer *mixer);
+const GList* gst_ossmixer_list_tracks (GstOssMixer * mixer);
+void gst_ossmixer_set_volume (GstOssMixer * mixer,
+ GstMixerTrack * track,
+ gint * volumes);
+void gst_ossmixer_get_volume (GstOssMixer * mixer,
+void gst_ossmixer_set_record (GstOssMixer * mixer,
+ gboolean record);
+void gst_ossmixer_set_mute (GstOssMixer * mixer,
+ gboolean mute);
+#define GST_IMPLEMENT_OSS_MIXER_METHODS(Type, interface_as_function) \
+static gboolean \
+interface_as_function ## _supported (Type *this, GType iface_type) \
+{ \
+ g_assert (iface_type == GST_TYPE_MIXER); \
+ \
+ return (this->mixer != NULL); \
+} \
+static const GList* \
+interface_as_function ## _list_tracks (GstMixer * mixer) \
+ Type *this = (Type*) mixer; \
+ g_return_val_if_fail (this != NULL, NULL); \
+ g_return_val_if_fail (this->mixer != NULL, NULL); \
+ return gst_ossmixer_list_tracks (this->mixer); \
+static void \
+interface_as_function ## _set_volume (GstMixer * mixer, GstMixerTrack * track, \
+ gint * volumes) \
+ g_return_if_fail (this != NULL); \
+ g_return_if_fail (this->mixer != NULL); \
+ gst_ossmixer_set_volume (this->mixer, track, volumes); \
+interface_as_function ## _get_volume (GstMixer * mixer, GstMixerTrack * track, \
+ gst_ossmixer_get_volume (this->mixer, track, volumes); \
+interface_as_function ## _set_record (GstMixer * mixer, GstMixerTrack * track, \
+ gboolean record) \
+ gst_ossmixer_set_record (this->mixer, track, record); \
+interface_as_function ## _set_mute (GstMixer * mixer, GstMixerTrack * track, \
+ gboolean mute) \
+ gst_ossmixer_set_mute (this->mixer, track, mute); \
+interface_as_function ## _interface_init (GstMixerClass * klass) \
+ GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE; \
+ /* set up the interface hooks */ \
+ klass->list_tracks = interface_as_function ## _list_tracks; \
+ klass->set_volume = interface_as_function ## _set_volume; \
+ klass->get_volume = interface_as_function ## _get_volume; \
+ klass->set_mute = interface_as_function ## _set_mute; \
+ klass->set_record = interface_as_function ## _set_record; \
#endif /* __GST_OSS_MIXER_H__ */
--- NEW FILE: gstossmixertrack.c ---
/* GStreamer OSS Mixer implementation
* Copyright (C) 2003 Ronald Bultje <rbultje@...>
*
* gstossmixer.c: mixer interface implementation for OSS
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdio.h>
#include <stdlib.h>
#include <fcntl.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include <gst/gst-i18n-plugin.h>
#include "gstossmixertrack.h"
#define MASK_BIT_IS_SET(mask, bit) \
(mask & (1 << bit))
G_DEFINE_TYPE (GstOssMixerTrack, gst_ossmixer_track, GST_TYPE_MIXER_TRACK);
static void
gst_ossmixer_track_class_init (GstOssMixerTrackClass * klass)
{
/* nop */
}
gst_ossmixer_track_init (GstOssMixerTrack * track)
track->lvol = track->rvol = 0;
track->track_num = 0;
static const gchar **labels = NULL;
/* three functions: firstly, OSS has the nasty habit of inserting
* spaces in the labels, we want to get rid of them. Secondly,
* i18n is impossible with OSS' way of providing us with mixer
* labels, so we make a 'given' list of i18n'ed labels. Thirdly, I
* personally don't like the "1337" names that OSS gives to their
* labels ("Vol", "Mic", "Rec"), I'd rather see full names. */
fill_labels (void)
gint i, pos;
gchar *origs[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_LABELS;
struct
{
gchar *given, *wanted;
}
cases[] = {
/* Note: this list is simply ripped from soundcard.h. For
* some people, some values might be missing (3D surround,
* etc.) - feel free to add them. That's the reason why
* I'm doing this in such a horribly complicated way. */
{
"Vol ", _("Volume")}
, {
"Bass ", _("Bass")}
"Trebl", _("Treble")}
"Synth", _("Synth")}
"Pcm ", _("PCM")}
"Spkr ", _("Speaker")}
"Line ", _("Line-in")}
"Mic ", _("Microphone")}
"CD ", _("CD")}
"Mix ", _("Mixer")}
"Pcm2 ", _("PCM-2")}
"Rec ", _("Record")}
"IGain", _("In-gain")}
"OGain", _("Out-gain")}
"Line1", _("Line-1")}
"Line2", _("Line-2")}
"Line3", _("Line-3")}
"Digital1", _("Digital-1")}
"Digital2", _("Digital-2")}
"Digital3", _("Digital-3")}
"PhoneIn", _("Phone-in")}
"PhoneOut", _("Phone-out")}
"Video", _("Video")}
"Radio", _("Radio")}
"Monitor", _("Monitor")}
NULL, NULL}
};
labels = g_malloc (sizeof (gchar *) * SOUND_MIXER_NRDEVICES);
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++) {
for (pos = 0; cases[pos].given != NULL; pos++) {
if (!strcmp (cases[pos].given, origs[i])) {
labels[i] = g_strdup (cases[pos].wanted);
break;
}
}
if (cases[pos].given == NULL)
labels[i] = g_strdup (origs[i]);
GstMixerTrack *
gst_ossmixer_track_new (gint mixer_fd,
gint track_num, gint max_chans, gint flags)
GstOssMixerTrack *osstrack;
GstMixerTrack *track;
gint volume;
if (!labels)
fill_labels ();
osstrack = g_object_new (GST_TYPE_OSSMIXER_TRACK, NULL);
track = GST_MIXER_TRACK (osstrack);
track->label = g_strdup (labels[track_num]);
track->num_channels = max_chans;
track->flags = flags;
track->min_volume = 0;
track->max_volume = 100;
osstrack->track_num = track_num;
/* volume */
if (ioctl (mixer_fd, MIXER_READ (osstrack->track_num), &volume) < 0) {
g_warning ("Error getting device (%d) volume: %s",
osstrack->track_num, strerror (errno));
volume = 0;
osstrack->lvol = (volume & 0xff);
if (track->num_channels == 2) {
osstrack->rvol = ((volume >> 8) & 0xff);
return track;
--- NEW FILE: gstossmixertrack.h ---
* gstossmixertrack.h: OSS mixer tracks
#ifndef __GST_OSS_MIXER_TRACK_H__
#define __GST_OSS_MIXER_TRACK_H__
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
#include "gstosshelper.h"
G_BEGIN_DECLS
#define GST_TYPE_OSSMIXER_TRACK \
(gst_ossmixer_track_get_type ())
#define GST_OSSMIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_OSSMIXER_TRACK, \
GstOssMixerTrack))
#define GST_OSSMIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_OSSMIXER_TRACK, \
GstOssMixerTrackClass))
#define GST_IS_OSSMIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_OSSMIXER_TRACK))
#define GST_IS_OSSMIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_OSSMIXER_TRACK))
typedef struct _GstOssMixerTrack {
GstMixerTrack parent;
gint lvol, rvol;
gint track_num;
} GstOssMixerTrack;
typedef struct _GstOssMixerTrackClass {
GstMixerTrackClass parent;
} GstOssMixerTrackClass;
GType gst_ossmixer_track_get_type (void);
GstMixerTrack* gst_ossmixer_track_new (gint mixer_fd,
gint track_num, gint max_chans, gint flags);
G_END_DECLS
#endif /* __GST_OSS_MIXER_TRACK_H__ */
Index: gstosssink.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssink.c,v
retrieving revision 1.104
retrieving revision 1.105
diff -u -d -r1.104 -r1.105
--- gstosssink.c 23 Aug 2005 09:46:29 -0000 1.104
+++ gstosssink.c 23 Aug 2005 13:26:21 -0000 1.105
@@ -162,27 +162,22 @@
GST_DEBUG ("initializing osssink");
- osssink->element = g_object_new (GST_TYPE_OSSELEMENT, NULL);
+ osssink->fd = -1;
static GstCaps *
gst_oss_sink_getcaps (GstBaseSink * bsink)
GstOssSink *osssink;
- GstOssElement *element;
GstCaps *caps;
osssink = GST_OSSSINK (bsink);
- element = osssink->element;
- gst_osselement_probe_caps (element);
- if (element->probed_caps == NULL) {
- caps =
- gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
+ if (osssink->fd == -1) {
+ caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD
(bsink)));
- caps = gst_caps_ref (element->probed_caps);
+ caps = gst_oss_helper_probe_caps (osssink->fd);
return caps;
Index: gstosssink.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssink.h,v
retrieving revision 1.22
retrieving revision 1.23
diff -u -d -r1.22 -r1.23
--- gstosssink.h 10 Jul 2005 12:52:20 -0000 1.22
+++ gstosssink.h 23 Aug 2005 13:26:21 -0000 1.23
@@ -28,7 +28,7 @@
#include <gst/audio/gstaudiosink.h>
@@ -44,8 +44,6 @@
struct _GstOssSink {
GstAudioSink sink;
gint fd;
gint bytes_per_sample;
};
Index: gstosssrc.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.c,v
retrieving revision 1.57
retrieving revision 1.58
diff -u -d -r1.57 -r1.58
--- gstosssrc.c 23 Aug 2005 09:46:29 -0000 1.57
+++ gstosssrc.c 23 Aug 2005 13:26:21 -0000 1.58
@@ -192,27 +192,23 @@
GST_DEBUG ("initializing osssrc");
+ osssrc->fd = -1;
osssrc->device = g_strdup ("/dev/dsp");
- osssrc->element = g_object_new (GST_TYPE_OSSELEMENT, NULL);
gst_oss_src_getcaps (GstBaseSrc * bsrc)
GstOssSrc *osssrc;
osssrc = GST_OSS_SRC (bsrc);
- element = osssrc->element;
- gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc)));
+ if (osssrc->fd == -1) {
+ caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD
+ (bsrc)));
+ caps = gst_oss_helper_probe_caps (osssrc->fd);
Index: gstosssrc.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.h,v
retrieving revision 1.15
retrieving revision 1.16
diff -u -d -r1.15 -r1.16
--- gstosssrc.h 23 Aug 2005 09:46:29 -0000 1.15
+++ gstosssrc.h 23 Aug 2005 13:26:21 -0000 1.16
#include <gst/audio/gstaudiosrc.h>
struct _GstOssSrc {
GstAudioSrc src;
|
|
From: <ensonic@fr...> - 2005-08-23 11:54:15
|
CVS Root: /cvs/gstreamer
Module: gstreamer
Changes by: ensonic
Date: Tue Aug 23 2005 04:54:10 PDT
Log message:
* docs/gst/gstreamer-sections.txt:
* docs/gst/tmpl/.cvsignore:
* docs/gst/tmpl/gstbin.sgml:
* docs/gst/tmpl/gstbuffer.sgml:
* gst/base/gstbasesrc.c:
* gst/gstbin.c: (gst_bin_get_type), (gst_bin_class_init):
* gst/gstbuffer.c:
* gst/gstbuffer.h:
* tools/gst-launch.1.in:
inlined more doc comments, added missing comments and fixed comments
fixed typos
Modified files:
. : ChangeLog
docs/gst : gstreamer-sections.txt
docs/gst/tmpl : .cvsignore
gst : gstbin.c gstbuffer.c gstbuffer.h
gst/base : gstbasesrc.c
tools : gst-launch.1.in
Removed files:
docs/gst/tmpl : gstbin.sgml gstbuffer.sgml
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/ChangeLog.diff?r1=1.1383&r2=1.1384
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/gstreamer-sections.txt.diff?r1=1.154&r2=1.155
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/.cvsignore.diff?r1=1.3&r2=1.4
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/gstbin.sgml
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/docs/gst/tmpl/gstbuffer.sgml
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstbin.c.diff?r1=1.252&r2=1.253
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstbuffer.c.diff?r1=1.103&r2=1.104
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstbuffer.h.diff?r1=1.89&r2=1.90
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/base/gstbasesrc.c.diff?r1=1.46&r2=1.47
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/tools/gst-launch.1.in.diff?r1=1.7&r2=1.8
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gstreamer/ChangeLog,v
retrieving revision 1.1383
retrieving revision 1.1384
diff -u -d -r1.1383 -r1.1384
--- ChangeLog 23 Aug 2005 11:38:28 -0000 1.1383
+++ ChangeLog 23 Aug 2005 11:53:57 -0000 1.1384
@@ -1,3 +1,17 @@
+2005-08-23 Stefan Kost <ensonic@...>
+
+ * docs/gst/gstreamer-sections.txt:
+ * docs/gst/tmpl/.cvsignore:
+ * docs/gst/tmpl/gstbin.sgml:
+ * docs/gst/tmpl/gstbuffer.sgml:
+ * gst/base/gstbasesrc.c:
+ * gst/gstbin.c: (gst_bin_get_type), (gst_bin_class_init):
+ * gst/gstbuffer.c:
+ * gst/gstbuffer.h:
+ * tools/gst-launch.1.in:
+ inlined more doc comments, added missing comments and fixed comments
+ fixed typos
2005-08-23 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/gstbuffer.c: (gst_buffer_new_and_alloc):
Index: gstreamer-sections.txt
RCS file: /cvs/gstreamer/gstreamer/docs/gst/gstreamer-sections.txt,v
retrieving revision 1.154
retrieving revision 1.155
diff -u -d -r1.154 -r1.155
--- gstreamer-sections.txt 22 Aug 2005 21:03:33 -0000 1.154
+++ gstreamer-sections.txt 23 Aug 2005 11:53:58 -0000 1.155
@@ -85,6 +85,7 @@
GST_BUFFER_SIZE
GST_BUFFER_TIMESTAMP
GST_BUFFER_DURATION
+GST_BUFFER_CAPS
GST_BUFFER_OFFSET
GST_BUFFER_OFFSET_END
@@ -109,17 +110,20 @@
GST_BUFFER_TIMESTAMP_IS_VALID
GST_BUFFER_OFFSET_IS_VALID
GST_BUFFER_OFFSET_END_IS_VALID
+GST_BUFFER_MALLOCDATA
-<SUBSECTION>
gst_buffer_stamp
gst_buffer_join
gst_buffer_merge
<SUBSECTION Standard>
GST_BUFFER
+GST_BUFFER_CLASS
GST_IS_BUFFER
+GST_IS_BUFFER_CLASS
GST_TYPE_BUFFER
GST_TYPE_BUFFER_FLAG
+GST_BUFFER_CAST
<SUBSECTION Private>
gst_buffer_get_type
gst_buffer_flag_get_type
Index: .cvsignore
RCS file: /cvs/gstreamer/gstreamer/docs/gst/tmpl/.cvsignore,v
retrieving revision 1.3
retrieving revision 1.4
diff -u -d -r1.3 -r1.4
--- .cvsignore 5 Aug 2005 13:42:09 -0000 1.3
+++ .cvsignore 23 Aug 2005 11:53:58 -0000 1.4
@@ -1,10 +1,14 @@
*.bak
unused-build.stamp
+gst.sgml
gstadapter.sgml
gstbasesink.sgml
gstbasesrc.sgml
gstbasetransform.sgml
+gstbin.sgml
+gstbuffer.sgml
gstcollectpads.sgml
+gstevent.sgml
gstfakesrc.sgml
gstfakesink.sgml
gstfilesrc.sgml
--- gstbin.sgml DELETED ---
--- gstbuffer.sgml DELETED ---
Index: gstbasesrc.c
RCS file: /cvs/gstreamer/gstreamer/gst/base/gstbasesrc.c,v
retrieving revision 1.46
retrieving revision 1.47
diff -u -d -r1.46 -r1.47
--- gstbasesrc.c 22 Aug 2005 19:22:34 -0000 1.46
+++ gstbasesrc.c 23 Aug 2005 11:53:58 -0000 1.47
@@ -251,7 +251,7 @@
}
/**
- * gst_base_src_get_live:
+ * gst_base_src_is_live:
* @src: base source instance
*
* Check if an element is in live mode.
Index: gstbin.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstbin.c,v
retrieving revision 1.252
retrieving revision 1.253
diff -u -d -r1.252 -r1.253
--- gstbin.c 22 Aug 2005 21:03:33 -0000 1.252
+++ gstbin.c 23 Aug 2005 11:53:58 -0000 1.253
@@ -22,6 +22,46 @@
* MT safe.
*/
+/**
+ * SECTION:gstbin
+ * @short_description: Base class for elements that contain other elements
+ *
+ * GstBin is the simplest of the container elements, allowing elements to
+ * become children of itself. Pads from the child elements can be ghosted to
+ * the bin, making the bin itself look transparently like any other element,
+ * allowing for deep nesting of predefined sub-pipelines.
+ * A new GstBin is created with gst_bin_new(). Use a #GstPipeline instead if you
+ * want to create a toplevel bin because a normal bin doesn't have a scheduler
+ * of its own.
+ *
+ * After the bin has been created you will typically add elements to it with
+ * gst_bin_add(). You can remove elements with gst_bin_remove().
+ * An element can be retrieved from a bin with gst_bin_get_by_name(), using the
+ * elements name. gst_bin_get_by_name_recurse_up() is mainly used for internal
+ * purposes and will query the parent bins when the element is not found in the
+ * current bin.
+ * The list of elements in a bin can be retrieved with gst_bin_get_list().
+ * After the bin has been set to the PLAYING state (with gst_element_set_state()),
+ * gst_bin_iterate() is used to process the elements in the bin.
+ * The "element_added" signal is fired whenever a new element is added to the
+ * bin. Likewise the "element_removed" signal is fired whenever an element is
+ * removed from the bin.
+ * gst_bin_destroy() is used to destroy the bin.
+ * To control the selection of the clock in a bin, you can use the following
+ * methods:
+ * gst_bin_auto_clock() to let the bin select a clock automatically,
+ * gst_bin_get_clock() to get the current clock of the bin and
+ * gst_bin_use_clock() to specify a clock explicitly.
+ * Note that the default behaviour is to automatically select a clock from one
+ * of the clock providers in the bin.
+ */
#include "gst_private.h"
@@ -183,10 +223,24 @@
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+ /**
+ * GstBin::element-added:
+ * @bin: the object which emitted the signal.
+ * @element: the element that was added to the bin
+ *
+ * Will be emitted if a new element was removed/added to this bin.
+ */
gst_bin_signals[ELEMENT_ADDED] =
g_signal_new ("element-added", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST, G_STRUCT_OFFSET (GstBinClass, element_added), NULL,
NULL, gst_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+ * GstBin::element-removed:
+ * @element: the element that was removed from the bin
+ * Will be emitted if an element was removed from this bin.
gst_bin_signals[ELEMENT_REMOVED] =
g_signal_new ("element-removed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST, G_STRUCT_OFFSET (GstBinClass, element_removed), NULL,
@@ -1688,7 +1742,7 @@
- * gst_bin_get_all_by_interface:
+ * gst_bin_iterate_all_by_interface:
* @bin: bin to find elements in
* @interface: interface to be implemented by interface
Index: gstbuffer.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstbuffer.c,v
retrieving revision 1.103
retrieving revision 1.104
diff -u -d -r1.103 -r1.104
--- gstbuffer.c 14 Aug 2005 22:29:07 -0000 1.103
+++ gstbuffer.c 23 Aug 2005 11:53:58 -0000 1.104
@@ -19,7 +19,74 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
-
+ * SECTION:gstbuffer
+ * @short_description: Data-passing buffer type, supporting sub-buffers.
+ * @see_also: #GstPad, #GstMiniObject
+ * Buffers are the basic unit of data transfer in GStreamer. The GstBuffer type
+ * provides all the state necessary to define a region of memory as part of a
+ * stream. Sub-buffers are also supported, allowing a smaller region of a
+ * buffer to become its own buffer, with mechanisms in place to ensure that
+ * neither memory space goes away.
+ * Buffers are usually created with gst_buffer_new(). After a buffer has been
+ * created one will typically allocate memory for it and set the size of the
+ * buffer data. The following example creates a buffer that can hold a given
+ * video frame with a given width, height and bits per plane.
+ * <example>
+ * <title>Creating a buffer for a video frame</title>
+ * <programlisting>
+ * GstBuffer *buffer;
+ * gint size, width, height, bpp;
+ *
+ * ...
+ * size = width * height * bpp;
+ * buffer = gst_buffer_new ();
+ * GST_BUFFER_SIZE (buffer) = size;
+ * GST_BUFFER_MALLOCDATA (buffer) = g_alloc (size);
+ * GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+ * </programlisting>
+ * </example>
+ * Alternatively, use gst_buffer_new_and_alloc()
+ * to create a buffer with preallocated data of a given size.
+ * If an element knows what pad you will push the buffer out on, it should use
+ * gst_pad_alloc_buffer() instead to create a buffer. This allows downstream
+ * elements to provide special buffers to write in, like hardware buffers.
+ * gst_buffer_ref() is used to increase the refcount of a buffer. This must be
+ * done when you want to keep a handle to the buffer after pushing it to the
+ * next element.
+ * To efficiently create a smaller buffer out of an existing one, you can
+ * use gst_buffer_create_sub().
+ * If the plug-in wants to modify the buffer in-place, it should first obtain
+ * a buffer that is safe to modify by using gst_buffer_make_writable(). This
+ * function is optimized so that a copy will only be made when it is necessary.
+ * Several flags of the buffer can be set and unset with the GST_BUFFER_FLAG_SET()
+ * and GST_BUFFER_FLAG_UNSET() macros. Use GST_BUFFER_FLAG_IS_SET() to test it
+ * a certain #GstBufferFlag is set.
+ * Buffers can be efficiently merged into a larger buffer with gst_buffer_merge()
+ * and gst_buffer_span() if the gst_buffer_is_span_fast() function returns TRUE.
+ * An element should either unref the buffer or push it out on a src pad
+ * using gst_pad_push() (see #GstPad).
+ * Buffers usually are freed by unreffing them with gst_buffer_unref().
+ * Do not use gst_buffer_free() : this function effectively frees the buffer
+ * regardless of the refcount, which is dangerous.
+ * Last reviewed on August 12th, 2004 (0.8.5)
#include "gstbuffer.h"
@@ -320,7 +387,7 @@
- * gst_buffer_create_subbuffer:
+ * gst_buffer_create_sub:
* @parent: a parent #GstBuffer to create a subbuffer from.
* @offset: the offset into parent #GstBuffer.
* @size: the size of the new #GstBuffer sub-buffer (with size > 0).
Index: gstbuffer.h
RCS file: /cvs/gstreamer/gstreamer/gst/gstbuffer.h,v
retrieving revision 1.89
retrieving revision 1.90
diff -u -d -r1.89 -r1.90
--- gstbuffer.h 27 Jul 2005 18:33:02 -0000 1.89
+++ gstbuffer.h 23 Aug 2005 11:53:58 -0000 1.90
@@ -33,6 +33,11 @@
typedef struct _GstBuffer GstBuffer;
typedef struct _GstBufferClass GstBufferClass;
+ * GST_BUFFER_TRACE_NAME:
+ * The name used for tracing memory allocations.
#define GST_BUFFER_TRACE_NAME "GstBuffer"
#define GST_TYPE_BUFFER (gst_buffer_get_type())
@@ -43,25 +48,157 @@
#define GST_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_BUFFER, GstBufferClass))
#define GST_BUFFER_CAST(obj) ((GstBuffer *)(obj))
+ * GST_BUFFER_FLAGS:
+ * @buf: a #GstBuffer to retrieve the flags from.
+ * Gets the flags from this buffer.
+ * Returns: the set of #GstBufferFlag items
#define GST_BUFFER_FLAGS(buf) GST_MINI_OBJECT_FLAGS(buf)
+ * GST_BUFFER_FLAG_IS_SET:
+ * @buf: a #GstBuffer to query flags of.
+ * @flag: the #GstBufferFlag to check.
+ * Gives the status of a given flag of a buffer.
+ * Returns: %TRUE if flag is set.
#define GST_BUFFER_FLAG_IS_SET(buf,flag) GST_MINI_OBJECT_FLAG_IS_SET (buf, flag)
+ * GST_BUFFER_FLAG_SET:
+ * @buf: a #GstBuffer to modify flags of.
+ * @flag: the #GstBufferFlag to set.
+ * Sets a buffer flag.
#define GST_BUFFER_FLAG_SET(buf,flag) GST_MINI_OBJECT_FLAG_SET (buf, flag)
+ * GST_BUFFER_FLAG_UNSET:
+ * @flag: the #GstBufferFlag to clear.
+ * Clears a buffer flag.
#define GST_BUFFER_FLAG_UNSET(buf,flag) GST_MINI_OBJECT_FLAG_UNSET (buf, flag)
+ * GST_BUFFER_DATA:
+ * @buf: a #GstBuffer to get data pointer of.
+ * Retrieves a pointer to the data element of this buffer.
+ * Returns: the pointer to the actual data contents of the buffer.
#define GST_BUFFER_DATA(buf) (GST_BUFFER_CAST(buf)->data)
+ * GST_BUFFER_SIZE:
+ * @buf: a #GstBuffer to get data size of.
+ * Gets the size of the data in this buffer.
+ * Returns: the buffer size in bytes
#define GST_BUFFER_SIZE(buf) (GST_BUFFER_CAST(buf)->size)
+ * GST_BUFFER_TIMESTAMP:
+ * @buf: a #GstBuffer to get the timestamp of.:
+ * Gets the timestamp for this buffer.
+ * Returns: the timestamp for this buffer
#define GST_BUFFER_TIMESTAMP(buf) (GST_BUFFER_CAST(buf)->timestamp)
+ * GST_BUFFER_DURATION:
+ * @buf: a #GstBuffer to get the duration from.
+ * Gets the duration in nanoseconds of the data in the buffer.
+ * Value will be %GST_CLOCK_TIME_NONE if the duration is unknown.
+ * Returns: the duration of the buffer
#define GST_BUFFER_DURATION(buf) (GST_BUFFER_CAST(buf)->duration)
+ * GST_BUFFER_CAPS:
+ * @buf: a #GstBuffer to get the caps of.
+ * Gets the caps for this buffer.
+ * Returns: the #GstCaps for this buffer
#define GST_BUFFER_CAPS(buf) (GST_BUFFER_CAST(buf)->caps)
+ * GST_BUFFER_OFFSET:
+ * @buf: a #GstBuffer to get the offset of.
+ * Gets the offset in the source file of the beginning of this buffer.
+ * Returns: the start offset for this buffer
#define GST_BUFFER_OFFSET(buf) (GST_BUFFER_CAST(buf)->offset)
+ * GST_BUFFER_OFFSET_END:
+ * Gets the offset in the source file of the end of this buffer.
+ * Returns: the end offset for this buffer
#define GST_BUFFER_OFFSET_END(buf) (GST_BUFFER_CAST(buf)->offset_end)
+ * GST_BUFFER_MALLOCDATA:
+ * @buf: a #GstBuffer to get access to the malloc_data field
+ * If the buffers data should be automatically freed by buffer management at the
+ * end of the buffers lifecycle, also set the data to the mallocdata field.
#define GST_BUFFER_MALLOCDATA(buf) (GST_BUFFER_CAST(buf)->malloc_data)
+ * GST_BUFFER_OFFSET_NONE:
+ * Constant for no-offset return results.
#define GST_BUFFER_OFFSET_NONE ((guint64)-1)
+ * GST_BUFFER_DURATION_IS_VALID:
+ * @buffer: the #GstBuffer to check for the duration
+ * Tests if the duration is known.
+ * Returns: %TRUE for success
#define GST_BUFFER_DURATION_IS_VALID(buffer) (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_DURATION (buffer)))
+ * GST_BUFFER_TIMESTAMP_IS_VALID:
+ * @buffer: the #GstBuffer to check for the timestamp
+ * Tests if the timestamp is known.
#define GST_BUFFER_TIMESTAMP_IS_VALID(buffer) (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buffer)))
+ * GST_BUFFER_OFFSET_IS_VALID:
+ * @buffer: the #GstBuffer to check for the start offset
+ * Tests if the start offset is known.
#define GST_BUFFER_OFFSET_IS_VALID(buffer) (GST_BUFFER_OFFSET (buffer) != GST_BUFFER_OFFSET_NONE)
+ * GST_BUFFER_OFFSET_END_IS_VALID:
+ * @buffer: the #GstBuffer to check for the end offset
+ * Tests if the end offset is known.
#define GST_BUFFER_OFFSET_END_IS_VALID(buffer) (GST_BUFFER_OFFSET_END (buffer) != GST_BUFFER_OFFSET_NONE)
@@ -115,9 +252,9 @@
guint64 offset;
guint64 offset_end;
- /*< private >*/
guint8 *malloc_data;
+ /*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
@@ -131,6 +268,14 @@
GstBuffer* gst_buffer_new (void);
GstBuffer* gst_buffer_new_and_alloc (guint size);
+ * gst_buffer_set_data:
+ * @buf: The buffer to modify
+ * @data: The data to set on the buffer
+ * @size: The size to set on the buffer
+ * A convenience function to set the data and size on a buffer.
#define gst_buffer_set_data(buf, data, size) \
G_STMT_START { \
GST_BUFFER_DATA (buf) = data; \
@@ -138,13 +283,59 @@
} G_STMT_END
/* refcounting */
+ * gst_buffer_ref:
+ * @buf: a #GstBuffer to increase the refcount of.
+ * Increases the refcount of the given buffer by one.
#define gst_buffer_ref(buf) GST_BUFFER_CAST (gst_mini_object_ref (GST_MINI_OBJECT (buf)))
+ * gst_buffer_unref:
+ * @buf: a #GstBuffer to decrease the refcount of.
+ * Decreases the refcount of the buffer. If the refcount reaches 0, the buffer
+ * will be freed.
#define gst_buffer_unref(buf) gst_mini_object_unref (GST_MINI_OBJECT (buf))
/* copy buffer */
+ * gst_buffer_copy:
+ * @buf: a #GstBuffer to copy.
+ * Copies the given buffer using the copy function of the parent #GstData
+ * structure.
+ * Returns: a new #GstBuffer copy of the buffer.
#define gst_buffer_copy(buf) GST_BUFFER_CAST (gst_mini_object_copy (GST_MINI_OBJECT (buf)))
+ * gst_buffer_is_writable:
+ * @buf: a #GstBuffer to check
+ * Tests if you can safely write data into a buffer's data array.
+ * Returns: %TRUE if buffer is writable
#define gst_buffer_is_writable(buf) gst_mini_object_is_writable (GST_MINI_OBJECT (buf))
+ * gst_buffer_make_writable:
+ * @buf: a #GstBuffer to make writable
+ * Makes a buffer writable.
+ * Returns: a #GstBuffer that is writable
#define gst_buffer_make_writable(buf) GST_BUFFER_CAST (gst_mini_object_make_writable (GST_MINI_OBJECT (buf)))
+ * gst_buffer_replace:
+ * @obuf: a #GstBuffer to receive the data
+ * @nbuf: a #GstBuffer to take the data from
+ * Replaces the data in @obuf with the one in @nbuf
#define gst_buffer_replace(obuf,nbuf) gst_mini_object_replace ((GstMiniObject **)(obuf), GST_MINI_OBJECT (nbuf))
GstCaps* gst_buffer_get_caps (GstBuffer *buffer);
Index: gst-launch.1.in
RCS file: /cvs/gstreamer/gstreamer/tools/gst-launch.1.in,v
retrieving revision 1.7
retrieving revision 1.8
diff -u -d -r1.7 -r1.8
--- gst-launch.1.in 8 Feb 2005 18:05:29 -0000 1.7
+++ gst-launch.1.in 23 Aug 2005 11:53:58 -0000 1.8
@@ -135,7 +135,7 @@
(second line) is a short cut for using the first line and "thread" as the
BINTYPE.
.br
-Please not the dot that has to be used after the BINTYPE.
+Please note the dot that has to be used after the BINTYPE.
.B Links
|
|
From: <thomasvs@fr...> - 2005-08-23 11:38:44
|
CVS Root: /cvs/gstreamer
Module: gstreamer
Changes by: thomasvs
Date: Tue Aug 23 2005 04:38:40 PDT
Log message:
* gst/gstbuffer.c: (gst_buffer_new_and_alloc):
some debugging
* gst/gstcaps.h:
whitespace fixes
* gst/gstpad.c: (gst_pad_activate_push), (gst_pad_alloc_buffer):
more debugging
* gst/gststructure.c: (gst_caps_structure_fixate_field_boolean):
* gst/gststructure.h:
add a fixate function for booleans; add a FIXME that these func
names should probably be gst_structure_fixate_*
Modified files:
. : ChangeLog
gst : gstcaps.h gstpad.c gststructure.c gststructure.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/ChangeLog.diff?r1=1.1382&r2=1.1383
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstcaps.h.diff?r1=1.89&r2=1.90
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gstpad.c.diff?r1=1.426&r2=1.427
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gststructure.c.diff?r1=1.48&r2=1.49
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gstreamer/gst/gststructure.h.diff?r1=1.23&r2=1.24
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gstreamer/ChangeLog,v
retrieving revision 1.1382
retrieving revision 1.1383
diff -u -d -r1.1382 -r1.1383
--- ChangeLog 22 Aug 2005 21:03:32 -0000 1.1382
+++ ChangeLog 23 Aug 2005 11:38:28 -0000 1.1383
@@ -1,3 +1,16 @@
+2005-08-23 Thomas Vander Stichele <thomas at apestaart dot org>
+
+ * gst/gstbuffer.c: (gst_buffer_new_and_alloc):
+ some debugging
+ * gst/gstcaps.h:
+ whitespace fixes
+ * gst/gstpad.c: (gst_pad_activate_push), (gst_pad_alloc_buffer):
+ more debugging
+ * gst/gststructure.c: (gst_caps_structure_fixate_field_boolean):
+ * gst/gststructure.h:
+ add a fixate function for booleans; add a FIXME that these func
+ names should probably be gst_structure_fixate_*
2005-08-23 Stefan Kost <ensonic@...>
* docs/gst/gstreamer-docs.sgml:
Index: gstcaps.h
RCS file: /cvs/gstreamer/gstreamer/gst/gstcaps.h,v
retrieving revision 1.89
retrieving revision 1.90
diff -u -d -r1.89 -r1.90
--- gstcaps.h 15 Jul 2005 16:10:41 -0000 1.89
+++ gstcaps.h 23 Aug 2005 11:38:28 -0000 1.90
@@ -92,10 +92,10 @@
va_list var_args);
/* reference counting */
-GstCaps * gst_caps_ref (GstCaps* caps);
+GstCaps * gst_caps_ref (GstCaps* caps);
GstCaps * gst_caps_copy (const GstCaps * caps);
-GstCaps * gst_caps_make_writable (GstCaps *caps);
-void gst_caps_unref (GstCaps* caps);
+GstCaps * gst_caps_make_writable (GstCaps *caps);
+void gst_caps_unref (GstCaps* caps);
GstCaps * gst_static_caps_get (GstStaticCaps *static_caps);
@@ -107,7 +107,7 @@
int gst_caps_get_size (const GstCaps *caps);
GstStructure * gst_caps_get_structure (const GstCaps *caps,
int index);
-GstCaps * gst_caps_copy_nth (const GstCaps * caps, gint nth);
+GstCaps * gst_caps_copy_nth (const GstCaps * caps, gint nth);
void gst_caps_truncate (GstCaps * caps);
void gst_caps_set_simple (GstCaps *caps,
char *field, ...);
@@ -120,11 +120,11 @@
gboolean gst_caps_is_empty (const GstCaps *caps);
gboolean gst_caps_is_fixed (const GstCaps *caps);
gboolean gst_caps_is_always_compatible (const GstCaps *caps1,
- const GstCaps *caps2);
+ const GstCaps *caps2);
gboolean gst_caps_is_subset (const GstCaps *subset,
const GstCaps *superset);
gboolean gst_caps_is_equal (const GstCaps *caps1,
+ const GstCaps *caps2);
/* operations */
GstCaps * gst_caps_intersect (const GstCaps *caps1,
Index: gstpad.c
RCS file: /cvs/gstreamer/gstreamer/gst/gstpad.c,v
retrieving revision 1.426
retrieving revision 1.427
diff -u -d -r1.426 -r1.427
--- gstpad.c 21 Aug 2005 15:01:18 -0000 1.426
+++ gstpad.c 23 Aug 2005 11:38:28 -0000 1.427
@@ -2302,9 +2302,9 @@
* @buf: a newly allocated buffer
*
* Allocates a new, empty buffer optimized to push to pad @pad. This
- * function only works if @pad is a source pad and has a peer.
+ * function only works if @pad is a source pad and has a peer.
- * You need to check the caps of the buffer after performing this
+ * You need to check the caps of the buffer after performing this
* function and renegotiate to the format if needed.
* A new, empty #GstBuffer will be put in the @buf argument.
@@ -2346,9 +2346,9 @@
goto flushing;
GST_CAT_DEBUG (GST_CAT_PADS,
- "calling bufferallocfunc &%s (@%p) of peer pad %s:%s",
+ "calling bufferallocfunc &%s (@%p) of peer pad %s:%s for size %d",
GST_DEBUG_FUNCPTR_NAME (bufferallocfunc),
- &bufferallocfunc, GST_DEBUG_PAD_NAME (peer));
+ &bufferallocfunc, GST_DEBUG_PAD_NAME (peer), size);
GST_UNLOCK (peer);
ret = bufferallocfunc (peer, offset, size, caps, buf);
Index: gststructure.c
RCS file: /cvs/gstreamer/gstreamer/gst/gststructure.c,v
retrieving revision 1.48
retrieving revision 1.49
diff -u -d -r1.48 -r1.49
--- gststructure.c 31 Jul 2005 11:59:32 -0000 1.48
+++ gststructure.c 23 Aug 2005 11:38:28 -0000 1.49
@@ -1592,6 +1592,7 @@
* Returns: TRUE if the structure could be fixated
*/
+/* FIXME: rename to gst_structure_... */
gboolean
gst_caps_structure_fixate_field_nearest_int (GstStructure * structure,
const char *field_name, int target)
@@ -1709,3 +1710,57 @@
return FALSE;
}
+/**
+ * gst_caps_structure_fixate_field_boolean:
+ * @structure: a #GstStructure
+ * @field_name: a field in @structure
+ * @target: the target value of the fixation
+ *
+ * Fixates a #GstStructure by changing the given @field_name field to the given
+ * @target boolean if that field is not fixed yet.
+ * Returns: TRUE if the structure could be fixated
+ */
+gboolean
+gst_caps_structure_fixate_field_boolean (GstStructure * structure,
+ const char *field_name, gboolean target)
+{
+ const GValue *value;
+ g_return_val_if_fail (gst_structure_has_field (structure, field_name), FALSE);
+ g_return_val_if_fail (IS_MUTABLE (structure), FALSE);
+ value = gst_structure_get_value (structure, field_name);
+ if (G_VALUE_TYPE (value) == G_TYPE_BOOLEAN) {
+ /* already fixed */
+ return FALSE;
+ } else if (G_VALUE_TYPE (value) == GST_TYPE_LIST) {
+ const GValue *list_value;
+ int i, n;
+ int best = 0;
+ int best_index = -1;
+ n = gst_value_list_get_size (value);
+ for (i = 0; i < n; i++) {
+ list_value = gst_value_list_get_value (value, i);
+ if (G_VALUE_TYPE (list_value) == G_TYPE_BOOLEAN) {
+ gboolean x = g_value_get_boolean (list_value);
+ if (best_index == -1 || x == target) {
+ best_index = i;
+ best = x;
+ }
+ }
+ }
+ if (best_index != -1) {
+ gst_structure_set (structure, field_name, G_TYPE_BOOLEAN, best, NULL);
+ return TRUE;
+ }
+ return FALSE;
+}
Index: gststructure.h
RCS file: /cvs/gstreamer/gstreamer/gst/gststructure.h,v
retrieving revision 1.23
retrieving revision 1.24
diff -u -d -r1.23 -r1.24
--- gststructure.h 31 Jul 2005 11:59:32 -0000 1.23
+++ gststructure.h 23 Aug 2005 11:38:28 -0000 1.24
@@ -144,6 +144,10 @@
const char *field_name,
double target);
+gboolean gst_caps_structure_fixate_field_boolean (GstStructure *structure,
+ const char *field_name,
+ gboolean target);
G_END_DECLS
|
|
From: <wingo@fr...> - 2005-08-23 09:46:44
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: wingo
Date: Tue Aug 23 2005 02:46:41 PDT
Log message:
2005-08-23 Andy Wingo <wingo@...>
* sys/oss/gstosssrc.h:
* sys/oss/gstosssrc.c: Totally ported, dude.
* sys/oss/Makefile.am:
* sys/oss/gstossaudio.c: Add osssrc.
* sys/oss/gstosssink.c: We do native byte order.
Modified files:
. : ChangeLog
sys/oss : Makefile.am gstossaudio.c gstosssink.c
gstosssrc.c gstosssrc.h
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1801&r2=1.1802
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/Makefile.am.diff?r1=1.27&r2=1.28
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstossaudio.c.diff?r1=1.16&r2=1.17
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssink.c.diff?r1=1.103&r2=1.104
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.c.diff?r1=1.56&r2=1.57
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/sys/oss/gstosssrc.h.diff?r1=1.14&r2=1.15
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1801
retrieving revision 1.1802
diff -u -d -r1.1801 -r1.1802
--- ChangeLog 23 Aug 2005 07:54:31 -0000 1.1801
+++ ChangeLog 23 Aug 2005 09:46:29 -0000 1.1802
@@ -1,3 +1,13 @@
+2005-08-23 Andy Wingo <wingo@...>
+
+ * sys/oss/gstosssrc.h:
+ * sys/oss/gstosssrc.c: Totally ported, dude.
+ * sys/oss/Makefile.am:
+ * sys/oss/gstossaudio.c: Add osssrc.
+
+ * sys/oss/gstosssink.c: We do native byte order.
2005-08-23 Owen Fraser-Green <owen@...>
* gst/realmedia/rmdemux.c (gst_rmdemux_src_event): Fixed bug
Index: Makefile.am
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/Makefile.am,v
retrieving revision 1.27
retrieving revision 1.28
diff -u -d -r1.27 -r1.28
--- Makefile.am 8 Jul 2005 11:19:19 -0000 1.27
+++ Makefile.am 23 Aug 2005 09:46:29 -0000 1.28
@@ -4,7 +4,8 @@
gstosselement.c \
gstosshelper.c \
gstossmixer.c \
- gstosssink.c
+ gstosssink.c \
+ gstosssrc.c
# gstossdmabuffer.c
Index: gstossaudio.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstossaudio.c,v
retrieving revision 1.16
retrieving revision 1.17
diff -u -d -r1.16 -r1.17
--- gstossaudio.c 4 Aug 2005 20:05:51 -0000 1.16
+++ gstossaudio.c 23 Aug 2005 09:46:29 -0000 1.17
@@ -35,9 +35,9 @@
plugin_init (GstPlugin * plugin)
{
if ( /*!gst_element_register (plugin, "ossmixer", GST_RANK_PRIMARY,
- GST_TYPE_OSSELEMENT) ||
- !gst_element_register (plugin, "osssrc", GST_RANK_PRIMARY,
- GST_TYPE_OSSSRC) || */
+ GST_TYPE_OSSELEMENT) || */
+ !gst_element_register (plugin, "osssrc", GST_RANK_PRIMARY,
+ GST_TYPE_OSS_SRC) ||
!gst_element_register (plugin, "osssink", GST_RANK_SECONDARY,
GST_TYPE_OSSSINK)) {
return FALSE;
Index: gstosssink.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssink.c,v
retrieving revision 1.103
retrieving revision 1.104
diff -u -d -r1.103 -r1.104
--- gstosssink.c 8 Aug 2005 16:43:47 -0000 1.103
+++ gstosssink.c 23 Aug 2005 09:46:29 -0000 1.104
@@ -69,10 +69,8 @@
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
- //"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- //"signed = (boolean) { TRUE, FALSE }, "
- "endianness = (int) LITTLE_ENDIAN, "
- "signed = (boolean) TRUE, "
+ "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+ "signed = (boolean) { TRUE, FALSE }, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
Index: gstosssrc.c
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.c,v
retrieving revision 1.56
retrieving revision 1.57
diff -u -d -r1.56 -r1.57
--- gstosssrc.c 10 Jul 2005 12:52:20 -0000 1.56
+++ gstosssrc.c 23 Aug 2005 09:46:29 -0000 1.57
@@ -1,6 +1,6 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@...>
- * 2000 Wim Taymans <wtay@...>
+ * 2000,2005 Wim Taymans <wim@...>
*
* gstosssrc.c:
@@ -23,62 +23,66 @@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
#include <sys/ioctl.h>
+#include <fcntl.h>
#include <errno.h>
#include <unistd.h>
#include <string.h>
-
-#ifdef HAVE_OSS_INCLUDE_IN_SYS
#include <sys/soundcard.h>
-#else
-#ifdef HAVE_OSS_INCLUDE_IN_ROOT
-#include <soundcard.h>
-#include <machine/soundcard.h>
-#endif /* HAVE_OSS_INCLUDE_IN_ROOT */
-#endif /* HAVE_OSS_INCLUDE_IN_SYS */
+#include "gstosssrc.h"
-#include <gstosssrc.h>
-#include <gstosselement.h>
-#include <gst/audio/audioclock.h>
-/* elementfactory information */
static GstElementDetails gst_oss_src_details =
GST_ELEMENT_DETAILS ("Audio Source (OSS)",
"Source/Audio",
- "Read from the sound card",
- "Erik Walthinsen <omega@...>");
+ "Capture from a sound card via OSS",
+ "Erik Walthinsen <omega@...>, " "Wim Taymans <wim@...>");
-/* OssSrc signals and args */
enum
- /* FILL ME */
- LAST_SIGNAL
+ PROP_0,
+ PROP_DEVICE,
+ PROP_DEVICE_NAME,
};
-enum
-{
- ARG_0,
- ARG_BUFFERSIZE,
- ARG_FRAGMENT
-};
+GST_BOILERPLATE (GstOssSrc, gst_oss_src, GstAudioSrc, GST_TYPE_AUDIO_SRC);
+/*
+GST_BOILERPLATE_WITH_INTERFACE (GstOssSrc, gst_oss_src, GstAudioSrc, GST_TYPE_AUDIO_SRC,
+ GstMixer, GST_TYPE_MIXER, gst_oss_src_mixer);
+GST_IMPLEMENT_OSS_MIXER_METHODS (GstOssSrc, gst_oss_src_mixer);
+*/
+static void gst_oss_src_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_oss_src_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_oss_src_dispose (GObject * object);
+static GstCaps *gst_oss_src_getcaps (GstBaseSrc * bsrc);
+static gboolean gst_oss_src_open (GstAudioSrc * asrc);
+static gboolean gst_oss_src_close (GstAudioSrc * asrc);
+static gboolean gst_oss_src_prepare (GstAudioSrc * asrc,
+ GstRingBufferSpec * spec);
+static gboolean gst_oss_src_unprepare (GstAudioSrc * asrc);
+static guint gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length);
+static guint gst_oss_src_delay (GstAudioSrc * asrc);
+static void gst_oss_src_reset (GstAudioSrc * asrc);
static GstStaticPadTemplate osssrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
"signed = (boolean) { TRUE, FALSE }, "
- "depth = (int) { 8, 16 }, "
+ "depth = (int) 16, "
"audio/x-raw-int, "
@@ -87,64 +91,11 @@
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]")
);
-static void gst_oss_src_base_init (gpointer g_class);
-static void gst_oss_src_class_init (GstOssSrcClass * klass);
-static void gst_oss_src_init (GstOssSrc * osssrc);
-static void gst_oss_src_dispose (GObject * object);
-static GstPadLinkReturn gst_oss_src_src_link (GstPad * pad, GstPad * peer);
-static GstCaps *gst_oss_src_getcaps (GstPad * pad);
-static const GstFormat *gst_oss_src_get_formats (GstPad * pad);
-static gboolean gst_oss_src_convert (GstPad * pad,
- GstFormat src_format, gint64 src_value,
- GstFormat * dest_format, gint64 * dest_value);
-static void gst_oss_src_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_oss_src_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-static GstElementStateReturn gst_oss_src_change_state (GstElement * element);
-static void gst_oss_src_set_clock (GstElement * element, GstClock * clock);
-static GstClock *gst_oss_src_get_clock (GstElement * element);
-static GstClockTime gst_oss_src_get_time (GstClock * clock, gpointer data);
-static const GstEventMask *gst_oss_src_get_event_masks (GstPad * pad);
-static gboolean gst_oss_src_src_event (GstPad * pad, GstEvent * event);
-static gboolean gst_oss_src_send_event (GstElement * element, GstEvent * event);
-static const GstQueryType *gst_oss_src_get_query_types (GstPad * pad);
-static gboolean gst_oss_src_src_query (GstPad * pad, GstQueryType type,
- GstFormat * format, gint64 * value);
-static void gst_oss_src_loop (GstPad * pad);
-static GstElementClass *parent_class = NULL;
-/*static guint gst_oss_src_signals[LAST_SIGNAL] = { 0 }; */
-GType
-gst_oss_src_get_type (void)
+static void
+gst_oss_src_dispose (GObject * object)
- static GType osssrc_type = 0;
- if (!osssrc_type) {
- static const GTypeInfo osssrc_info = {
- sizeof (GstOssSrcClass),
- gst_oss_src_base_init,
- NULL,
- (GClassInitFunc) gst_oss_src_class_init,
- sizeof (GstOssSrc),
- 0,
- (GInstanceInitFunc) gst_oss_src_init,
- };
- osssrc_type =
- g_type_register_static (GST_TYPE_OSSELEMENT, "GstOssSrc", &osssrc_info,
- 0);
- }
- return osssrc_type;
+ G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
@@ -153,6 +104,7 @@
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_oss_src_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&osssrc_src_factory));
@@ -161,435 +113,336 @@
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstBaseSrcClass *gstbasesrc_class;
+ GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+ GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ gstbasesrc_class = (GstBaseSrcClass *) klass;
+ gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+ gstaudiosrc_class = (GstAudioSrcClass *) klass;
- parent_class = g_type_class_ref (GST_TYPE_OSSELEMENT);
- gobject_class->set_property = gst_oss_src_set_property;
- gobject_class->get_property = gst_oss_src_get_property;
+ gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_oss_src_dispose);
+ gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_oss_src_get_property);
+ gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_oss_src_set_property);
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BUFFERSIZE,
- g_param_spec_ulong ("buffersize", "Buffer Size",
- "The size of the buffers with samples", 0, G_MAXULONG, 0,
- G_PARAM_READWRITE));
- g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FRAGMENT,
- g_param_spec_int ("fragment", "Fragment",
- "The fragment as 0xMMMMSSSS (MMMM = total fragments, 2^SSSS = fragment size)",
- 0, G_MAXINT, 6, G_PARAM_READWRITE));
+ gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_src_getcaps);
- gobject_class->dispose = gst_oss_src_dispose;
+ gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_oss_src_open);
+ gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_src_prepare);
+ gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_src_unprepare);
+ gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_oss_src_close);
+ gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_oss_src_read);
+ gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_oss_src_delay);
+ gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_oss_src_reset);
- gstelement_class->change_state = gst_oss_src_change_state;
- gstelement_class->send_event = gst_oss_src_send_event;
+ g_object_class_install_property (gobject_class, PROP_DEVICE,
+ g_param_spec_string ("device", "Device",
+ "OSS device (usually /dev/dspN)", "/dev/dsp", G_PARAM_READWRITE));
- gstelement_class->set_clock = gst_oss_src_set_clock;
- gstelement_class->get_clock = gst_oss_src_get_clock;
+ g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,
+ g_param_spec_string ("device-name", "Device name",
+ "Human-readable name of the sound device", "", G_PARAM_READABLE));
-gst_oss_src_init (GstOssSrc * osssrc)
+gst_oss_src_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
- osssrc->srcpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&osssrc_src_factory), "src");
- gst_pad_set_loop_function (osssrc->srcpad, gst_oss_src_loop);
- gst_pad_set_getcaps_function (osssrc->srcpad, gst_oss_src_getcaps);
- gst_pad_set_link_function (osssrc->srcpad, gst_oss_src_src_link);
- gst_pad_set_convert_function (osssrc->srcpad, gst_oss_src_convert);
- gst_pad_set_formats_function (osssrc->srcpad, gst_oss_src_get_formats);
- gst_pad_set_event_function (osssrc->srcpad, gst_oss_src_src_event);
- gst_pad_set_event_mask_function (osssrc->srcpad, gst_oss_src_get_event_masks);
- gst_pad_set_query_function (osssrc->srcpad, gst_oss_src_src_query);
- gst_pad_set_query_type_function (osssrc->srcpad, gst_oss_src_get_query_types);
- gst_element_add_pad (GST_ELEMENT (osssrc), osssrc->srcpad);
- osssrc->buffersize = 4096;
- osssrc->curoffset = 0;
+ GstOssSrc *src;
- osssrc->provided_clock =
- gst_audio_clock_new ("ossclock", gst_oss_src_get_time, osssrc);
- gst_object_set_parent (GST_OBJECT (osssrc->provided_clock),
- GST_OBJECT (osssrc));
+ src = GST_OSS_SRC (object);
- osssrc->clock = NULL;
+ switch (prop_id) {
+ case PROP_DEVICE:
+ if (src->device)
+ g_free (src->device);
+ src->device = g_strdup (g_value_get_string (value));
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ }
-gst_oss_src_dispose (GObject * object)
+gst_oss_src_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
- GstOssSrc *osssrc = (GstOssSrc *) object;
- if (osssrc->provided_clock != NULL) {
- gst_object_unparent (GST_OBJECT (osssrc->provided_clock));
- osssrc->provided_clock = NULL;
+ g_value_set_string (value, src->device);
+ case PROP_DEVICE_NAME:
+ g_value_set_string (value, src->device_name);
}
+}
- G_OBJECT_CLASS (parent_class)->dispose (object);
+gst_oss_src_init (GstOssSrc * osssrc)
+{
+ GST_DEBUG ("initializing osssrc");
+ osssrc->device = g_strdup ("/dev/dsp");
+ osssrc->element = g_object_new (GST_TYPE_OSSELEMENT, NULL);
static GstCaps *
-gst_oss_src_getcaps (GstPad * pad)
+gst_oss_src_getcaps (GstBaseSrc * bsrc)
- GstOssSrc *src;
+ GstOssSrc *osssrc;
+ GstOssElement *element;
GstCaps *caps;
- src = GST_OSSSRC (GST_PAD_PARENT (pad));
+ osssrc = GST_OSS_SRC (bsrc);
+ element = osssrc->element;
- gst_osselement_probe_caps (GST_OSSELEMENT (src));
+ gst_osselement_probe_caps (element);
- if (GST_OSSELEMENT (src)->probed_caps == NULL) {
- caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+ if (element->probed_caps == NULL) {
+ caps =
+ gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc)));
} else {
- caps = gst_caps_copy (GST_OSSELEMENT (src)->probed_caps);
+ caps = gst_caps_ref (element->probed_caps);
return caps;
-static GstPadLinkReturn
-gst_oss_src_src_link (GstPad * pad, GstPad * peer)
+static gint
+ilog2 (gint x)
- return GST_RPAD_LINKFUNC (peer) (peer, pad);
+ /* well... hacker's delight explains... */
+ x = x | (x >> 1);
+ x = x | (x >> 2);
+ x = x | (x >> 4);
+ x = x | (x >> 8);
+ x = x | (x >> 16);
+ x = x - ((x >> 1) & 0x55555555);
+ x = (x & 0x33333333) + ((x >> 2) & 0x33333333);
+ x = (x + (x >> 4)) & 0x0f0f0f0f;
+ x = x + (x >> 8);
+ x = x + (x >> 16);
+ return (x & 0x0000003f) - 1;
-static gboolean
-gst_oss_src_negotiate (GstPad * pad)
- GstCaps *allowed;
- //allowed = gst_pad_get_allowed_caps (pad);
- allowed = NULL;
- if (!gst_osselement_merge_fixed_caps (GST_OSSELEMENT (src), allowed))
- return FALSE;
+#define SET_PARAM(_oss, _label, _name, _val) \
+G_STMT_START { \
+ int _tmp = _val; \
+ if (ioctl(_oss->fd, _name, &_tmp) == -1) { \
+ perror(_label); \
+ return FALSE; \
+ } \
+ GST_DEBUG_OBJECT (_oss, _label " %d", _tmp); \
+} G_STMT_END
- if (!gst_osselement_sync_parms (GST_OSSELEMENT (src)))
+#define GET_PARAM(oss, label, name, val) \
+ if (ioctl(oss->fd, name, val) == -1) { \
+ perror(label); \
- /* set caps on src pad */
- GST_PAD_CAPS (src->srcpad) =
- gst_caps_new_simple ("audio/x-raw-int",
- "endianness", G_TYPE_INT, GST_OSSELEMENT (src)->endianness,
- "signed", G_TYPE_BOOLEAN, GST_OSSELEMENT (src)->sign,
- "width", G_TYPE_INT, GST_OSSELEMENT (src)->width,
- "depth", G_TYPE_INT, GST_OSSELEMENT (src)->depth,
- "rate", G_TYPE_INT, GST_OSSELEMENT (src)->rate,
- "channels", G_TYPE_INT, GST_OSSELEMENT (src)->channels, NULL);
+gst_oss_src_get_format (GstBufferFormat fmt)
+ gint result;
- return TRUE;
+ switch (fmt) {
+ case GST_MU_LAW:
+ result = AFMT_MU_LAW;
+ case GST_A_LAW:
+ result = AFMT_A_LAW;
+ case GST_IMA_ADPCM:
+ result = AFMT_IMA_ADPCM;
+ case GST_U8:
+ result = AFMT_U8;
+ case GST_S16_LE:
+ result = AFMT_S16_LE;
+ case GST_S16_BE:
+ result = AFMT_S16_BE;
+ case GST_S8:
+ result = AFMT_S8;
+ case GST_U16_LE:
+ result = AFMT_U16_LE;
+ case GST_U16_BE:
+ result = AFMT_U16_BE;
+ case GST_MPEG:
+ result = AFMT_MPEG;
+ result = 0;
+ return result;
-static GstClockTime
-gst_oss_src_get_time (GstClock * clock, gpointer data)
+static gboolean
+gst_oss_src_open (GstAudioSrc * asrc)
- GstOssSrc *osssrc = GST_OSSSRC (data);
- audio_buf_info info;
- if (!GST_OSSELEMENT (osssrc)->bps)
- return 0;
- if (ioctl (GST_OSSELEMENT (osssrc)->fd, SNDCTL_DSP_GETISPACE, &info) < 0)
+ GstOssSrc *oss;
+ int mode;
- return (osssrc->curoffset * GST_OSSELEMENT (osssrc)->sample_width +
- info.bytes) * GST_SECOND / GST_OSSELEMENT (osssrc)->bps;
-}
+ oss = GST_OSS_SRC (asrc);
-static GstClock *
-gst_oss_src_get_clock (GstElement * element)
- GstOssSrc *osssrc;
+ mode = O_RDONLY;
+ mode |= O_NONBLOCK;
- osssrc = GST_OSSSRC (element);
+ oss->fd = open (oss->device, mode, 0);
+ if (oss->fd == -1) {
+ perror (oss->device);
+ return FALSE;
- return GST_CLOCK (osssrc->provided_clock);
+ return TRUE;
-static void
-gst_oss_src_set_clock (GstElement * element, GstClock * clock)
+gst_oss_src_close (GstAudioSrc * asrc)
- osssrc->clock = clock;
+ close (GST_OSS_SRC (asrc)->fd);
-gst_oss_src_loop (GstPad * pad)
+gst_oss_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
- GstBuffer *buf;
- glong readbytes;
- glong readsamples;
- GST_DEBUG ("attempting to read something from the soundcard");
- if (src->need_eos) {
- src->need_eos = FALSE;
- gst_pad_push_event (pad, gst_event_new (GST_EVENT_EOS));
- return;
- buf = gst_buffer_new_and_alloc (src->buffersize);
- if (!GST_PAD_CAPS (pad)) {
- /* nothing was negotiated, we can decide on a format */
- if (!gst_oss_src_negotiate (pad)) {
- gst_buffer_unref (buf);
- GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL), (NULL));
- return;
- }
- if (GST_OSSELEMENT (src)->bps == 0) {
- gst_buffer_unref (buf);
- GST_ELEMENT_ERROR (src, CORE, NEGOTIATION, (NULL),
- ("format wasn't negotiated before chain function"));
+ struct audio_buf_info info;
+ int tmp;
- readbytes = read (GST_OSSELEMENT (src)->fd, GST_BUFFER_DATA (buf),
- src->buffersize);
- if (readbytes < 0) {
- GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
- if (readbytes == 0) {
+ mode = fcntl (oss->fd, F_GETFL);
+ mode &= ~O_NONBLOCK;
+ if (fcntl (oss->fd, F_SETFL, mode) == -1) {
- readsamples = readbytes * GST_OSSELEMENT (src)->rate /
- GST_OSSELEMENT (src)->bps;
+ tmp = gst_oss_src_get_format (spec->format);
+ if (tmp == 0)
+ goto wrong_format;
- GST_BUFFER_SIZE (buf) = readbytes;
- GST_BUFFER_OFFSET (buf) = src->curoffset;
- GST_BUFFER_OFFSET_END (buf) = src->curoffset + readsamples;
- GST_BUFFER_DURATION (buf) =
- readsamples * GST_SECOND / GST_OSSELEMENT (src)->rate;
+ tmp = ilog2 (spec->segsize);
+ tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;
+ GST_DEBUG ("set segsize: %d, segtotal: %d, value: %08x", spec->segsize,
+ spec->segtotal, tmp);
- /* if we have a clock */
- if (src->clock) {
- if (src->clock == src->provided_clock) {
- /* if it's our own clock, we can be very accurate */
- GST_BUFFER_TIMESTAMP (buf) =
- src->curoffset * GST_SECOND / GST_OSSELEMENT (src)->rate;
- } else {
- /* somebody elses clock, timestamp with that clock, no discontinuity in
- * the stream since the OFFSET is updated correctly. Elements can stretch
- * to match timestamps */
- gst_element_get_time (GST_ELEMENT (src)) - GST_BUFFER_DURATION (buf);
- } else {
- /* no clock, no timestamp */
- GST_BUFFER_TIMESTAMP (buf) = GST_CLOCK_TIME_NONE;
+ SET_PARAM (oss, "SETFRAGMENT", SNDCTL_DSP_SETFRAGMENT, tmp);
- src->curoffset += readsamples;
+ SET_PARAM (oss, "RESET", SNDCTL_DSP_RESET, 0);
- GST_DEBUG ("pushed buffer from soundcard of %ld bytes, timestamp %"
- G_GINT64_FORMAT, readbytes, GST_BUFFER_TIMESTAMP (buf));
+ SET_PARAM (oss, "SETFMT", SNDCTL_DSP_SETFMT, tmp);
+ if (spec->channels == 2)
+ SET_PARAM (oss, "STEREO", SNDCTL_DSP_STEREO, 1);
+ SET_PARAM (oss, "CHANNELS", SNDCTL_DSP_CHANNELS, spec->channels);
+ SET_PARAM (oss, "SPEED", SNDCTL_DSP_SPEED, spec->rate);
- gst_pad_push (pad, buf);
+ GET_PARAM (oss, "GETISPACE", SNDCTL_DSP_GETISPACE, &info);
- return;
+ spec->segsize = info.fragsize;
+ spec->segtotal = info.fragstotal;
+ spec->bytes_per_sample = 4;
+ oss->bytes_per_sample = 4;
+ memset (spec->silence_sample, 0, spec->bytes_per_sample);
-gst_oss_src_set_property (GObject * object, guint prop_id, const GValue * value,
- GParamSpec * pspec)
+ GST_DEBUG ("got segsize: %d, segtotal: %d, value: %08x", spec->segsize,
- src = GST_OSSSRC (object);
- switch (prop_id) {
- case ARG_BUFFERSIZE:
- src->buffersize = g_value_get_ulong (value);
- break;
- case ARG_FRAGMENT:
- GST_OSSELEMENT (src)->fragment = g_value_get_int (value);
- gst_osselement_sync_parms (GST_OSSELEMENT (src));
- default:
+wrong_format:
+ {
+ GST_DEBUG ("wrong format %d\n", spec->format);
-gst_oss_src_get_property (GObject * object, guint prop_id, GValue * value,
+gst_oss_src_unprepare (GstAudioSrc * asrc)
+ /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */
- g_value_set_ulong (value, src->buffersize);
- g_value_set_int (value, GST_OSSELEMENT (src)->fragment);
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ if (!gst_oss_src_close (asrc))
+ goto couldnt_close;
-static GstElementStateReturn
-gst_oss_src_change_state (GstElement * element)
- GstOssSrc *osssrc = GST_OSSSRC (element);
+ if (!gst_oss_src_open (asrc))
+ goto couldnt_reopen;
- GST_DEBUG ("osssrc: state change");
- switch (GST_STATE_TRANSITION (element)) {
- case GST_STATE_READY_TO_PAUSED:
- osssrc->curoffset = 0;
- case GST_STATE_PAUSED_TO_PLAYING:
- gst_audio_clock_set_active (GST_AUDIO_CLOCK (osssrc->provided_clock),
- TRUE);
- case GST_STATE_PLAYING_TO_PAUSED:
- FALSE);
- case GST_STATE_PAUSED_TO_READY:
- if (GST_FLAG_IS_SET (element, GST_OSSSRC_OPEN))
- ioctl (GST_OSSELEMENT (osssrc)->fd, SNDCTL_DSP_RESET, 0);
+couldnt_close:
+ GST_DEBUG ("Could not close the audio device");
+couldnt_reopen:
+ GST_DEBUG ("Could not reopen the audio device");
- if (GST_ELEMENT_CLASS (parent_class)->change_state)
- return GST_ELEMENT_CLASS (parent_class)->change_state (element);
- return GST_STATE_SUCCESS;
-static const GstFormat *
-gst_oss_src_get_formats (GstPad * pad)
- static const GstFormat formats[] = {
- GST_FORMAT_TIME,
- GST_FORMAT_DEFAULT,
- GST_FORMAT_BYTES,
- 0
- };
- return formats;
-gst_oss_src_convert (GstPad * pad, GstFormat src_format, gint64 src_value,
- GstFormat * dest_format, gint64 * dest_value)
- osssrc = GST_OSSSRC (GST_PAD_PARENT (pad));
- return gst_osselement_convert (GST_OSSELEMENT (osssrc), src_format, src_value,
- dest_format, dest_value);
-static const GstEventMask *
-gst_oss_src_get_event_masks (GstPad * pad)
+static guint
+gst_oss_src_read (GstAudioSrc * asrc, gpointer data, guint length)
- static const GstEventMask gst_oss_src_src_event_masks[] = {
- {GST_EVENT_EOS, 0},
- {GST_EVENT_SIZE, 0},
- {0,}
- return gst_oss_src_src_event_masks;
+ return read (GST_OSS_SRC (asrc)->fd, data, length);
-gst_oss_src_src_event (GstPad * pad, GstEvent * event)
+gst_oss_src_delay (GstAudioSrc * asrc)
- gboolean retval = FALSE;
+ gint delay = 0;
+ gint ret;
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:
- osssrc->need_eos = TRUE;
- retval = TRUE;
- case GST_EVENT_SIZE:
- {
- GstFormat format;
- gint64 value;
+#ifdef SNDCTL_DSP_GETODELAY
+ ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);
+#else
+ ret = -1;
+#endif
+ if (ret < 0) {
+ audio_buf_info info;
- format = GST_FORMAT_BYTES;
+ ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info);
- /* convert to bytes */
- if (gst_osselement_convert (GST_OSSELEMENT (osssrc),
- GST_EVENT_SIZE_FORMAT (event),
- GST_EVENT_SIZE_VALUE (event), &format, &value)) {
- osssrc->buffersize = GST_EVENT_SIZE_VALUE (event);
- g_object_notify (G_OBJECT (osssrc), "buffersize");
- retval = TRUE;
- }
+ delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);
- gst_event_unref (event);
- return retval;
-gst_oss_src_send_event (GstElement * element, GstEvent * event)
- return gst_oss_src_src_event (osssrc->srcpad, event);
+ return delay / oss->bytes_per_sample;
-static const GstQueryType *
-gst_oss_src_get_query_types (GstPad * pad)
+gst_oss_src_reset (GstAudioSrc * asrc)
- static const GstQueryType query_types[] = {
- GST_QUERY_POSITION,
- 0,
- return query_types;
-gst_oss_src_src_query (GstPad * pad, GstQueryType type, GstFormat * format,
- gint64 * value)
- gboolean res = FALSE;
+ //gint ret;
- switch (type) {
- case GST_QUERY_POSITION:
- res = gst_osselement_convert (GST_OSSELEMENT (osssrc),
- GST_FORMAT_DEFAULT, osssrc->curoffset, format, value);
- return res;
+ /* deadlocks on my machine... */
+ //ret = ioctl (oss->fd, SNDCTL_DSP_RESET, 0);
Index: gstosssrc.h
RCS file: /cvs/gstreamer/gst-plugins/sys/oss/gstosssrc.h,v
retrieving revision 1.14
retrieving revision 1.15
diff -u -d -r1.14 -r1.15
--- gstosssrc.h 10 Jul 2005 12:52:20 -0000 1.14
+++ gstosssrc.h 23 Aug 2005 09:46:29 -0000 1.15
@@ -21,58 +21,44 @@
*/
-#ifndef __GST_OSSSRC_H__
-#define __GST_OSSSRC_H__
+#ifndef __GST_OSS_SRC_H__
+#define __GST_OSS_SRC_H__
#include <gst/gst.h>
+#include <gst/audio/gstaudiosrc.h>
#include "gstosselement.h"
G_BEGIN_DECLS
-#define GST_TYPE_OSSSRC \
- (gst_oss_src_get_type())
-#define GST_OSSSRC(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSSSRC,GstOssSrc))
-#define GST_OSSSRC_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSSSRC,GstOssSrcClass))
-#define GST_IS_OSSSRC(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSSSRC))
-#define GST_IS_OSSSRC_CLASS(obj) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSSSRC))
-typedef enum {
- GST_OSSSRC_OPEN = GST_ELEMENT_FLAG_LAST,
- GST_OSSSRC_FLAG_LAST = GST_ELEMENT_FLAG_LAST+2,
-} GstOssSrcFlags;
+#define GST_TYPE_OSS_SRC (gst_oss_src_get_type())
+#define GST_OSS_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_OSS_SRC,GstOssSrc))
+#define GST_OSS_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_OSS_SRC,GstOssSrcClass))
+#define GST_IS_OSS_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_OSS_SRC))
+#define GST_IS_OSS_SRC_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_OSS_SRC))
typedef struct _GstOssSrc GstOssSrc;
typedef struct _GstOssSrcClass GstOssSrcClass;
struct _GstOssSrc {
- GstOssElement element;
+ GstAudioSrc src;
- /* pads */
- GstPad *srcpad;
- gboolean need_eos; /* Do we need to emit an EOS? */
-
- /* blocking.
- * curoffset is in *samples*. */
- gulong curoffset;
- gulong buffersize;
+ gint fd;
+ gint bytes_per_sample;
- /* clocks */
- GstClock *provided_clock, *clock;
+ gchar *device;
+ gchar *device_name;
struct _GstOssSrcClass {
- GstOssElementClass parent_class;
+ GstAudioSrcClass parent_class;
GType gst_oss_src_get_type(void);
G_END_DECLS
-#endif /* __GST_OSSSRC_H__ */
+#endif /* __GST_OSS_SRC_H__ */
|
|
From: <ow3n@fr...> - 2005-08-23 07:54:46
|
CVS Root: /cvs/gstreamer
Module: gst-plugins
Changes by: ow3n
Date: Tue Aug 23 2005 00:54:43 PDT
Log message:
Fixed mishandling events and incorrect audio skipping after seek.
Modified files:
. : ChangeLog
gst/realmedia : rmdemux.c
Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/ChangeLog.diff?r1=1.1800&r2=1.1801
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins/gst/realmedia/rmdemux.c.diff?r1=1.37&r2=1.38
====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins/ChangeLog,v
retrieving revision 1.1800
retrieving revision 1.1801
diff -u -d -r1.1800 -r1.1801
--- ChangeLog 22 Aug 2005 19:58:57 -0000 1.1800
+++ ChangeLog 23 Aug 2005 07:54:31 -0000 1.1801
@@ -1,3 +1,9 @@
+2005-08-23 Owen Fraser-Green <owen@...>
+
+ * gst/realmedia/rmdemux.c (gst_rmdemux_src_event): Fixed bug
+ causing events to be passed wrong way.
+ (gst_rmdemux_parse_packet): Avoid accidentally skipping audio.
2005-08-22 Jan Schmidt <thaytan@...>
* ext/mad/gstid3tag.c: (gst_id3_tag_init),
Index: rmdemux.c
RCS file: /cvs/gstreamer/gst-plugins/gst/realmedia/rmdemux.c,v
retrieving revision 1.37
retrieving revision 1.38
diff -u -d -r1.37 -r1.38
--- rmdemux.c 22 Aug 2005 07:33:00 -0000 1.37
+++ rmdemux.c 23 Aug 2005 07:54:31 -0000 1.38
@@ -264,7 +264,7 @@
default:
GST_LOG_OBJECT (rmdemux, "Event on sink: type=%d",
GST_EVENT_TYPE (event));
- ret = gst_pad_event_default (rmdemux->sinkpad, event);
+ ret = gst_pad_event_default (pad, event);
break;
}
@@ -331,7 +331,7 @@
}
GST_LOG_OBJECT (rmdemux, "Event on src: type=%d", GST_EVENT_TYPE (event));
@@ -1460,7 +1460,7 @@
stream = gst_rmdemux_get_stream_by_id (rmdemux, id);
- if (rmdemux->offset >= stream->seek_offset) {
+ if ((rmdemux->offset + packet_size) > stream->seek_offset) {
if (gst_pad_alloc_buffer (stream->pad, GST_BUFFER_OFFSET_NONE,
packet_size, stream->caps, &buffer) != GST_FLOW_OK) {
GST_WARNING_OBJECT (rmdemux, "failed to alloc src buffer for stream %d",
@@ -1478,7 +1478,7 @@
} else {
GST_DEBUG_OBJECT (rmdemux,
- "Stream %d is skipping: seek_offset=%d, offset=%d", stream->id,
- stream->seek_offset, rmdemux->offset);
+ "Stream %d is skipping: seek_offset=%d, offset=%d, packet_size",
+ stream->id, stream->seek_offset, rmdemux->offset, packet_size);
}
|