mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-02-20 04:56:24 +00:00
Code Issues 1 Projects Releases Wiki Activity

Added audiosource base classes.

Browse source 
Original commit message from CVS:
Added audiosource base classes.
Ported alsasrc, still very basic.
This commit is contained in:
Wim Taymans 2005-07-06 15:27:17 +00:00
parent a46a991d26
commit ceb88a7777
14 changed files with 2041 additions and 742 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

51
ChangeLog
View file

@ -1,3 +1,54 @@
2005-07-06 Wim Taymans <wim@fluendo.com>
* ext/alsa/Makefile.am:
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_type),
(gst_alsasrc_dispose), (gst_alsasrc_base_init),
(gst_alsasrc_class_init), (gst_alsasrc_init),
(gst_alsasrc_getcaps), (set_hwparams), (set_swparams),
(alsasrc_parse_spec), (gst_alsasrc_open), (gst_alsasrc_close),
(xrun_recovery), (gst_alsasrc_read), (gst_alsasrc_delay),
(gst_alsasrc_reset):
* ext/alsa/gstalsasrc.h:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(gst_audioringbuffer_start):
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_get_type),
(gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
(gst_audioringbuffer_init), (gst_audioringbuffer_dispose),
(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release), (gst_audioringbuffer_start),
(gst_audioringbuffer_stop), (gst_audioringbuffer_delay),
(gst_audiosrc_base_init), (gst_audiosrc_class_init),
(gst_audiosrc_init), (gst_audiosrc_create_ringbuffer):
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
(gst_baseaudiosink_get_time), (gst_baseaudiosink_setcaps),
(gst_baseaudiosink_preroll), (gst_baseaudiosink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_baseaudiosrc_base_init), (gst_baseaudiosrc_class_init),
(gst_baseaudiosrc_init), (gst_baseaudiosrc_get_clock),
(gst_baseaudiosrc_get_time), (gst_baseaudiosrc_set_property),
(gst_baseaudiosrc_get_property), (gst_baseaudiosrc_fixate),
(gst_baseaudiosrc_setcaps), (gst_baseaudiosrc_get_times),
(gst_baseaudiosrc_event), (gst_baseaudiosrc_create),
(gst_baseaudiosrc_create_ringbuffer), (gst_baseaudiosrc_callback),
(gst_baseaudiosrc_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ringbuffer_debug_spec_caps), (gst_ringbuffer_debug_spec_buff),
(gst_ringbuffer_parse_caps), (gst_ringbuffer_start),
(gst_ringbuffer_pause), (gst_ringbuffer_stop),
(gst_ringbuffer_samples_done), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit), (gst_ringbuffer_read),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance):
* gst-libs/gst/audio/gstringbuffer.h:
Added audiosource base classes.
Ported alsasrc, still very basic.
2005-07-06 Wim Taymans <wim@fluendo.com> 2005-07-06 Wim Taymans <wim@fluendo.com>
* ext/theora/theoradec.c: (theora_dec_src_getcaps), * ext/theora/theoradec.c: (theora_dec_src_getcaps),

6
ext/alsa/Makefile.am
View file

@ -3,7 +3,8 @@ plugin_LTLIBRARIES = libgstalsa.la
libgstalsa_la_SOURCES = \ libgstalsa_la_SOURCES = \
gstalsaplugin.c \ gstalsaplugin.c \
gstalsasink.c gstalsasink.c \
gstalsasrc.c
# port alsa stuff then add the _SOURCES above # port alsa stuff then add the _SOURCES above
EXTRA_DIST = \ EXTRA_DIST = \
@ -11,8 +12,7 @@ EXTRA_DIST = \
gstalsamixertrack.c \ gstalsamixertrack.c \
gstalsamixeroptions.c \ gstalsamixeroptions.c \
gstalsa.c \ gstalsa.c \
gstalsaclock.c \ gstalsaclock.c
gstalsasrc.c
libgstalsa_la_CFLAGS = $(GST_CFLAGS) $(ALSA_CFLAGS) libgstalsa_la_CFLAGS = $(GST_CFLAGS) $(ALSA_CFLAGS)
libgstalsa_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) \ libgstalsa_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) \

2
ext/alsa/gstalsaplugin.c
View file

@ -60,10 +60,10 @@ plugin_init (GstPlugin * plugin)
if (!gst_element_register (plugin, "alsamixer", GST_RANK_NONE, if (!gst_element_register (plugin, "alsamixer", GST_RANK_NONE,
GST_TYPE_ALSA_MIXER)) GST_TYPE_ALSA_MIXER))
return FALSE; return FALSE;
*/
if (!gst_element_register (plugin, "alsasrc", GST_RANK_NONE, if (!gst_element_register (plugin, "alsasrc", GST_RANK_NONE,
GST_TYPE_ALSA_SRC)) GST_TYPE_ALSA_SRC))
return FALSE; return FALSE;
*/
if (!gst_element_register (plugin, "alsasink", GST_RANK_NONE, if (!gst_element_register (plugin, "alsasink", GST_RANK_NONE,
GST_TYPE_ALSA_SINK)) GST_TYPE_ALSA_SINK))
return FALSE; return FALSE;

979
ext/alsa/gstalsasrc.c
View file

File diff suppressed because it is too large Load diff

64
ext/alsa/gstalsasrc.h
View file

@ -1,8 +1,7 @@
/* /* GStreamer
* Copyright (C) 2001 CodeFactory AB * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2001 Thomas Nyberg <thomas@codefactory.se> *
* Copyright (C) 2001-2002 Andy Wingo <apwingo@eos.ncsu.edu> * gstalsasrc.h:
* Copyright (C) 2003 Benjamin Otte <in7y118@public.uni-hamburg.de>
* *
* This library is free software; you can redistribute it and/or * This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public * modify it under the terms of the GNU Library General Public
@ -15,41 +14,58 @@
* Library General Public License for more details. * Library General Public License for more details.
* *
* You should have received a copy of the GNU Library General Public * You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the Free * License along with this library; if not, write to the
* Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/ */
#ifndef __GST_ALSA_SRC_H__
#define __GST_ALSA_SRC_H__
#include "gstalsamixer.h" #ifndef __GST_ALSASRC_H__
#define __GST_ALSASRC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiosrc.h>
#include <alsa/asoundlib.h>
G_BEGIN_DECLS G_BEGIN_DECLS
#define GST_ALSA_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_ALSA_SRC, GstAlsaSrc)) #define GST_TYPE_ALSA_SRC (gst_alsasrc_get_type())
#define GST_ALSA_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_ALSA_SRC, GstAlsaSrcClass)) #define GST_ALSA_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_ALSA_SRC,GstAlsaSrc))
#define GST_IS_ALSA_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_ALSA_SRC)) #define GST_ALSA_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_ALSA_SRC,GstAlsaSrcClass))
#define GST_IS_ALSA_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_ALSA_SRC)) #define GST_IS_ALSA_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_ALSA_SRC))
#define GST_TYPE_ALSA_SRC (gst_alsa_src_get_type()) #define GST_IS_ALSA_SRC_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_ALSA_SRC))
typedef struct _GstAlsaSrc GstAlsaSrc; typedef struct _GstAlsaSrc GstAlsaSrc;
typedef struct _GstAlsaSrcClass GstAlsaSrcClass; typedef struct _GstAlsaSrcClass GstAlsaSrcClass;
struct _GstAlsaSrc { struct _GstAlsaSrc {
GstAlsaMixer parent; GstAudioSrc src;
GstBuffer *buf[GST_ALSA_MAX_TRACKS];
snd_pcm_status_t *status; gchar *device;
GstClockTime base_time; /* FIXME: move this up ? already present in element ? */
snd_pcm_t *handle;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_access_t access;
snd_pcm_format_t format;
guint rate;
guint channels;
gint bytes_per_sample;
guint buffer_time;
guint period_time;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
}; };
struct _GstAlsaSrcClass { struct _GstAlsaSrcClass {
GstAlsaMixerClass parent_class; GstAudioSrcClass parent_class;
}; };
GType gst_alsa_src_get_type (void); GType gst_alsasrc_get_type(void);
gboolean gst_alsa_src_factory_init (GstPlugin *plugin);
G_END_DECLS G_END_DECLS
#endif /* __GST_ALSA_SRC_H__ */ #endif /* __GST_ALSASRC_H__ */

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

@ -17,7 +17,9 @@ CLEANFILES = gstaudiofilterexample.c \
libgstaudio_@GST_MAJORMINOR@_la_SOURCES = audio.c gstaudioclock.c \ libgstaudio_@GST_MAJORMINOR@_la_SOURCES = audio.c gstaudioclock.c \
multichannel.c \ multichannel.c \
gstaudiosink.c \ gstaudiosink.c \
gstaudiosrc.c \
gstbaseaudiosink.c \ gstbaseaudiosink.c \
gstbaseaudiosrc.c \
gstringbuffer.c gstringbuffer.c
nodist_libgstaudio_@GST_MAJORMINOR@_la_SOURCES = $(built_sources) $(built_headers) nodist_libgstaudio_@GST_MAJORMINOR@_la_SOURCES = $(built_sources) $(built_headers)
@ -27,7 +29,9 @@ libgstaudio_@GST_MAJORMINOR@include_HEADERS = \
gstaudioclock.h \ gstaudioclock.h \
gstaudiofilter.h \ gstaudiofilter.h \
gstaudiosink.h \ gstaudiosink.h \
gstaudiosrc.h \
gstbaseaudiosink.h \ gstbaseaudiosink.h \
gstbaseaudiosrc.h \
gstringbuffer.h \ gstringbuffer.h \
multichannel.h multichannel.h

16
gst-libs/gst/audio/gstaudiosink.c
View file

@ -73,12 +73,12 @@ static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf, static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec); GstRingBufferSpec * spec);
static gboolean gst_audioringbuffer_release (GstRingBuffer * buf); static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_play (GstRingBuffer * buf); static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf); static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
static guint gst_audioringbuffer_delay (GstRingBuffer * buf); static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
/* ringbuffer abstract base class */ /* ringbuffer abstract base class */
GType static GType
gst_audioringbuffer_get_type (void) gst_audioringbuffer_get_type (void)
{ {
static GType ringbuffer_type = 0; static GType ringbuffer_type = 0;
@ -98,7 +98,7 @@ gst_audioringbuffer_get_type (void)
}; };
ringbuffer_type = ringbuffer_type =
g_type_register_static (GST_TYPE_RINGBUFFER, "GstAudioRingBuffer", g_type_register_static (GST_TYPE_RINGBUFFER, "GstAudioSinkRingBuffer",
&ringbuffer_info, 0); &ringbuffer_info, 0);
} }
return ringbuffer_type; return ringbuffer_type;
@ -124,8 +124,8 @@ gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire); GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire);
gstringbuffer_class->release = gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_release); GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
gstringbuffer_class->play = GST_DEBUG_FUNCPTR (gst_audioringbuffer_play); gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_play); gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop); gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay); gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay);
@ -136,7 +136,7 @@ typedef guint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
/* this internal thread does nothing else but write samples to the audio device. /* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play * It will write each segment in the ringbuffer and will update the play
* pointer. * pointer.
* The play/stop methods control the thread. * The start/stop methods control the thread.
*/ */
static void static void
audioringbuffer_thread_func (GstRingBuffer * buf) audioringbuffer_thread_func (GstRingBuffer * buf)
@ -307,13 +307,13 @@ gst_audioringbuffer_release (GstRingBuffer * buf)
} }
static gboolean static gboolean
gst_audioringbuffer_play (GstRingBuffer * buf) gst_audioringbuffer_start (GstRingBuffer * buf)
{ {
GstAudioSink *sink; GstAudioSink *sink;
sink = GST_AUDIOSINK (GST_OBJECT_PARENT (buf)); sink = GST_AUDIOSINK (GST_OBJECT_PARENT (buf));
GST_DEBUG ("play, sending signal"); GST_DEBUG ("start, sending signal");
GST_AUDIORINGBUFFER_SIGNAL (buf); GST_AUDIORINGBUFFER_SIGNAL (buf);
return TRUE; return TRUE;

418
gst-libs/gst/audio/gstaudiosrc.c Normal file
View file

@ -0,0 +1,418 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudiosrc.c: simple audio src base class
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include "gstaudiosrc.h"
GST_DEBUG_CATEGORY_STATIC (gst_audiosrc_debug);
#define GST_CAT_DEFAULT gst_audiosrc_debug
#define GST_TYPE_AUDIORINGBUFFER \
(gst_audioringbuffer_get_type())
#define GST_AUDIORINGBUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORINGBUFFER,GstAudioRingBuffer))
#define GST_AUDIORINGBUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORINGBUFFER,GstAudioRingBufferClass))
#define GST_AUDIORINGBUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORINGBUFFER, GstAudioRingBufferClass))
#define GST_IS_AUDIORINGBUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORINGBUFFER))
#define GST_IS_AUDIORINGBUFFER_CLASS(obj)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORINGBUFFER))
typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
#define GST_AUDIORINGBUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
#define GST_AUDIORINGBUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORINGBUFFER_GET_COND (buf), GST_GET_LOCK (buf)))
#define GST_AUDIORINGBUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORINGBUFFER_GET_COND (buf)))
#define GST_AUDIORINGBUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORINGBUFFER_GET_COND (buf)))
struct _GstAudioRingBuffer
{
GstRingBuffer object;
gboolean running;
gint queuedseg;
GCond *cond;
};
struct _GstAudioRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer);
static void gst_audioringbuffer_dispose (GObject * object);
static void gst_audioringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
/* ringbuffer abstract base class */
static GType
gst_audioringbuffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstAudioRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_audioringbuffer_class_init,
NULL,
NULL,
sizeof (GstAudioRingBuffer),
0,
(GInstanceInitFunc) gst_audioringbuffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RINGBUFFER, "GstAudioSrcRingBuffer",
&ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_ref (GST_TYPE_RINGBUFFER);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioringbuffer_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioringbuffer_finalize);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay);
}
typedef guint (*ReadFunc) (GstAudioSrc * src, gpointer data, guint length);
/* this internal thread does nothing else but read samples from the audio device.
* It will read each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
GstAudioRingBuffer *abuf = GST_AUDIORINGBUFFER (buf);
ReadFunc readfunc;
src = GST_AUDIOSRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIOSRC_GET_CLASS (src);
GST_DEBUG ("enter thread");
readfunc = csrc->read;
if (readfunc == NULL)
goto no_function;
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
if (gst_ringbuffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint read = 0;
left = len;
do {
GST_DEBUG ("transfer %d bytes to segment %d", left, readseg);
read = readfunc (src, readptr + read, left);
GST_DEBUG ("transfered %d bytes", read);
if (read < 0 || read > left) {
GST_WARNING ("error reading data (reason: %s), skipping segment\n",
strerror (errno));
break;
}
left -= read;
} while (left > 0);
/* we read one segment */
gst_ringbuffer_advance (buf, 1);
} else {
GST_LOCK (abuf);
if (!abuf->running)
goto stop_running;
GST_DEBUG ("signal wait");
GST_AUDIORINGBUFFER_SIGNAL (buf);
GST_DEBUG ("wait for action");
GST_AUDIORINGBUFFER_WAIT (buf);
GST_DEBUG ("got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG ("continue running");
GST_UNLOCK (abuf);
}
}
GST_DEBUG ("exit thread");
return;
/* ERROR */
no_function:
{
GST_DEBUG ("no write function, exit thread");
return;
}
stop_running:
{
GST_UNLOCK (abuf);
GST_DEBUG ("stop running, exit thread");
return;
}
}
static void
gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer)
{
ringbuffer->running = FALSE;
ringbuffer->queuedseg = 0;
ringbuffer->cond = g_cond_new ();
}
static void
gst_audioringbuffer_dispose (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static void
gst_audioringbuffer_finalize (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
static gboolean
gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
src = GST_AUDIOSRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIOSRC_GET_CLASS (src);
if (csrc->open)
result = csrc->open (src, spec);
if (!result)
goto could_not_open;
/* allocate one more segment as we need some headroom */
spec->segtotal++;
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
abuf = GST_AUDIORINGBUFFER (buf);
abuf->running = TRUE;
src->thread =
g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
NULL);
GST_AUDIORINGBUFFER_WAIT (buf);
return result;
could_not_open:
{
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_audioringbuffer_release (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
src = GST_AUDIOSRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIOSRC_GET_CLASS (src);
abuf = GST_AUDIORINGBUFFER (buf);
abuf->running = FALSE;
GST_AUDIORINGBUFFER_SIGNAL (buf);
GST_UNLOCK (buf);
/* join the thread */
g_thread_join (src->thread);
GST_LOCK (buf);
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
if (csrc->close)
result = csrc->close (src);
return result;
}
static gboolean
gst_audioringbuffer_start (GstRingBuffer * buf)
{
GstAudioSrc *src;
src = GST_AUDIOSRC (GST_OBJECT_PARENT (buf));
GST_DEBUG ("start, sending signal");
GST_AUDIORINGBUFFER_SIGNAL (buf);
return TRUE;
}
static gboolean
gst_audioringbuffer_stop (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
src = GST_AUDIOSRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIOSRC_GET_CLASS (src);
/* unblock any pending writes to the audio device */
if (csrc->reset) {
GST_DEBUG ("reset...");
csrc->reset (src);
GST_DEBUG ("reset done");
}
GST_DEBUG ("stop, waiting...");
GST_AUDIORINGBUFFER_WAIT (buf);
GST_DEBUG ("stoped");
return TRUE;
}
static guint
gst_audioringbuffer_delay (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
guint res = 0;
src = GST_AUDIOSRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIOSRC_GET_CLASS (src);
if (csrc->delay)
res = csrc->delay (src);
return res;
}
/* AudioSrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
};
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audiosrc_debug, "audiosrc", 0, "audiosrc element");
GST_BOILERPLATE_FULL (GstAudioSrc, gst_audiosrc, GstBaseAudioSrc,
GST_TYPE_BASEAUDIOSRC, _do_init);
static GstRingBuffer *gst_audiosrc_create_ringbuffer (GstBaseAudioSrc * src);
static void
gst_audiosrc_base_init (gpointer g_class)
{
}
static void
gst_audiosrc_class_init (GstAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstPushSrcClass *gstpushsrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstpushsrc_class = (GstPushSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audiosrc_create_ringbuffer);
}
static void
gst_audiosrc_init (GstAudioSrc * audiosrc)
{
gst_base_src_set_live (GST_BASE_SRC (audiosrc), TRUE);
}
static GstRingBuffer *
gst_audiosrc_create_ringbuffer (GstBaseAudioSrc * src)
{
GstRingBuffer *buffer;
GST_DEBUG ("creating ringbuffer");
buffer = g_object_new (GST_TYPE_AUDIORINGBUFFER, NULL);
GST_DEBUG ("created ringbuffer @%p", buffer);
return buffer;
}

84
gst-libs/gst/audio/gstaudiosrc.h Normal file
View file

@ -0,0 +1,84 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudiosrc.h:
*
* 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.
*/
/* a base class for simple audio srcs.
*
* This base class only requires subclasses to implement a set
* of simple functions.
*
* - open: open the device with the specified caps
* - read: read samples to the audio device
* - close: close the device
* - delay: the number of samples queued in the device
* - reset: unblock a read to the device and reset.
*
* All scheduling of samples and timestamps is done in this
* base class.
*/
#ifndef __GST_AUDIOSRC_H__
#define __GST_AUDIOSRC_H__
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosrc.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIOSRC (gst_audiosrc_get_type())
#define GST_AUDIOSRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIOSRC,GstAudioSrc))
#define GST_AUDIOSRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIOSRC,GstAudioSrcClass))
#define GST_AUDIOSRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIOSRC,GstAudioSrcClass))
#define GST_IS_AUDIOSRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIOSRC))
#define GST_IS_AUDIOSRC_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOSRC))
typedef struct _GstAudioSrc GstAudioSrc;
typedef struct _GstAudioSrcClass GstAudioSrcClass;
struct _GstAudioSrc {
GstBaseAudioSrc element;
/*< private >*/ /* with LOCK */
GThread *thread;
};
struct _GstAudioSrcClass {
GstBaseAudioSrcClass parent_class;
/* vtable */
/* open the device with given specs */
gboolean (*open) (GstAudioSrc *src, GstRingBufferSpec *spec);
/* close the device */
gboolean (*close) (GstAudioSrc *src);
/* read samples from the device */
guint (*read) (GstAudioSrc *src, gpointer data, guint length);
/* get number of samples queued in the device */
guint (*delay) (GstAudioSrc *src);
/* reset the audio device, unblock from a write */
void (*reset) (GstAudioSrc *src);
};
GType gst_audiosrc_get_type(void);
G_END_DECLS
#endif /* __GST_AUDIOSRC_H__ */

210
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -160,7 +160,7 @@ gst_baseaudiosink_get_time (GstClock * clock, GstBaseAudioSink * sink)
if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0) if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
return 0; return 0;
samples = gst_ringbuffer_played_samples (sink->ringbuffer); samples = gst_ringbuffer_samples_done (sink->ringbuffer);
result = samples * GST_SECOND / sink->ringbuffer->spec.rate; result = samples * GST_SECOND / sink->ringbuffer->spec.rate;
result += GST_ELEMENT (sink)->base_time; result += GST_ELEMENT (sink)->base_time;
@ -210,204 +210,31 @@ gst_baseaudiosink_get_property (GObject * object, guint prop_id, GValue * value,
} }
} }
static int linear_formats[4 * 2 * 2] = {
GST_S8,
GST_S8,
GST_U8,
GST_U8,
GST_S16_LE,
GST_S16_BE,
GST_U16_LE,
GST_U16_BE,
GST_S24_LE,
GST_S24_BE,
GST_U24_LE,
GST_U24_BE,
GST_S32_LE,
GST_S32_BE,
GST_U32_LE,
GST_U32_BE
};
static int linear24_formats[3 * 2 * 2] = {
GST_S24_3LE,
GST_S24_3BE,
GST_U24_3LE,
GST_U24_3BE,
GST_S20_3LE,
GST_S20_3BE,
GST_U20_3LE,
GST_U20_3BE,
GST_S18_3LE,
GST_S18_3BE,
GST_U18_3LE,
GST_U18_3BE,
};
static GstBufferFormat
build_linear_format (int depth, int width, int unsignd, int big_endian)
{
if (width == 24) {
switch (depth) {
case 24:
depth = 0;
break;
case 20:
depth = 1;
break;
case 18:
depth = 2;
break;
default:
return GST_UNKNOWN;
}
return ((int (*)[2][2]) linear24_formats)[depth][!!unsignd][!!big_endian];
} else {
switch (depth) {
case 8:
depth = 0;
break;
case 16:
depth = 1;
break;
case 24:
depth = 2;
break;
case 32:
depth = 3;
break;
default:
return GST_UNKNOWN;
}
}
return ((int (*)[2][2]) linear_formats)[depth][!!unsignd][!!big_endian];
}
static void
debug_spec_caps (GstBaseAudioSink * sink, GstRingBufferSpec * spec)
{
GST_DEBUG ("spec caps: %p %" GST_PTR_FORMAT, spec->caps, spec->caps);
GST_DEBUG ("parsed caps: type: %d", spec->type);
GST_DEBUG ("parsed caps: format: %d", spec->format);
GST_DEBUG ("parsed caps: width: %d", spec->width);
GST_DEBUG ("parsed caps: depth: %d", spec->depth);
GST_DEBUG ("parsed caps: sign: %d", spec->sign);
GST_DEBUG ("parsed caps: bigend: %d", spec->bigend);
GST_DEBUG ("parsed caps: rate: %d", spec->rate);
GST_DEBUG ("parsed caps: channels: %d", spec->channels);
GST_DEBUG ("parsed caps: sample bytes: %d", spec->bytes_per_sample);
}
static void
debug_spec_buffer (GstBaseAudioSink * sink, GstRingBufferSpec * spec)
{
GST_DEBUG ("acquire ringbuffer: buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG ("acquire ringbuffer: latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG ("acquire ringbuffer: total segments: %d", spec->segtotal);
GST_DEBUG ("acquire ringbuffer: segment size: %d bytes = %d samples",
spec->segsize, spec->segsize / spec->bytes_per_sample);
GST_DEBUG ("acquire ringbuffer: buffer size: %d bytes = %d samples",
spec->segsize * spec->segtotal,
spec->segsize * spec->segtotal / spec->bytes_per_sample);
}
static gboolean static gboolean
gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps) gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps)
{ {
GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink); GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
GstRingBufferSpec *spec; GstRingBufferSpec *spec;
const gchar *mimetype;
GstStructure *structure;
spec = &sink->ringbuffer->spec; spec = &sink->ringbuffer->spec;
structure = gst_caps_get_structure (caps, 0); GST_DEBUG ("release old ringbuffer");
/* we have to differentiate between int and float formats */ /* release old ringbuffer */
mimetype = gst_structure_get_name (structure); gst_ringbuffer_release (sink->ringbuffer);
if (!strncmp (mimetype, "audio/x-raw-int", 15)) { GST_DEBUG ("parse caps");
gint endianness;
spec->type = GST_BUFTYPE_LINEAR;
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "width", &spec->width) &&
gst_structure_get_int (structure, "depth", &spec->depth) &&
gst_structure_get_boolean (structure, "signed", &spec->sign)))
goto parse_error;
/* extract endianness if needed */
if (spec->width > 8) {
if (!gst_structure_get_int (structure, "endianness", &endianness))
goto parse_error;
} else {
endianness = G_BYTE_ORDER;
}
spec->bigend = endianness == G_LITTLE_ENDIAN ? FALSE : TRUE;
spec->format =
build_linear_format (spec->depth, spec->width, spec->sign ? 0 : 1,
spec->bigend ? 1 : 0);
} else if (!strncmp (mimetype, "audio/x-raw-float", 17)) {
spec->type = GST_BUFTYPE_FLOAT;
/* get layout */
if (!gst_structure_get_int (structure, "width", &spec->width))
goto parse_error;
/* match layout to format wrt to endianness */
switch (spec->width) {
case 32:
spec->format =
G_BYTE_ORDER == G_LITTLE_ENDIAN ? GST_FLOAT32_LE : GST_FLOAT32_BE;
break;
case 64:
spec->format =
G_BYTE_ORDER == G_LITTLE_ENDIAN ? GST_FLOAT64_LE : GST_FLOAT64_BE;
break;
default:
goto parse_error;
}
} else if (!strncmp (mimetype, "audio/x-alaw", 12)) {
spec->type = GST_BUFTYPE_A_LAW;
spec->format = GST_A_LAW;
} else if (!strncmp (mimetype, "audio/x-mulaw", 13)) {
spec->type = GST_BUFTYPE_MU_LAW;
spec->format = GST_MU_LAW;
} else {
goto parse_error;
}
/* get rate and channels */
if (!(gst_structure_get_int (structure, "rate", &spec->rate) &&
gst_structure_get_int (structure, "channels", &spec->channels)))
goto parse_error;
spec->bytes_per_sample = (spec->width >> 3) * spec->channels;
gst_caps_replace (&spec->caps, caps);
debug_spec_caps (sink, spec);
spec->buffer_time = sink->buffer_time; spec->buffer_time = sink->buffer_time;
spec->latency_time = sink->latency_time; spec->latency_time = sink->latency_time;
/* calculate suggested segsize and segtotal */ /* parse new caps */
spec->segsize = if (!gst_ringbuffer_parse_caps (spec, caps))
spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND; goto parse_error;
spec->segtotal = spec->buffer_time / spec->latency_time;
GST_DEBUG ("release old ringbuffer"); gst_ringbuffer_debug_spec_buff (spec);
gst_ringbuffer_release (sink->ringbuffer); GST_DEBUG ("acquire new ringbuffer");
debug_spec_buffer (sink, spec);
if (!gst_ringbuffer_acquire (sink->ringbuffer, spec)) if (!gst_ringbuffer_acquire (sink->ringbuffer, spec))
goto acquire_error; goto acquire_error;
@ -419,7 +246,7 @@ gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps)
spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate * spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate *
spec->bytes_per_sample); spec->bytes_per_sample);
debug_spec_buffer (sink, spec); gst_ringbuffer_debug_spec_buff (spec);
return TRUE; return TRUE;
@ -487,10 +314,23 @@ gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
static GstFlowReturn static GstFlowReturn
gst_baseaudiosink_preroll (GstBaseSink * bsink, GstBuffer * buffer) gst_baseaudiosink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
{ {
GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
if (!gst_ringbuffer_is_acquired (sink->ringbuffer))
goto wrong_state;
/* we don't really do anything when prerolling. We could make a /* we don't really do anything when prerolling. We could make a
* property to play this buffer to have some sort of scrubbing * property to play this buffer to have some sort of scrubbing
* support. */ * support. */
return GST_FLOW_OK; return GST_FLOW_OK;
wrong_state:
{
GST_DEBUG ("ringbuffer in wrong state");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink not negotiated."), (NULL));
return GST_FLOW_ERROR;
}
} }
static GstFlowReturn static GstFlowReturn
@ -514,6 +354,8 @@ gst_baseaudiosink_render (GstBaseSink * bsink, GstBuffer * buf)
wrong_state: wrong_state:
{ {
GST_DEBUG ("ringbuffer in wrong state"); GST_DEBUG ("ringbuffer in wrong state");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
("sink not negotiated."), (NULL));
return GST_FLOW_ERROR; return GST_FLOW_ERROR;
} }
} }

403
gst-libs/gst/audio/gstbaseaudiosrc.c Normal file
View file

@ -0,0 +1,403 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosrc.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include "gstbaseaudiosrc.h"
GST_DEBUG_CATEGORY_STATIC (gst_baseaudiosrc_debug);
#define GST_CAT_DEFAULT gst_baseaudiosrc_debug
/* BaseAudioSrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_BUFFER_TIME 500 * GST_USECOND
#define DEFAULT_LATENCY_TIME 10 * GST_USECOND
enum
{
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
};
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_baseaudiosrc_debug, "baseaudiosrc", 0, "baseaudiosrc element");
GST_BOILERPLATE_FULL (GstBaseAudioSrc, gst_baseaudiosrc, GstPushSrc,
GST_TYPE_PUSHSRC, _do_init);
static void gst_baseaudiosrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_baseaudiosrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_baseaudiosrc_fixate (GstPad * pad, GstCaps * caps);
static GstElementStateReturn gst_baseaudiosrc_change_state (GstElement *
element);
static GstClock *gst_baseaudiosrc_get_clock (GstElement * elem);
static GstClockTime gst_baseaudiosrc_get_time (GstClock * clock,
GstBaseAudioSrc * src);
static GstFlowReturn gst_baseaudiosrc_create (GstPushSrc * psrc,
GstBuffer ** buf);
static gboolean gst_baseaudiosrc_event (GstBaseSrc * bsrc, GstEvent * event);
static void gst_baseaudiosrc_get_times (GstBaseSrc * bsrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_baseaudiosrc_setcaps (GstBaseSrc * bsrc, GstCaps * caps);
//static guint gst_baseaudiosrc_signals[LAST_SIGNAL] = { 0 };
static void
gst_baseaudiosrc_base_init (gpointer g_class)
{
}
static void
gst_baseaudiosrc_class_init (GstBaseAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstPushSrcClass *gstpushsrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstpushsrc_class = (GstPushSrcClass *) klass;
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_baseaudiosrc_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_baseaudiosrc_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_TIME,
g_param_spec_int64 ("buffer-time", "Buffer Time",
"Size of audio buffer in milliseconds (-1 = default)",
-1, G_MAXINT64, DEFAULT_BUFFER_TIME, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LATENCY_TIME,
g_param_spec_int64 ("latency-time", "Latency Time",
"Audio latency in milliseconds (-1 = default)",
-1, G_MAXINT64, DEFAULT_LATENCY_TIME, G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_baseaudiosrc_change_state);
gstelement_class->get_clock = GST_DEBUG_FUNCPTR (gst_baseaudiosrc_get_clock);
gstbasesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_baseaudiosrc_setcaps);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_baseaudiosrc_event);
gstbasesrc_class->get_times = GST_DEBUG_FUNCPTR (gst_baseaudiosrc_get_times);
gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_baseaudiosrc_create);
}
static void
gst_baseaudiosrc_init (GstBaseAudioSrc * baseaudiosrc)
{
baseaudiosrc->buffer_time = DEFAULT_BUFFER_TIME;
baseaudiosrc->latency_time = DEFAULT_LATENCY_TIME;
baseaudiosrc->clock = gst_audio_clock_new ("clock",
(GstAudioClockGetTimeFunc) gst_baseaudiosrc_get_time, baseaudiosrc);
gst_pad_set_fixatecaps_function (GST_BASE_SRC_PAD (baseaudiosrc),
gst_baseaudiosrc_fixate);
}
static GstClock *
gst_baseaudiosrc_get_clock (GstElement * elem)
{
GstBaseAudioSrc *src;
src = GST_BASEAUDIOSRC (elem);
return GST_CLOCK (gst_object_ref (GST_OBJECT (src->clock)));
}
static GstClockTime
gst_baseaudiosrc_get_time (GstClock * clock, GstBaseAudioSrc * src)
{
guint64 samples;
GstClockTime result;
if (src->ringbuffer == NULL || src->ringbuffer->spec.rate == 0)
return 0;
samples = gst_ringbuffer_samples_done (src->ringbuffer);
result = samples * GST_SECOND / src->ringbuffer->spec.rate;
result += GST_ELEMENT (src)->base_time;
return result;
}
static void
gst_baseaudiosrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASEAUDIOSRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
src->buffer_time = g_value_get_int64 (value);
break;
case PROP_LATENCY_TIME:
src->latency_time = g_value_get_int64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_baseaudiosrc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBaseAudioSrc *src;
src = GST_BASEAUDIOSRC (object);
switch (prop_id) {
case PROP_BUFFER_TIME:
g_value_set_int64 (value, src->buffer_time);
break;
case PROP_LATENCY_TIME:
g_value_set_int64 (value, src->latency_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_baseaudiosrc_fixate (GstPad * pad, GstCaps * caps)
{
GstStructure *s;
s = gst_caps_get_structure (caps, 0);
gst_caps_structure_fixate_field_nearest_int (s, "rate", 44100);
gst_caps_structure_fixate_field_nearest_int (s, "channels", 2);
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);
}
static gboolean
gst_baseaudiosrc_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstBaseAudioSrc *src = GST_BASEAUDIOSRC (bsrc);
GstRingBufferSpec *spec;
spec = &src->ringbuffer->spec;
spec->buffer_time = src->buffer_time;
spec->latency_time = src->latency_time;
if (!gst_ringbuffer_parse_caps (spec, caps))
goto parse_error;
/* calculate suggested segsize and segtotal */
spec->segsize =
spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND;
spec->segtotal = spec->buffer_time / spec->latency_time;
GST_DEBUG ("release old ringbuffer");
gst_ringbuffer_release (src->ringbuffer);
gst_ringbuffer_debug_spec_buff (spec);
GST_DEBUG ("acquire new ringbuffer");
if (!gst_ringbuffer_acquire (src->ringbuffer, spec))
goto acquire_error;
/* calculate actual latency and buffer times */
spec->latency_time =
spec->segsize * GST_MSECOND / (spec->rate * spec->bytes_per_sample);
spec->buffer_time =
spec->segtotal * spec->segsize * GST_MSECOND / (spec->rate *
spec->bytes_per_sample);
gst_ringbuffer_debug_spec_buff (spec);
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG ("could not parse caps");
return FALSE;
}
acquire_error:
{
GST_DEBUG ("could not acquire ringbuffer");
return FALSE;
}
}
static void
gst_baseaudiosrc_get_times (GstBaseSrc * bsrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* ne need to sync to a clock here, we schedule the samples based
* on our own clock for the moment. FIXME, implement this when
* we are not using our own clock */
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
static gboolean
gst_baseaudiosrc_event (GstBaseSrc * bsrc, GstEvent * event)
{
GstBaseAudioSrc *src = GST_BASEAUDIOSRC (bsrc);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH:
if (GST_EVENT_FLUSH_DONE (event)) {
} else {
gst_ringbuffer_pause (src->ringbuffer);
}
break;
default:
break;
}
return TRUE;
}
static GstFlowReturn
gst_baseaudiosrc_create (GstPushSrc * psrc, GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASEAUDIOSRC (psrc);
GstBuffer *buf;
guchar *data;
guint len;
guint res;
if (!gst_ringbuffer_is_acquired (src->ringbuffer))
goto wrong_state;
buf = gst_buffer_new_and_alloc (src->ringbuffer->spec.segsize);
data = GST_BUFFER_DATA (buf);
len = GST_BUFFER_SIZE (buf);
res = gst_ringbuffer_read (src->ringbuffer, -1, data, len);
if (res == -1)
goto stopped;
gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (psrc)));
*outbuf = buf;
return GST_FLOW_OK;
wrong_state:
{
GST_DEBUG ("ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG ("ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
GstRingBuffer *
gst_baseaudiosrc_create_ringbuffer (GstBaseAudioSrc * src)
{
GstBaseAudioSrcClass *bclass;
GstRingBuffer *buffer = NULL;
bclass = GST_BASEAUDIOSRC_GET_CLASS (src);
if (bclass->create_ringbuffer)
buffer = bclass->create_ringbuffer (src);
if (buffer) {
gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (src));
}
return buffer;
}
void
gst_baseaudiosrc_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
gpointer user_data)
{
//GstBaseAudioSrc *src = GST_BASEAUDIOSRC (data);
}
static GstElementStateReturn
gst_baseaudiosrc_change_state (GstElement * element)
{
GstElementStateReturn ret = GST_STATE_SUCCESS;
GstBaseAudioSrc *src = GST_BASEAUDIOSRC (element);
GstElementState transition = GST_STATE_TRANSITION (element);
switch (transition) {
case GST_STATE_NULL_TO_READY:
break;
case GST_STATE_READY_TO_PAUSED:
src->ringbuffer = gst_baseaudiosrc_create_ringbuffer (src);
gst_ringbuffer_set_callback (src->ringbuffer, gst_baseaudiosrc_callback,
src);
break;
case GST_STATE_PAUSED_TO_PLAYING:
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
switch (transition) {
case GST_STATE_PLAYING_TO_PAUSED:
gst_ringbuffer_pause (src->ringbuffer);
break;
case GST_STATE_PAUSED_TO_READY:
gst_ringbuffer_stop (src->ringbuffer);
gst_ringbuffer_release (src->ringbuffer);
gst_object_unref (GST_OBJECT (src->ringbuffer));
src->ringbuffer = NULL;
break;
case GST_STATE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}

77
gst-libs/gst/audio/gstbaseaudiosrc.h Normal file
View file

@ -0,0 +1,77 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosrc.h:
*
* 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.
*/
/* a base class for audio sources.
*/
#ifndef __GST_BASEAUDIOSRC_H__
#define __GST_BASEAUDIOSRC_H__
#include <gst/gst.h>
#include <gst/base/gstpushsrc.h>
#include "gstringbuffer.h"
#include "gstaudioclock.h"
G_BEGIN_DECLS
#define GST_TYPE_BASEAUDIOSRC (gst_baseaudiosrc_get_type())
#define GST_BASEAUDIOSRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASEAUDIOSRC,GstBaseAudioSrc))
#define GST_BASEAUDIOSRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASEAUDIOSRC,GstBaseAudioSrcClass))
#define GST_BASEAUDIOSRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASEAUDIOSRC, GstBaseAudioSrcClass))
#define GST_IS_BASEAUDIOSRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASEAUDIOSRC))
#define GST_IS_BASEAUDIOSRC_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASEAUDIOSRC))
#define GST_BASEAUDIOSRC_CLOCK(obj) (GST_BASEAUDIOSRC (obj)->clock)
#define GST_BASEAUDIOSRC_PAD(obj) (GST_BASESRC (obj)->srcpad)
typedef struct _GstBaseAudioSrc GstBaseAudioSrc;
typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass;
struct _GstBaseAudioSrc {
GstPushSrc element;
/*< protected >*/ /* with LOCK */
/* our ringbuffer */
GstRingBuffer *ringbuffer;
/* required buffer and latency */
GstClockTime buffer_time;
GstClockTime latency_time;
/* clock */
GstClock *clock;
};
struct _GstBaseAudioSrcClass {
GstPushSrcClass parent_class;
/* subclass ringbuffer allocation */
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src);
};
GType gst_baseaudiosrc_get_type(void);
GstRingBuffer *gst_baseaudiosrc_create_ringbuffer (GstBaseAudioSrc *src);
G_END_DECLS
#endif /* __GST_BASEAUDIOSRC_H__ */

443
gst-libs/gst/audio/gstringbuffer.c
View file

@ -106,6 +106,204 @@ gst_ringbuffer_finalize (GObject * object)
G_OBJECT_CLASS (parent_class)->finalize (G_OBJECT (ringbuffer)); G_OBJECT_CLASS (parent_class)->finalize (G_OBJECT (ringbuffer));
} }
static int linear_formats[4 * 2 * 2] = {
GST_S8,
GST_S8,
GST_U8,
GST_U8,
GST_S16_LE,
GST_S16_BE,
GST_U16_LE,
GST_U16_BE,
GST_S24_LE,
GST_S24_BE,
GST_U24_LE,
GST_U24_BE,
GST_S32_LE,
GST_S32_BE,
GST_U32_LE,
GST_U32_BE
};
static int linear24_formats[3 * 2 * 2] = {
GST_S24_3LE,
GST_S24_3BE,
GST_U24_3LE,
GST_U24_3BE,
GST_S20_3LE,
GST_S20_3BE,
GST_U20_3LE,
GST_U20_3BE,
GST_S18_3LE,
GST_S18_3BE,
GST_U18_3LE,
GST_U18_3BE,
};
static GstBufferFormat
build_linear_format (int depth, int width, int unsignd, int big_endian)
{
if (width == 24) {
switch (depth) {
case 24:
depth = 0;
break;
case 20:
depth = 1;
break;
case 18:
depth = 2;
break;
default:
return GST_UNKNOWN;
}
return ((int (*)[2][2]) linear24_formats)[depth][!!unsignd][!!big_endian];
} else {
switch (depth) {
case 8:
depth = 0;
break;
case 16:
depth = 1;
break;
case 24:
depth = 2;
break;
case 32:
depth = 3;
break;
default:
return GST_UNKNOWN;
}
}
return ((int (*)[2][2]) linear_formats)[depth][!!unsignd][!!big_endian];
}
void
gst_ringbuffer_debug_spec_caps (GstRingBufferSpec * spec)
{
GST_DEBUG ("spec caps: %p %" GST_PTR_FORMAT, spec->caps, spec->caps);
GST_DEBUG ("parsed caps: type: %d", spec->type);
GST_DEBUG ("parsed caps: format: %d", spec->format);
GST_DEBUG ("parsed caps: width: %d", spec->width);
GST_DEBUG ("parsed caps: depth: %d", spec->depth);
GST_DEBUG ("parsed caps: sign: %d", spec->sign);
GST_DEBUG ("parsed caps: bigend: %d", spec->bigend);
GST_DEBUG ("parsed caps: rate: %d", spec->rate);
GST_DEBUG ("parsed caps: channels: %d", spec->channels);
GST_DEBUG ("parsed caps: sample bytes: %d", spec->bytes_per_sample);
}
void
gst_ringbuffer_debug_spec_buff (GstRingBufferSpec * spec)
{
GST_DEBUG ("acquire ringbuffer: buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG ("acquire ringbuffer: latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG ("acquire ringbuffer: total segments: %d", spec->segtotal);
GST_DEBUG ("acquire ringbuffer: segment size: %d bytes = %d samples",
spec->segsize, spec->segsize / spec->bytes_per_sample);
GST_DEBUG ("acquire ringbuffer: buffer size: %d bytes = %d samples",
spec->segsize * spec->segtotal,
spec->segsize * spec->segtotal / spec->bytes_per_sample);
}
gboolean
gst_ringbuffer_parse_caps (GstRingBufferSpec * spec, GstCaps * caps)
{
const gchar *mimetype;
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
/* we have to differentiate between int and float formats */
mimetype = gst_structure_get_name (structure);
if (!strncmp (mimetype, "audio/x-raw-int", 15)) {
gint endianness;
spec->type = GST_BUFTYPE_LINEAR;
/* extract the needed information from the cap */
if (!(gst_structure_get_int (structure, "width", &spec->width) &&
gst_structure_get_int (structure, "depth", &spec->depth) &&
gst_structure_get_boolean (structure, "signed", &spec->sign)))
goto parse_error;
/* extract endianness if needed */
if (spec->width > 8) {
if (!gst_structure_get_int (structure, "endianness", &endianness))
goto parse_error;
} else {
endianness = G_BYTE_ORDER;
}
spec->bigend = endianness == G_LITTLE_ENDIAN ? FALSE : TRUE;
spec->format =
build_linear_format (spec->depth, spec->width, spec->sign ? 0 : 1,
spec->bigend ? 1 : 0);
} else if (!strncmp (mimetype, "audio/x-raw-float", 17)) {
spec->type = GST_BUFTYPE_FLOAT;
/* get layout */
if (!gst_structure_get_int (structure, "width", &spec->width))
goto parse_error;
/* match layout to format wrt to endianness */
switch (spec->width) {
case 32:
spec->format =
G_BYTE_ORDER == G_LITTLE_ENDIAN ? GST_FLOAT32_LE : GST_FLOAT32_BE;
break;
case 64:
spec->format =
G_BYTE_ORDER == G_LITTLE_ENDIAN ? GST_FLOAT64_LE : GST_FLOAT64_BE;
break;
default:
goto parse_error;
}
} else if (!strncmp (mimetype, "audio/x-alaw", 12)) {
spec->type = GST_BUFTYPE_A_LAW;
spec->format = GST_A_LAW;
} else if (!strncmp (mimetype, "audio/x-mulaw", 13)) {
spec->type = GST_BUFTYPE_MU_LAW;
spec->format = GST_MU_LAW;
} else {
goto parse_error;
}
/* get rate and channels */
if (!(gst_structure_get_int (structure, "rate", &spec->rate) &&
gst_structure_get_int (structure, "channels", &spec->channels)))
goto parse_error;
spec->bytes_per_sample = (spec->width >> 3) * spec->channels;
gst_caps_replace (&spec->caps, caps);
g_return_val_if_fail (spec->latency_time != 0, FALSE);
/* calculate suggested segsize and segtotal */
spec->segsize =
spec->rate * spec->bytes_per_sample * spec->latency_time / GST_MSECOND;
spec->segtotal = spec->buffer_time / spec->latency_time;
gst_ringbuffer_debug_spec_caps (spec);
gst_ringbuffer_debug_spec_buff (spec);
return TRUE;
/* ERRORS */
parse_error:
{
GST_DEBUG ("could not parse caps");
return FALSE;
}
}
/** /**
* gst_ringbuffer_set_callback: * gst_ringbuffer_set_callback:
* @buf: the #GstRingBuffer to set the callback on * @buf: the #GstRingBuffer to set the callback on
@ -264,17 +462,17 @@ gst_ringbuffer_is_acquired (GstRingBuffer * buf)
/** /**
* gst_ringbuffer_play: * gst_ringbuffer_start:
* @buf: the #GstRingBuffer to play * @buf: the #GstRingBuffer to start
* *
* Start playing samples from the ringbuffer. * Start processing samples from the ringbuffer.
* *
* Returns: TRUE if the device could be started, FALSE on error. * Returns: TRUE if the device could be started, FALSE on error.
* *
* MT safe. * MT safe.
*/ */
gboolean gboolean
gst_ringbuffer_play (GstRingBuffer * buf) gst_ringbuffer_start (GstRingBuffer * buf)
{ {
gboolean res = FALSE; gboolean res = FALSE;
GstRingBufferClass *rclass; GstRingBufferClass *rclass;
@ -283,16 +481,16 @@ gst_ringbuffer_play (GstRingBuffer * buf)
g_return_val_if_fail (buf != NULL, FALSE); g_return_val_if_fail (buf != NULL, FALSE);
GST_LOCK (buf); GST_LOCK (buf);
/* if paused, set to playing */ /* if stopped, set to started */
res = g_atomic_int_compare_and_exchange (&buf->state, res = g_atomic_int_compare_and_exchange (&buf->state,
GST_RINGBUFFER_STATE_STOPPED, GST_RINGBUFFER_STATE_PLAYING); GST_RINGBUFFER_STATE_STOPPED, GST_RINGBUFFER_STATE_STARTED);
if (!res) { if (!res) {
/* was not stopped, try from paused */ /* was not stopped, try from paused */
res = g_atomic_int_compare_and_exchange (&buf->state, res = g_atomic_int_compare_and_exchange (&buf->state,
GST_RINGBUFFER_STATE_PAUSED, GST_RINGBUFFER_STATE_PLAYING); GST_RINGBUFFER_STATE_PAUSED, GST_RINGBUFFER_STATE_STARTED);
if (!res) { if (!res) {
/* was not paused either, must be playing then */ /* was not paused either, must be started then */
res = TRUE; res = TRUE;
goto done; goto done;
} }
@ -304,8 +502,8 @@ gst_ringbuffer_play (GstRingBuffer * buf)
if (rclass->resume) if (rclass->resume)
res = rclass->resume (buf); res = rclass->resume (buf);
} else { } else {
if (rclass->play) if (rclass->start)
res = rclass->play (buf); res = rclass->start (buf);
} }
if (!res) { if (!res) {
@ -322,7 +520,7 @@ done:
* gst_ringbuffer_pause: * gst_ringbuffer_pause:
* @buf: the #GstRingBuffer to pause * @buf: the #GstRingBuffer to pause
* *
* Pause playing samples from the ringbuffer. * Pause processing samples from the ringbuffer.
* *
* Returns: TRUE if the device could be paused, FALSE on error. * Returns: TRUE if the device could be paused, FALSE on error.
* *
@ -337,12 +535,12 @@ gst_ringbuffer_pause (GstRingBuffer * buf)
g_return_val_if_fail (buf != NULL, FALSE); g_return_val_if_fail (buf != NULL, FALSE);
GST_LOCK (buf); GST_LOCK (buf);
/* if playing, set to paused */ /* if started, set to paused */
res = g_atomic_int_compare_and_exchange (&buf->state, res = g_atomic_int_compare_and_exchange (&buf->state,
GST_RINGBUFFER_STATE_PLAYING, GST_RINGBUFFER_STATE_PAUSED); GST_RINGBUFFER_STATE_STARTED, GST_RINGBUFFER_STATE_PAUSED);
if (!res) { if (!res) {
/* was not playing */ /* was not started */
res = TRUE; res = TRUE;
goto done; goto done;
} }
@ -355,7 +553,7 @@ gst_ringbuffer_pause (GstRingBuffer * buf)
res = rclass->pause (buf); res = rclass->pause (buf);
if (!res) { if (!res) {
buf->state = GST_RINGBUFFER_STATE_PLAYING; buf->state = GST_RINGBUFFER_STATE_STARTED;
} }
done: done:
GST_UNLOCK (buf); GST_UNLOCK (buf);
@ -367,7 +565,7 @@ done:
* gst_ringbuffer_stop: * gst_ringbuffer_stop:
* @buf: the #GstRingBuffer to stop * @buf: the #GstRingBuffer to stop
* *
* Stop playing samples from the ringbuffer. * Stop processing samples from the ringbuffer.
* *
* Returns: TRUE if the device could be stopped, FALSE on error. * Returns: TRUE if the device could be stopped, FALSE on error.
* *
@ -382,12 +580,12 @@ gst_ringbuffer_stop (GstRingBuffer * buf)
g_return_val_if_fail (buf != NULL, FALSE); g_return_val_if_fail (buf != NULL, FALSE);
GST_LOCK (buf); GST_LOCK (buf);
/* if playing, set to stopped */ /* if started, set to stopped */
res = g_atomic_int_compare_and_exchange (&buf->state, res = g_atomic_int_compare_and_exchange (&buf->state,
GST_RINGBUFFER_STATE_PLAYING, GST_RINGBUFFER_STATE_STOPPED); GST_RINGBUFFER_STATE_STARTED, GST_RINGBUFFER_STATE_STOPPED);
if (!res) { if (!res) {
/* was not playing, must be stopped then */ /* was not started, must be stopped then */
res = TRUE; res = TRUE;
goto done; goto done;
} }
@ -400,7 +598,7 @@ gst_ringbuffer_stop (GstRingBuffer * buf)
res = rclass->stop (buf); res = rclass->stop (buf);
if (!res) { if (!res) {
buf->state = GST_RINGBUFFER_STATE_PLAYING; buf->state = GST_RINGBUFFER_STATE_STARTED;
} else { } else {
gst_ringbuffer_set_sample (buf, 0); gst_ringbuffer_set_sample (buf, 0);
} }
@ -442,38 +640,38 @@ gst_ringbuffer_delay (GstRingBuffer * buf)
} }
/** /**
* gst_ringbuffer_played_samples: * gst_ringbuffer_samples_done:
* @buf: the #GstRingBuffer to query * @buf: the #GstRingBuffer to query
* *
* Get the number of samples that were played by the ringbuffer * Get the number of samples that were processed by the ringbuffer
* since it was last started. * since it was last started.
* *
* Returns: The number of samples played by the ringbuffer. * Returns: The number of samples processed by the ringbuffer.
* *
* MT safe. * MT safe.
*/ */
guint64 guint64
gst_ringbuffer_played_samples (GstRingBuffer * buf) gst_ringbuffer_samples_done (GstRingBuffer * buf)
{ {
gint segplayed; gint segdone;
guint64 raw, samples; guint64 raw, samples;
guint delay; guint delay;
g_return_val_if_fail (buf != NULL, 0); g_return_val_if_fail (buf != NULL, 0);
/* get the amount of segments we played */ /* get the amount of segments we processed */
segplayed = g_atomic_int_get (&buf->segplayed); segdone = g_atomic_int_get (&buf->segdone);
/* and the number of samples not yet played */ /* and the number of samples not yet processed */
delay = gst_ringbuffer_delay (buf); delay = gst_ringbuffer_delay (buf);
samples = (segplayed * buf->samples_per_seg); samples = (segdone * buf->samples_per_seg);
raw = samples; raw = samples;
if (samples >= delay) if (samples >= delay)
samples -= delay; samples -= delay;
GST_DEBUG ("played samples: raw %llu, delay %u, real %llu", raw, delay, GST_DEBUG ("processed samples: raw %llu, delay %u, real %llu", raw, delay,
samples); samples);
return samples; return samples;
@ -505,47 +703,47 @@ gst_ringbuffer_set_sample (GstRingBuffer * buf, guint64 sample)
/* FIXME, we assume the ringbuffer can restart at a random /* FIXME, we assume the ringbuffer can restart at a random
* position, round down to the beginning and keep track of * position, round down to the beginning and keep track of
* offset when calculating the played samples. */ * offset when calculating the processed samples. */
buf->segplayed = sample / buf->samples_per_seg; buf->segdone = sample / buf->samples_per_seg;
buf->next_sample = sample; buf->next_sample = sample;
for (i = 0; i < buf->spec.segtotal; i++) { for (i = 0; i < buf->spec.segtotal; i++) {
gst_ringbuffer_clear (buf, i); gst_ringbuffer_clear (buf, i);
} }
GST_DEBUG ("setting sample to %llu, segplayed %d", sample, buf->segplayed); GST_DEBUG ("setting sample to %llu, segdone %d", sample, buf->segdone);
} }
static gboolean static gboolean
wait_segment (GstRingBuffer * buf) wait_segment (GstRingBuffer * buf)
{ {
/* buffer must be playing now or we deadlock since nobody is reading */ /* buffer must be started now or we deadlock since nobody is reading */
if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_PLAYING) { if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_STARTED) {
GST_DEBUG ("play!"); GST_DEBUG ("start!");
gst_ringbuffer_play (buf); gst_ringbuffer_start (buf);
} }
/* take lock first, then update our waiting flag */ /* take lock first, then update our waiting flag */
GST_LOCK (buf); GST_LOCK (buf);
if (g_atomic_int_compare_and_exchange (&buf->waiting, 0, 1)) { if (g_atomic_int_compare_and_exchange (&buf->waiting, 0, 1)) {
GST_DEBUG ("waiting.."); GST_DEBUG ("waiting..");
if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_PLAYING) if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_STARTED)
goto not_playing; goto not_started;
GST_RINGBUFFER_WAIT (buf); GST_RINGBUFFER_WAIT (buf);
if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_PLAYING) if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_STARTED)
goto not_playing; goto not_started;
} }
GST_UNLOCK (buf); GST_UNLOCK (buf);
return TRUE; return TRUE;
/* ERROR */ /* ERROR */
not_playing: not_started:
{ {
GST_UNLOCK (buf); GST_UNLOCK (buf);
GST_DEBUG ("stopped playing"); GST_DEBUG ("stopped processing");
return FALSE; return FALSE;
} }
} }
@ -573,7 +771,7 @@ guint
gst_ringbuffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data, gst_ringbuffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
guint len) guint len)
{ {
gint segplayed; gint segdone;
gint segsize, segtotal, bps, sps; gint segsize, segtotal, bps, sps;
guint8 *dest; guint8 *dest;
@ -582,7 +780,7 @@ gst_ringbuffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
g_return_val_if_fail (data != NULL, -1); g_return_val_if_fail (data != NULL, -1);
if (sample == -1) { if (sample == -1) {
/* play aligned with last sample */ /* process aligned with last sample */
sample = buf->next_sample; sample = buf->next_sample;
} else { } else {
if (sample != buf->next_sample) { if (sample != buf->next_sample) {
@ -614,17 +812,17 @@ gst_ringbuffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
while (TRUE) { while (TRUE) {
gint diff; gint diff;
/* get the currently playing segment */ /* get the currently processed segment */
segplayed = g_atomic_int_get (&buf->segplayed); segdone = g_atomic_int_get (&buf->segdone);
/* see how far away it is from the write segment */ /* see how far away it is from the write segment */
diff = writeseg - segplayed; diff = writeseg - segdone;
GST_DEBUG GST_DEBUG
("pointer at %d, sample %llu, write to %d-%d, len %d, diff %d, segtotal %d, segsize %d", ("pointer at %d, sample %llu, write to %d-%d, len %d, diff %d, segtotal %d, segsize %d",
segplayed, sample, writeseg, writeoff, len, diff, segtotal, segsize); segdone, sample, writeseg, writeoff, len, diff, segtotal, segsize);
/* play segment too far ahead, we need to drop */ /* segment too far ahead, we need to drop */
if (diff < 0) { if (diff < 0) {
/* we need to drop one segment at a time, pretend we wrote a /* we need to drop one segment at a time, pretend we wrote a
* segment. */ * segment. */
@ -639,7 +837,7 @@ gst_ringbuffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
/* else we need to wait for the segment to become writable. */ /* else we need to wait for the segment to become writable. */
if (!wait_segment (buf)) if (!wait_segment (buf))
goto not_playing; goto not_started;
} }
/* we can write now */ /* we can write now */
@ -660,9 +858,127 @@ gst_ringbuffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
return len; return len;
/* ERRORS */ /* ERRORS */
not_playing: not_started:
{ {
GST_DEBUG ("stopped playing"); GST_DEBUG ("stopped processing");
return -1;
}
}
/**
* gst_ringbuffer_read:
* @buf: the #GstRingBuffer to read from
* @sample: the sample position of the data
* @data: where the data should be read
* @len: the length of the data to read
*
* Read @length samples from the ringbuffer into the memory pointed
* to by @data.
* The first sample should be read from position @sample in
* the ringbuffer.
*
* @len should not be a multiple of the segment size of the ringbuffer
* although it is recommended.
*
* Returns: The number of samples read from the ringbuffer or -1 on
* error.
*
* MT safe.
*/
guint
gst_ringbuffer_read (GstRingBuffer * buf, guint64 sample, guchar * data,
guint len)
{
gint segdone;
gint segsize, segtotal, bps, sps;
guint8 *dest;
g_return_val_if_fail (buf != NULL, -1);
g_return_val_if_fail (buf->data != NULL, -1);
g_return_val_if_fail (data != NULL, -1);
if (sample == -1) {
/* process aligned with last sample */
sample = buf->next_sample;
} else {
if (sample != buf->next_sample) {
GST_WARNING ("discontinuity found got %" G_GUINT64_FORMAT
", expected %" G_GUINT64_FORMAT, sample, buf->next_sample);
}
}
dest = GST_BUFFER_DATA (buf->data);
segsize = buf->spec.segsize;
segtotal = buf->spec.segtotal;
bps = buf->spec.bytes_per_sample;
sps = buf->samples_per_seg;
/* we assume the complete buffer will be consumed and the next sample
* should be written after this */
buf->next_sample = sample + len / bps;
/* read enough bytes */
while (len > 0) {
gint readlen;
gint readseg, readoff;
/* figure out the segment and the offset inside the segment where
* the sample should be written. */
readseg = sample / sps;
readoff = (sample % sps) * bps;
while (TRUE) {
gint diff;
/* get the currently processed segment */
segdone = g_atomic_int_get (&buf->segdone);
/* see how far away it is from the read segment */
diff = segdone - readseg;
GST_DEBUG
("pointer at %d, sample %llu, read from %d-%d, len %d, diff %d, segtotal %d, segsize %d",
segdone, sample, readseg, readoff, len, diff, segtotal, segsize);
/* segment too far ahead, we need to drop */
if (diff < 0) {
/* we need to drop one segment at a time, pretend we wrote a
* segment. */
readlen = MIN (segsize, len);
goto next;
}
/* read segment is within readable range, we can break the loop and
* start reading the data. */
if (diff > 0 && diff < segtotal)
break;
/* else we need to wait for the segment to become readable. */
if (!wait_segment (buf))
goto not_started;
}
/* we can read now */
readseg = readseg % segtotal;
readlen = MIN (segsize - readoff, len);
GST_DEBUG ("read @%p seg %d, off %d, len %d",
dest + readseg * segsize, readseg, readoff, readlen);
memcpy (data, dest + readseg * segsize + readoff, readlen);
next:
len -= readlen;
data += readlen;
sample += readlen / bps;
}
return len;
/* ERRORS */
not_started:
{
GST_DEBUG ("stopped processing");
return -1; return -1;
} }
} }
@ -677,7 +993,7 @@ not_playing:
* Returns a pointer to memory where the data from segment @segment * Returns a pointer to memory where the data from segment @segment
* can be found. This function is used by subclasses. * can be found. This function is used by subclasses.
* *
* Returns: FALSE if the buffer is not playing. * Returns: FALSE if the buffer is not started.
* *
* MT safe. * MT safe.
*/ */
@ -686,23 +1002,24 @@ gst_ringbuffer_prepare_read (GstRingBuffer * buf, gint * segment,
guint8 ** readptr, gint * len) guint8 ** readptr, gint * len)
{ {
guint8 *data; guint8 *data;
gint segplayed; gint segdone;
/* buffer must be playing */ /* buffer must be started */
if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_PLAYING) if (g_atomic_int_get (&buf->state) != GST_RINGBUFFER_STATE_STARTED)
return FALSE; return FALSE;
g_return_val_if_fail (buf != NULL, FALSE); g_return_val_if_fail (buf != NULL, FALSE);
g_return_val_if_fail (buf->data != NULL, FALSE); g_return_val_if_fail (buf->data != NULL, FALSE);
g_return_val_if_fail (segment != NULL, FALSE);
g_return_val_if_fail (readptr != NULL, FALSE); g_return_val_if_fail (readptr != NULL, FALSE);
g_return_val_if_fail (len != NULL, FALSE); g_return_val_if_fail (len != NULL, FALSE);
data = GST_BUFFER_DATA (buf->data); data = GST_BUFFER_DATA (buf->data);
/* get the position of the play pointer */ /* get the position of the pointer */
segplayed = g_atomic_int_get (&buf->segplayed); segdone = g_atomic_int_get (&buf->segdone);
*segment = segplayed % buf->spec.segtotal; *segment = segdone % buf->spec.segtotal;
*len = buf->spec.segsize; *len = buf->spec.segsize;
*readptr = data + *segment * *len; *readptr = data + *segment * *len;
@ -711,7 +1028,7 @@ gst_ringbuffer_prepare_read (GstRingBuffer * buf, gint * segment,
buf->callback (buf, *readptr, *len, buf->cb_data); buf->callback (buf, *readptr, *len, buf->cb_data);
GST_DEBUG ("prepare read from segment %d (real %d) @%p", GST_DEBUG ("prepare read from segment %d (real %d) @%p",
*segment, segplayed, *readptr); *segment, segdone, *readptr);
return TRUE; return TRUE;
} }
@ -722,7 +1039,7 @@ gst_ringbuffer_prepare_read (GstRingBuffer * buf, gint * segment,
* @advance: the number of segments written * @advance: the number of segments written
* *
* Subclasses should call this function to notify the fact that * Subclasses should call this function to notify the fact that
* @advance segments are now played by the device. * @advance segments are now processed by the device.
* *
* MT safe. * MT safe.
*/ */
@ -732,7 +1049,7 @@ gst_ringbuffer_advance (GstRingBuffer * buf, guint advance)
g_return_if_fail (buf != NULL); g_return_if_fail (buf != NULL);
/* update counter */ /* update counter */
g_atomic_int_add (&buf->segplayed, advance); g_atomic_int_add (&buf->segdone, advance);
/* the lock is already taken when the waiting flag is set, /* the lock is already taken when the waiting flag is set,
* we grab the lock as well to make sure the waiter is actually * we grab the lock as well to make sure the waiter is actually

20
gst-libs/gst/audio/gstringbuffer.h
View file

@ -44,7 +44,7 @@ typedef void (*GstRingBufferCallback) (GstRingBuffer *rbuf, guint8* data, guint
typedef enum { typedef enum {
GST_RINGBUFFER_STATE_STOPPED, GST_RINGBUFFER_STATE_STOPPED,
GST_RINGBUFFER_STATE_PAUSED, GST_RINGBUFFER_STATE_PAUSED,
GST_RINGBUFFER_STATE_PLAYING, GST_RINGBUFFER_STATE_STARTED,
} GstRingBufferState; } GstRingBufferState;
typedef enum { typedef enum {
@ -160,11 +160,11 @@ struct _GstRingBuffer {
/*< public >*/ /* ATOMIC */ /*< public >*/ /* ATOMIC */
gint state; /* state of the buffer */ gint state; /* state of the buffer */
gint segplayed; /* number of segments played since last start */ gint segdone; /* number of segments processed since last start */
gint waiting; /* when waiting for a segment to be freed */ gint waiting; /* when waiting for a segment to be freed */
/*< private >*/ /*< private >*/
guint64 next_sample; /* the next sample we need to write */ guint64 next_sample; /* the next sample we need to process */
GstRingBufferCallback callback; GstRingBufferCallback callback;
gpointer cb_data; gpointer cb_data;
}; };
@ -179,7 +179,7 @@ struct _GstRingBufferClass {
gboolean (*release) (GstRingBuffer *buf); gboolean (*release) (GstRingBuffer *buf);
/* playback control */ /* playback control */
gboolean (*play) (GstRingBuffer *buf); gboolean (*start) (GstRingBuffer *buf);
gboolean (*pause) (GstRingBuffer *buf); gboolean (*pause) (GstRingBuffer *buf);
gboolean (*resume) (GstRingBuffer *buf); gboolean (*resume) (GstRingBuffer *buf);
gboolean (*stop) (GstRingBuffer *buf); gboolean (*stop) (GstRingBuffer *buf);
@ -194,6 +194,10 @@ GType gst_ringbuffer_get_type(void);
void gst_ringbuffer_set_callback (GstRingBuffer *buf, GstRingBufferCallback cb, void gst_ringbuffer_set_callback (GstRingBuffer *buf, GstRingBufferCallback cb,
gpointer user_data); gpointer user_data);
gboolean gst_ringbuffer_parse_caps (GstRingBufferSpec *spec, GstCaps *caps);
void gst_ringbuffer_debug_spec_caps (GstRingBufferSpec *spec);
void gst_ringbuffer_debug_spec_buff (GstRingBufferSpec *spec);
/* allocate resources */ /* allocate resources */
gboolean gst_ringbuffer_acquire (GstRingBuffer *buf, GstRingBufferSpec *spec); gboolean gst_ringbuffer_acquire (GstRingBuffer *buf, GstRingBufferSpec *spec);
gboolean gst_ringbuffer_release (GstRingBuffer *buf); gboolean gst_ringbuffer_release (GstRingBuffer *buf);
@ -201,21 +205,25 @@ gboolean gst_ringbuffer_release (GstRingBuffer *buf);
gboolean gst_ringbuffer_is_acquired (GstRingBuffer *buf); gboolean gst_ringbuffer_is_acquired (GstRingBuffer *buf);
/* playback/pause */ /* playback/pause */
gboolean gst_ringbuffer_play (GstRingBuffer *buf); gboolean gst_ringbuffer_start (GstRingBuffer *buf);
gboolean gst_ringbuffer_pause (GstRingBuffer *buf); gboolean gst_ringbuffer_pause (GstRingBuffer *buf);
gboolean gst_ringbuffer_stop (GstRingBuffer *buf); gboolean gst_ringbuffer_stop (GstRingBuffer *buf);
/* get status */ /* get status */
guint gst_ringbuffer_delay (GstRingBuffer *buf); guint gst_ringbuffer_delay (GstRingBuffer *buf);
guint64 gst_ringbuffer_played_samples (GstRingBuffer *buf); guint64 gst_ringbuffer_samples_done (GstRingBuffer *buf);
void gst_ringbuffer_set_sample (GstRingBuffer *buf, guint64 sample); void gst_ringbuffer_set_sample (GstRingBuffer *buf, guint64 sample);
/* commit samples */ /* commit samples */
guint gst_ringbuffer_commit (GstRingBuffer *buf, guint64 sample, guint gst_ringbuffer_commit (GstRingBuffer *buf, guint64 sample,
guchar *data, guint len); guchar *data, guint len);
/* read samples */
guint gst_ringbuffer_read (GstRingBuffer *buf, guint64 sample,
guchar *data, guint len);
/* mostly protected */ /* mostly protected */
gboolean gst_ringbuffer_prepare_write (GstRingBuffer *buf, gint *segment, guint8 **writeptr, gint *len);
gboolean gst_ringbuffer_prepare_read (GstRingBuffer *buf, gint *segment, guint8 **readptr, gint *len); gboolean gst_ringbuffer_prepare_read (GstRingBuffer *buf, gint *segment, guint8 **readptr, gint *len);
void gst_ringbuffer_clear (GstRingBuffer *buf, gint segment); void gst_ringbuffer_clear (GstRingBuffer *buf, gint segment);
void gst_ringbuffer_advance (GstRingBuffer *buf, guint advance); void gst_ringbuffer_advance (GstRingBuffer *buf, guint advance);