gstreamer/gst-libs/gst/audio/gstaudiosrc.c

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/* 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.
*/
/**
* SECTION:gstaudiosrc
* @short_description: Simple base class for audio sources
* @see_also: #GstBaseAudioSrc, #GstRingBuffer, #GstAudioSrc.
*
* This is the most simple base class for audio sources that only requires
* subclasses to implement a set of simple functions:
*
* <variablelist>
* <varlistentry>
* <term>open()</term>
* <listitem><para>Open the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>prepare()</term>
* <listitem><para>Configure the device with the specified format.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>read()</term>
* <listitem><para>Read samples from the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>reset()</term>
* <listitem><para>Unblock reads and flush the device.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>delay()</term>
* <listitem><para>Get the number of samples in the device but not yet read.
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>unprepare()</term>
* <listitem><para>Undo operations done by prepare.</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>close()</term>
* <listitem><para>Close the device.</para></listitem>
* </varlistentry>
* </variablelist>
*
* All scheduling of samples and timestamps is done in this base class
* together with #GstBaseAudioSrc using a default implementation of a
* #GstRingBuffer that uses threads.
*
* Last reviewed on 2006-09-27 (0.10.12)
*/
#include <string.h>
#include "gstaudiosrc.h"
#include "gst/glib-compat-private.h"
GST_DEBUG_CATEGORY_STATIC (gst_audio_src_debug);
#define GST_CAT_DEFAULT gst_audio_src_debug
#define GST_TYPE_AUDIORING_BUFFER \
(gst_audioringbuffer_get_type())
#define GST_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer))
#define GST_AUDIORING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass))
#define GST_AUDIORING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass))
#define GST_IS_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER))
#define GST_IS_AUDIORING_BUFFER_CLASS(klass)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER))
typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
#define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
#define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
#define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf)))
#define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_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,
GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_dispose (GObject * object);
static void gst_audioringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf);
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_RING_BUFFER, "GstAudioSrcRingBuffer",
&ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_class_init): * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init): * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init): * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_class_init): * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init): * gst-libs/gst/interfaces/colorbalancechannel.c: (gst_color_balance_channel_class_init): * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/interfaces/tunerchannel.c: (gst_tuner_channel_class_init): * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init): * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_stream_selector_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * sys/v4l/gstv4lcolorbalance.c: (gst_v4l_color_balance_channel_class_init): * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init), (gst_v4l_tuner_norm_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:02:53 +00:00
ring_parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_audioringbuffer_dispose;
gobject_class->finalize = gst_audioringbuffer_finalize;
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_close_device);
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_AUDIORING_BUFFER (buf);
ReadFunc readfunc;
GstMessage *message;
GValue val = { 0 };
src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_GET_CLASS (src);
GST_DEBUG_OBJECT (src, "enter thread");
readfunc = csrc->read;
if (readfunc == NULL)
goto no_function;
/* FIXME: maybe we should at least use a custom pointer type here? */
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, src->thread);
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (src));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (src, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (src), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint read;
left = len;
do {
read = readfunc (src, readptr, left);
GST_LOG_OBJECT (src, "transfered %d bytes of %d to segment %d", read,
left, readseg);
if (read < 0 || read > left) {
GST_WARNING_OBJECT (src,
"error reading data %d (reason: %s), skipping segment", read,
g_strerror (errno));
break;
}
left -= read;
readptr += read;
} while (left > 0);
/* we read one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (src, "signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (src, "wait for action");
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (src, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (src, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
g_assert_not_reached ();
return;
/* ERROR */
no_function:
{
GST_DEBUG ("no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG ("stop running, exit thread");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (src));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (src, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (src), message);
return;
}
}
static void
gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
GstAudioRingBufferClass * g_class)
{
ringbuffer->running = FALSE;
ringbuffer->queuedseg = 0;
ringbuffer->cond = g_cond_new ();
}
static void
gst_audioringbuffer_dispose (GObject * object)
{
GstAudioRingBuffer *ringbuffer = GST_AUDIORING_BUFFER (object);
if (ringbuffer->cond) {
g_cond_free (ringbuffer->cond);
ringbuffer->cond = NULL;
}
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_open_device (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
gboolean result = TRUE;
src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_GET_CLASS (src);
if (csrc->open)
result = csrc->open (src);
if (!result)
goto could_not_open;
return result;
could_not_open:
{
return FALSE;
}
}
static gboolean
gst_audioringbuffer_close_device (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
gboolean result = TRUE;
src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_GET_CLASS (src);
if (csrc->close)
result = csrc->close (src);
if (!result)
goto could_not_open;
return result;
could_not_open:
{
return FALSE;
}
}
static gboolean
gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_GET_CLASS (src);
if (csrc->prepare)
result = csrc->prepare (src, spec);
if (!result)
goto could_not_open;
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_AUDIORING_BUFFER (buf);
abuf->running = TRUE;
/* FIXME: handle thread creation failure */
#if !GLIB_CHECK_VERSION (2, 31, 0)
src->thread =
g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
NULL);
#else
src->thread = g_thread_try_new ("audiosrc-ringbuffer",
(GThreadFunc) audioringbuffer_thread_func, buf, NULL);
#endif
GST_AUDIORING_BUFFER_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_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_GET_CLASS (src);
abuf = GST_AUDIORING_BUFFER (buf);
abuf->running = FALSE;
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (buf);
/* join the thread */
g_thread_join (src->thread);
GST_OBJECT_LOCK (buf);
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
if (csrc->unprepare)
result = csrc->unprepare (src);
return result;
}
static gboolean
gst_audioringbuffer_start (GstRingBuffer * buf)
{
GST_DEBUG ("start, sending signal");
GST_AUDIORING_BUFFER_SIGNAL (buf);
return TRUE;
}
static gboolean
gst_audioringbuffer_stop (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_GET_CLASS (src);
/* unblock any pending writes to the audio device */
if (csrc->reset) {
GST_DEBUG ("reset...");
csrc->reset (src);
GST_DEBUG ("reset done");
}
#if 0
GST_DEBUG ("stop, waiting...");
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG ("stoped");
#endif
return TRUE;
}
static guint
gst_audioringbuffer_delay (GstRingBuffer * buf)
{
GstAudioSrc *src;
GstAudioSrcClass *csrc;
guint res = 0;
src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
csrc = GST_AUDIO_SRC_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_audio_src_debug, "audiosrc", 0, "audiosrc element");
GST_BOILERPLATE_FULL (GstAudioSrc, gst_audio_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init);
static GstRingBuffer *gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src);
static void
gst_audio_src_base_init (gpointer g_class)
{
}
static void
gst_audio_src_class_init (GstAudioSrcClass * klass)
{
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audio_src_create_ringbuffer);
g_type_class_ref (GST_TYPE_AUDIORING_BUFFER);
}
static void
gst_audio_src_init (GstAudioSrc * audiosrc, GstAudioSrcClass * g_class)
{
}
static GstRingBuffer *
gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstRingBuffer *buffer;
GST_DEBUG ("creating ringbuffer");
buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL);
GST_DEBUG ("created ringbuffer @%p", buffer);
return buffer;
}