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547 lines
15 KiB
C
547 lines
15 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2005 Wim Taymans <wim@fluendo.com>
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*
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* gstaudiosrc.c: simple audio src base class
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstaudiosrc
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* @short_description: Simple base class for audio sources
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* @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer, #GstAudioSrc.
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*
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* This is the most simple base class for audio sources that only requires
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* subclasses to implement a set of simple functions:
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*
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* <variablelist>
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* <varlistentry>
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* <term>open()</term>
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* <listitem><para>Open the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>prepare()</term>
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* <listitem><para>Configure the device with the specified format.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>read()</term>
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* <listitem><para>Read samples from the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>reset()</term>
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* <listitem><para>Unblock reads and flush the device.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>delay()</term>
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* <listitem><para>Get the number of samples in the device but not yet read.
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* </para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>unprepare()</term>
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* <listitem><para>Undo operations done by prepare.</para></listitem>
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* </varlistentry>
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* <varlistentry>
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* <term>close()</term>
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* <listitem><para>Close the device.</para></listitem>
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* </varlistentry>
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* </variablelist>
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*
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* All scheduling of samples and timestamps is done in this base class
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* together with #GstAudioBaseSrc using a default implementation of a
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* #GstAudioRingBuffer that uses threads.
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*
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* Last reviewed on 2006-09-27 (0.10.12)
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*/
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#include <string.h>
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#include <gst/audio/audio.h>
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#include "gstaudiosrc.h"
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GST_DEBUG_CATEGORY_STATIC (gst_audio_src_debug);
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#define GST_CAT_DEFAULT gst_audio_src_debug
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#define GST_TYPE_AUDIO_SRC_RING_BUFFER \
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(gst_audio_src_ring_buffer_get_type())
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#define GST_AUDIO_SRC_RING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_SRC_RING_BUFFER,GstAudioSrcRingBuffer))
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#define GST_AUDIO_SRC_RING_BUFFER_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_SRC_RING_BUFFER,GstAudioSrcRingBufferClass))
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#define GST_AUDIO_SRC_RING_BUFFER_GET_CLASS(obj) \
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(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_SRC_RING_BUFFER, GstAudioSrcRingBufferClass))
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#define GST_IS_AUDIO_SRC_RING_BUFFER(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_SRC_RING_BUFFER))
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#define GST_IS_AUDIO_SRC_RING_BUFFER_CLASS(klass)\
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_SRC_RING_BUFFER))
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typedef struct _GstAudioSrcRingBuffer GstAudioSrcRingBuffer;
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typedef struct _GstAudioSrcRingBufferClass GstAudioSrcRingBufferClass;
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#define GST_AUDIO_SRC_RING_BUFFER_GET_COND(buf) (&(((GstAudioSrcRingBuffer *)buf)->cond))
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#define GST_AUDIO_SRC_RING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIO_SRC_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
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#define GST_AUDIO_SRC_RING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIO_SRC_RING_BUFFER_GET_COND (buf)))
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#define GST_AUDIO_SRC_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_SRC_RING_BUFFER_GET_COND (buf)))
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struct _GstAudioSrcRingBuffer
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{
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GstAudioRingBuffer object;
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gboolean running;
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gint queuedseg;
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GCond cond;
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};
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struct _GstAudioSrcRingBufferClass
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{
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GstAudioRingBufferClass parent_class;
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};
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static void gst_audio_src_ring_buffer_class_init (GstAudioSrcRingBufferClass *
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klass);
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static void gst_audio_src_ring_buffer_init (GstAudioSrcRingBuffer * ringbuffer,
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GstAudioSrcRingBufferClass * klass);
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static void gst_audio_src_ring_buffer_dispose (GObject * object);
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static void gst_audio_src_ring_buffer_finalize (GObject * object);
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static GstAudioRingBufferClass *ring_parent_class = NULL;
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static gboolean gst_audio_src_ring_buffer_open_device (GstAudioRingBuffer *
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buf);
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static gboolean gst_audio_src_ring_buffer_close_device (GstAudioRingBuffer *
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buf);
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static gboolean gst_audio_src_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec);
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static gboolean gst_audio_src_ring_buffer_release (GstAudioRingBuffer * buf);
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static gboolean gst_audio_src_ring_buffer_start (GstAudioRingBuffer * buf);
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static gboolean gst_audio_src_ring_buffer_stop (GstAudioRingBuffer * buf);
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static guint gst_audio_src_ring_buffer_delay (GstAudioRingBuffer * buf);
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/* ringbuffer abstract base class */
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static GType
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gst_audio_src_ring_buffer_get_type (void)
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{
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static GType ringbuffer_type = 0;
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if (!ringbuffer_type) {
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static const GTypeInfo ringbuffer_info = {
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sizeof (GstAudioSrcRingBufferClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_audio_src_ring_buffer_class_init,
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NULL,
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NULL,
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sizeof (GstAudioSrcRingBuffer),
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0,
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(GInstanceInitFunc) gst_audio_src_ring_buffer_init,
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NULL
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};
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ringbuffer_type =
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g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
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"GstAudioSrcRingBuffer", &ringbuffer_info, 0);
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}
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return ringbuffer_type;
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}
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static void
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gst_audio_src_ring_buffer_class_init (GstAudioSrcRingBufferClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioRingBufferClass *gstringbuffer_class;
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gobject_class = (GObjectClass *) klass;
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gstringbuffer_class = (GstAudioRingBufferClass *) klass;
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ring_parent_class = g_type_class_peek_parent (klass);
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gobject_class->dispose = gst_audio_src_ring_buffer_dispose;
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gobject_class->finalize = gst_audio_src_ring_buffer_finalize;
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gstringbuffer_class->open_device =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_open_device);
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gstringbuffer_class->close_device =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_close_device);
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gstringbuffer_class->acquire =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_acquire);
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gstringbuffer_class->release =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_release);
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gstringbuffer_class->start =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_start);
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gstringbuffer_class->resume =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_start);
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gstringbuffer_class->stop =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_stop);
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gstringbuffer_class->delay =
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GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_delay);
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}
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typedef guint (*ReadFunc)
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(GstAudioSrc * src, gpointer data, guint length, GstClockTime * timestamp);
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/* this internal thread does nothing else but read samples from the audio device.
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* It will read each segment in the ringbuffer and will update the play
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* pointer.
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* The start/stop methods control the thread.
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*/
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static void
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audioringbuffer_thread_func (GstAudioRingBuffer * buf)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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GstAudioSrcRingBuffer *abuf = GST_AUDIO_SRC_RING_BUFFER (buf);
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ReadFunc readfunc;
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GstMessage *message;
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GValue val = { 0 };
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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GST_DEBUG_OBJECT (src, "enter thread");
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if ((readfunc = csrc->read) == NULL)
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goto no_function;
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message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
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GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (src));
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g_value_init (&val, GST_TYPE_G_THREAD);
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g_value_set_boxed (&val, src->thread);
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gst_message_set_stream_status_object (message, &val);
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g_value_unset (&val);
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GST_DEBUG_OBJECT (src, "posting ENTER stream status");
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gst_element_post_message (GST_ELEMENT_CAST (src), message);
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while (TRUE) {
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gint left, len;
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guint8 *readptr;
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gint readseg;
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GstClockTime timestamp = GST_CLOCK_TIME_NONE;
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if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
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gint read;
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left = len;
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do {
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read = readfunc (src, readptr, left, ×tamp);
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GST_LOG_OBJECT (src, "transfered %d bytes of %d to segment %d", read,
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left, readseg);
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if (read < 0 || read > left) {
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GST_WARNING_OBJECT (src,
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"error reading data %d (reason: %s), skipping segment", read,
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g_strerror (errno));
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break;
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}
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left -= read;
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readptr += read;
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} while (left > 0);
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/* Update timestamp on buffer if required */
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gst_audio_ring_buffer_set_timestamp (buf, readseg, timestamp);
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/* we read one segment */
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gst_audio_ring_buffer_advance (buf, 1);
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} else {
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GST_OBJECT_LOCK (abuf);
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if (!abuf->running)
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goto stop_running;
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if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
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GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
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GST_OBJECT_UNLOCK (abuf);
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continue;
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}
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GST_DEBUG_OBJECT (src, "signal wait");
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GST_AUDIO_SRC_RING_BUFFER_SIGNAL (buf);
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GST_DEBUG_OBJECT (src, "wait for action");
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GST_AUDIO_SRC_RING_BUFFER_WAIT (buf);
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GST_DEBUG_OBJECT (src, "got signal");
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if (!abuf->running)
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goto stop_running;
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GST_DEBUG_OBJECT (src, "continue running");
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GST_OBJECT_UNLOCK (abuf);
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}
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}
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/* Will never be reached */
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g_assert_not_reached ();
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return;
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/* ERROR */
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no_function:
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{
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GST_DEBUG ("no write function, exit thread");
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return;
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}
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stop_running:
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{
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GST_OBJECT_UNLOCK (abuf);
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GST_DEBUG ("stop running, exit thread");
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message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
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GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (src));
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g_value_init (&val, GST_TYPE_G_THREAD);
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g_value_set_boxed (&val, src->thread);
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gst_message_set_stream_status_object (message, &val);
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g_value_unset (&val);
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GST_DEBUG_OBJECT (src, "posting LEAVE stream status");
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gst_element_post_message (GST_ELEMENT_CAST (src), message);
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return;
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}
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}
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static void
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gst_audio_src_ring_buffer_init (GstAudioSrcRingBuffer * ringbuffer,
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GstAudioSrcRingBufferClass * g_class)
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{
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ringbuffer->running = FALSE;
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ringbuffer->queuedseg = 0;
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g_cond_init (&ringbuffer->cond);
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}
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static void
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gst_audio_src_ring_buffer_dispose (GObject * object)
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{
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GstAudioSrcRingBuffer *ringbuffer = GST_AUDIO_SRC_RING_BUFFER (object);
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g_cond_clear (&ringbuffer->cond);
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G_OBJECT_CLASS (ring_parent_class)->dispose (object);
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}
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static void
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gst_audio_src_ring_buffer_finalize (GObject * object)
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{
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G_OBJECT_CLASS (ring_parent_class)->finalize (object);
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}
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static gboolean
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gst_audio_src_ring_buffer_open_device (GstAudioRingBuffer * buf)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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gboolean result = TRUE;
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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if (csrc->open)
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result = csrc->open (src);
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if (!result)
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goto could_not_open;
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return result;
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could_not_open:
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{
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return FALSE;
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}
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}
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static gboolean
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gst_audio_src_ring_buffer_close_device (GstAudioRingBuffer * buf)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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gboolean result = TRUE;
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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if (csrc->close)
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result = csrc->close (src);
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if (!result)
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goto could_not_open;
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return result;
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could_not_open:
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{
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return FALSE;
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}
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}
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static gboolean
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gst_audio_src_ring_buffer_acquire (GstAudioRingBuffer * buf,
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GstAudioRingBufferSpec * spec)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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GstAudioSrcRingBuffer *abuf;
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gboolean result = FALSE;
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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if (csrc->prepare)
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result = csrc->prepare (src, spec);
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if (!result)
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goto could_not_open;
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buf->size = spec->segtotal * spec->segsize;
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buf->memory = g_malloc0 (buf->size);
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abuf = GST_AUDIO_SRC_RING_BUFFER (buf);
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abuf->running = TRUE;
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/* FIXME: handle thread creation failure */
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src->thread = g_thread_try_new ("audiosrc-ringbuffer",
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(GThreadFunc) audioringbuffer_thread_func, buf, NULL);
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GST_AUDIO_SRC_RING_BUFFER_WAIT (buf);
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return result;
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could_not_open:
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{
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return FALSE;
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}
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}
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/* function is called with LOCK */
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static gboolean
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gst_audio_src_ring_buffer_release (GstAudioRingBuffer * buf)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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GstAudioSrcRingBuffer *abuf;
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gboolean result = FALSE;
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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abuf = GST_AUDIO_SRC_RING_BUFFER (buf);
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abuf->running = FALSE;
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GST_AUDIO_SRC_RING_BUFFER_SIGNAL (buf);
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GST_OBJECT_UNLOCK (buf);
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/* join the thread */
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g_thread_join (src->thread);
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GST_OBJECT_LOCK (buf);
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/* free the buffer */
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g_free (buf->memory);
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buf->memory = NULL;
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if (csrc->unprepare)
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result = csrc->unprepare (src);
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return result;
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}
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static gboolean
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gst_audio_src_ring_buffer_start (GstAudioRingBuffer * buf)
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{
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GST_DEBUG ("start, sending signal");
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GST_AUDIO_SRC_RING_BUFFER_SIGNAL (buf);
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return TRUE;
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}
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static gboolean
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gst_audio_src_ring_buffer_stop (GstAudioRingBuffer * buf)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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/* unblock any pending writes to the audio device */
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if (csrc->reset) {
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GST_DEBUG ("reset...");
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csrc->reset (src);
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GST_DEBUG ("reset done");
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}
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#if 0
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GST_DEBUG ("stop, waiting...");
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GST_AUDIO_SRC_RING_BUFFER_WAIT (buf);
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GST_DEBUG ("stoped");
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#endif
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return TRUE;
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}
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static guint
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gst_audio_src_ring_buffer_delay (GstAudioRingBuffer * buf)
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{
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GstAudioSrc *src;
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GstAudioSrcClass *csrc;
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guint res = 0;
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src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));
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csrc = GST_AUDIO_SRC_GET_CLASS (src);
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|
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if (csrc->delay)
|
|
res = csrc->delay (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* AudioSrc signals and args */
|
|
enum
|
|
{
|
|
/* FILL ME */
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
};
|
|
|
|
#define _do_init \
|
|
GST_DEBUG_CATEGORY_INIT (gst_audio_src_debug, "audiosrc", 0, "audiosrc element");
|
|
#define gst_audio_src_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (GstAudioSrc, gst_audio_src,
|
|
GST_TYPE_AUDIO_BASE_SRC, _do_init);
|
|
|
|
static GstAudioRingBuffer *gst_audio_src_create_ringbuffer (GstAudioBaseSrc *
|
|
src);
|
|
|
|
static void
|
|
gst_audio_src_class_init (GstAudioSrcClass * klass)
|
|
{
|
|
GstAudioBaseSrcClass *gstaudiobasesrc_class;
|
|
|
|
gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
|
|
|
|
gstaudiobasesrc_class->create_ringbuffer =
|
|
GST_DEBUG_FUNCPTR (gst_audio_src_create_ringbuffer);
|
|
|
|
g_type_class_ref (GST_TYPE_AUDIO_SRC_RING_BUFFER);
|
|
}
|
|
|
|
static void
|
|
gst_audio_src_init (GstAudioSrc * audiosrc)
|
|
{
|
|
}
|
|
|
|
static GstAudioRingBuffer *
|
|
gst_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
|
|
{
|
|
GstAudioRingBuffer *buffer;
|
|
|
|
GST_DEBUG ("creating ringbuffer");
|
|
buffer = g_object_new (GST_TYPE_AUDIO_SRC_RING_BUFFER, NULL);
|
|
GST_DEBUG ("created ringbuffer @%p", buffer);
|
|
|
|
return buffer;
|
|
}
|