/* 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:gstaudiosrc

 * @title: GstAudioSrc

 * @short_description: Simple base class for audio sources

 * @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer, #GstAudioSrc.

 *

 * This is the most simple base class for audio sources that only requires

 * subclasses to implement a set of simple functions:

 *

 * * `open()` :Open the device.

 * * `prepare()` :Configure the device with the specified format.

 * * `read()` :Read samples from the device.

 * * `reset()` :Unblock reads and flush the device.

 * * `delay()` :Get the number of samples in the device but not yet read.

 * * `unprepare()` :Undo operations done by prepare.

 * * `close()` :Close the device.

 *

 * All scheduling of samples and timestamps is done in this base class

 * together with #GstAudioBaseSrc using a default implementation of a

 * #GstAudioRingBuffer that uses threads.

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <string.h>

#include <gst/audio/audio.h>

#include "gstaudiosrc.h"

#include "gstaudioutilsprivate.h"

GST_DEBUG_CATEGORY_STATIC (gst_audio_src_debug);

#define GST_CAT_DEFAULT gst_audio_src_debug

#define GST_TYPE_AUDIO_SRC_RING_BUFFER        \

        (gst_audio_src_ring_buffer_get_type())

#define GST_AUDIO_SRC_RING_BUFFER(obj)        \

        (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_SRC_RING_BUFFER,GstAudioSrcRingBuffer))

#define GST_AUDIO_SRC_RING_BUFFER_CLASS(klass) \

        (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_SRC_RING_BUFFER,GstAudioSrcRingBufferClass))

#define GST_AUDIO_SRC_RING_BUFFER_GET_CLASS(obj) \

        (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_SRC_RING_BUFFER, GstAudioSrcRingBufferClass))

#define GST_IS_AUDIO_SRC_RING_BUFFER(obj)     \

        (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_SRC_RING_BUFFER))

#define GST_IS_AUDIO_SRC_RING_BUFFER_CLASS(klass)\

        (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_SRC_RING_BUFFER))

typedef struct _GstAudioSrcRingBuffer GstAudioSrcRingBuffer;

typedef struct _GstAudioSrcRingBufferClass GstAudioSrcRingBufferClass;

#define GST_AUDIO_SRC_RING_BUFFER_GET_COND(buf) (&(((GstAudioSrcRingBuffer *)buf)->cond))

#define GST_AUDIO_SRC_RING_BUFFER_WAIT(buf)     (g_cond_wait (GST_AUDIO_SRC_RING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))

#define GST_AUDIO_SRC_RING_BUFFER_SIGNAL(buf)   (g_cond_signal (GST_AUDIO_SRC_RING_BUFFER_GET_COND (buf)))

#define GST_AUDIO_SRC_RING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIO_SRC_RING_BUFFER_GET_COND (buf)))

struct _GstAudioSrcRingBuffer

{

  GstAudioRingBuffer object;

  gboolean running;

  gint queuedseg;

  GCond cond;

};

struct _GstAudioSrcRingBufferClass

{

  GstAudioRingBufferClass parent_class;

};

static void gst_audio_src_ring_buffer_class_init (GstAudioSrcRingBufferClass *

    klass);

static void gst_audio_src_ring_buffer_init (GstAudioSrcRingBuffer * ringbuffer,

    GstAudioSrcRingBufferClass * klass);

static void gst_audio_src_ring_buffer_dispose (GObject * object);

static void gst_audio_src_ring_buffer_finalize (GObject * object);

static GstAudioRingBufferClass *ring_parent_class = NULL;

static gboolean gst_audio_src_ring_buffer_open_device (GstAudioRingBuffer *

    buf);

static gboolean gst_audio_src_ring_buffer_close_device (GstAudioRingBuffer *

    buf);

static gboolean gst_audio_src_ring_buffer_acquire (GstAudioRingBuffer * buf,

    GstAudioRingBufferSpec * spec);

static gboolean gst_audio_src_ring_buffer_release (GstAudioRingBuffer * buf);

static gboolean gst_audio_src_ring_buffer_start (GstAudioRingBuffer * buf);

static gboolean gst_audio_src_ring_buffer_stop (GstAudioRingBuffer * buf);

static guint gst_audio_src_ring_buffer_delay (GstAudioRingBuffer * buf);

/* ringbuffer abstract base class */

static GType

gst_audio_src_ring_buffer_get_type (void)

{

  static GType ringbuffer_type = 0;

  if (!ringbuffer_type) {

    static const GTypeInfo ringbuffer_info = {

      sizeof (GstAudioSrcRingBufferClass),

      NULL,

      NULL,

      (GClassInitFunc) gst_audio_src_ring_buffer_class_init,

      NULL,

      NULL,

      sizeof (GstAudioSrcRingBuffer),

      0,

      (GInstanceInitFunc) gst_audio_src_ring_buffer_init,

      NULL

    };

    ringbuffer_type =

        g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,

        "GstAudioSrcRingBuffer", &ringbuffer_info, 0);

  }

  return ringbuffer_type;

}

static void

gst_audio_src_ring_buffer_class_init (GstAudioSrcRingBufferClass * klass)

{

  GObjectClass *gobject_class;

  GstAudioRingBufferClass *gstringbuffer_class;

  gobject_class = (GObjectClass *) klass;

  gstringbuffer_class = (GstAudioRingBufferClass *) klass;

  ring_parent_class = g_type_class_peek_parent (klass);

  gobject_class->dispose = gst_audio_src_ring_buffer_dispose;

  gobject_class->finalize = gst_audio_src_ring_buffer_finalize;

  gstringbuffer_class->open_device =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_open_device);

  gstringbuffer_class->close_device =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_close_device);

  gstringbuffer_class->acquire =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_acquire);

  gstringbuffer_class->release =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_release);

  gstringbuffer_class->start =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_start);

  gstringbuffer_class->resume =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_start);

  gstringbuffer_class->stop =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_stop);

  gstringbuffer_class->delay =

      GST_DEBUG_FUNCPTR (gst_audio_src_ring_buffer_delay);

}

typedef guint (*ReadFunc)

  (GstAudioSrc * src, gpointer data, guint length, GstClockTime * timestamp);

/* 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 (GstAudioRingBuffer * buf)

{

  GstAudioSrc *src;

  GstAudioSrcClass *csrc;

  GstAudioSrcRingBuffer *abuf = GST_AUDIO_SRC_RING_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");

  if ((readfunc = csrc->read) == NULL)

    goto no_function;

  if (G_UNLIKELY (!__gst_audio_set_thread_priority ()))

    GST_WARNING_OBJECT (src, "failed to set thread priority");

  message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),

      GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (src));

  g_value_init (&val, GST_TYPE_G_THREAD);

  g_value_set_boxed (&val, g_thread_self ());

  gst_message_set_stream_status_object (message, &val;;

  g_value_unset (&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;

    GstClockTime timestamp = GST_CLOCK_TIME_NONE;

    if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {

      gint read;

      left = len;

      do {

        read = readfunc (src, readptr, left, &timestamp);

        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 && g_atomic_int_get (&abuf->running));

      /* Update timestamp on buffer if required */

      gst_audio_ring_buffer_set_timestamp (buf, readseg, timestamp);

      /* we read one segment */

      gst_audio_ring_buffer_advance (buf, 1);

    } else {

      GST_OBJECT_LOCK (abuf);

      if (!abuf->running)

        goto stop_running;

      if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==

              GST_AUDIO_RING_BUFFER_STATE_STARTED)) {

        GST_OBJECT_UNLOCK (abuf);

        continue;

      }

      GST_DEBUG_OBJECT (src, "signal wait");

      GST_AUDIO_SRC_RING_BUFFER_SIGNAL (buf);

      GST_DEBUG_OBJECT (src, "wait for action");

      GST_AUDIO_SRC_RING_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));

    g_value_init (&val, GST_TYPE_G_THREAD);

    g_value_set_boxed (&val, g_thread_self ());

    gst_message_set_stream_status_object (message, &val;;

    g_value_unset (&val;;

    GST_DEBUG_OBJECT (src, "posting LEAVE stream status");

    gst_element_post_message (GST_ELEMENT_CAST (src), message);

    return;

  }

}

static void

gst_audio_src_ring_buffer_init (GstAudioSrcRingBuffer * ringbuffer,

    GstAudioSrcRingBufferClass * g_class)

{

  ringbuffer->running = FALSE;

  ringbuffer->queuedseg = 0;

  g_cond_init (&ringbuffer->cond);

}

static void

gst_audio_src_ring_buffer_dispose (GObject * object)

{

  GstAudioSrcRingBuffer *ringbuffer = GST_AUDIO_SRC_RING_BUFFER (object);

  g_cond_clear (&ringbuffer->cond);

  G_OBJECT_CLASS (ring_parent_class)->dispose (object);

}

static void

gst_audio_src_ring_buffer_finalize (GObject * object)

{

  G_OBJECT_CLASS (ring_parent_class)->finalize (object);

}

static gboolean

gst_audio_src_ring_buffer_open_device (GstAudioRingBuffer * 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_audio_src_ring_buffer_close_device (GstAudioRingBuffer * 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_audio_src_ring_buffer_acquire (GstAudioRingBuffer * buf,

    GstAudioRingBufferSpec * spec)

{

  GstAudioSrc *src;

  GstAudioSrcClass *csrc;

  GstAudioSrcRingBuffer *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->size = spec->segtotal * spec->segsize;

  buf->memory = g_malloc (buf->size);

  if (buf->spec.type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {

    gst_audio_format_fill_silence (buf->spec.info.finfo, buf->memory,

        buf->size);

  } else {

    /* FIXME, non-raw formats get 0 as the empty sample */

    memset (buf->memory, 0, buf->size);

  }

  abuf = GST_AUDIO_SRC_RING_BUFFER (buf);

  abuf->running = TRUE;

  /* FIXME: handle thread creation failure */

  src->thread = g_thread_try_new ("audiosrc-ringbuffer",

      (GThreadFunc) audioringbuffer_thread_func, buf, NULL);

  GST_AUDIO_SRC_RING_BUFFER_WAIT (buf);

  return result;

could_not_open:

  {

    return FALSE;

  }

}

/* function is called with LOCK */

static gboolean

gst_audio_src_ring_buffer_release (GstAudioRingBuffer * buf)

{

  GstAudioSrc *src;

  GstAudioSrcClass *csrc;

  GstAudioSrcRingBuffer *abuf;

  gboolean result = FALSE;

  src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf));

  csrc = GST_AUDIO_SRC_GET_CLASS (src);

  abuf = GST_AUDIO_SRC_RING_BUFFER (buf);

  abuf->running = FALSE;

  GST_AUDIO_SRC_RING_BUFFER_SIGNAL (buf);

  GST_OBJECT_UNLOCK (buf);

  /* join the thread */

  g_thread_join (src->thread);

  GST_OBJECT_LOCK (buf);

  /* free the buffer */

  g_free (buf->memory);

  buf->memory = NULL;

  if (csrc->unprepare)

    result = csrc->unprepare (src);

  return result;

}

static gboolean

gst_audio_src_ring_buffer_start (GstAudioRingBuffer * buf)

{

  GST_DEBUG ("start, sending signal");

  GST_AUDIO_SRC_RING_BUFFER_SIGNAL (buf);

  return TRUE;

}

static gboolean

gst_audio_src_ring_buffer_stop (GstAudioRingBuffer * 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_AUDIO_SRC_RING_BUFFER_WAIT (buf);

  GST_DEBUG ("stoped");

#endif

  return TRUE;

}

static guint

gst_audio_src_ring_buffer_delay (GstAudioRingBuffer * 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 \

    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;

}