/* GStreamer

 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>

 *               2000,2005 Wim Taymans <wim@fluendo.com>

 *

 * gstosssink.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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:element-osssink

 *

 * This element lets you output sound using the Open Sound System (OSS).

 *

 * Note that you should almost always use generic audio conversion elements

 * like audioconvert and audioresample in front of an audiosink to make sure

 * your pipeline works under all circumstances (those conversion elements will

 * act in passthrough-mode if no conversion is necessary).

 *

 * <refsect2>

 * <title>Example pipelines</title>

 * |[

 * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink

 * ]| will output a sine wave (continuous beep sound) to your sound card (with

 * a very low volume as precaution).

 * |[

 * gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink

 * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System.

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <sys/ioctl.h>

#include <fcntl.h>

#include <errno.h>

#include <unistd.h>

#include <string.h>

#ifdef HAVE_OSS_INCLUDE_IN_SYS

# include <sys/soundcard.h>

#else

# ifdef HAVE_OSS_INCLUDE_IN_ROOT

#  include <soundcard.h>

# else

#  ifdef HAVE_OSS_INCLUDE_IN_MACHINE

#   include <machine/soundcard.h>

#  else

#   error "What to include?"

#  endif /* HAVE_OSS_INCLUDE_IN_MACHINE */

# endif /* HAVE_OSS_INCLUDE_IN_ROOT */

#endif /* HAVE_OSS_INCLUDE_IN_SYS */

#include "common.h"

#include "gstosssink.h"

#include <gst/gst-i18n-plugin.h>

GST_DEBUG_CATEGORY_EXTERN (oss_debug);

#define GST_CAT_DEFAULT oss_debug

static void gst_oss_sink_dispose (GObject * object);

static void gst_oss_sink_finalise (GObject * object);

static void gst_oss_sink_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec);

static void gst_oss_sink_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec);

static GstCaps *gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter);

static gboolean gst_oss_sink_open (GstAudioSink * asink);

static gboolean gst_oss_sink_close (GstAudioSink * asink);

static gboolean gst_oss_sink_prepare (GstAudioSink * asink,

    GstAudioRingBufferSpec * spec);

static gboolean gst_oss_sink_unprepare (GstAudioSink * asink);

static gint gst_oss_sink_write (GstAudioSink * asink, gpointer data,

    guint length);

static guint gst_oss_sink_delay (GstAudioSink * asink);

static void gst_oss_sink_reset (GstAudioSink * asink);

/* OssSink signals and args */

enum

{

  LAST_SIGNAL

};

#define DEFAULT_DEVICE  "/dev/dsp"

enum

{

  PROP_0,

  PROP_DEVICE,

};

#define FORMATS "{" GST_AUDIO_NE(S16)","GST_AUDIO_NE(U16)", S8, U8 }"

static GstStaticPadTemplate osssink_sink_factory =

    GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-raw, "

        "format = (string) " FORMATS ", "

        "layout = (string) interleaved, "

        "rate = (int) [ 1, MAX ], "

        "channels = (int) 1; "

        "audio/x-raw, "

        "format = (string) " FORMATS ", "

        "layout = (string) interleaved, "

        "rate = (int) [ 1, MAX ], "

        "channels = (int) 2, " "channel-mask = (bitmask) 0x3")

    );

/* static guint gst_oss_sink_signals[LAST_SIGNAL] = { 0 }; */

#define gst_oss_sink_parent_class parent_class

G_DEFINE_TYPE (GstOssSink, gst_oss_sink, GST_TYPE_AUDIO_SINK);

static void

gst_oss_sink_dispose (GObject * object)

{

  GstOssSink *osssink = GST_OSSSINK (object);

  if (osssink->probed_caps) {

    gst_caps_unref (osssink->probed_caps);

    osssink->probed_caps = NULL;

  }

  G_OBJECT_CLASS (parent_class)->dispose (object);

}

static void

gst_oss_sink_class_init (GstOssSinkClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstBaseSinkClass *gstbasesink_class;

  GstAudioSinkClass *gstaudiosink_class;

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstbasesink_class = (GstBaseSinkClass *) klass;

  gstaudiosink_class = (GstAudioSinkClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  gobject_class->dispose = gst_oss_sink_dispose;

  gobject_class->finalize = gst_oss_sink_finalise;

  gobject_class->get_property = gst_oss_sink_get_property;

  gobject_class->set_property = gst_oss_sink_set_property;

  g_object_class_install_property (gobject_class, PROP_DEVICE,

      g_param_spec_string ("device", "Device",

          "OSS device (usually /dev/dspN)", DEFAULT_DEVICE,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_oss_sink_getcaps);

  gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_oss_sink_open);

  gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_oss_sink_close);

  gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_oss_sink_prepare);

  gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_oss_sink_unprepare);

  gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_oss_sink_write);

  gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_oss_sink_delay);

  gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_oss_sink_reset);

  gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (OSS)",

      "Sink/Audio",

      "Output to a sound card via OSS",

      "Erik Walthinsen <omega@cse.ogi.edu>, "

      "Wim Taymans <wim.taymans@chello.be>");

  gst_element_class_add_static_pad_template (gstelement_class,

      &osssink_sink_factory);

}

static void

gst_oss_sink_init (GstOssSink * osssink)

{

  const gchar *device;

  GST_DEBUG_OBJECT (osssink, "initializing osssink");

  device = g_getenv ("AUDIODEV");

  if (device == NULL)

    device = DEFAULT_DEVICE;

  osssink->device = g_strdup (device);

  osssink->fd = -1;

}

static void

gst_oss_sink_finalise (GObject * object)

{

  GstOssSink *osssink = GST_OSSSINK (object);

  g_free (osssink->device);

  G_OBJECT_CLASS (parent_class)->finalize ((GObject *) (object));

}

static void

gst_oss_sink_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec)

{

  GstOssSink *sink;

  sink = GST_OSSSINK (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_free (sink->device);

      sink->device = g_value_dup_string (value);

      if (sink->probed_caps) {

        gst_caps_unref (sink->probed_caps);

        sink->probed_caps = NULL;

      }

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_oss_sink_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec)

{

  GstOssSink *sink;

  sink = GST_OSSSINK (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_value_set_string (value, sink->device);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static GstCaps *

gst_oss_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)

{

  GstOssSink *osssink;

  GstCaps *caps;

  osssink = GST_OSSSINK (bsink);

  if (osssink->fd == -1) {

    caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink));

  } else if (osssink->probed_caps) {

    caps = gst_caps_ref (osssink->probed_caps);

  } else {

    caps = gst_oss_helper_probe_caps (osssink->fd);

    if (caps && !gst_caps_is_empty (caps)) {

      osssink->probed_caps = gst_caps_ref (caps);

    }

  }

  if (filter && caps) {

    GstCaps *intersection;

    intersection =

        gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (caps);

    return intersection;

  } else {

    return caps;

  }

}

static gint

ilog2 (gint x)

{

  /* well... hacker's delight explains... */

  x = x | (x >> 1);

  x = x | (x >> 2);

  x = x | (x >> 4);

  x = x | (x >> 8);

  x = x | (x >> 16);

  x = x - ((x >> 1) & 0x55555555);

  x = (x & 0x33333333) + ((x >> 2) & 0x33333333);

  x = (x + (x >> 4)) & 0x0f0f0f0f;

  x = x + (x >> 8);

  x = x + (x >> 16);

  return (x & 0x0000003f) - 1;

}

static gint

gst_oss_sink_get_format (GstAudioRingBufferFormatType fmt, GstAudioFormat rfmt)

{

  gint result;

  switch (fmt) {

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW:

      result = AFMT_MU_LAW;

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW:

      result = AFMT_A_LAW;

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM:

      result = AFMT_IMA_ADPCM;

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:

      result = AFMT_MPEG;

      break;

    case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW:

    {

      switch (rfmt) {

        case GST_AUDIO_FORMAT_U8:

          result = AFMT_U8;

          break;

        case GST_AUDIO_FORMAT_S16LE:

          result = AFMT_S16_LE;

          break;

        case GST_AUDIO_FORMAT_S16BE:

          result = AFMT_S16_BE;

          break;

        case GST_AUDIO_FORMAT_S8:

          result = AFMT_S8;

          break;

        case GST_AUDIO_FORMAT_U16LE:

          result = AFMT_U16_LE;

          break;

        case GST_AUDIO_FORMAT_U16BE:

          result = AFMT_U16_BE;

          break;

        default:

          result = 0;

          break;

      }

      break;

    }

    default:

      result = 0;

      break;

  }

  return result;

}

static gboolean

gst_oss_sink_open (GstAudioSink * asink)

{

  GstOssSink *oss;

  int mode;

  oss = GST_OSSSINK (asink);

  mode = O_WRONLY;

  mode |= O_NONBLOCK;

  oss->fd = open (oss->device, mode, 0);

  if (oss->fd == -1) {

    switch (errno) {

      case EBUSY:

        goto busy;

      case EACCES:

        goto no_permission;

      default:

        goto open_failed;

    }

  }

  return TRUE;

  /* ERRORS */

busy:

  {

    GST_ELEMENT_ERROR (oss, RESOURCE, BUSY,

        (_("Could not open audio device for playback. "

                "Device is being used by another application.")), (NULL));

    return FALSE;

  }

no_permission:

  {

    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,

        (_("Could not open audio device for playback. "

                "You don't have permission to open the device.")),

        GST_ERROR_SYSTEM);

    return FALSE;

  }

open_failed:

  {

    GST_ELEMENT_ERROR (oss, RESOURCE, OPEN_WRITE,

        (_("Could not open audio device for playback.")), GST_ERROR_SYSTEM);

    return FALSE;

  }

}

static gboolean

gst_oss_sink_close (GstAudioSink * asink)

{

  close (GST_OSSSINK (asink)->fd);

  GST_OSSSINK (asink)->fd = -1;

  return TRUE;

}

static gboolean

gst_oss_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)

{

  GstOssSink *oss;

  struct audio_buf_info info;

  int mode;

  int tmp;

  guint width, rate, channels;

  oss = GST_OSSSINK (asink);

  /* we opened non-blocking so that we can detect if the device is available

   * without hanging forever. We now want to remove the non-blocking flag. */

  mode = fcntl (oss->fd, F_GETFL);

  mode &= ~O_NONBLOCK;

  if (fcntl (oss->fd, F_SETFL, mode) == -1) {

    /* some drivers do no support unsetting the non-blocking flag, try to

     * close/open the device then. This is racy but we error out properly. */

    gst_oss_sink_close (asink);

    if ((oss->fd = open (oss->device, O_WRONLY, 0)) == -1)

      goto non_block;

  }

  tmp = gst_oss_sink_get_format (spec->type,

      GST_AUDIO_INFO_FORMAT (&spec->info));

  if (tmp == 0)

    goto wrong_format;

  width = GST_AUDIO_INFO_WIDTH (&spec->info);

  rate = GST_AUDIO_INFO_RATE (&spec->info);

  channels = GST_AUDIO_INFO_CHANNELS (&spec->info);

  if (width != 16 && width != 8)

    goto dodgy_width;

  SET_PARAM (oss, SNDCTL_DSP_SETFMT, tmp, "SETFMT");

  if (channels == 2)

    SET_PARAM (oss, SNDCTL_DSP_STEREO, 1, "STEREO");

  SET_PARAM (oss, SNDCTL_DSP_CHANNELS, channels, "CHANNELS");

  SET_PARAM (oss, SNDCTL_DSP_SPEED, rate, "SPEED");

  tmp = ilog2 (spec->segsize);

  tmp = ((spec->segtotal & 0x7fff) << 16) | tmp;

  GST_DEBUG_OBJECT (oss, "set segsize: %d, segtotal: %d, value: %08x",

      spec->segsize, spec->segtotal, tmp);

  SET_PARAM (oss, SNDCTL_DSP_SETFRAGMENT, tmp, "SETFRAGMENT");

  GET_PARAM (oss, SNDCTL_DSP_GETOSPACE, &info, "GETOSPACE");

  spec->segsize = info.fragsize;

  spec->segtotal = info.fragstotal;

  oss->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);

  GST_DEBUG_OBJECT (oss, "got segsize: %d, segtotal: %d, value: %08x",

      spec->segsize, spec->segtotal, tmp);

  return TRUE;

  /* ERRORS */

non_block:

  {

    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),

        ("Unable to set device %s in non blocking mode: %s",

            oss->device, g_strerror (errno)));

    return FALSE;

  }

wrong_format:

  {

    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),

        ("Unable to get format (%d, %d)", spec->type,

            GST_AUDIO_INFO_FORMAT (&spec->info)));

    return FALSE;

  }

dodgy_width:

  {

    GST_ELEMENT_ERROR (oss, RESOURCE, SETTINGS, (NULL),

        ("unexpected width %d", width));

    return FALSE;

  }

}

static gboolean

gst_oss_sink_unprepare (GstAudioSink * asink)

{

  /* could do a SNDCTL_DSP_RESET, but the OSS manual recommends a close/open */

  if (!gst_oss_sink_close (asink))

    goto couldnt_close;

  if (!gst_oss_sink_open (asink))

    goto couldnt_reopen;

  return TRUE;

  /* ERRORS */

couldnt_close:

  {

    GST_DEBUG_OBJECT (asink, "Could not close the audio device");

    return FALSE;

  }

couldnt_reopen:

  {

    GST_DEBUG_OBJECT (asink, "Could not reopen the audio device");

    return FALSE;

  }

}

static gint

gst_oss_sink_write (GstAudioSink * asink, gpointer data, guint length)

{

  return write (GST_OSSSINK (asink)->fd, data, length);

}

static guint

gst_oss_sink_delay (GstAudioSink * asink)

{

  GstOssSink *oss;

  gint delay = 0;

  gint ret;

  oss = GST_OSSSINK (asink);

#ifdef SNDCTL_DSP_GETODELAY

  ret = ioctl (oss->fd, SNDCTL_DSP_GETODELAY, &delay);

#else

  ret = -1;

#endif

  if (ret < 0) {

    audio_buf_info info;

    ret = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &info;;

    delay = (ret < 0 ? 0 : (info.fragstotal * info.fragsize) - info.bytes);

  }

  return delay / oss->bytes_per_sample;

}

static void

gst_oss_sink_reset (GstAudioSink * asink)

{

  /* There's nothing we can do here really: OSS can't handle access to the

   * same device/fd from multiple threads and might deadlock or blow up in

   * other ways if we try an ioctl SNDCTL_DSP_RESET or similar */

}