/* GStreamer

 * Copyright (C) 2005 Wim Taymans <wim@fluendo.com>

 * Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>

 *

 * gstalsasink.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.

 */

/**

 * SECTION:element-alsasink

 * @short_description: play audio to an ALSA device

 * @see_also: alsasrc, alsamixer

 *

 * <refsect2>

 * <para>

 * This element renders raw audio samples using the ALSA api.

 * </para>

 * <title>Example pipelines</title>

 * <para>

 * Play an Ogg/Vorbis file.

 * </para>

 * <programlisting>

 * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink

 * </programlisting>

 * </refsect2>

 *

 * Last reviewed on 2006-03-01 (0.10.4)

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <sys/ioctl.h>

#include <fcntl.h>

#include <errno.h>

#include <unistd.h>

#include <string.h>

#include <getopt.h>

#include <alsa/asoundlib.h>

#include "gstalsa.h"

#include "gstalsasink.h"

#include "gstalsadeviceprobe.h"

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

/* elementfactory information */

static const GstElementDetails gst_alsasink_details =

GST_ELEMENT_DETAILS ("Audio sink (ALSA)",

    "Sink/Audio",

    "Output to a sound card via ALSA",

    "Wim Taymans <wim@fluendo.com>");

#define DEFAULT_DEVICE		"default"

#define DEFAULT_DEVICE_NAME	""

#define SPDIF_PERIOD_SIZE 1536

#define SPDIF_BUFFER_SIZE 15360

enum

{

  PROP_0,

  PROP_DEVICE,

  PROP_DEVICE_NAME

};

static void gst_alsasink_init_interfaces (GType type);

GST_BOILERPLATE_FULL (GstAlsaSink, gst_alsasink, GstAudioSink,

    GST_TYPE_AUDIO_SINK, gst_alsasink_init_interfaces);

static void gst_alsasink_finalise (GObject * object);

static void gst_alsasink_set_property (GObject * object,

    guint prop_id, const GValue * value, GParamSpec * pspec);

static void gst_alsasink_get_property (GObject * object,

    guint prop_id, GValue * value, GParamSpec * pspec);

static GstCaps *gst_alsasink_getcaps (GstBaseSink * bsink);

static gboolean gst_alsasink_open (GstAudioSink * asink);

static gboolean gst_alsasink_prepare (GstAudioSink * asink,

    GstRingBufferSpec * spec);

static gboolean gst_alsasink_unprepare (GstAudioSink * asink);

static gboolean gst_alsasink_close (GstAudioSink * asink);

static guint gst_alsasink_write (GstAudioSink * asink, gpointer data,

    guint length);

static guint gst_alsasink_delay (GstAudioSink * asink);

static void gst_alsasink_reset (GstAudioSink * asink);

static gint output_ref;         /* 0    */

static snd_output_t *output;    /* NULL */

static GStaticMutex output_mutex = G_STATIC_MUTEX_INIT;

#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)

# define ALSA_SINK_FACTORY_ENDIANNESS	"LITTLE_ENDIAN, BIG_ENDIAN"

#else

# define ALSA_SINK_FACTORY_ENDIANNESS	"BIG_ENDIAN, LITTLE_ENDIAN"

#endif

static GstStaticPadTemplate alsasink_sink_factory =

    GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-raw-int, "

        "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

        "signed = (boolean) { TRUE, FALSE }, "

        "width = (int) 32, "

        "depth = (int) 32, "

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

        "audio/x-raw-int, "

        "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

        "signed = (boolean) { TRUE, FALSE }, "

        "width = (int) 24, "

        "depth = (int) 24, "

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

        "audio/x-raw-int, "

        "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

        "signed = (boolean) { TRUE, FALSE }, "

        "width = (int) 32, "

        "depth = (int) 24, "

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

        "audio/x-raw-int, "

        "endianness = (int) { " ALSA_SINK_FACTORY_ENDIANNESS " }, "

        "signed = (boolean) { TRUE, FALSE }, "

        "width = (int) 16, "

        "depth = (int) 16, "

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

        "audio/x-raw-int, "

        "signed = (boolean) { TRUE, FALSE }, "

        "width = (int) 8, "

        "depth = (int) 8, "

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

        "audio/x-iec958")

    );

static void

gst_alsasink_finalise (GObject * object)

{

  GstAlsaSink *sink = GST_ALSA_SINK (object);

  g_free (sink->device);

  g_mutex_free (sink->alsa_lock);

  g_static_mutex_lock (&output_mutex);

  --output_ref;

  if (output_ref == 0) {

    snd_output_close (output);

    output = NULL;

  }

  g_static_mutex_unlock (&output_mutex);

  G_OBJECT_CLASS (parent_class)->finalize (object);

}

static void

gst_alsasink_init_interfaces (GType type)

{

  gst_alsa_type_add_device_property_probe_interface (type);

}

static void

gst_alsasink_base_init (gpointer g_class)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

  gst_element_class_set_details (element_class, &gst_alsasink_details);

  gst_element_class_add_pad_template (element_class,

      gst_static_pad_template_get (&alsasink_sink_factory));

}

static void

gst_alsasink_class_init (GstAlsaSinkClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstBaseSinkClass *gstbasesink_class;

  GstBaseAudioSinkClass *gstbaseaudiosink_class;

  GstAudioSinkClass *gstaudiosink_class;

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstbasesink_class = (GstBaseSinkClass *) klass;

  gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;

  gstaudiosink_class = (GstAudioSinkClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasink_finalise);

  gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasink_get_property);

  gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasink_set_property);

  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasink_getcaps);

  gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_alsasink_open);

  gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasink_prepare);

  gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasink_unprepare);

  gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_alsasink_close);

  gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_alsasink_write);

  gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_alsasink_delay);

  gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_alsasink_reset);

  g_object_class_install_property (gobject_class, PROP_DEVICE,

      g_param_spec_string ("device", "Device",

          "ALSA device, as defined in an asound configuration file",

          DEFAULT_DEVICE, G_PARAM_READWRITE));

  g_object_class_install_property (gobject_class, PROP_DEVICE_NAME,

      g_param_spec_string ("device-name", "Device name",

          "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,

          G_PARAM_READABLE));

}

static void

gst_alsasink_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec)

{

  GstAlsaSink *sink;

  sink = GST_ALSA_SINK (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_free (sink->device);

      sink->device = g_value_dup_string (value);

      /* setting NULL restores the default device */

      if (sink->device == NULL) {

        sink->device = g_strdup (DEFAULT_DEVICE);

      }

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_alsasink_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec)

{

  GstAlsaSink *sink;

  sink = GST_ALSA_SINK (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_value_set_string (value, sink->device);

      break;

    case PROP_DEVICE_NAME:

      g_value_take_string (value,

          gst_alsa_find_device_name (GST_OBJECT_CAST (sink),

              sink->device, sink->handle, SND_PCM_STREAM_PLAYBACK));

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_alsasink_init (GstAlsaSink * alsasink, GstAlsaSinkClass * g_class)

{

  GST_DEBUG_OBJECT (alsasink, "initializing alsasink");

  alsasink->device = g_strdup (DEFAULT_DEVICE);

  alsasink->handle = NULL;

  alsasink->cached_caps = NULL;

  alsasink->alsa_lock = g_mutex_new ();

  g_static_mutex_lock (&output_mutex);

  if (output_ref == 0) {

    snd_output_stdio_attach (&output, stdout, 0);

    ++output_ref;

  }

  g_static_mutex_unlock (&output_mutex);

}

#define CHECK(call, error) \

G_STMT_START {                  \

if ((err = call) < 0)           \

  goto error;                   \

} G_STMT_END;

static GstCaps *

gst_alsasink_getcaps (GstBaseSink * bsink)

{

  GstElementClass *element_class;

  GstPadTemplate *pad_template;

  GstAlsaSink *sink = GST_ALSA_SINK (bsink);

  GstCaps *caps;

  if (sink->handle == NULL) {

    GST_DEBUG_OBJECT (sink, "device not open, using template caps");

    return NULL;                /* base class will get template caps for us */

  }

  if (sink->cached_caps) {

    GST_LOG_OBJECT (sink, "Returning cached caps");

    return gst_caps_ref (sink->cached_caps);

  }

  element_class = GST_ELEMENT_GET_CLASS (sink);

  pad_template = gst_element_class_get_pad_template (element_class, "sink");

  g_return_val_if_fail (pad_template != NULL, NULL);

  caps = gst_alsa_probe_supported_formats (GST_OBJECT (sink), sink->handle,

      gst_pad_template_get_caps (pad_template));

  if (caps) {

    sink->cached_caps = gst_caps_ref (caps);

  }

  GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, caps);

  return caps;

}

static int

set_hwparams (GstAlsaSink * alsa)

{

  guint rrate;

  gint err, dir;

  snd_pcm_hw_params_t *params;

  guint period_time, buffer_time;

  snd_pcm_hw_params_malloc (&params);

  GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "

      "SPDIF (%d)", alsa->channels, alsa->rate,

      snd_pcm_format_name (alsa->format), alsa->iec958);

  /* start with requested values, if we cannot configure alsa for those values,

   * we set these values to -1, which will leave the default alsa values */

  buffer_time = alsa->buffer_time;

  period_time = alsa->period_time;

retry:

  /* choose all parameters */

  CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);

  /* set the interleaved read/write format */

  CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),

      wrong_access);

  /* set the sample format */

  if (alsa->iec958) {

    /* Try to use big endian first else fallback to le and swap bytes */

    if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {

      alsa->format = SND_PCM_FORMAT_S16_LE;

      alsa->need_swap = TRUE;

      GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");

    } else {

      alsa->need_swap = FALSE;

    }

  }

  CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),

      no_sample_format);

  /* set the count of channels */

  CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),

      no_channels);

  /* set the stream rate */

  rrate = alsa->rate;

  CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),

      no_rate);

  if (rrate != alsa->rate)

    goto rate_match;

  /* get and dump some limits */

  {

    guint min, max;

    snd_pcm_hw_params_get_buffer_time_min (params, &min, &dir);

    snd_pcm_hw_params_get_buffer_time_max (params, &max, &dir);

    GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",

        alsa->buffer_time, min, max);

    snd_pcm_hw_params_get_period_time_min (params, &min, &dir);

    snd_pcm_hw_params_get_period_time_max (params, &max, &dir);

    GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",

        alsa->period_time, min, max);

    snd_pcm_hw_params_get_periods_min (params, &min, &dir);

    snd_pcm_hw_params_get_periods_max (params, &max, &dir);

    GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);

  }

  /* now try to configure the buffer time and period time, if one

   * of those fail, we fall back to the defaults and emit a warning. */

  if (buffer_time != -1 && !alsa->iec958) {

    /* set the buffer time */

    if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,

                &buffer_time, &dir)) < 0) {

      GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),

          ("Unable to set buffer time %i for playback: %s",

              buffer_time, snd_strerror (err)));

      /* disable buffer_time the next round */

      buffer_time = -1;

      goto retry;

    }

    GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);

  }

  if (period_time != -1 && !alsa->iec958) {

    /* set the period time */

    if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,

                &period_time, &dir)) < 0) {

      GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),

          ("Unable to set period time %i for playback: %s",

              period_time, snd_strerror (err)));

      /* disable period_time the next round */

      period_time = -1;

      goto retry;

    }

    GST_DEBUG_OBJECT (alsa, "period time %u", period_time);

  }

  /* Set buffer size and period size manually for SPDIF */

  if (G_UNLIKELY (alsa->iec958)) {

    snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;

    snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;

    CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,

            &buffer_size), buffer_size);

    CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,

            &period_size, NULL), period_size);

  }

  /* write the parameters to device */

  CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);

  /* now get the configured values */

  CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),

      buffer_size);

  CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir),

      period_size);

  GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,

      alsa->period_size);

  snd_pcm_hw_params_free (params);

  return 0;

  /* ERRORS */

no_config:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Broken configuration for playback: no configurations available: %s",

            snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

wrong_access:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Access type not available for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

no_sample_format:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Sample format not available for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

no_channels:

  {

    gchar *msg = NULL;

    if ((alsa->channels) == 1)

      msg = g_strdup (_("Could not open device for playback in mono mode."));

    if ((alsa->channels) == 2)

      msg = g_strdup (_("Could not open device for playback in stereo mode."));

    if ((alsa->channels) > 2)

      msg =

          g_strdup_printf (_

          ("Could not open device for playback in %d-channel mode."),

          alsa->channels);

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err)));

    g_free (msg);

    snd_pcm_hw_params_free (params);

    return err;

  }

no_rate:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Rate %iHz not available for playback: %s",

            alsa->rate, snd_strerror (err)));

    return err;

  }

rate_match:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err));

    snd_pcm_hw_params_free (params);

    return -EINVAL;

  }

buffer_size:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to get buffer size for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

period_size:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to get period size for playback: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

set_hw_params:

  {

    GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),

        ("Unable to set hw params for playback: %s", snd_strerror (err)));