/* GStreamer

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

 *

 * gstalsasrc.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-alsasrc

 * @short_description: capture audio from an alsa device

 * @see_also: alsasink, alsamixer

 *

 * <refsect2>

 * <para>

 * This element reads data from an audio card using the ALSA API.

 * </para>

 * <title>Example pipelines</title>

 * <para>

 * Record from a sound card using ALSA and encode to Ogg/Vorbis.

 * </para>

 * <programlisting>

 * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg

 * </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 "gstalsasrc.h"

#include "gstalsadeviceprobe.h"

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

/* elementfactory information */

static const GstElementDetails gst_alsasrc_details =

GST_ELEMENT_DETAILS ("Audio source (ALSA)",

    "Source/Audio",

    "Read from a sound card via ALSA",

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

#define DEFAULT_PROP_DEVICE		"default"

#define DEFAULT_PROP_DEVICE_NAME	""

enum

{

  PROP_0,

  PROP_DEVICE,

  PROP_DEVICE_NAME,

};

static void gst_alsasrc_init_interfaces (GType type);

GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc,

    GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces);

GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer);

static void gst_alsasrc_finalize (GObject * object);

static void gst_alsasrc_set_property (GObject * object,

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

static void gst_alsasrc_get_property (GObject * object,

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

static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc);

static gboolean gst_alsasrc_open (GstAudioSrc * asrc);

static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc,

    GstRingBufferSpec * spec);

static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc);

static gboolean gst_alsasrc_close (GstAudioSrc * asrc);

static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length);

static guint gst_alsasrc_delay (GstAudioSrc * asrc);

static void gst_alsasrc_reset (GstAudioSrc * asrc);

/* AlsaSrc signals and args */

enum

{

  LAST_SIGNAL

};

#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)

# define ALSA_SRC_FACTORY_ENDIANNESS   "LITTLE_ENDIAN, BIG_ENDIAN"

#else

# define ALSA_SRC_FACTORY_ENDIANNESS   "BIG_ENDIAN, LITTLE_ENDIAN"

#endif

static GstStaticPadTemplate alsasrc_src_factory =

    GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

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

        "endianness = (int) { " ALSA_SRC_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_SRC_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_SRC_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_SRC_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 ]")

    );

static void

gst_alsasrc_finalize (GObject * object)

{

  GstAlsaSrc *src = GST_ALSA_SRC (object);

  g_free (src->device);

  g_mutex_free (src->alsa_lock);

  G_OBJECT_CLASS (parent_class)->finalize (object);

}

static gboolean

gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type)

{

  /* only support this one interface (wrapped by GstImplementsInterface) */

  g_assert (interface_type == GST_TYPE_MIXER);

  return gst_alsasrc_mixer_supported (this, interface_type);

}

static void

gst_implements_interface_init (GstImplementsInterfaceClass * klass)

{

  klass->supported = (gpointer) gst_alsasrc_interface_supported;

}

static void

gst_alsasrc_init_interfaces (GType type)

{

  static const GInterfaceInfo implements_iface_info = {

    (GInterfaceInitFunc) gst_implements_interface_init,

    NULL,

    NULL,

  };

  static const GInterfaceInfo mixer_iface_info = {

    (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init,

    NULL,

    NULL,

  };

  g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,

      &implements_iface_info);

  g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);

  gst_alsa_type_add_device_property_probe_interface (type);

}

static void

gst_alsasrc_base_init (gpointer g_class)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

  gst_element_class_set_details (element_class, &gst_alsasrc_details);

  gst_element_class_add_pad_template (element_class,

      gst_static_pad_template_get (&alsasrc_src_factory));

}

static void

gst_alsasrc_class_init (GstAlsaSrcClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstBaseSrcClass *gstbasesrc_class;

  GstBaseAudioSrcClass *gstbaseaudiosrc_class;

  GstAudioSrcClass *gstaudiosrc_class;

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstbasesrc_class = (GstBaseSrcClass *) klass;

  gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;

  gstaudiosrc_class = (GstAudioSrcClass *) klass;

  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasrc_finalize);

  gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasrc_get_property);

  gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasrc_set_property);

  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps);

  gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open);

  gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare);

  gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare);

  gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close);

  gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read);

  gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay);

  gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_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_PROP_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_PROP_DEVICE_NAME, G_PARAM_READABLE));

}

static void

gst_alsasrc_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec)

{

  GstAlsaSrc *src;

  src = GST_ALSA_SRC (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_free (src->device);

      src->device = g_value_dup_string (value);

      if (src->device == NULL) {

        src->device = g_strdup (DEFAULT_PROP_DEVICE);

      }

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_alsasrc_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec)

{

  GstAlsaSrc *src;

  src = GST_ALSA_SRC (object);

  switch (prop_id) {

    case PROP_DEVICE:

      g_value_set_string (value, src->device);

      break;

    case PROP_DEVICE_NAME:

      g_value_take_string (value,

          gst_alsa_find_device_name (GST_OBJECT_CAST (src),

              src->device, src->handle, SND_PCM_STREAM_CAPTURE));

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class)

{

  GST_DEBUG_OBJECT (alsasrc, "initializing");

  alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE);

  alsasrc->cached_caps = NULL;

  alsasrc->alsa_lock = g_mutex_new ();

}

#define CHECK(call, error) \

G_STMT_START {                  \

if ((err = call) < 0)           \

  goto error;                   \

} G_STMT_END;

static GstCaps *

gst_alsasrc_getcaps (GstBaseSrc * bsrc)

{

  GstElementClass *element_class;

  GstPadTemplate *pad_template;

  GstAlsaSrc *src;

  GstCaps *caps;

  src = GST_ALSA_SRC (bsrc);

  if (src->handle == NULL) {

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

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

  }

  if (src->cached_caps) {

    GST_LOG_OBJECT (src, "Returning cached caps");

    return gst_caps_ref (src->cached_caps);

  }

  element_class = GST_ELEMENT_GET_CLASS (src);

  pad_template = gst_element_class_get_pad_template (element_class, "src");

  g_return_val_if_fail (pad_template != NULL, NULL);

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

      gst_pad_template_get_caps (pad_template));

  if (caps) {

    src->cached_caps = gst_caps_ref (caps);

  }

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

  return caps;

}

static int

set_hwparams (GstAlsaSrc * alsa)

{

  guint rrate;

  gint err, dir;

  snd_pcm_hw_params_t *params;

  snd_pcm_hw_params_malloc (&params);

  /* 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 */

  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;

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

    /* set the buffer time */

    CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,

            &alsa->buffer_time, &dir), buffer_time);

  }

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

    /* set the period time */

    CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params,

            &alsa->period_time, &dir), period_time);

  }

  /* write the parameters to device */

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

  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);

  snd_pcm_hw_params_free (params);

  return 0;

  /* ERRORS */

no_config:

  {

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

        ("Broken configuration for recording: 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 recording: %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 recording: %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 recording in mono mode."));

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

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

    if ((alsa->channels) > 2)

      msg =

          g_strdup_printf (_

          ("Could not open device for recording 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 recording: %s",

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

    snd_pcm_hw_params_free (params);

    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_time:

  {

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

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

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

    snd_pcm_hw_params_free (params);

    return err;

  }

buffer_size:

  {

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

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

    snd_pcm_hw_params_free (params);

    return err;

  }

period_time:

  {

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

        ("Unable to set period time %i for recording: %s", alsa->period_time,

            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 recording: %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 recording: %s", snd_strerror (err)));

    snd_pcm_hw_params_free (params);

    return err;

  }

}

static int

set_swparams (GstAlsaSrc * alsa)

{

  int err;

  snd_pcm_sw_params_t *params;

  snd_pcm_sw_params_malloc (&params);

  /* get the current swparams */

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

  /* allow the transfer when at least period_size samples can be processed */

  CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params,

          alsa->period_size), set_avail);

  /* start the transfer on first read */

  CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params,

          0), start_threshold);

#if GST_CHECK_ALSA_VERSION(1,0,16)

  /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */

#else

  /* align all transfers to 1 sample */

  CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align);

#endif

  /* write the parameters to the recording device */

  CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params);

  snd_pcm_sw_params_free (params);

  return 0;

  /* ERRORS */

no_config:

  {

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

        ("Unable to determine current swparams for playback: %s",

            snd_strerror (err)));

    snd_pcm_sw_params_free (params);

    return err;

  }

start_threshold: