/* GStreamer

 * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>

 *

 * 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:gstbasertpaudiopayload

 * @short_description: Base class for audio RTP payloader

 *

 * <refsect2>

 * <para>

 * Provides a base class for audio RTP payloaders for frame or sample based

 * audio codecs (constant bitrate)

 * </para>

 * <para>

 * This class derives from GstBaseRTPPayload. It can be used for payloading

 * audio codecs. It will only work with constant bitrate codecs. It supports

 * both frame based and sample based codecs. It takes care of packing up the

 * audio data into RTP packets and filling up the headers accordingly. The

 * payloading is done based on the maximum MTU (mtu) and the maximum time per

 * packet (max-ptime). The general idea is to divide large data buffers into

 * smaller RTP packets. The RTP packet size is the minimum of either the MTU,

 * max-ptime (if set) or available data. The RTP packet size is always larger or

 * equal to min-ptime (if set). If min-ptime is not set, any residual data is

 * sent in a last RTP packet. In the case of frame based codecs, the resulting

 * RTP packets always contain full frames.

 * </para>

 * <title>Usage</title>

 * <para>

 * To use this base class, your child element needs to call either

 * gst_base_rtp_audio_payload_set_frame_based() or

 * gst_base_rtp_audio_payload_set_sample_based(). This is usually done in the

 * element's _init() function. Then, the child element must call either

 * gst_base_rtp_audio_payload_set_frame_options(),

 * gst_base_rtp_audio_payload_set_sample_options() or

 * gst_base_rtp_audio_payload_set_samplebits_options. Since

 * GstBaseRTPAudioPayload derives from GstBaseRTPPayload, the child element

 * must set any variables or call/override any functions required by that base

 * class. The child element does not need to override any other functions

 * specific to GstBaseRTPAudioPayload.

 * </para>

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <stdlib.h>

#include <string.h>

#include <gst/rtp/gstrtpbuffer.h>

#include <gst/base/gstadapter.h>

#include "gstbasertpaudiopayload.h"

GST_DEBUG_CATEGORY_STATIC (basertpaudiopayload_debug);

#define GST_CAT_DEFAULT (basertpaudiopayload_debug)

typedef enum

{

  AUDIO_CODEC_TYPE_NONE,

  AUDIO_CODEC_TYPE_FRAME_BASED,

  AUDIO_CODEC_TYPE_SAMPLE_BASED

} AudioCodecType;

struct _GstBaseRTPAudioPayloadPrivate

{

  AudioCodecType type;

  GstAdapter *adapter;

  guint64 min_ptime;

};

#define GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE(o) \

  (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_BASE_RTP_AUDIO_PAYLOAD, \

                                GstBaseRTPAudioPayloadPrivate))

static void gst_base_rtp_audio_payload_finalize (GObject * object);

static GstFlowReturn gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload

    * payload, GstBuffer * buffer);

static GstFlowReturn

gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *

    basepayload, GstBuffer * buffer);

static GstFlowReturn

gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *

    basepayload, GstBuffer * buffer);

static GstStateChangeReturn

gst_base_rtp_payload_audio_change_state (GstElement * element,

    GstStateChange transition);

static gboolean

gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event);

GST_BOILERPLATE (GstBaseRTPAudioPayload, gst_base_rtp_audio_payload,

    GstBaseRTPPayload, GST_TYPE_BASE_RTP_PAYLOAD);

static void

gst_base_rtp_audio_payload_base_init (gpointer klass)

{

}

static void

gst_base_rtp_audio_payload_class_init (GstBaseRTPAudioPayloadClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstBaseRTPPayloadClass *gstbasertppayload_class;

  g_type_class_add_private (klass, sizeof (GstBaseRTPAudioPayloadPrivate));

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;

  gobject_class->finalize =

      GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_finalize);

  gstelement_class->change_state =

      GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_change_state);

  gstbasertppayload_class->handle_buffer =

      GST_DEBUG_FUNCPTR (gst_base_rtp_audio_payload_handle_buffer);

  gstbasertppayload_class->handle_event =

      GST_DEBUG_FUNCPTR (gst_base_rtp_payload_audio_handle_event);

  GST_DEBUG_CATEGORY_INIT (basertpaudiopayload_debug, "basertpaudiopayload", 0,

      "base audio RTP payloader");

}

static void

gst_base_rtp_audio_payload_init (GstBaseRTPAudioPayload * basertpaudiopayload,

    GstBaseRTPAudioPayloadClass * klass)

{

  basertpaudiopayload->priv =

      GST_BASE_RTP_AUDIO_PAYLOAD_GET_PRIVATE (basertpaudiopayload);

  basertpaudiopayload->base_ts = 0;

  basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_NONE;

  /* these need to be set by child object if frame based */

  basertpaudiopayload->frame_size = 0;

  basertpaudiopayload->frame_duration = 0;

  /* these need to be set by child object if sample based */

  basertpaudiopayload->sample_size = 0;

  basertpaudiopayload->priv->adapter = gst_adapter_new ();

}

static void

gst_base_rtp_audio_payload_finalize (GObject * object)

{

  GstBaseRTPAudioPayload *basertpaudiopayload;

  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (object);

  g_object_unref (basertpaudiopayload->priv->adapter);

  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));

}

/**

 * gst_base_rtp_audio_payload_set_frame_based:

 * @basertpaudiopayload: a pointer to the element.

 *

 * Tells #GstBaseRTPAudioPayload that the child element is for a frame based

 * audio codec

 *

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *

    basertpaudiopayload)

{

  g_return_if_fail (basertpaudiopayload != NULL);

  g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);

  basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_FRAME_BASED;

}

/**

 * gst_base_rtp_audio_payload_set_sample_based:

 * @basertpaudiopayload: a pointer to the element.

 *

 * Tells #GstBaseRTPAudioPayload that the child element is for a sample based

 * audio codec

 *

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *

    basertpaudiopayload)

{

  g_return_if_fail (basertpaudiopayload != NULL);

  g_return_if_fail (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_NONE);

  basertpaudiopayload->priv->type = AUDIO_CODEC_TYPE_SAMPLE_BASED;

}

/**

 * gst_base_rtp_audio_payload_set_frame_options:

 * @basertpaudiopayload: a pointer to the element.

 * @frame_duration: The duraction of an audio frame in milliseconds.

 * @frame_size: The size of an audio frame in bytes.

 *

 * Sets the options for frame based audio codecs.

 *

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload

    * basertpaudiopayload, gint frame_duration, gint frame_size)

{

  g_return_if_fail (basertpaudiopayload != NULL);

  basertpaudiopayload->frame_size = frame_size;

  basertpaudiopayload->frame_duration = frame_duration;

  if (basertpaudiopayload->priv->adapter) {

    gst_adapter_clear (basertpaudiopayload->priv->adapter);

  }

}

/**

 * gst_base_rtp_audio_payload_set_sample_options:

 * @basertpaudiopayload: a pointer to the element.

 * @sample_size: Size per sample in bytes.

 *

 * Sets the options for sample based audio codecs.

 *

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload

    * basertpaudiopayload, gint sample_size)

{

  g_return_if_fail (basertpaudiopayload != NULL);

  /* sample_size is in bits internally */

  basertpaudiopayload->sample_size = sample_size * 8;

  if (basertpaudiopayload->priv->adapter) {

    gst_adapter_clear (basertpaudiopayload->priv->adapter);

  }

}

/**

 * gst_base_rtp_audio_payload_set_samplebits_options:

 * @basertpaudiopayload: a pointer to the element.

 * @sample_size: Size per sample in bits.

 *

 * Sets the options for sample based audio codecs.

 *

 * Since: 0.10.18

 */

#ifdef __SYMBIAN32__

EXPORT_C

#endif

void

gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload

    * basertpaudiopayload, gint sample_size)

{

  g_return_if_fail (basertpaudiopayload != NULL);

  basertpaudiopayload->sample_size = sample_size;

  if (basertpaudiopayload->priv->adapter) {

    gst_adapter_clear (basertpaudiopayload->priv->adapter);

  }

}

static GstFlowReturn

gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload * basepayload,

    GstBuffer * buffer)

{

  GstFlowReturn ret;

  GstBaseRTPAudioPayload *basertpaudiopayload;

  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);

  ret = GST_FLOW_ERROR;

  if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_FRAME_BASED) {

    ret = gst_base_rtp_audio_payload_handle_frame_based_buffer (basepayload,

        buffer);

  } else if (basertpaudiopayload->priv->type == AUDIO_CODEC_TYPE_SAMPLE_BASED) {

    ret = gst_base_rtp_audio_payload_handle_sample_based_buffer (basepayload,

        buffer);

  } else {

    GST_DEBUG_OBJECT (basertpaudiopayload, "Audio codec type not set");

  }

  return ret;

}

/* this assumes all frames have a constant duration and a constant size */

static GstFlowReturn

gst_base_rtp_audio_payload_handle_frame_based_buffer (GstBaseRTPPayload *

    basepayload, GstBuffer * buffer)

{

  GstBaseRTPAudioPayload *basertpaudiopayload;

  guint payload_len;

  const guint8 *data = NULL;

  GstFlowReturn ret;

  guint available;

  gint frame_size, frame_duration;

  guint maxptime_octets = G_MAXUINT;

  guint minptime_octets = 0;

  guint min_payload_len;

  guint max_payload_len;

  gboolean use_adapter = FALSE;

  guint minptime_ms;

  ret = GST_FLOW_OK;

  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);

  if (basertpaudiopayload->frame_size == 0 ||

      basertpaudiopayload->frame_duration == 0) {

    GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");

    gst_buffer_unref (buffer);

    return GST_FLOW_ERROR;

  }

  frame_size = basertpaudiopayload->frame_size;

  frame_duration = basertpaudiopayload->frame_duration;

  /* max number of bytes based on given ptime, has to be multiple of

   * frame_duration */

  if (basepayload->max_ptime != -1) {

    guint ptime_ms = basepayload->max_ptime / 1000000;

    maxptime_octets = frame_size * (int) (ptime_ms / frame_duration);

    if (maxptime_octets == 0) {

      GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %d is smaller than"

          " minimum %d ms, overwriting to minimum", ptime_ms, frame_duration);

      maxptime_octets = frame_size;

    }

  }

  max_payload_len = MIN (

      /* MTU max */

      (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU

              (basertpaudiopayload), 0, 0) / frame_size) * frame_size,

      /* ptime max */

      maxptime_octets);

  /* min number of bytes based on a given ptime, has to be a multiple

     of frame duration */

  minptime_ms = basepayload->min_ptime / 1000000;

  minptime_octets = frame_size * (int) (minptime_ms / frame_duration);

  min_payload_len = MAX (minptime_octets, frame_size);

  if (min_payload_len > max_payload_len) {

    min_payload_len = max_payload_len;

  }

  GST_DEBUG_OBJECT (basertpaudiopayload,

      "Calculated min_payload_len %u and max_payload_len %u",

      min_payload_len, max_payload_len);

  if (basertpaudiopayload->priv->adapter &&

      gst_adapter_available (basertpaudiopayload->priv->adapter)) {

    /* If there is always data in the adapter, we have to use it */

    gst_adapter_push (basertpaudiopayload->priv->adapter, buffer);

    available = gst_adapter_available (basertpaudiopayload->priv->adapter);

    use_adapter = TRUE;

  } else {

    /* let's set the base timestamp */

    basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buffer);

    /* If buffer fits on an RTP packet, let's just push it through */

    /* this will check against max_ptime and max_mtu */

    if (GST_BUFFER_SIZE (buffer) >= min_payload_len &&

        GST_BUFFER_SIZE (buffer) <= max_payload_len) {

      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,

          GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer),

          GST_BUFFER_TIMESTAMP (buffer));

      gst_buffer_unref (buffer);

      return ret;

    }

    available = GST_BUFFER_SIZE (buffer);

    data = (guint8 *) GST_BUFFER_DATA (buffer);

  }

  /* as long as we have full frames */

  while (available >= min_payload_len) {

    gfloat ts_inc;

    /* We send as much as we can */

    payload_len = MIN (max_payload_len, (available / frame_size) * frame_size);

    if (use_adapter) {

      data = gst_adapter_peek (basertpaudiopayload->priv->adapter, payload_len);

    }

    ret =

        gst_base_rtp_audio_payload_push (basertpaudiopayload, data, payload_len,

        basertpaudiopayload->base_ts);

    ts_inc = (payload_len * frame_duration) / frame_size;

    ts_inc = ts_inc * GST_MSECOND;

    basertpaudiopayload->base_ts += gst_gdouble_to_guint64 (ts_inc);

    if (use_adapter) {

      gst_adapter_flush (basertpaudiopayload->priv->adapter, payload_len);

      available = gst_adapter_available (basertpaudiopayload->priv->adapter);

    } else {

      available -= payload_len;

      data += payload_len;

    }

  }

  if (!use_adapter) {

    if (available != 0 && basertpaudiopayload->priv->adapter) {

      GstBuffer *buf;

      buf = gst_buffer_create_sub (buffer,

          GST_BUFFER_SIZE (buffer) - available, available);

      gst_adapter_push (basertpaudiopayload->priv->adapter, buf);

    }

    gst_buffer_unref (buffer);

  }

  return ret;

}

static GstFlowReturn

gst_base_rtp_audio_payload_handle_sample_based_buffer (GstBaseRTPPayload *

    basepayload, GstBuffer * buffer)

{

  GstBaseRTPAudioPayload *basertpaudiopayload;

  guint payload_len;

  const guint8 *data = NULL;

  GstFlowReturn ret;

  guint available;

  guint maxptime_octets = G_MAXUINT;

  guint minptime_octets = 0;

  guint min_payload_len;

  guint max_payload_len;

  gboolean use_adapter = FALSE;

  guint fragment_size;

  ret = GST_FLOW_OK;

  basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);

  if (basertpaudiopayload->sample_size == 0) {

    GST_DEBUG_OBJECT (basertpaudiopayload, "Required options not set");

    gst_buffer_unref (buffer);

    return GST_FLOW_ERROR;

  }

  /* sample_size is in bits and is converted into multiple bytes */

  fragment_size = basertpaudiopayload->sample_size;

  while ((fragment_size % 8) != 0)

    fragment_size += fragment_size;

  fragment_size /= 8;

  /* max number of bytes based on given ptime */

  if (basepayload->max_ptime != -1) {

    maxptime_octets = 8 * basepayload->max_ptime * basepayload->clock_rate /

        (basertpaudiopayload->sample_size * GST_SECOND);

  }

  max_payload_len = MIN (

      /* MTU max */

      gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU

          (basertpaudiopayload), 0, 0),

      /* ptime max */

      maxptime_octets);

  /* min number of bytes based on a given ptime, has to be a multiple

     of sample rate */

  minptime_octets = 8 * basepayload->min_ptime * basepayload->clock_rate /

      (basertpaudiopayload->sample_size * GST_SECOND);

  min_payload_len = MAX (minptime_octets, fragment_size);

  if (min_payload_len > max_payload_len) {

    min_payload_len = max_payload_len;

  }

  GST_DEBUG_OBJECT (basertpaudiopayload,