/* GStreamer

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

 *

 * 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.

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <string.h>

#include <stdlib.h>

#include <gst/gst.h>

#include <gst/audio/audio.h>

#define GST_CAT_DEFAULT audio_rate_debug

GST_DEBUG_CATEGORY_STATIC (audio_rate_debug);

#define GST_TYPE_AUDIO_RATE \

  (gst_audio_rate_get_type())

#define GST_AUDIO_RATE(obj) \

  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RATE,GstAudioRate))

#define GST_AUDIO_RATE_CLASS(klass) \

  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RATE,GstAudioRate))

#define GST_IS_AUDIO_RATE(obj) \

  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RATE))

#define GST_IS_AUDIO_RATE_CLASS(klass) \

  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RATE))

typedef struct _GstAudioRate GstAudioRate;

typedef struct _GstAudioRateClass GstAudioRateClass;

struct _GstAudioRate

{

  GstElement element;

  GstPad *sinkpad, *srcpad;

  /* audio format */

  gint bytes_per_sample;

  gint rate;

  /* stats */

  guint64 in, out, add, drop;

  gboolean silent;

  /* audio state */

  guint64 next_offset;

  guint64 next_ts;

  gboolean discont;

  gboolean new_segment;

  /* we accept all formats on the sink */

  GstSegment sink_segment;

  /* we output TIME format on the src */

  GstSegment src_segment;

};

struct _GstAudioRateClass

{

  GstElementClass parent_class;

};

/* elementfactory information */

static const GstElementDetails audio_rate_details =

GST_ELEMENT_DETAILS ("Audio rate adjuster",

    "Filter/Effect/Audio",

    "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",

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

/* GstAudioRate signals and args */

enum

{

  /* FILL ME */

  LAST_SIGNAL

};

#define DEFAULT_SILENT  TRUE

enum

{

  ARG_0,

  ARG_IN,

  ARG_OUT,

  ARG_ADD,

  ARG_DROP,

  ARG_SILENT,

  /* FILL ME */

};

static GstStaticPadTemplate gst_audio_rate_src_template =

    GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"

        GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)

    );

static GstStaticPadTemplate gst_audio_rate_sink_template =

    GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"

        GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)

    );

static void gst_audio_rate_base_init (gpointer g_class);

static void gst_audio_rate_class_init (GstAudioRateClass * klass);

static void gst_audio_rate_init (GstAudioRate * audiorate);

static gboolean gst_audio_rate_sink_event (GstPad * pad, GstEvent * event);

static gboolean gst_audio_rate_src_event (GstPad * pad, GstEvent * event);

static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);

static void gst_audio_rate_set_property (GObject * object,

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

static void gst_audio_rate_get_property (GObject * object,

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

static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element,

    GstStateChange transition);

static GstElementClass *parent_class = NULL;

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

static GType

gst_audio_rate_get_type (void)

{

  static GType audio_rate_type = 0;

  if (!audio_rate_type) {

    static const GTypeInfo audio_rate_info = {

      sizeof (GstAudioRateClass),

      gst_audio_rate_base_init,

      NULL,

      (GClassInitFunc) gst_audio_rate_class_init,

      NULL,

      NULL,

      sizeof (GstAudioRate),

      0,

      (GInstanceInitFunc) gst_audio_rate_init,

    };

    audio_rate_type = g_type_register_static (GST_TYPE_ELEMENT,

        "GstAudioRate", &audio_rate_info, 0);

  }

  return audio_rate_type;

}

static void

gst_audio_rate_base_init (gpointer g_class)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

  gst_element_class_set_details (element_class, &audio_rate_details);

  gst_element_class_add_pad_template (element_class,

      gst_static_pad_template_get (&gst_audio_rate_sink_template));

  gst_element_class_add_pad_template (element_class,

      gst_static_pad_template_get (&gst_audio_rate_src_template));

}

static void

gst_audio_rate_class_init (GstAudioRateClass * klass)

{

  GObjectClass *object_class = G_OBJECT_CLASS (klass);

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  parent_class = g_type_class_peek_parent (klass);

  object_class->set_property = gst_audio_rate_set_property;

  object_class->get_property = gst_audio_rate_get_property;

  g_object_class_install_property (object_class, ARG_IN,

      g_param_spec_uint64 ("in", "In",

          "Number of input samples", 0, G_MAXUINT64, 0,

          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (object_class, ARG_OUT,

      g_param_spec_uint64 ("out", "Out", "Number of output samples", 0,

          G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (object_class, ARG_ADD,

      g_param_spec_uint64 ("add", "Add", "Number of added samples", 0,

          G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (object_class, ARG_DROP,

      g_param_spec_uint64 ("drop", "Drop", "Number of dropped samples", 0,

          G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (object_class, ARG_SILENT,

      g_param_spec_boolean ("silent", "silent",

          "Don't emit notify for dropped and duplicated frames", DEFAULT_SILENT,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  element_class->change_state = gst_audio_rate_change_state;

}

static void

gst_audio_rate_reset (GstAudioRate * audiorate)

{

  audiorate->next_offset = -1;

  audiorate->next_ts = -1;

  audiorate->discont = TRUE;

  gst_segment_init (&audiorate->sink_segment, GST_FORMAT_UNDEFINED);

  gst_segment_init (&audiorate->src_segment, GST_FORMAT_TIME);

  GST_DEBUG_OBJECT (audiorate, "handle reset");

}

static gboolean

gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)

{

  GstAudioRate *audiorate;

  GstStructure *structure;

  GstPad *otherpad;

  gboolean ret = FALSE;

  gint channels, width, rate;

  audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));

  structure = gst_caps_get_structure (caps, 0);

  if (!gst_structure_get_int (structure, "channels", &channels))

    goto wrong_caps;

  if (!gst_structure_get_int (structure, "width", &width))

    goto wrong_caps;

  if (!gst_structure_get_int (structure, "rate", &rate))

    goto wrong_caps;

  audiorate->bytes_per_sample = channels * (width / 8);

  if (audiorate->bytes_per_sample == 0)

    goto wrong_format;

  audiorate->rate = rate;

  /* the format is correct, configure caps on other pad */

  otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :

      audiorate->srcpad;

  ret = gst_pad_set_caps (otherpad, caps);

done:

  gst_object_unref (audiorate);

  return ret;

  /* ERRORS */

wrong_caps:

  {

    GST_DEBUG_OBJECT (audiorate, "could not get channels/width from caps");

    goto done;

  }

wrong_format:

  {

    GST_DEBUG_OBJECT (audiorate, "bytes_per_samples gave 0");

    goto done;

  }

}

static void

gst_audio_rate_init (GstAudioRate * audiorate)

{

  audiorate->sinkpad =

      gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink");

  gst_pad_set_event_function (audiorate->sinkpad, gst_audio_rate_sink_event);

  gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain);

  gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_setcaps);

  gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);

  gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);

  audiorate->srcpad =

      gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src");

  gst_pad_set_event_function (audiorate->srcpad, gst_audio_rate_src_event);

  gst_pad_set_setcaps_function (audiorate->srcpad, gst_audio_rate_setcaps);

  gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);

  gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);

  audiorate->in = 0;

  audiorate->out = 0;

  audiorate->drop = 0;

  audiorate->add = 0;

  audiorate->silent = DEFAULT_SILENT;

}

static gboolean

gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)

{

  gboolean res;

  GstAudioRate *audiorate;

  audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));

  switch (GST_EVENT_TYPE (event)) {

    case GST_EVENT_FLUSH_STOP:

      GST_DEBUG_OBJECT (audiorate, "handling FLUSH_STOP");

      gst_audio_rate_reset (audiorate);

      res = gst_pad_push_event (audiorate->srcpad, event);

      break;

    case GST_EVENT_NEWSEGMENT:

    {

      GstFormat format;

      gdouble rate, arate;

      gint64 start, stop, time;

      gboolean update;

      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,

          &start, &stop, &time);

      GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");

      /* FIXME:

       * - sparse stream support. For this, the update flag is TRUE and the

       *   start/time positions are updated, meaning that time progressed by

       *   time - old_time amount and we need to fill that gap with empty

       *   samples.

       * - fill the current segment if it has a valid stop position. This

       *   happens when the update flag is FALSE. With the segment helper we can

       *   calculate the accumulated time and compare this to the next_offset.

       */

      if (!update) {

        /* a new segment starts. We need to figure out what will be the next

         * sample offset. We mark the offsets as invalid so that the _chain

         * function will perform this calculation. */

        audiorate->next_offset = -1;

        audiorate->next_ts = -1;

      }

      /* we accept all formats */

      gst_segment_set_newsegment_full (&audiorate->sink_segment, update, rate,

          arate, format, start, stop, time);

#ifndef __SYMBIAN32__

      GST_DEBUG_OBJECT (audiorate, "updated segment: %" GST_SEGMENT_FORMAT,

          &audiorate->sink_segment);

#endif

      if (format == GST_FORMAT_TIME) {

        /* TIME formats can be copied to src and forwarded */

        res = gst_pad_push_event (audiorate->srcpad, event);

        memcpy (&audiorate->src_segment, &audiorate->sink_segment,

            sizeof (GstSegment));

      } else {

        /* other formats will be handled in the _chain function */

        gst_event_unref (event);

        res = TRUE;

      }

      break;

    }

    case GST_EVENT_EOS:

      /* FIXME, fill last segment */

      res = gst_pad_push_event (audiorate->srcpad, event);

      break;

    default:

      res = gst_pad_push_event (audiorate->srcpad, event);

      break;

  }

  gst_object_unref (audiorate);

  return res;

}

static gboolean

gst_audio_rate_src_event (GstPad * pad, GstEvent * event)

{

  gboolean res;

  GstAudioRate *audiorate;

  audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));

  switch (GST_EVENT_TYPE (event)) {

    default:

      res = gst_pad_push_event (audiorate->sinkpad, event);

      break;

  }

  gst_object_unref (audiorate);

  return res;

}

static gboolean

gst_audio_rate_convert (GstAudioRate * audiorate,

    GstFormat src_fmt, gint64 src_val, GstFormat dest_fmt, gint64 * dest_val)

{

  if (src_fmt == dest_fmt) {

    *dest_val = src_val;

    return TRUE;

  }

  switch (src_fmt) {

    case GST_FORMAT_DEFAULT:

      switch (dest_fmt) {

        case GST_FORMAT_BYTES:

          *dest_val = src_val * audiorate->bytes_per_sample;

          break;

        case GST_FORMAT_TIME:

          *dest_val =

              gst_util_uint64_scale_int (src_val, GST_SECOND, audiorate->rate);

          break;

        default:

          return FALSE;;

      }

      break;

    case GST_FORMAT_BYTES:

      switch (dest_fmt) {

        case GST_FORMAT_DEFAULT:

          *dest_val = src_val / audiorate->bytes_per_sample;

          break;

        case GST_FORMAT_TIME:

          *dest_val = gst_util_uint64_scale_int (src_val, GST_SECOND,

              audiorate->rate * audiorate->bytes_per_sample);

          break;

        default:

          return FALSE;;

      }

      break;

    case GST_FORMAT_TIME:

      switch (dest_fmt) {

        case GST_FORMAT_BYTES:

          *dest_val = gst_util_uint64_scale_int (src_val,

              audiorate->rate * audiorate->bytes_per_sample, GST_SECOND);

          break;

        case GST_FORMAT_DEFAULT:

          *dest_val =

              gst_util_uint64_scale_int (src_val, audiorate->rate, GST_SECOND);

          break;

        default:

          return FALSE;;

      }

      break;

    default:

      return FALSE;

  }

  return TRUE;

}

static gboolean

gst_audio_rate_convert_segments (GstAudioRate * audiorate)

{

  GstFormat src_fmt, dst_fmt;

  src_fmt = audiorate->sink_segment.format;

  dst_fmt = audiorate->src_segment.format;

#define CONVERT_VAL(field) gst_audio_rate_convert (audiorate, \

		src_fmt, audiorate->sink_segment.field,       \

		dst_fmt, &audiorate->src_segment.field);

  audiorate->sink_segment.rate = audiorate->src_segment.rate;

  audiorate->sink_segment.abs_rate = audiorate->src_segment.abs_rate;

  audiorate->sink_segment.flags = audiorate->src_segment.flags;

  audiorate->sink_segment.applied_rate = audiorate->src_segment.applied_rate;

  CONVERT_VAL (start);

  CONVERT_VAL (stop);

  CONVERT_VAL (time);

  CONVERT_VAL (accum);

  CONVERT_VAL (last_stop);

#undef CONVERT_VAL

  return TRUE;

}

static GstFlowReturn

gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)

{

  GstAudioRate *audiorate;

  GstClockTime in_time, in_duration, in_stop, run_time;

  guint64 in_offset, in_offset_end, in_samples;

  guint in_size;

  GstFlowReturn ret = GST_FLOW_OK;

  audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));

  /* need to be negotiated now */

  if (audiorate->bytes_per_sample == 0)

    goto not_negotiated;

  /* we have a new pending segment */

  if (audiorate->next_offset == -1) {

    gint64 pos;

    /* update the TIME segment */

    gst_audio_rate_convert_segments (audiorate);

    /* first buffer, we are negotiated and we have a segment, calculate the

     * current expected offsets based on the segment.start, which is the first

     * media time of the segment and should match the media time of the first

     * buffer in that segment, which is the offset expressed in DEFAULT units.

     */

    /* convert first timestamp of segment to sample position */

    pos = gst_util_uint64_scale_int (audiorate->src_segment.start,

        audiorate->rate, GST_SECOND);

    GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);

    audiorate->next_offset = pos;

    audiorate->next_ts = gst_util_uint64_scale_int (audiorate->next_offset,

        GST_SECOND, audiorate->rate);

  }

  audiorate->in++;

  in_time = GST_BUFFER_TIMESTAMP (buf);

  if (in_time == GST_CLOCK_TIME_NONE) {

    GST_DEBUG_OBJECT (audiorate, "no timestamp, using expected next time");

    in_time = audiorate->next_ts;

  }

  in_size = GST_BUFFER_SIZE (buf);

  in_samples = in_size / audiorate->bytes_per_sample;

  /* get duration from the size because we can and it's more accurate */

  in_duration =

      gst_util_uint64_scale_int (in_samples, GST_SECOND, audiorate->rate);

  in_stop = in_time + in_duration;

  /* Figure out the total accumulated segment time. */