/* 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., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

/**

 * SECTION:element-audiorate

 * @title: audiorate

 * @see_also: #GstVideoRate

 *

 * This element takes an incoming stream of timestamped raw audio frames and

 * produces a perfect stream by inserting or dropping samples as needed.

 *

 * This operation may be of use to link to elements that require or otherwise

 * implicitly assume a perfect stream as they do not store timestamps,

 * but derive this by some means (e.g. bitrate for some AVI cases).

 *

 * The properties #GstAudioRate:in, #GstAudioRate:out, #GstAudioRate:add

 * and #GstAudioRate:drop can be read to obtain information about number of

 * input samples, output samples, dropped samples (i.e. the number of unused

 * input samples) and inserted samples (i.e. the number of samples added to

 * stream).

 *

 * When the #GstAudioRate:silent property is set to FALSE, a GObject property

 * notification will be emitted whenever one of the #GstAudioRate:add or

 * #GstAudioRate:drop values changes.

 * This can potentially cause performance degradation.

 * Note that property notification will happen from the streaming thread, so

 * applications should be prepared for this.

 *

 * If the #GstAudioRate:tolerance property is non-zero, and an incoming buffer's

 * timestamp deviates less than the property indicates from what would make a

 * 'perfect time', then no samples will be added or dropped.

 * Note that the output is still guaranteed to be a perfect stream, which means

 * that the incoming data is then simply shifted (by less than the indicated

 * tolerance) to a perfect time.

 *

 * ## Example pipelines

 * |[

 * gst-launch-1.0 -v autoaudiosrc ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav

 * ]|

 *  Capture audio from the sound card and turn it into a perfect stream

 * for saving in a raw audio file.

 * |[

 * gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav

 * ]|

 *  Decodes an audio file and transforms it into a perfect stream for saving

 * in a raw audio WAV file. Without the audio rate, the timing might not be

 * preserved correctly in the WAV file in case the decoded stream is jittery

 * or there are samples missing.

 *

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <string.h>

#include <stdlib.h>

#include "gstaudiorate.h"

#define GST_CAT_DEFAULT audio_rate_debug

GST_DEBUG_CATEGORY_STATIC (audio_rate_debug);

/* GstAudioRate signals and args */

enum

{

  /* FILL ME */

  LAST_SIGNAL

};

#define DEFAULT_SILENT     TRUE

#define DEFAULT_TOLERANCE  (40 * GST_MSECOND)

#define DEFAULT_SKIP_TO_FIRST FALSE

enum

{

  PROP_0,

  PROP_IN,

  PROP_OUT,

  PROP_ADD,

  PROP_DROP,

  PROP_SILENT,

  PROP_TOLERANCE,

  PROP_SKIP_TO_FIRST

};

static GstStaticPadTemplate gst_audio_rate_src_template =

GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)

        ", layout = (string) { interleaved, non-interleaved }")

    );

static GstStaticPadTemplate gst_audio_rate_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)

        ", layout = (string) { interleaved, non-interleaved }")

    );

static gboolean gst_audio_rate_sink_event (GstPad * pad, GstObject * parent,

    GstEvent * event);

static gboolean gst_audio_rate_src_event (GstPad * pad, GstObject * parent,

    GstEvent * event);

static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstObject * parent,

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

static GParamSpec *pspec_drop = NULL;

static GParamSpec *pspec_add = NULL;

#define gst_audio_rate_parent_class parent_class

G_DEFINE_TYPE (GstAudioRate, gst_audio_rate, GST_TYPE_ELEMENT);

static void

gst_audio_rate_class_init (GstAudioRateClass * klass)

{

  GObjectClass *object_class = G_OBJECT_CLASS (klass);

  GstElementClass *element_class = GST_ELEMENT_CLASS (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, PROP_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, PROP_OUT,

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

          G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));

  pspec_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, PROP_ADD, pspec_add);

  pspec_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, PROP_DROP, pspec_drop);

  g_object_class_install_property (object_class, PROP_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));

  /**

   * GstAudioRate:tolerance:

   *

   * The difference between incoming timestamp and next timestamp must exceed

   * the given value for audiorate to add or drop samples.

   */

  g_object_class_install_property (object_class, PROP_TOLERANCE,

      g_param_spec_uint64 ("tolerance", "tolerance",

          "Only act if timestamp jitter/imperfection exceeds indicated tolerance (ns)",

          0, G_MAXUINT64, DEFAULT_TOLERANCE,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  /**

   * GstAudioRate:skip-to-first:

   *

   * Don't produce buffers before the first one we receive.

   */

  g_object_class_install_property (object_class, PROP_SKIP_TO_FIRST,

      g_param_spec_boolean ("skip-to-first", "Skip to first buffer",

          "Don't produce buffers before the first one we receive",

          DEFAULT_SKIP_TO_FIRST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  gst_element_class_set_static_metadata (element_class,

      "Audio rate adjuster", "Filter/Effect/Audio",

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

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

  gst_element_class_add_static_pad_template (element_class,

      &gst_audio_rate_sink_template);

  gst_element_class_add_static_pad_template (element_class,

      &gst_audio_rate_src_template);

  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 (GstAudioRate * audiorate, GstCaps * caps)

{

  GstAudioInfo info;

  if (!gst_audio_info_from_caps (&info, caps))

    goto wrong_caps;

  audiorate->info = info;

  return TRUE;

  /* ERRORS */

wrong_caps:

  {

    GST_DEBUG_OBJECT (audiorate, "could not parse caps");

    return FALSE;

  }

}

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_PROXY_CAPS (audiorate->sinkpad);

  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_PROXY_CAPS (audiorate->srcpad);

  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;

  audiorate->tolerance = DEFAULT_TOLERANCE;

}

static void

gst_audio_rate_fill_to_time (GstAudioRate * audiorate, GstClockTime time)

{

  GstBuffer *buf;

  GST_DEBUG_OBJECT (audiorate, "next_ts: %" GST_TIME_FORMAT

      ", filling to %" GST_TIME_FORMAT, GST_TIME_ARGS (audiorate->next_ts),

      GST_TIME_ARGS (time));

  if (!GST_CLOCK_TIME_IS_VALID (time) ||

      !GST_CLOCK_TIME_IS_VALID (audiorate->next_ts))

    return;

  /* feed an empty buffer to chain with the given timestamp,

   * it will take care of filling */

  buf = gst_buffer_new ();

  GST_BUFFER_TIMESTAMP (buf) = time;

  gst_audio_rate_chain (audiorate->sinkpad, GST_OBJECT_CAST (audiorate), buf);

}

static gboolean

gst_audio_rate_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)

{

  gboolean res;

  GstAudioRate *audiorate;

  audiorate = GST_AUDIO_RATE (parent);

  switch (GST_EVENT_TYPE (event)) {

    case GST_EVENT_CAPS:

    {

      GstCaps *caps;

      gst_event_parse_caps (event, &caps);

      if ((res = gst_audio_rate_setcaps (audiorate, caps))) {

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

      } else {

        gst_event_unref (event);

      }

      break;

    }

    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_SEGMENT:

    {

      gst_event_copy_segment (event, &audiorate->sink_segment);

      GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");

#if 0

      /* FIXME: bad things will likely happen if rate < 0 ... */

      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. */

        gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);

#endif

        audiorate->next_offset = -1;

        audiorate->next_ts = -1;

#if 0

      } else {

        gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.start);

      }

#endif

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

          &audiorate->sink_segment);

      if (audiorate->sink_segment.format == GST_FORMAT_TIME) {

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

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

        gst_segment_copy_into (&audiorate->sink_segment,

            &audiorate->src_segment);

      } else {

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

        gst_event_unref (event);

        res = TRUE;

      }

      break;

    }

    case GST_EVENT_EOS:

      /* Fill segment until the end */

      if (GST_CLOCK_TIME_IS_VALID (audiorate->src_segment.stop))

        gst_audio_rate_fill_to_time (audiorate, audiorate->src_segment.stop);

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

      break;

    case GST_EVENT_GAP:

    {

      /* Fill until end of gap */

      GstClockTime timestamp, duration;

      gst_event_parse_gap (event, &timestamp, &duration);

      gst_event_unref (event);

      if (GST_CLOCK_TIME_IS_VALID (timestamp)) {

        if (GST_CLOCK_TIME_IS_VALID (duration))

          timestamp += duration;

        gst_audio_rate_fill_to_time (audiorate, timestamp);

      }

      res = TRUE;

      break;

    }

    default:

      res = gst_pad_event_default (pad, parent, event);

      break;

  }

  return res;

}

static gboolean

gst_audio_rate_src_event (GstPad * pad, GstObject * parent, GstEvent * event)

{

  gboolean res;

  GstAudioRate *audiorate;

  audiorate = GST_AUDIO_RATE (parent);

  switch (GST_EVENT_TYPE (event)) {

    default:

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

      break;

  }

  return res;

}

static gboolean

gst_audio_rate_convert (GstAudioRate * audiorate,

    GstFormat src_fmt, guint64 src_val, GstFormat dest_fmt, guint64 * dest_val)

{

  return gst_audio_info_convert (&audiorate->info, src_fmt, src_val, dest_fmt,

      (gint64 *) dest_val);

}

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.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 (base);

  CONVERT_VAL (position);

#undef CONVERT_VAL

  return TRUE;

}

static void

gst_audio_rate_notify_drop (GstAudioRate * audiorate)

{

  g_object_notify_by_pspec ((GObject *) audiorate, pspec_drop);

}

static void

gst_audio_rate_notify_add (GstAudioRate * audiorate)

{

  g_object_notify_by_pspec ((GObject *) audiorate, pspec_add);

}

static GstFlowReturn

gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)

{

  GstAudioRate *audiorate;

  GstClockTime in_time;

  guint64 in_offset, in_offset_end, in_samples;

  guint in_size;

  GstFlowReturn ret = GST_FLOW_OK;

  GstClockTimeDiff diff;

  gint rate, bpf;

  audiorate = GST_AUDIO_RATE (parent);

  rate = GST_AUDIO_INFO_RATE (&audiorate->info);

  bpf = GST_AUDIO_INFO_BPF (&audiorate->info);

  /* need to be negotiated now */

  if (bpf == 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_round (audiorate->src_segment.start,

        GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND);

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

    /* resyncing is a discont */

    audiorate->discont = TRUE;

    audiorate->next_offset = pos;

    audiorate->next_ts =

        gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND,

        GST_AUDIO_INFO_RATE (&audiorate->info));

    if (audiorate->skip_to_first && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {

      GST_DEBUG_OBJECT (audiorate, "but skipping to first buffer instead");

      pos = gst_util_uint64_scale_int_round (GST_BUFFER_TIMESTAMP (buf),

          GST_AUDIO_INFO_RATE (&audiorate->info), GST_SECOND);

      GST_DEBUG_OBJECT (audiorate, "so resync to offset %" G_GINT64_FORMAT,

          pos);

      audiorate->next_offset = pos;

      audiorate->next_ts = GST_BUFFER_TIMESTAMP (buf);

    }

  }

  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_get_size (buf);

  in_samples = in_size / bpf;

  audiorate->in += in_samples;

  /* calculate the buffer offset */

  in_offset = gst_util_uint64_scale_int_round (in_time, rate, GST_SECOND);

  in_offset_end = in_offset + in_samples;

  GST_LOG_OBJECT (audiorate,

      "in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT

      ", in_size:%u, in_offset:%" G_GUINT64_FORMAT ", in_offset_end:%"

      G_GUINT64_FORMAT ", ->next_offset:%" G_GUINT64_FORMAT ", ->next_ts:%"

      GST_TIME_FORMAT, GST_TIME_ARGS (in_time),

      GST_TIME_ARGS (GST_FRAMES_TO_CLOCK_TIME (in_samples, rate)),

      in_size, in_offset, in_offset_end, audiorate->next_offset,

      GST_TIME_ARGS (audiorate->next_ts));

  diff = in_time - audiorate->next_ts;

  if (diff <= (GstClockTimeDiff) audiorate->tolerance &&

      diff >= (GstClockTimeDiff) - audiorate->tolerance) {

    /* buffer time close enough to expected time,

     * so produce a perfect stream by simply 'shifting'

     * it to next ts and offset and sending */

    GST_LOG_OBJECT (audiorate, "within tolerance %" GST_TIME_FORMAT,

        GST_TIME_ARGS (audiorate->tolerance));

    /* The outgoing buffer's offset will be set to ->next_offset, we also

     * need to adjust the offset_end value accordingly */

    in_offset_end = audiorate->next_offset + in_samples;

    audiorate->out += in_samples;

    goto send;

  }

  /* do we need to insert samples */

  if (in_offset > audiorate->next_offset) {

    GstBuffer *fill;

    gint fillsize;

    guint64 fillsamples;

    /* We don't want to allocate a single unreasonably huge buffer - it might

       be hundreds of megabytes. So, limit each output buffer to one second of

       audio */

    fillsamples = in_offset - audiorate->next_offset;

    while (fillsamples > 0) {

      guint64 cursamples = MIN (fillsamples, rate);

      GstMapInfo fillmap;

      fillsamples -= cursamples;

      fillsize = cursamples * bpf;

      fill = gst_buffer_new_and_alloc (fillsize);

      gst_buffer_map (fill, &fillmap, GST_MAP_WRITE);

      gst_audio_format_fill_silence (audiorate->info.finfo, fillmap.data,

          fillmap.size);

      gst_buffer_unmap (fill, &fillmap);

      GST_DEBUG_OBJECT (audiorate, "inserting %" G_GUINT64_FORMAT " samples",

          cursamples);

      GST_BUFFER_OFFSET (fill) = audiorate->next_offset;

      audiorate->next_offset += cursamples;

      GST_BUFFER_OFFSET_END (fill) = audiorate->next_offset;

      /* Use next timestamp, then calculate following timestamp based on 

       * offset to get duration. Necessary complexity to get 'perfect' 

       * streams */

      GST_BUFFER_TIMESTAMP (fill) = audiorate->next_ts;

      audiorate->next_ts =

          gst_util_uint64_scale_int_round (audiorate->next_offset, GST_SECOND,

          rate);

      GST_BUFFER_DURATION (fill) =

          audiorate->next_ts - GST_BUFFER_TIMESTAMP (fill);

      /* we created this buffer to fill a gap */

      GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);

      /* set discont if it's pending, this is mostly done for the first buffer 

       * and after a flushing seek */

      if (audiorate->discont) {

        GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);

        audiorate->discont = FALSE;

      }

      fill = gst_audio_buffer_clip (fill, &audiorate->src_segment, rate, bpf);

      if (fill)

        ret = gst_pad_push (audiorate->srcpad, fill);

      if (ret != GST_FLOW_OK)

        goto beach;

      audiorate->out += cursamples;

      audiorate->add += cursamples;

      if (!audiorate->silent)

        gst_audio_rate_notify_add (audiorate);

    }

  } else if (in_offset < audiorate->next_offset) {

    /* need to remove samples */

    if (in_offset_end <= audiorate->next_offset) {

      guint64 drop = in_size / bpf;

      audiorate->drop += drop;

      GST_DEBUG_OBJECT (audiorate, "dropping %" G_GUINT64_FORMAT " samples",

          drop);

      /* we can drop the buffer completely */

      gst_buffer_unref (buf);

      buf = NULL;

      if (!audiorate->silent)

        gst_audio_rate_notify_drop (audiorate);

      goto beach;

    } else {

      guint64 truncsamples;

      guint truncsize, leftsize;

      GstBuffer *trunc;

      /* truncate buffer */

      truncsamples = audiorate->next_offset - in_offset;

      truncsize = truncsamples * bpf;

      leftsize = in_size - truncsize;

      trunc =

          gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, truncsize,

          leftsize);

      gst_buffer_unref (buf);

      buf = trunc;

      audiorate->drop += truncsamples;

      audiorate->out += (leftsize / bpf);

      GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples",

          truncsamples);

      if (!audiorate->silent)

        gst_audio_rate_notify_drop (audiorate);

    }

  }

send:

  if (gst_buffer_get_size (buf) == 0)

    goto beach;

  /* Now calculate parameters for whichever buffer (either the original

   * or truncated one) we're pushing. */

  GST_BUFFER_OFFSET (buf) = audiorate->next_offset;

  GST_BUFFER_OFFSET_END (buf) = in_offset_end;

  GST_BUFFER_TIMESTAMP (buf) = audiorate->next_ts;

  audiorate->next_ts = gst_util_uint64_scale_int_round (in_offset_end,

      GST_SECOND, rate);

  GST_BUFFER_DURATION (buf) = audiorate->next_ts - GST_BUFFER_TIMESTAMP (buf);

  if (audiorate->discont) {

    /* we need to output a discont buffer, do so now */

    GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer");

    buf = gst_buffer_make_writable (buf);

    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);

    audiorate->discont = FALSE;

  } else if (GST_BUFFER_IS_DISCONT (buf)) {

    /* else we make everything continuous so we can safely remove the DISCONT

     * flag from the buffer if there was one */

    GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer");

    buf = gst_buffer_make_writable (buf);

    GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);

  }

  buf = gst_audio_buffer_clip (buf, &audiorate->src_segment, rate, bpf);

  if (buf) {

    /* set last_stop on segment */

    audiorate->src_segment.position =

        GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);

    ret = gst_pad_push (audiorate->srcpad, buf);

  }

  buf = NULL;

  audiorate->next_offset = in_offset_end;

beach:

  if (buf)

    gst_buffer_unref (buf);

  return ret;

  /* ERRORS */

not_negotiated:

  {

    gst_buffer_unref (buf);

    GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT,

        (NULL), ("pipeline error, format was not negotiated"));

    return GST_FLOW_NOT_NEGOTIATED;

  }

}

static void

gst_audio_rate_set_property (GObject * object,

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

{

  GstAudioRate *audiorate = GST_AUDIO_RATE (object);

  switch (prop_id) {

    case PROP_SILENT:

      audiorate->silent = g_value_get_boolean (value);

      break;

    case PROP_TOLERANCE:

      audiorate->tolerance = g_value_get_uint64 (value);

      break;

    case PROP_SKIP_TO_FIRST:

      audiorate->skip_to_first = g_value_get_boolean (value);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_audio_rate_get_property (GObject * object,

    guint prop_id, GValue * value, GParamSpec * pspec)

{

  GstAudioRate *audiorate = GST_AUDIO_RATE (object);

  switch (prop_id) {

    case PROP_IN:

      g_value_set_uint64 (value, audiorate->in);

      break;

    case PROP_OUT:

      g_value_set_uint64 (value, audiorate->out);

      break;

    case PROP_ADD:

      g_value_set_uint64 (value, audiorate->add);

      break;

    case PROP_DROP:

      g_value_set_uint64 (value, audiorate->drop);

      break;

    case PROP_SILENT:

      g_value_set_boolean (value, audiorate->silent);

      break;

    case PROP_TOLERANCE:

      g_value_set_uint64 (value, audiorate->tolerance);

      break;

    case PROP_SKIP_TO_FIRST:

      g_value_set_boolean (value, audiorate->skip_to_first);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static GstStateChangeReturn

gst_audio_rate_change_state (GstElement * element, GstStateChange transition)

{

  GstAudioRate *audiorate = GST_AUDIO_RATE (element);

  switch (transition) {

    case GST_STATE_CHANGE_PAUSED_TO_READY:

      break;

    case GST_STATE_CHANGE_READY_TO_PAUSED:

      audiorate->in = 0;

      audiorate->out = 0;

      audiorate->drop = 0;

      audiorate->add = 0;

      gst_audio_info_init (&audiorate->info);

      gst_audio_rate_reset (audiorate);

      break;

    default:

      break;

  }

  return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

}

static gboolean

plugin_init (GstPlugin * plugin)

{

  GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0,

      "AudioRate stream fixer");

  return gst_element_register (plugin, "audiorate", GST_RANK_NONE,

      GST_TYPE_AUDIO_RATE);

}

GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,

    GST_VERSION_MINOR,

    audiorate,

    "Adjusts audio frames",

    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)