/*
 * GStreamer
 * Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
 *
 * 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.
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include "gstaudiobuffersplit.h"

#define GST_CAT_DEFAULT gst_audio_buffer_split_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-raw")
    );

static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-raw")
    );

enum
{
  PROP_0,
  PROP_OUTPUT_BUFFER_DURATION,
  PROP_ALIGNMENT_THRESHOLD,
  PROP_DISCONT_WAIT,
  PROP_STRICT_BUFFER_SIZE,
  PROP_GAPLESS,
  LAST_PROP
};

#define DEFAULT_OUTPUT_BUFFER_DURATION_N (1)
#define DEFAULT_OUTPUT_BUFFER_DURATION_D (50)
#define DEFAULT_ALIGNMENT_THRESHOLD   (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_STRICT_BUFFER_SIZE (FALSE)
#define DEFAULT_GAPLESS (FALSE)

#define parent_class gst_audio_buffer_split_parent_class
G_DEFINE_TYPE (GstAudioBufferSplit, gst_audio_buffer_split, GST_TYPE_ELEMENT);

static GstFlowReturn gst_audio_buffer_split_sink_chain (GstPad * pad,
    GstObject * parent, GstBuffer * buffer);
static gboolean gst_audio_buffer_split_sink_event (GstPad * pad,
    GstObject * parent, GstEvent * event);
static gboolean gst_audio_buffer_split_src_query (GstPad * pad,
    GstObject * parent, GstQuery * query);

static void gst_audio_buffer_split_finalize (GObject * object);
static void gst_audio_buffer_split_get_property (GObject * object,
    guint property_id, GValue * value, GParamSpec * pspec);
static void gst_audio_buffer_split_set_property (GObject * object,
    guint property_id, const GValue * value, GParamSpec * pspec);

static GstStateChangeReturn gst_audio_buffer_split_change_state (GstElement *
    element, GstStateChange transition);

static void
gst_audio_buffer_split_class_init (GstAudioBufferSplitClass * klass)
{
  GObjectClass *gobject_class = (GObjectClass *) klass;
  GstElementClass *gstelement_class = (GstElementClass *) klass;

  gobject_class->set_property = gst_audio_buffer_split_set_property;
  gobject_class->get_property = gst_audio_buffer_split_get_property;
  gobject_class->finalize = gst_audio_buffer_split_finalize;

  g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
      gst_param_spec_fraction ("output-buffer-duration",
          "Output Buffer Duration", "Output block size in seconds", 1, G_MAXINT,
          G_MAXINT, 1, DEFAULT_OUTPUT_BUFFER_DURATION_N,
          DEFAULT_OUTPUT_BUFFER_DURATION_D,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
          GST_PARAM_MUTABLE_READY));

  g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
      g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
          "Timestamp alignment threshold in nanoseconds", 0,
          G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
          GST_PARAM_MUTABLE_READY));

  g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
      g_param_spec_uint64 ("discont-wait", "Discont Wait",
          "Window of time in nanoseconds to wait before "
          "creating a discontinuity", 0,
          G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
          GST_PARAM_MUTABLE_READY));

  g_object_class_install_property (gobject_class, PROP_STRICT_BUFFER_SIZE,
      g_param_spec_boolean ("strict-buffer-size", "Strict buffer size",
          "Discard the last samples at EOS or discont if they are too "
          "small to fill a buffer", DEFAULT_STRICT_BUFFER_SIZE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
          GST_PARAM_MUTABLE_READY));

  g_object_class_install_property (gobject_class, PROP_GAPLESS,
      g_param_spec_boolean ("gapless", "Gapless",
          "Insert silence/drop samples instead of creating a discontinuity",
          DEFAULT_GAPLESS,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
          GST_PARAM_MUTABLE_READY));

  gst_element_class_set_static_metadata (gstelement_class,
      "Audio Buffer Split", "Audio/Filter",
      "Splits raw audio buffers into equal sized chunks",
      "Sebastian Dröge <sebastian@centricular.com>");

  gst_element_class_add_pad_template (gstelement_class,
      gst_static_pad_template_get (&src_template));
  gst_element_class_add_pad_template (gstelement_class,
      gst_static_pad_template_get (&sink_template));

  gstelement_class->change_state = gst_audio_buffer_split_change_state;
}

static void
gst_audio_buffer_split_init (GstAudioBufferSplit * self)
{
  self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
  gst_pad_set_chain_function (self->sinkpad,
      GST_DEBUG_FUNCPTR (gst_audio_buffer_split_sink_chain));
  gst_pad_set_event_function (self->sinkpad,
      GST_DEBUG_FUNCPTR (gst_audio_buffer_split_sink_event));
  GST_PAD_SET_PROXY_CAPS (self->sinkpad);
  gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);

  self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
  gst_pad_set_query_function (self->srcpad,
      GST_DEBUG_FUNCPTR (gst_audio_buffer_split_src_query));
  GST_PAD_SET_PROXY_CAPS (self->srcpad);
  gst_pad_use_fixed_caps (self->srcpad);
  gst_element_add_pad (GST_ELEMENT (self), self->srcpad);

  self->output_buffer_duration_n = DEFAULT_OUTPUT_BUFFER_DURATION_N;
  self->output_buffer_duration_d = DEFAULT_OUTPUT_BUFFER_DURATION_D;
  self->strict_buffer_size = DEFAULT_STRICT_BUFFER_SIZE;
  self->gapless = DEFAULT_GAPLESS;

  self->adapter = gst_adapter_new ();

  self->stream_align =
      gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD,
      DEFAULT_DISCONT_WAIT);
}

static void
gst_audio_buffer_split_finalize (GObject * object)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);

  if (self->adapter) {
    gst_object_unref (self->adapter);
    self->adapter = NULL;
  }

  if (self->stream_align) {
    gst_audio_stream_align_free (self->stream_align);
    self->stream_align = NULL;
  }

  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static gboolean
gst_audio_buffer_split_update_samples_per_buffer (GstAudioBufferSplit * self)
{
  gboolean ret = TRUE;

  GST_OBJECT_LOCK (self);

  /* For a later time */
  if (!self->info.finfo
      || GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
    self->samples_per_buffer = 0;
    goto out;
  }

  self->samples_per_buffer =
      (((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
      self->output_buffer_duration_n) / self->output_buffer_duration_d;
  if (self->samples_per_buffer == 0) {
    ret = FALSE;
    goto out;
  }

  self->error_per_buffer =
      (((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
      self->output_buffer_duration_n) % self->output_buffer_duration_d;
  self->accumulated_error = 0;

  GST_DEBUG_OBJECT (self, "Buffer duration: %u/%u",
      self->output_buffer_duration_n, self->output_buffer_duration_d);
  GST_DEBUG_OBJECT (self, "Samples per buffer: %u (error: %u/%u)",
      self->samples_per_buffer, self->error_per_buffer,
      self->output_buffer_duration_d);
out:
  GST_OBJECT_UNLOCK (self);

  return ret;
}

static void
gst_audio_buffer_split_set_property (GObject * object, guint property_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);

  switch (property_id) {
    case PROP_OUTPUT_BUFFER_DURATION:
      self->output_buffer_duration_n = gst_value_get_fraction_numerator (value);
      self->output_buffer_duration_d =
          gst_value_get_fraction_denominator (value);
      gst_audio_buffer_split_update_samples_per_buffer (self);
      break;
    case PROP_ALIGNMENT_THRESHOLD:
      GST_OBJECT_LOCK (self);
      gst_audio_stream_align_set_alignment_threshold (self->stream_align,
          g_value_get_uint64 (value));
      GST_OBJECT_UNLOCK (self);
      break;
    case PROP_DISCONT_WAIT:
      GST_OBJECT_LOCK (self);
      gst_audio_stream_align_set_discont_wait (self->stream_align,
          g_value_get_uint64 (value));
      GST_OBJECT_UNLOCK (self);
      break;
    case PROP_STRICT_BUFFER_SIZE:
      self->strict_buffer_size = g_value_get_boolean (value);
      break;
    case PROP_GAPLESS:
      self->gapless = g_value_get_boolean (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
      break;
  }
}

static void
gst_audio_buffer_split_get_property (GObject * object, guint property_id,
    GValue * value, GParamSpec * pspec)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);

  switch (property_id) {
    case PROP_OUTPUT_BUFFER_DURATION:
      gst_value_set_fraction (value, self->output_buffer_duration_n,
          self->output_buffer_duration_d);
      break;
    case PROP_ALIGNMENT_THRESHOLD:
      GST_OBJECT_LOCK (self);
      g_value_set_uint64 (value,
          gst_audio_stream_align_get_alignment_threshold (self->stream_align));
      GST_OBJECT_UNLOCK (self);
      break;
    case PROP_DISCONT_WAIT:
      GST_OBJECT_LOCK (self);
      g_value_set_uint64 (value,
          gst_audio_stream_align_get_discont_wait (self->stream_align));
      GST_OBJECT_UNLOCK (self);
      break;
    case PROP_STRICT_BUFFER_SIZE:
      g_value_set_boolean (value, self->strict_buffer_size);
      break;
    case PROP_GAPLESS:
      g_value_set_boolean (value, self->gapless);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
      break;
  }
}

static GstStateChangeReturn
gst_audio_buffer_split_change_state (GstElement * element,
    GstStateChange transition)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (element);
  GstStateChangeReturn state_ret;

  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      gst_audio_info_init (&self->info);
      gst_segment_init (&self->segment, GST_FORMAT_TIME);
      GST_OBJECT_LOCK (self);
      gst_audio_stream_align_mark_discont (self->stream_align);
      GST_OBJECT_UNLOCK (self);
      self->current_offset = -1;
      self->accumulated_error = 0;
      self->samples_per_buffer = 0;
      break;
    default:
      break;
  }

  state_ret =
      GST_ELEMENT_CLASS (gst_audio_buffer_split_parent_class)->change_state
      (element, transition);
  if (state_ret == GST_STATE_CHANGE_FAILURE)
    return state_ret;

  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      gst_adapter_clear (self->adapter);
      GST_OBJECT_LOCK (self);
      gst_audio_stream_align_mark_discont (self->stream_align);
      GST_OBJECT_UNLOCK (self);
      break;
    default:
      break;
  }

  return state_ret;
}

static GstFlowReturn
gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
    gint rate, gint bpf, guint samples_per_buffer)
{
  gint size, avail;
  GstFlowReturn ret = GST_FLOW_OK;
  GstClockTime resync_time;

  resync_time = self->resync_time;
  size = samples_per_buffer * bpf;

  /* If we accumulated enough error for one sample, include one
   * more sample in this buffer. Accumulated error is updated below */
  if (self->error_per_buffer + self->accumulated_error >=
      self->output_buffer_duration_d)
    size += bpf;

  while ((avail = gst_adapter_available (self->adapter)) >= size || (force
          && avail > 0)) {
    GstBuffer *buffer;
    GstClockTime resync_time_diff;

    size = MIN (size, avail);
    buffer = gst_adapter_take_buffer (self->adapter, size);

    /* After a reset we have to set the discont flag */
    if (self->current_offset == 0)
      GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);

    resync_time_diff =
        gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
    if (self->segment.rate < 0.0) {
      if (resync_time > resync_time_diff)
        GST_BUFFER_TIMESTAMP (buffer) = resync_time - resync_time_diff;
      else
        GST_BUFFER_TIMESTAMP (buffer) = 0;
      GST_BUFFER_DURATION (buffer) =
          gst_util_uint64_scale (size / bpf, GST_SECOND, rate);

      self->current_offset += size / bpf;
    } else {
      GST_BUFFER_TIMESTAMP (buffer) = resync_time + resync_time_diff;
      self->current_offset += size / bpf;
      resync_time_diff =
          gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
      GST_BUFFER_DURATION (buffer) =
          resync_time_diff - (GST_BUFFER_TIMESTAMP (buffer) - resync_time);
    }

    GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
    GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;

    self->accumulated_error =
        (self->accumulated_error +
        self->error_per_buffer) % self->output_buffer_duration_d;

    GST_LOG_OBJECT (self,
        "Outputting buffer at timestamp %" GST_TIME_FORMAT " with duration %"
        GST_TIME_FORMAT " (%u samples)",
        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
        GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), size / bpf);

    ret = gst_pad_push (self->srcpad, buffer);
    if (ret != GST_FLOW_OK)
      break;

    /* Update the size based on the accumulated error we have now after
     * taking out a buffer. Same code as above */
    size = samples_per_buffer * bpf;
    if (self->error_per_buffer + self->accumulated_error >=
        self->output_buffer_duration_d)
      size += bpf;
  }

  return ret;
}

static GstFlowReturn
gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
    GstBuffer * buffer, GstAudioFormat format, gint rate, gint bpf,
    guint samples_per_buffer)
{
  gboolean discont;
  GstFlowReturn ret = GST_FLOW_OK;

  GST_OBJECT_LOCK (self);
  discont =
      gst_audio_stream_align_process (self->stream_align,
      self->segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
      || GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC),
      GST_BUFFER_PTS (buffer), gst_buffer_get_size (buffer) / bpf, NULL, NULL,
      NULL);
  GST_OBJECT_UNLOCK (self);

  if (discont) {
    guint avail = gst_adapter_available (self->adapter);
    guint avail_samples = avail / bpf;
    guint64 new_offset;
    GstClockTime current_timestamp;
    GstClockTime current_timestamp_end;

    /* Reset */
    self->drop_samples = 0;

    if (self->segment.rate < 0.0) {
      current_timestamp =
          self->resync_time - gst_util_uint64_scale (self->current_offset +
          avail_samples, GST_SECOND, rate);
      current_timestamp_end =
          self->resync_time - gst_util_uint64_scale (self->current_offset,
          GST_SECOND, rate);
    } else {
      current_timestamp =
          self->resync_time + gst_util_uint64_scale (self->current_offset,
          GST_SECOND, rate);
      current_timestamp_end =
          self->resync_time + gst_util_uint64_scale (self->current_offset +
          avail_samples, GST_SECOND, rate);
    }

    if (self->gapless) {
      if (self->current_offset == -1) {
        /* We only set resync time on the very first buffer */
        self->current_offset = 0;
        self->resync_time = GST_BUFFER_PTS (buffer);
      } else {
        GST_DEBUG_OBJECT (self,
            "Got discont in gapless mode: Current timestamp %" GST_TIME_FORMAT
            ", current end timestamp %" GST_TIME_FORMAT
            ", timestamp after discont %" GST_TIME_FORMAT,
            GST_TIME_ARGS (current_timestamp),
            GST_TIME_ARGS (current_timestamp_end),
            GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));

        new_offset =
            gst_util_uint64_scale (GST_BUFFER_PTS (buffer) - self->resync_time,
            rate, GST_SECOND);
        if (GST_BUFFER_PTS (buffer) < self->resync_time) {
          guint64 drop_samples;

          new_offset =
              gst_util_uint64_scale (self->resync_time -
              GST_BUFFER_PTS (buffer), rate, GST_SECOND);
          drop_samples = self->current_offset + avail_samples + new_offset;

          GST_DEBUG_OBJECT (self,
              "Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
              drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
                      GST_SECOND, rate)));
        } else if (new_offset > self->current_offset + avail_samples) {
          guint64 silence_samples =
              new_offset - (self->current_offset + avail_samples);
          const GstAudioFormatInfo *info = gst_audio_format_get_info (format);

          GST_DEBUG_OBJECT (self,
              "Inserting %" G_GUINT64_FORMAT " samples of silence (%"
              GST_TIME_FORMAT ")", silence_samples,
              GST_TIME_ARGS (gst_util_uint64_scale (silence_samples, GST_SECOND,
                      rate)));

          /* Insert silence buffers to fill the gap in 1s chunks */
          while (silence_samples > 0) {
            guint n_samples = MIN (silence_samples, rate);
            GstBuffer *silence;
            GstMapInfo map;

            silence = gst_buffer_new_and_alloc (n_samples * bpf);
            GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
            gst_buffer_map (silence, &map, GST_MAP_WRITE);
            gst_audio_format_fill_silence (info, map.data, map.size);
            gst_buffer_unmap (silence, &map);

            gst_adapter_push (self->adapter, silence);
            ret =
                gst_audio_buffer_split_output (self, FALSE, rate, bpf,
                samples_per_buffer);
            if (ret != GST_FLOW_OK)
              return ret;

            silence_samples -= n_samples;
          }
        } else if (new_offset < self->current_offset + avail_samples) {
          guint64 drop_samples =
              self->current_offset + avail_samples - new_offset;

          GST_DEBUG_OBJECT (self,
              "Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
              drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
                      GST_SECOND, rate)));
          self->drop_samples = drop_samples;
        }
      }
    } else {
      GST_DEBUG_OBJECT (self,
          "Got discont: Current timestamp %" GST_TIME_FORMAT
          ", current end timestamp %" GST_TIME_FORMAT
          ", timestamp after discont %" GST_TIME_FORMAT,
          GST_TIME_ARGS (current_timestamp),
          GST_TIME_ARGS (current_timestamp_end),
          GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));

      if (self->strict_buffer_size) {
        gst_adapter_clear (self->adapter);
        ret = GST_FLOW_OK;
      } else {
        ret =
            gst_audio_buffer_split_output (self, TRUE, rate, bpf,
            samples_per_buffer);
      }

      self->current_offset = 0;
      self->accumulated_error = 0;
      self->resync_time = GST_BUFFER_PTS (buffer);
    }
  }

  return ret;
}

static GstBuffer *
gst_audio_buffer_split_clip_buffer (GstAudioBufferSplit * self,
    GstBuffer * buffer, const GstSegment * segment, gint rate, gint bpf)
{
  return gst_audio_buffer_clip (buffer, segment, rate, bpf);
}

static GstBuffer *
gst_audio_buffer_split_clip_buffer_start_for_gapless (GstAudioBufferSplit *
    self, GstBuffer * buffer, gint rate, gint bpf)
{
  guint nsamples;

  if (!self->gapless || self->drop_samples == 0)
    return buffer;

  nsamples = gst_buffer_get_size (buffer) / bpf;

  GST_DEBUG_OBJECT (self, "Have to drop %" G_GUINT64_FORMAT
      " samples, got %u samples", self->drop_samples, nsamples);

  if (nsamples <= self->drop_samples) {
    gst_buffer_unref (buffer);
    self->drop_samples -= nsamples;
    return NULL;
  }

  if (self->segment.rate < 0.0) {
    buffer =
        gst_audio_buffer_truncate (buffer, bpf, 0,
        nsamples - self->drop_samples);
    self->drop_samples = 0;
    return buffer;
  } else {
    buffer = gst_audio_buffer_truncate (buffer, bpf, self->drop_samples, -1);
    self->drop_samples = 0;
    return buffer;
  }

  return buffer;
}

static GstFlowReturn
gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
    GstBuffer * buffer)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
  GstFlowReturn ret;
  GstAudioFormat format;
  gint rate, bpf, samples_per_buffer;

  GST_OBJECT_LOCK (self);
  format =
      self->info.
      finfo ? GST_AUDIO_INFO_FORMAT (&self->info) : GST_AUDIO_FORMAT_UNKNOWN;
  rate = GST_AUDIO_INFO_RATE (&self->info);
  bpf = GST_AUDIO_INFO_BPF (&self->info);
  samples_per_buffer = self->samples_per_buffer;
  GST_OBJECT_UNLOCK (self);

  if (format == GST_AUDIO_FORMAT_UNKNOWN || samples_per_buffer == 0) {
    gst_buffer_unref (buffer);
    return GST_FLOW_NOT_NEGOTIATED;
  }

  buffer =
      gst_audio_buffer_split_clip_buffer (self, buffer, &self->segment, rate,
      bpf);
  if (!buffer)
    return GST_FLOW_OK;

  ret =
      gst_audio_buffer_split_handle_discont (self, buffer, format, rate, bpf,
      samples_per_buffer);
  if (ret != GST_FLOW_OK) {
    gst_buffer_unref (buffer);
    return ret;
  }

  buffer =
      gst_audio_buffer_split_clip_buffer_start_for_gapless (self, buffer, rate,
      bpf);
  if (!buffer)
    return GST_FLOW_OK;

  gst_adapter_push (self->adapter, buffer);

  return gst_audio_buffer_split_output (self, FALSE, rate, bpf,
      samples_per_buffer);
}

static gboolean
gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
    GstEvent * event)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
  gboolean ret = FALSE;

  switch (GST_EVENT_TYPE (event)) {
    case GST_EVENT_CAPS:{
      GstCaps *caps;
      GstAudioInfo info;

      gst_event_parse_caps (event, &caps);

      ret = gst_audio_info_from_caps (&info, caps);
      if (ret) {
        GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);

        if (!gst_audio_info_is_equal (&info, &self->info)) {
          if (self->strict_buffer_size) {
            gst_adapter_clear (self->adapter);
          } else {
            GstAudioFormat format;
            gint rate, bpf, samples_per_buffer;

            GST_OBJECT_LOCK (self);
            format =
                self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
                GST_AUDIO_FORMAT_UNKNOWN;
            rate = GST_AUDIO_INFO_RATE (&self->info);
            bpf = GST_AUDIO_INFO_BPF (&self->info);
            samples_per_buffer = self->samples_per_buffer;
            GST_OBJECT_UNLOCK (self);

            if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
              gst_audio_buffer_split_output (self, TRUE, rate, bpf,
                  samples_per_buffer);
          }
        }
        self->info = info;
        GST_OBJECT_LOCK (self);
        gst_audio_stream_align_set_rate (self->stream_align, self->info.rate);
        GST_OBJECT_UNLOCK (self);
        ret = gst_audio_buffer_split_update_samples_per_buffer (self);
      } else {
        ret = FALSE;
      }

      if (ret)
        ret = gst_pad_event_default (pad, parent, event);
      else
        gst_event_unref (event);

      break;
    }
    case GST_EVENT_FLUSH_STOP:
      gst_segment_init (&self->segment, GST_FORMAT_TIME);
      GST_OBJECT_LOCK (self);
      gst_audio_stream_align_mark_discont (self->stream_align);
      GST_OBJECT_UNLOCK (self);
      self->current_offset = -1;
      self->accumulated_error = 0;
      gst_adapter_clear (self->adapter);
      ret = gst_pad_event_default (pad, parent, event);
      break;
    case GST_EVENT_SEGMENT:
      gst_event_copy_segment (event, &self->segment);
      if (self->segment.format != GST_FORMAT_TIME) {
        gst_event_unref (event);
        ret = FALSE;
      } else {
        ret = gst_pad_event_default (pad, parent, event);
      }
      break;
    case GST_EVENT_EOS:
      if (self->strict_buffer_size) {
        gst_adapter_clear (self->adapter);
      } else {
        GstAudioFormat format;
        gint rate, bpf, samples_per_buffer;

        GST_OBJECT_LOCK (self);
        format =
            self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
            GST_AUDIO_FORMAT_UNKNOWN;
        rate = GST_AUDIO_INFO_RATE (&self->info);
        bpf = GST_AUDIO_INFO_BPF (&self->info);
        samples_per_buffer = self->samples_per_buffer;
        GST_OBJECT_UNLOCK (self);

        if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
          gst_audio_buffer_split_output (self, TRUE, rate, bpf,
              samples_per_buffer);
      }
      ret = gst_pad_event_default (pad, parent, event);
      break;
    default:
      ret = gst_pad_event_default (pad, parent, event);
      break;
  }

  return ret;
}

static gboolean
gst_audio_buffer_split_src_query (GstPad * pad,
    GstObject * parent, GstQuery * query)
{
  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
  gboolean ret = FALSE;

  switch (GST_QUERY_TYPE (query)) {
    case GST_QUERY_LATENCY:{
      if ((ret = gst_pad_peer_query (self->sinkpad, query))) {
        GstClockTime latency;
        GstClockTime min, max;
        gboolean live;

        gst_query_parse_latency (query, &live, &min, &max);

        GST_DEBUG_OBJECT (self, "Peer latency: min %"
            GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
            GST_TIME_ARGS (min), GST_TIME_ARGS (max));

        latency =
            gst_util_uint64_scale (GST_SECOND, self->output_buffer_duration_n,
            self->output_buffer_duration_d);

        GST_DEBUG_OBJECT (self, "Our latency: min %" GST_TIME_FORMAT
            ", max %" GST_TIME_FORMAT,
            GST_TIME_ARGS (latency), GST_TIME_ARGS (latency));

        min += latency;
        if (max != GST_CLOCK_TIME_NONE)
          max += latency;

        GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
            GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
            GST_TIME_ARGS (min), GST_TIME_ARGS (max));

        gst_query_set_latency (query, live, min, max);
      }

      break;
    }
    default:
      ret = gst_pad_query_default (pad, parent, query);
      break;
  }

  return ret;
}

static gboolean
plugin_init (GstPlugin * plugin)
{
  GST_DEBUG_CATEGORY_INIT (gst_audio_buffer_split_debug, "audiobuffersplit",
      0, "Audio buffer splitter");

  gst_element_register (plugin, "audiobuffersplit", GST_RANK_NONE,
      GST_TYPE_AUDIO_BUFFER_SPLIT);

  return TRUE;
}

GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
    GST_VERSION_MINOR,
    audiobuffersplit,
    "Audio buffer splitter",
    plugin_init, VERSION, "LGPL", PACKAGE_NAME, GST_PACKAGE_ORIGIN)