static void
gst_audio_aggregator_init (GstAudioAggregator * aagg)
{
  aagg->priv =
      G_TYPE_INSTANCE_GET_PRIVATE (aagg, GST_TYPE_AUDIO_AGGREGATOR,
      GstAudioAggregatorPrivate);

  g_mutex_init (&aagg->priv->mutex);

  aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
  aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
  aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;

  aagg->current_caps = NULL;
  gst_audio_info_init (&aagg->info);

  gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
      aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration);
}
static GstFlowReturn
gst_gl_stereo_mix_aggregate_frames (GstVideoAggregator * vagg,
    GstBuffer * outbuf)
{
  GstGLStereoMix *mix = GST_GL_STEREO_MIX (vagg);
  /* If we're operating in frame-by-frame mode, push
   * the primary view now, and let the parent class
   * push the remaining auxilliary view */
  if (GST_VIDEO_INFO_MULTIVIEW_MODE (&vagg->info) ==
      GST_VIDEO_MULTIVIEW_MODE_FRAME_BY_FRAME) {
    /* Transfer the timestamps video-agg put on the aux buffer */
    gst_buffer_copy_into (mix->primary_out, outbuf,
        GST_BUFFER_COPY_TIMESTAMPS, 0, -1);
    gst_aggregator_finish_buffer (GST_AGGREGATOR (vagg), mix->primary_out);
    mix->primary_out = NULL;

    /* And actually, we don't want timestamps on the aux buffer */
    GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
    GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
  }
  return GST_FLOW_OK;
}
/* GstElement vmethods implementations */
static GstStateChangeReturn
_change_state (GstElement * element, GstStateChange transition)
{
  GstStateChangeReturn ret;
  GstAggregator *self = GST_AGGREGATOR (element);
  GstAggregatorClass *agg_class = GST_AGGREGATOR_GET_CLASS (self);


  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      agg_class->start (self);
      break;
    default:
      break;
  }

  if ((ret =
          GST_ELEMENT_CLASS (aggregator_parent_class)->change_state (element,
              transition)) == GST_STATE_CHANGE_FAILURE)
    goto failure;


  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      agg_class->stop (self);
      break;
    default:
      break;
  }

  return ret;

failure:
  {
    GST_ERROR_OBJECT (element, "parent failed state change");
    return ret;
  }
}
static GstBuffer *
gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg,
    guint num_frames)
{
  GstAllocator *allocator;
  GstAllocationParams params;
  GstBuffer *outbuf;
  GstMapInfo outmap;

  gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, &params);

  outbuf = gst_buffer_new_allocate (allocator, num_frames *
      GST_AUDIO_INFO_BPF (&aagg->info), &params);

  if (allocator)
    gst_object_unref (allocator);

  gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
  gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size);
  gst_buffer_unmap (outbuf, &outmap);

  return outbuf;
}
static GstPad *
gst_aggregator_request_new_pad (GstElement * element, GstPadTemplate * templ,
    const gchar * unused, const GstCaps * caps)
{
  GstAggregator *aggregator = GST_AGGREGATOR (element);
  gchar *name;
  GstPad *newpad;
  gint padcount;

  if (templ->direction != GST_PAD_SINK)
    return NULL;

  /* create new pad */
  padcount = g_atomic_int_add (&aggregator->padcount, 1);
  name = g_strdup_printf ("sink_%u", padcount);
  newpad = gst_pad_new_from_template (templ, name);
  g_free (name);

  gst_collect_pads_add_pad (aggregator->collect, newpad,
      sizeof (GstCollectData), NULL, TRUE);

  /* takes ownership of the pad */
  if (!gst_element_add_pad (GST_ELEMENT (aggregator), newpad))
    goto could_not_add;

  GST_DEBUG_OBJECT (aggregator, "added new pad %s", GST_OBJECT_NAME (newpad));
  return newpad;

  /* errors */
could_not_add:
  {
    GST_DEBUG_OBJECT (aggregator, "could not add pad");
    gst_collect_pads_remove_pad (aggregator->collect, newpad);
    gst_object_unref (newpad);
    return NULL;
  }
}
static void
gst_audio_aggregator_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);

  switch (prop_id) {
    case PROP_OUTPUT_BUFFER_DURATION:
      aagg->priv->output_buffer_duration = g_value_get_uint64 (value);
      gst_aggregator_set_latency (GST_AGGREGATOR (aagg),
          aagg->priv->output_buffer_duration,
          aagg->priv->output_buffer_duration);
      break;
    case PROP_ALIGNMENT_THRESHOLD:
      aagg->priv->alignment_threshold = g_value_get_uint64 (value);
      break;
    case PROP_DISCONT_WAIT:
      aagg->priv->discont_wait = g_value_get_uint64 (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}
/* the first caps we receive on any of the sinkpads will define the caps for all
 * the other sinkpads because we can only mix streams with the same caps.
 */
static gboolean
gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad,
    GstCaps * orig_caps)
{
  GstAggregator *agg = GST_AGGREGATOR (audiomixer);
  GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (audiomixer);
  GstCaps *caps;
  GstAudioInfo info;
  GstStructure *s;
  gint channels = 0;

  caps = gst_caps_copy (orig_caps);

  s = gst_caps_get_structure (caps, 0);
  if (gst_structure_get_int (s, "channels", &channels))
    if (channels <= 2)
      gst_structure_remove_field (s, "channel-mask");

  if (!gst_audio_info_from_caps (&info, caps))
    goto invalid_format;

  if (channels == 1) {
    GstCaps *filter;
    GstCaps *downstream_caps;

    if (audiomixer->filter_caps)
      filter = gst_caps_intersect_full (caps, audiomixer->filter_caps,
          GST_CAPS_INTERSECT_FIRST);
    else
      filter = gst_caps_ref (caps);

    downstream_caps = gst_pad_peer_query_caps (agg->srcpad, filter);
    gst_caps_unref (filter);

    if (downstream_caps) {
      gst_caps_unref (caps);
      caps = downstream_caps;

      if (gst_caps_is_empty (caps)) {
        gst_caps_unref (caps);
        return FALSE;
      }
      caps = gst_caps_fixate (caps);
    }
  }

  GST_OBJECT_LOCK (audiomixer);
  /* don't allow reconfiguration for now; there's still a race between the
   * different upstream threads doing query_caps + accept_caps + sending
   * (possibly different) CAPS events, but there's not much we can do about
   * that, upstream needs to deal with it. */
  if (aagg->current_caps != NULL) {
    if (gst_audio_info_is_equal (&info, &aagg->info)) {
      GST_OBJECT_UNLOCK (audiomixer);
      gst_caps_unref (caps);
      gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
          orig_caps);
      return TRUE;
    } else {
      GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
          "current caps are %" GST_PTR_FORMAT, caps, aagg->current_caps);
      GST_OBJECT_UNLOCK (audiomixer);
      gst_pad_push_event (pad, gst_event_new_reconfigure ());
      gst_caps_unref (caps);
      return FALSE;
    }
  } else {
    gst_caps_replace (&aagg->current_caps, caps);
    aagg->info = info;
    gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (agg));
  }
  GST_OBJECT_UNLOCK (audiomixer);

  gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
      orig_caps);

  GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);

  gst_caps_unref (caps);

  return TRUE;

  /* ERRORS */
invalid_format:
  {
    gst_caps_unref (caps);
    GST_WARNING_OBJECT (audiomixer, "invalid format set as caps");
    return FALSE;
  }
}
static GstFlowReturn
_chain (GstPad * pad, GstObject * object, GstBuffer * buffer)
{
  GstBuffer *actual_buf = buffer;
  GstAggregator *self = GST_AGGREGATOR (object);
  GstAggregatorPrivate *priv = self->priv;
  GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
  GstAggregatorClass *aggclass = GST_AGGREGATOR_GET_CLASS (object);
  GstClockTime timeout = gst_aggregator_get_timeout (self);
  GstClockTime now;

  GST_DEBUG_OBJECT (aggpad, "Start chaining a buffer %" GST_PTR_FORMAT, buffer);
  if (aggpad->priv->timeout_id) {
    gst_clock_id_unschedule (aggpad->priv->timeout_id);
    gst_clock_id_unref (aggpad->priv->timeout_id);
    aggpad->priv->timeout_id = NULL;
  }
  g_atomic_int_set (&aggpad->unresponsive, FALSE);

  PAD_STREAM_LOCK (aggpad);

  if (g_atomic_int_get (&aggpad->priv->flushing) == TRUE)
    goto flushing;

  if (g_atomic_int_get (&aggpad->priv->pending_eos) == TRUE)
    goto eos;

  PAD_LOCK_EVENT (aggpad);

  if (aggpad->buffer) {
    GST_DEBUG_OBJECT (aggpad, "Waiting for buffer to be consumed");
    PAD_WAIT_EVENT (aggpad);
  }
  PAD_UNLOCK_EVENT (aggpad);

  if (g_atomic_int_get (&aggpad->priv->flushing) == TRUE)
    goto flushing;

  if (aggclass->clip) {
    aggclass->clip (self, aggpad, buffer, &actual_buf);
  }

  PAD_LOCK_EVENT (aggpad);
  if (aggpad->buffer)
    gst_buffer_unref (aggpad->buffer);
  aggpad->buffer = actual_buf;
  PAD_UNLOCK_EVENT (aggpad);
  PAD_STREAM_UNLOCK (aggpad);

  QUEUE_PUSH (self);

  if (GST_CLOCK_TIME_IS_VALID (timeout)) {
    now = gst_clock_get_time (self->clock);
    aggpad->priv->timeout_id =
        gst_clock_new_single_shot_id (self->clock, now + timeout);
    gst_clock_id_wait_async (aggpad->priv->timeout_id, _unresponsive_timeout,
        gst_object_ref (aggpad), gst_object_unref);
  }

  GST_DEBUG_OBJECT (aggpad, "Done chaining");

  return priv->flow_return;

flushing:
  PAD_STREAM_UNLOCK (aggpad);

  gst_buffer_unref (buffer);
  GST_DEBUG_OBJECT (aggpad, "We are flushing");

  return GST_FLOW_FLUSHING;

eos:
  PAD_STREAM_UNLOCK (aggpad);

  gst_buffer_unref (buffer);
  GST_DEBUG_OBJECT (pad, "We are EOS already...");

  return GST_FLOW_EOS;
}
/* Called with the object lock for both the element and pad held,
 * as well as the aagg lock
 */
static gboolean
gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
    GstAudioAggregatorPad * pad, GstBuffer * inbuf)
{
  GstClockTime start_time, end_time;
  gboolean discont = FALSE;
  guint64 start_offset, end_offset;
  gint rate, bpf;

  GstAggregator *agg = GST_AGGREGATOR (aagg);
  GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);

  g_assert (pad->priv->buffer == NULL);

  rate = GST_AUDIO_INFO_RATE (&pad->info);
  bpf = GST_AUDIO_INFO_BPF (&pad->info);

  pad->priv->position = 0;
  pad->priv->size = gst_buffer_get_size (inbuf) / bpf;

  if (!GST_BUFFER_PTS_IS_VALID (inbuf)) {
    if (pad->priv->output_offset == -1)
      pad->priv->output_offset = aagg->priv->offset;
    if (pad->priv->next_offset == -1)
      pad->priv->next_offset = pad->priv->size;
    else
      pad->priv->next_offset += pad->priv->size;
    goto done;
  }

  start_time = GST_BUFFER_PTS (inbuf);
  end_time =
      start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
      rate);

  /* Clipping should've ensured this */
  g_assert (start_time >= aggpad->segment.start);

  start_offset =
      gst_util_uint64_scale (start_time - aggpad->segment.start, rate,
      GST_SECOND);
  end_offset = start_offset + pad->priv->size;

  if (GST_BUFFER_IS_DISCONT (inbuf)
      || GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
      || pad->priv->new_segment || pad->priv->next_offset == -1) {
    discont = TRUE;
    pad->priv->new_segment = FALSE;
  } else {
    guint64 diff, max_sample_diff;

    /* Check discont, based on audiobasesink */
    if (start_offset <= pad->priv->next_offset)
      diff = pad->priv->next_offset - start_offset;
    else
      diff = start_offset - pad->priv->next_offset;

    max_sample_diff =
        gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate,
        GST_SECOND);

    /* Discont! */
    if (G_UNLIKELY (diff >= max_sample_diff)) {
      if (aagg->priv->discont_wait > 0) {
        if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) {
          pad->priv->discont_time = start_time;
        } else if (start_time - pad->priv->discont_time >=
            aagg->priv->discont_wait) {
          discont = TRUE;
          pad->priv->discont_time = GST_CLOCK_TIME_NONE;
        }
      } else {
        discont = TRUE;
      }
    } else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) {
      /* we have had a discont, but are now back on track! */
      pad->priv->discont_time = GST_CLOCK_TIME_NONE;
    }
  }

  if (discont) {
    /* Have discont, need resync */
    if (pad->priv->next_offset != -1)
      GST_INFO_OBJECT (pad, "Have discont. Expected %"
          G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
          pad->priv->next_offset, start_offset);
    pad->priv->output_offset = -1;
    pad->priv->next_offset = end_offset;
  } else {
    pad->priv->next_offset += pad->priv->size;
  }

  if (pad->priv->output_offset == -1) {
    GstClockTime start_running_time;
    GstClockTime end_running_time;
    guint64 start_output_offset;
    guint64 end_output_offset;

    start_running_time =
        gst_segment_to_running_time (&aggpad->segment,
        GST_FORMAT_TIME, start_time);
    end_running_time =
        gst_segment_to_running_time (&aggpad->segment,
        GST_FORMAT_TIME, end_time);

    /* Convert to position in the output segment */
    start_output_offset =
        gst_segment_to_position (&agg->segment, GST_FORMAT_TIME,
        start_running_time);
    if (start_output_offset != -1)
      start_output_offset =
          gst_util_uint64_scale (start_output_offset - agg->segment.start, rate,
          GST_SECOND);

    end_output_offset =
        gst_segment_to_position (&agg->segment, GST_FORMAT_TIME,
        end_running_time);
    if (end_output_offset != -1)
      end_output_offset =
          gst_util_uint64_scale (end_output_offset - agg->segment.start, rate,
          GST_SECOND);

    if (start_output_offset == -1 && end_output_offset == -1) {
      /* Outside output segment, drop */
      gst_buffer_unref (inbuf);
      pad->priv->buffer = NULL;
      pad->priv->position = 0;
      pad->priv->size = 0;
      pad->priv->output_offset = -1;
      GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
      return FALSE;
    }

    /* Calculate end_output_offset if it was outside the output segment */
    if (end_output_offset == -1)
      end_output_offset = start_output_offset + pad->priv->size;

    if (end_output_offset < aagg->priv->offset) {
      /* Before output segment, drop */
      gst_buffer_unref (inbuf);
      pad->priv->buffer = NULL;
      pad->priv->position = 0;
      pad->priv->size = 0;
      pad->priv->output_offset = -1;
      GST_DEBUG_OBJECT (pad,
          "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
          G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
      return FALSE;
    }

    if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) {
      guint diff;

      if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
        diff = pad->priv->size - end_output_offset + aagg->priv->offset;
      } else if (start_output_offset == -1) {
        start_output_offset = end_output_offset - pad->priv->size;

        if (start_output_offset < aagg->priv->offset)
          diff = aagg->priv->offset - start_output_offset;
        else
          diff = 0;
      } else {
        diff = aagg->priv->offset - start_output_offset;
      }

      pad->priv->position += diff;
      if (pad->priv->position >= pad->priv->size) {
        /* Empty buffer, drop */
        gst_buffer_unref (inbuf);
        pad->priv->buffer = NULL;
        pad->priv->position = 0;
        pad->priv->size = 0;
        pad->priv->output_offset = -1;
        GST_DEBUG_OBJECT (pad,
            "Buffer before segment or current position: %" G_GUINT64_FORMAT
            " < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset);
        return FALSE;
      }
    }

    if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
      pad->priv->output_offset = aagg->priv->offset;
    else
      pad->priv->output_offset = start_output_offset;

    GST_DEBUG_OBJECT (pad,
        "Buffer resynced: Pad offset %" G_GUINT64_FORMAT
        ", current audio aggregator offset %" G_GINT64_FORMAT,
        pad->priv->output_offset, aagg->priv->offset);
  }

done:

  GST_LOG_OBJECT (pad,
      "Queued new buffer at offset %" G_GUINT64_FORMAT,
      pad->priv->output_offset);
  pad->priv->buffer = inbuf;

  return TRUE;
}