Пример #1
0
static gboolean
aggregate_func (GstAggregator * self)
{
  GstAggregatorPrivate *priv = self->priv;
  GstAggregatorClass *klass = GST_AGGREGATOR_GET_CLASS (self);

  GST_LOG_OBJECT (self, "Checking aggregate");
  while (priv->send_eos && gst_aggregator_iterate_sinkpads (self,
          (GstAggregatorPadForeachFunc) _check_all_pads_with_data_or_eos,
          NULL) && priv->running) {
    GST_TRACE_OBJECT (self, "Actually aggregating!");

    priv->flow_return = klass->aggregate (self);

    if (priv->flow_return == GST_FLOW_EOS) {
      g_main_context_wakeup (self->priv->mcontext);
      _remove_all_sources (self);
      _push_eos (self);
    }

    if (priv->flow_return == GST_FLOW_FLUSHING &&
        g_atomic_int_get (&priv->flush_seeking))
      priv->flow_return = GST_FLOW_OK;

    GST_LOG_OBJECT (self, "flow return is %s",
        gst_flow_get_name (priv->flow_return));

    if (priv->flow_return != GST_FLOW_OK)
      break;
  }

  return G_SOURCE_REMOVE;
}
Пример #2
0
static gboolean
_stop (GstAggregator * agg)
{
  _reset_flow_values (agg);

  gst_aggregator_iterate_sinkpads (agg,
      (GstAggregatorPadForeachFunc) _flush_pad, NULL);

  return TRUE;
}
Пример #3
0
static gboolean
gst_gl_mixer_stop (GstAggregator * agg)
{
  GstGLMixer *mix = GST_GL_MIXER (agg);
  GstGLMixerClass *mixer_class = GST_GL_MIXER_GET_CLASS (mix);

  if (!GST_AGGREGATOR_CLASS (parent_class)->stop (agg))
    return FALSE;

  GST_OBJECT_LOCK (agg);
  g_ptr_array_free (mix->frames, TRUE);
  mix->frames = NULL;
  g_ptr_array_free (mix->array_buffers, TRUE);
  mix->array_buffers = NULL;
  GST_OBJECT_UNLOCK (agg);

  if (mixer_class->reset)
    mixer_class->reset (mix);
  if (mix->fbo) {
    gst_gl_context_del_fbo (mix->context, mix->fbo, mix->depthbuffer);
    mix->fbo = 0;
    mix->depthbuffer = 0;
  }
  if (mix->download) {
    gst_object_unref (mix->download);
    mix->download = NULL;
  }

  gst_aggregator_iterate_sinkpads (GST_AGGREGATOR (mix), _clean_upload, NULL);

  if (mix->priv->query) {
    gst_query_unref (mix->priv->query);
    mix->priv->query = NULL;
  }

  if (mix->priv->pool) {
    gst_object_unref (mix->priv->pool);
    mix->priv->pool = NULL;
  }

  if (mix->display) {
    gst_object_unref (mix->display);
    mix->display = NULL;
  }

  if (mix->context) {
    gst_object_unref (mix->context);
    mix->context = NULL;
  }
  gst_gl_mixer_reset (mix);

  return TRUE;
}
Пример #4
0
static gboolean
_stop (GstAggregator * agg)
{
  _reset_flow_values (agg);

  gst_aggregator_iterate_sinkpads (agg,
      (GstAggregatorPadForeachFunc) _stop_pad, NULL);

  if (agg->priv->tags)
    gst_tag_list_unref (agg->priv->tags);
  agg->priv->tags = NULL;

  return TRUE;
}
Пример #5
0
static void
_reset_gl (GstGLContext * context, GstGLVideoMixer * video_mixer)
{
  const GstGLFuncs *gl = GST_GL_BASE_MIXER (video_mixer)->context->gl_vtable;

  if (video_mixer->vao) {
    gl->DeleteVertexArrays (1, &video_mixer->vao);
    video_mixer->vao = 0;
  }

  if (video_mixer->vbo_indices) {
    gl->DeleteBuffers (1, &video_mixer->vbo_indices);
    video_mixer->vbo_indices = 0;
  }

  gst_aggregator_iterate_sinkpads (GST_AGGREGATOR (video_mixer), _reset_pad_gl,
      NULL);
}
Пример #6
0
static gboolean
gst_aggregator_query_latency (GstAggregator * self, GstQuery * query)
{
  LatencyData data;

  data.min = 0;
  data.max = GST_CLOCK_TIME_NONE;
  data.live = FALSE;

  /* query upstream's latency */
  gst_aggregator_iterate_sinkpads (self,
      (GstAggregatorPadForeachFunc) _latency_query, &data);

  if (data.live && GST_CLOCK_TIME_IS_VALID (self->timeout) &&
      self->timeout > data.max) {
    GST_ELEMENT_WARNING (self, CORE, NEGOTIATION,
        ("%s", "Timeout too big"),
        ("The requested timeout value is too big for the latency in the "
            "current pipeline.  Limiting to %" G_GINT64_FORMAT, data.max));
    self->timeout = data.max;
  }

  self->priv->latency_live = data.live;
  self->priv->latency_min = data.min;
  self->priv->latency_max = data.max;

  /* add our own */
  if (GST_CLOCK_TIME_IS_VALID (self->timeout)) {
    if (GST_CLOCK_TIME_IS_VALID (data.min))
      data.min += self->timeout;
    if (GST_CLOCK_TIME_IS_VALID (data.max))
      data.max += self->timeout;
  }

  GST_DEBUG_OBJECT (self, "configured latency live:%s min:%" G_GINT64_FORMAT
      " max:%" G_GINT64_FORMAT, data.live ? "true" : "false", data.min,
      data.max);

  gst_query_set_latency (query, data.live, data.min, data.max);

  return TRUE;
}
Пример #7
0
static void
aggregate_func (GstAggregator * self)
{
  GstAggregatorPrivate *priv = self->priv;
  GstAggregatorClass *klass = GST_AGGREGATOR_GET_CLASS (self);

  if (self->priv->running == FALSE) {
    GST_DEBUG_OBJECT (self, "Not running anymore");

    return;
  }

  QUEUE_POP (self);

  GST_LOG_OBJECT (self, "Checking aggregate");
  while (priv->send_eos && gst_aggregator_iterate_sinkpads (self,
          (GstAggregatorPadForeachFunc)
          _check_all_pads_with_data_or_eos_or_timeout, NULL) && priv->running) {
    GST_TRACE_OBJECT (self, "Actually aggregating!");

    priv->flow_return = klass->aggregate (self);

    if (priv->flow_return == GST_FLOW_EOS) {
      QUEUE_FLUSH (self);
      _push_eos (self);
    }

    if (priv->flow_return == GST_FLOW_FLUSHING &&
        g_atomic_int_get (&priv->flush_seeking))
      priv->flow_return = GST_FLOW_OK;

    GST_LOG_OBJECT (self, "flow return is %s",
        gst_flow_get_name (priv->flow_return));

    if (priv->flow_return != GST_FLOW_OK)
      break;
  }

}
Пример #8
0
static gboolean
_forward_event_to_all_sinkpads (GstAggregator * self, GstEvent * event,
    gboolean flush)
{
  EventData evdata;

  evdata.event = event;
  evdata.result = TRUE;
  evdata.flush = flush;

  /* We first need to set all pads as flushing in a first pass
   * as flush_start flush_stop is sometimes sent synchronously
   * while we send the seek event */
  if (flush)
    gst_aggregator_iterate_sinkpads (self,
        (GstAggregatorPadForeachFunc) _set_flush_pending, NULL);
  gst_pad_forward (self->srcpad, (GstPadForwardFunction) event_forward_func,
      &evdata);

  gst_event_unref (event);

  return evdata.result;
}
Пример #9
0
static GstFlowReturn
gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
  /* Get all pads that have data for us and store them in a
   * new list.
   *
   * Calculate the current output offset/timestamp and
   * offset_end/timestamp_end. Allocate a silence buffer
   * for this and store it.
   *
   * For all pads:
   * 1) Once per input buffer (cached)
   *   1) Check discont (flag and timestamp with tolerance)
   *   2) If discont or new, resync. That means:
   *     1) Drop all start data of the buffer that comes before
   *        the current position/offset.
   *     2) Calculate the offset (output segment!) that the first
   *        frame of the input buffer corresponds to. Base this on
   *        the running time.
   *
   * 2) If the current pad's offset/offset_end overlaps with the output
   *    offset/offset_end, mix it at the appropiate position in the output
   *    buffer and advance the pad's position. Remember if this pad needs
   *    a new buffer to advance behind the output offset_end.
   *
   * 3) If we had no pad with a buffer, go EOS.
   *
   * 4) If we had at least one pad that did not advance behind output
   *    offset_end, let collected be called again for the current
   *    output offset/offset_end.
   */
  GstElement *element;
  GstAudioAggregator *aagg;
  GList *iter;
  GstFlowReturn ret;
  GstBuffer *outbuf = NULL;
  gint64 next_offset;
  gint64 next_timestamp;
  gint rate, bpf;
  gboolean dropped = FALSE;
  gboolean is_eos = TRUE;
  gboolean is_done = TRUE;
  guint blocksize;

  element = GST_ELEMENT (agg);
  aagg = GST_AUDIO_AGGREGATOR (agg);

  /* Sync pad properties to the stream time */
  gst_aggregator_iterate_sinkpads (agg,
      (GstAggregatorPadForeachFunc) GST_DEBUG_FUNCPTR (sync_pad_values), NULL);

  GST_AUDIO_AGGREGATOR_LOCK (aagg);
  GST_OBJECT_LOCK (agg);

  /* Update position from the segment start/stop if needed */
  if (agg->segment.position == -1) {
    if (agg->segment.rate > 0.0)
      agg->segment.position = agg->segment.start;
    else
      agg->segment.position = agg->segment.stop;
  }

  if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
    if (timeout) {
      GST_DEBUG_OBJECT (aagg,
          "Got timeout before receiving any caps, don't output anything");

      /* Advance position */
      if (agg->segment.rate > 0.0)
        agg->segment.position += aagg->priv->output_buffer_duration;
      else if (agg->segment.position > aagg->priv->output_buffer_duration)
        agg->segment.position -= aagg->priv->output_buffer_duration;
      else
        agg->segment.position = 0;

      GST_OBJECT_UNLOCK (agg);
      GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
      return GST_FLOW_OK;
    } else {
      GST_OBJECT_UNLOCK (agg);
      goto not_negotiated;
    }
  }

  if (aagg->priv->send_caps) {
    GST_OBJECT_UNLOCK (agg);
    gst_aggregator_set_src_caps (agg, aagg->current_caps);
    GST_OBJECT_LOCK (agg);

    aagg->priv->send_caps = FALSE;
  }

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

  if (aagg->priv->offset == -1) {
    aagg->priv->offset =
        gst_util_uint64_scale (agg->segment.position - agg->segment.start, rate,
        GST_SECOND);
    GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT,
        aagg->priv->offset);
  }

  blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
      rate, GST_SECOND);
  blocksize = MAX (1, blocksize);

  /* for the next timestamp, use the sample counter, which will
   * never accumulate rounding errors */

  /* FIXME: Reverse mixing does not work at all yet */
  if (agg->segment.rate > 0.0) {
    next_offset = aagg->priv->offset + blocksize;
  } else {
    next_offset = aagg->priv->offset - blocksize;
  }

  next_timestamp =
      agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
      rate);

  if (aagg->priv->current_buffer == NULL) {
    GST_OBJECT_UNLOCK (agg);
    aagg->priv->current_buffer =
        GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
        blocksize);
    /* Be careful, some things could have changed ? */
    GST_OBJECT_LOCK (agg);
    GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
  }
  outbuf = aagg->priv->current_buffer;

  GST_LOG_OBJECT (agg,
      "Starting to mix %u samples for offset %" G_GINT64_FORMAT
      " with timestamp %" GST_TIME_FORMAT, blocksize,
      aagg->priv->offset, GST_TIME_ARGS (agg->segment.position));

  for (iter = element->sinkpads; iter; iter = iter->next) {
    GstBuffer *inbuf;
    GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
    GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
    gboolean drop_buf = FALSE;
    gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);

    if (!pad_eos)
      is_eos = FALSE;

    inbuf = gst_aggregator_pad_get_buffer (aggpad);

    GST_OBJECT_LOCK (pad);
    if (!inbuf) {
      if (timeout) {
        if (pad->priv->output_offset < next_offset) {
          gint64 diff = next_offset - pad->priv->output_offset;
          GST_LOG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT " frames (%"
              GST_TIME_FORMAT ")", diff,
              GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
                      GST_AUDIO_INFO_RATE (&aagg->info))));
        }
      } else if (!pad_eos) {
        is_done = FALSE;
      }
      GST_OBJECT_UNLOCK (pad);
      continue;
    }

    g_assert (!pad->priv->buffer || pad->priv->buffer == inbuf);

    /* New buffer? */
    if (!pad->priv->buffer) {
      /* Takes ownership of buffer */
      if (!gst_audio_aggregator_fill_buffer (aagg, pad, inbuf)) {
        dropped = TRUE;
        GST_OBJECT_UNLOCK (pad);
        gst_aggregator_pad_drop_buffer (aggpad);
        continue;
      }
    } else {
      gst_buffer_unref (inbuf);
    }

    if (!pad->priv->buffer && !dropped && pad_eos) {
      GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
      GST_OBJECT_UNLOCK (pad);
      continue;
    }

    g_assert (pad->priv->buffer);

    /* This pad is lacking behind, we need to update the offset
     * and maybe drop the current buffer */
    if (pad->priv->output_offset < aagg->priv->offset) {
      gint64 diff = aagg->priv->offset - pad->priv->output_offset;
      gint64 odiff = diff;

      if (pad->priv->position + diff > pad->priv->size)
        diff = pad->priv->size - pad->priv->position;
      pad->priv->position += diff;
      pad->priv->output_offset += diff;

      if (pad->priv->position == pad->priv->size) {
        GST_LOG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
            ", dropping %" GST_PTR_FORMAT,
            GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
                    GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
        /* Buffer done, drop it */
        gst_buffer_replace (&pad->priv->buffer, NULL);
        dropped = TRUE;
        GST_OBJECT_UNLOCK (pad);
        gst_aggregator_pad_drop_buffer (aggpad);
        continue;
      }
    }


    if (pad->priv->output_offset >= aagg->priv->offset
        && pad->priv->output_offset <
        aagg->priv->offset + blocksize && pad->priv->buffer) {
      GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
      drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
          outbuf);
      if (pad->priv->output_offset >= next_offset) {
        GST_DEBUG_OBJECT (pad,
            "Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
            G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
      } else {
        is_done = FALSE;
      }
    }

    GST_OBJECT_UNLOCK (pad);
    if (drop_buf)
      gst_aggregator_pad_drop_buffer (aggpad);

  }
  GST_OBJECT_UNLOCK (agg);

  if (dropped) {
    /* We dropped a buffer, retry */
    GST_INFO_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
    GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
    return GST_FLOW_OK;
  }

  if (!is_done && !is_eos) {
    /* Get more buffers */
    GST_INFO_OBJECT (aagg,
        "We're not done yet for the current offset," " waiting for more data");
    GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
    return GST_FLOW_OK;
  }

  if (is_eos) {
    gint64 max_offset = 0;

    GST_DEBUG_OBJECT (aagg, "We're EOS");

    GST_OBJECT_LOCK (agg);
    for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
      GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);

      max_offset = MAX ((gint64) max_offset, (gint64) pad->priv->output_offset);
    }
    GST_OBJECT_UNLOCK (agg);

    /* This means EOS or nothing mixed in at all */
    if (aagg->priv->offset == max_offset) {
      gst_buffer_replace (&aagg->priv->current_buffer, NULL);
      GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
      return GST_FLOW_EOS;
    }

    if (max_offset <= next_offset) {
      GST_DEBUG_OBJECT (aagg,
          "Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
          G_GINT64_FORMAT, max_offset, next_offset);
      next_offset = max_offset;
      next_timestamp =
          agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
          rate);

      if (next_offset > aagg->priv->offset)
        gst_buffer_resize (outbuf, 0, (next_offset - aagg->priv->offset) * bpf);
    }
  }

  /* set timestamps on the output buffer */
  GST_OBJECT_LOCK (agg);
  if (agg->segment.rate > 0.0) {
    GST_BUFFER_PTS (outbuf) = agg->segment.position;
    GST_BUFFER_OFFSET (outbuf) = aagg->priv->offset;
    GST_BUFFER_OFFSET_END (outbuf) = next_offset;
    GST_BUFFER_DURATION (outbuf) = next_timestamp - agg->segment.position;
  } else {
    GST_BUFFER_PTS (outbuf) = next_timestamp;
    GST_BUFFER_OFFSET (outbuf) = next_offset;
    GST_BUFFER_OFFSET_END (outbuf) = aagg->priv->offset;
    GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp;
  }

  GST_OBJECT_UNLOCK (agg);

  /* send it out */
  GST_LOG_OBJECT (aagg,
      "pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
      G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
      GST_BUFFER_OFFSET (outbuf));

  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);

  ret = gst_aggregator_finish_buffer (agg, aagg->priv->current_buffer);
  aagg->priv->current_buffer = NULL;

  GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret));

  GST_AUDIO_AGGREGATOR_LOCK (aagg);
  GST_OBJECT_LOCK (agg);
  aagg->priv->offset = next_offset;
  agg->segment.position = next_timestamp;

  /* If there was a timeout and there was a gap in data in out of the streams,
   * then it's a very good time to for a resync with the timestamps.
   */
  if (timeout) {
    for (iter = element->sinkpads; iter; iter = iter->next) {
      GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);

      GST_OBJECT_LOCK (pad);
      if (pad->priv->output_offset < aagg->priv->offset)
        pad->priv->output_offset = -1;
      GST_OBJECT_UNLOCK (pad);
    }
  }
  GST_OBJECT_UNLOCK (agg);
  GST_AUDIO_AGGREGATOR_UNLOCK (aagg);

  return ret;
  /* ERRORS */
not_negotiated:
  {
    GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
    GST_ELEMENT_ERROR (aagg, STREAM, FORMAT, (NULL),
        ("Unknown data received, not negotiated"));
    return GST_FLOW_NOT_NEGOTIATED;
  }
}