コード例 #1
0
static GstStateChangeReturn
gst_dv1394_src_change_state (GstElement * element, GstStateChange transition)
{
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
  GstDV1394Src *src = GST_DV1394SRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      gst_element_post_message (element,
          gst_message_new_clock_lost (GST_OBJECT_CAST (element),
              GST_CLOCK_CAST (src->provided_clock)));
      break;
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
  if (ret == GST_STATE_CHANGE_FAILURE)
    return ret;

  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      gst_element_post_message (element,
          gst_message_new_clock_provide (GST_OBJECT_CAST (element),
              GST_CLOCK_CAST (src->provided_clock), TRUE));
      break;
    default:
      break;
  }

  return ret;
}
コード例 #2
0
ファイル: rbgst-message.c プロジェクト: msakai/ruby-gnome2
static VALUE
clock_provide_initialize(VALUE self, VALUE src, VALUE clock, VALUE ready)
{
    G_INITIALIZE(self, gst_message_new_clock_provide(RVAL2GST_OBJ(src),
                                                     RVAL2GST_CLOCK(clock),
                                                     RVAL2CBOOL(ready)));
    return Qnil;
}
コード例 #3
0
static GstStateChangeReturn
gst_decklink_video_src_change_state (GstElement * element,
    GstStateChange transition)
{
  GstDecklinkVideoSrc *self = GST_DECKLINK_VIDEO_SRC_CAST (element);
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      if (!gst_decklink_video_src_open (self)) {
        ret = GST_STATE_CHANGE_FAILURE;
        goto out;
      }
      if (self->mode == GST_DECKLINK_MODE_AUTO &&
          self->video_format != GST_DECKLINK_VIDEO_FORMAT_AUTO) {
        GST_WARNING_OBJECT (self, "Warning: mode=auto and format!=auto may \
                            not work");
      }
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      g_mutex_lock (&self->input->lock);
      self->input->clock_start_time = GST_CLOCK_TIME_NONE;
      self->input->clock_epoch += self->input->clock_last_time;
      self->input->clock_last_time = 0;
      self->input->clock_offset = 0;
      g_mutex_unlock (&self->input->lock);
      gst_element_post_message (element,
          gst_message_new_clock_provide (GST_OBJECT_CAST (element),
              self->input->clock, TRUE));
      self->flushing = FALSE;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      GstClock *clock;

      clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
      if (clock) {
        if (clock != self->input->clock) {
          gst_clock_set_master (self->input->clock, clock);
        }

        gst_object_unref (clock);
      } else {
        GST_ELEMENT_ERROR (self, STREAM, FAILED,
            (NULL), ("Need a clock to go to PLAYING"));
        ret = GST_STATE_CHANGE_FAILURE;
      }

      break;
    }
    default:
      break;
  }
コード例 #4
0
static GstStateChangeReturn
gst_decklink_video_sink_change_state (GstElement * element,
    GstStateChange transition)
{
  GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;

  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      g_mutex_lock (&self->output->lock);
      self->output->clock_start_time = GST_CLOCK_TIME_NONE;
      self->output->clock_epoch += self->output->clock_last_time;
      self->output->clock_last_time = 0;
      self->output->clock_offset = 0;
      g_mutex_unlock (&self->output->lock);
      gst_element_post_message (element,
          gst_message_new_clock_provide (GST_OBJECT_CAST (element),
              self->output->clock, TRUE));
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      GstClock *clock, *audio_clock;

      clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
      if (clock) {
        audio_clock = gst_decklink_output_get_audio_clock (self->output);
        if (clock && clock != self->output->clock && clock != audio_clock) {
          gst_clock_set_master (self->output->clock, clock);
        }
        gst_object_unref (clock);
        if (audio_clock)
          gst_object_unref (audio_clock);
      } else {
        GST_ELEMENT_ERROR (self, STREAM, FAILED,
            (NULL), ("Need a clock to go to PLAYING"));
        ret = GST_STATE_CHANGE_FAILURE;
      }

      break;
    }
    default:
      break;
  }

  if (ret == GST_STATE_CHANGE_FAILURE)
    return ret;
  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
  if (ret == GST_STATE_CHANGE_FAILURE)
    return ret;

  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      gst_element_post_message (element,
          gst_message_new_clock_lost (GST_OBJECT_CAST (element),
              self->output->clock));
      gst_clock_set_master (self->output->clock, NULL);
      // Reset calibration to make the clock reusable next time we use it
      gst_clock_set_calibration (self->output->clock, 0, 0, 1, 1);
      g_mutex_lock (&self->output->lock);
      self->output->clock_start_time = GST_CLOCK_TIME_NONE;
      self->output->clock_epoch += self->output->clock_last_time;
      self->output->clock_last_time = 0;
      self->output->clock_offset = 0;
      g_mutex_unlock (&self->output->lock);
      gst_decklink_video_sink_stop (self);
      break;
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:{
      if (gst_decklink_video_sink_stop_scheduled_playback (self) ==
          GST_STATE_CHANGE_FAILURE)
        ret = GST_STATE_CHANGE_FAILURE;
      break;
    }
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      g_mutex_lock (&self->output->lock);
      if (self->output->start_scheduled_playback)
        self->output->start_scheduled_playback (self->output->videosink);
      g_mutex_unlock (&self->output->lock);
      break;
    }
    default:
      break;
  }

  return ret;
}
コード例 #5
0
static GstStateChangeReturn
gst_decklink_video_sink_change_state (GstElement * element,
    GstStateChange transition)
{
  GstDecklinkVideoSink *self = GST_DECKLINK_VIDEO_SINK_CAST (element);
  GstStateChangeReturn ret;

  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      g_mutex_lock (&self->output->lock);
      self->output->clock_start_time = GST_CLOCK_TIME_NONE;
      self->output->clock_last_time = 0;
      self->output->clock_offset = 0;
      g_mutex_unlock (&self->output->lock);
      gst_element_post_message (element,
          gst_message_new_clock_provide (GST_OBJECT_CAST (element),
              self->output->clock, TRUE));
      self->last_render_time = GST_CLOCK_TIME_NONE;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      GstClock *clock, *audio_clock;

      clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
      audio_clock = gst_decklink_output_get_audio_clock (self->output);
      if (clock && clock != self->output->clock && clock != audio_clock) {
        gst_clock_set_master (self->output->clock, clock);
      }
      if (clock)
        gst_object_unref (clock);
      if (audio_clock)
        gst_object_unref (audio_clock);

      break;
    }
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
  if (ret == GST_STATE_CHANGE_FAILURE)
    return ret;

  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      gst_element_post_message (element,
          gst_message_new_clock_lost (GST_OBJECT_CAST (element),
              self->output->clock));
      gst_clock_set_master (self->output->clock, NULL);
      g_mutex_lock (&self->output->lock);
      self->output->clock_start_time = GST_CLOCK_TIME_NONE;
      self->output->clock_last_time = 0;
      self->output->clock_offset = 0;
      g_mutex_unlock (&self->output->lock);
      break;
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:{
      GstClockTime start_time;
      HRESULT res;

      // FIXME: start time is the same for the complete pipeline,
      // but what we need here is the start time of this element!
      start_time = gst_element_get_base_time (element);
      if (start_time != GST_CLOCK_TIME_NONE)
        start_time = gst_clock_get_time (GST_ELEMENT_CLOCK (self)) - start_time;

      // FIXME: This will probably not work
      if (start_time == GST_CLOCK_TIME_NONE)
        start_time = 0;

      convert_to_internal_clock (self, &start_time, NULL);

      // The start time is now the running time when we stopped
      // playback

      GST_DEBUG_OBJECT (self,
          "Stopping scheduled playback at %" GST_TIME_FORMAT,
          GST_TIME_ARGS (start_time));

      g_mutex_lock (&self->output->lock);
      self->output->started = FALSE;
      g_mutex_unlock (&self->output->lock);
      res =
          self->output->output->StopScheduledPlayback (start_time, 0,
          GST_SECOND);
      if (res != S_OK) {
        GST_ELEMENT_ERROR (self, STREAM, FAILED,
            (NULL), ("Failed to stop scheduled playback: 0x%08x", res));
        ret = GST_STATE_CHANGE_FAILURE;
      }
      self->internal_base_time = GST_CLOCK_TIME_NONE;
      self->external_base_time = GST_CLOCK_TIME_NONE;
      break;
    }
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      g_mutex_lock (&self->output->lock);
      if (self->output->start_scheduled_playback)
        self->output->start_scheduled_playback (self->output->videosink);
      g_mutex_unlock (&self->output->lock);
      break;
    }
    default:
      break;
  }

  return ret;
}
コード例 #6
0
ファイル: pulsesrc.c プロジェクト: PeterXu/gst-mobile
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
  GstStateChangeReturn ret;
  GstPulseSrc *this = GST_PULSESRC_CAST (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      if (!(this->mainloop = pa_threaded_mainloop_new ()))
        goto mainloop_failed;
      if (pa_threaded_mainloop_start (this->mainloop) < 0) {
        pa_threaded_mainloop_free (this->mainloop);
        this->mainloop = NULL;
        goto mainloop_start_failed;
      }
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      gst_element_post_message (element,
          gst_message_new_clock_provide (GST_OBJECT_CAST (element),
              GST_AUDIO_BASE_SRC (this)->clock, TRUE));
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      /* uncork and start recording */
      gst_pulsesrc_play (this);
      break;
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      /* stop recording ASAP by corking */
      pa_threaded_mainloop_lock (this->mainloop);
      GST_DEBUG_OBJECT (this, "corking");
      gst_pulsesrc_set_corked (this, TRUE, FALSE);
      pa_threaded_mainloop_unlock (this->mainloop);
      break;
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  switch (transition) {
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      /* now make sure we get out of the _read method */
      gst_pulsesrc_pause (this);
      break;
    case GST_STATE_CHANGE_READY_TO_NULL:
      if (this->mainloop)
        pa_threaded_mainloop_stop (this->mainloop);

      gst_pulsesrc_destroy_context (this);

      if (this->mainloop) {
        pa_threaded_mainloop_free (this->mainloop);
        this->mainloop = NULL;
      }
      break;
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      /* format_lost is reset in release() in baseaudiosink */
      gst_element_post_message (element,
          gst_message_new_clock_lost (GST_OBJECT_CAST (element),
              GST_AUDIO_BASE_SRC (this)->clock));
      break;
    default:
      break;
  }

  return ret;

  /* ERRORS */
mainloop_failed:
  {
    GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
        ("pa_threaded_mainloop_new() failed"), (NULL));
    return GST_STATE_CHANGE_FAILURE;
  }
mainloop_start_failed:
  {
    GST_ELEMENT_ERROR (this, RESOURCE, FAILED,
        ("pa_threaded_mainloop_start() failed"), (NULL));
    return GST_STATE_CHANGE_FAILURE;
  }
}
コード例 #7
0
static GstStateChangeReturn
gst_decklink_video_src_change_state (GstElement * element,
    GstStateChange transition)
{
  GstDecklinkVideoSrc *self = GST_DECKLINK_VIDEO_SRC_CAST (element);
  GstStateChangeReturn ret;

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      if (!gst_decklink_video_src_open (self)) {
        ret = GST_STATE_CHANGE_FAILURE;
        goto out;
      }
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      g_mutex_lock (&self->input->lock);
      self->input->clock_start_time = GST_CLOCK_TIME_NONE;
      self->input->clock_last_time = 0;
      self->input->clock_offset = 0;
      g_mutex_unlock (&self->input->lock);
      gst_element_post_message (element,
          gst_message_new_clock_provide (GST_OBJECT_CAST (element),
              self->input->clock, TRUE));
      self->flushing = FALSE;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      GstClock *clock;

      clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
      if (clock && clock != self->input->clock) {
        gst_clock_set_master (self->input->clock, clock);
      }
      if (clock)
        gst_object_unref (clock);

      break;
    }
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
  if (ret == GST_STATE_CHANGE_FAILURE)
    return ret;

  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      gst_element_post_message (element,
          gst_message_new_clock_lost (GST_OBJECT_CAST (element),
              self->input->clock));
      gst_clock_set_master (self->input->clock, NULL);
      // Reset calibration to make the clock reusable next time we use it
      gst_clock_set_calibration (self->input->clock, 0, 0, 1, 1);
      g_mutex_lock (&self->input->lock);
      self->input->clock_start_time = GST_CLOCK_TIME_NONE;
      self->input->clock_last_time = 0;
      self->input->clock_offset = 0;
      g_mutex_unlock (&self->input->lock);

      g_queue_foreach (&self->current_frames, (GFunc) capture_frame_free, NULL);
      g_queue_clear (&self->current_frames);
      self->caps_mode = GST_DECKLINK_MODE_AUTO;
      break;
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:{
      HRESULT res;

      GST_DEBUG_OBJECT (self, "Stopping streams");
      g_mutex_lock (&self->input->lock);
      self->input->started = FALSE;
      g_mutex_unlock (&self->input->lock);

      res = self->input->input->StopStreams ();
      if (res != S_OK) {
        GST_ELEMENT_ERROR (self, STREAM, FAILED,
            (NULL), ("Failed to stop streams: 0x%08x", res));
        ret = GST_STATE_CHANGE_FAILURE;
      }
      self->internal_base_time = GST_CLOCK_TIME_NONE;
      self->external_base_time = GST_CLOCK_TIME_NONE;
      break;
    }
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
      g_mutex_lock (&self->input->lock);
      if (self->input->start_streams)
        self->input->start_streams (self->input->videosrc);
      g_mutex_unlock (&self->input->lock);

      break;
    }
    case GST_STATE_CHANGE_READY_TO_NULL:
      gst_decklink_video_src_close (self);
      break;
    default:
      break;
  }
out:

  return ret;
}
コード例 #8
0
static GstStateChangeReturn
gst_base_audio_src_change_state (GstElement * element,
    GstStateChange transition)
{
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
  GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      GST_DEBUG_OBJECT (src, "NULL->READY");
      GST_OBJECT_LOCK (src);
      if (src->ringbuffer == NULL) {
        gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
        src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
      }
      GST_OBJECT_UNLOCK (src);
      if (!gst_ring_buffer_open_device (src->ringbuffer))
        goto open_failed;
      break;
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      GST_DEBUG_OBJECT (src, "READY->PAUSED");
      src->next_sample = -1;
      gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
      gst_ring_buffer_may_start (src->ringbuffer, FALSE);
      /* Only post clock-provide messages if this is the clock that
       * we've created. If the subclass has overriden it the subclass
       * should post this messages whenever necessary */
      if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
          GST_AUDIO_CLOCK_CAST (src->clock)->func ==
          (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
        gst_element_post_message (element,
            gst_message_new_clock_provide (GST_OBJECT_CAST (element),
                src->clock, TRUE));
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
      gst_ring_buffer_may_start (src->ringbuffer, TRUE);
      break;
    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
      GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
      gst_ring_buffer_may_start (src->ringbuffer, FALSE);
      gst_ring_buffer_pause (src->ringbuffer);
      break;
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      GST_DEBUG_OBJECT (src, "PAUSED->READY");
      /* Only post clock-lost messages if this is the clock that
       * we've created. If the subclass has overriden it the subclass
       * should post this messages whenever necessary */
      if (src->clock && GST_IS_AUDIO_CLOCK (src->clock) &&
          GST_AUDIO_CLOCK_CAST (src->clock)->func ==
          (GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
        gst_element_post_message (element,
            gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock));
      gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
      break;
    default:
      break;
  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  switch (transition) {
    case GST_STATE_CHANGE_PAUSED_TO_READY:
      GST_DEBUG_OBJECT (src, "PAUSED->READY");
      gst_ring_buffer_release (src->ringbuffer);
      break;
    case GST_STATE_CHANGE_READY_TO_NULL:
      GST_DEBUG_OBJECT (src, "READY->NULL");
      gst_ring_buffer_close_device (src->ringbuffer);
      GST_OBJECT_LOCK (src);
      gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
      src->ringbuffer = NULL;
      GST_OBJECT_UNLOCK (src);
      break;
    default:
      break;
  }

  return ret;

  /* ERRORS */
open_failed:
  {
    /* subclass must post a meaningfull error message */
    GST_DEBUG_OBJECT (src, "open failed");
    return GST_STATE_CHANGE_FAILURE;
  }

}