예제 #1
0
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstRingBuffer * buf)
{
  GstJackAudioSrc *src;
  GstJackRingBuffer *abuf;
  gint res;

  abuf = GST_JACK_RING_BUFFER_CAST (buf);
  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));

  GST_DEBUG_OBJECT (src, "release");

  if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
    /* we only warn, this means the server is probably shut down and the client
     * is gone anyway. */
    GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
        ("Could not deactivate Jack client (%d)", res));
  }

  abuf->channels = -1;
  abuf->buffer_size = -1;
  abuf->sample_rate = -1;

  /* free the buffer */
  gst_buffer_unref (buf->data);
  buf->data = NULL;

  return TRUE;
}
예제 #2
0
static void
gst_jack_ring_buffer_finalize (GObject * object)
{
  GstJackRingBuffer *ringbuffer;
  ringbuffer = GST_JACK_RING_BUFFER_CAST (object);
  G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
예제 #3
0
/* this is the callback of jack. This should be RT-safe.
 * Writes samples from the jack input port's buffer to the gst ring buffer.
 */
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
  GstJackAudioSrc *src;
  GstRingBuffer *buf;
  GstJackRingBuffer *abuf;
  gint len, givenLen;
  guint8 *writeptr, *dataStart;
  gint writeseg;
  gint channels, i, j;
  sample_t **buffers, *data;

  buf = GST_RING_BUFFER_CAST (arg);
  abuf = GST_JACK_RING_BUFFER_CAST (arg);
  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));

  channels = buf->spec.channels;
  len = sizeof (sample_t) * nframes * channels;

  /* alloc pointers to samples */
  buffers = g_alloca (sizeof (sample_t *) * channels);
  data = g_alloca (len);

  /* get input buffers */
  for (i = 0; i < channels; i++)
    buffers[i] = (sample_t *) jack_port_get_buffer (src->ports[i], nframes);

  //writeptr = data; 
  dataStart = (guint8 *) data;

  /* the samples in the jack input buffers have to be interleaved into the 
   * ringbuffer 
   */

  for (i = 0; i < nframes; ++i)
    for (j = 0; j < channels; ++j)
      *data++ = buffers[j][i];

  if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &givenLen)) {
    memcpy (writeptr, (char *) dataStart, givenLen);

    GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
        len / channels, channels);

    /* clear written samples in the ringbuffer */
    // gst_ring_buffer_clear(buf, 0);

    /* we wrote one segment */
    gst_ring_buffer_advance (buf, 1);
  }
  return 0;
}
예제 #4
0
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
  GstJackAudioSrc *src;
  GstJackRingBuffer *abuf;

  abuf = GST_JACK_RING_BUFFER_CAST (arg);
  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));

  if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
    goto not_supported;

  return 0;

  /* ERRORS */
not_supported:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
        (NULL), ("Jack changed the buffer size, which is not supported"));
    return 1;
  }
}
예제 #5
0
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
  GstJackAudioSink *sink;
  GstJackRingBuffer *abuf;

  abuf = GST_JACK_RING_BUFFER_CAST (arg);
  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));

  if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
    goto not_supported;

  return 0;

  /* ERRORS */
not_supported:
  {
    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
        (NULL), ("Jack changed the sample rate, which is not supported"));
    return 1;
  }
}
예제 #6
0
/* allocate a buffer and setup resources to process the audio samples of
 * the format as specified in @spec.
 *
 * We allocate N jack ports, one for each channel. If we are asked to
 * automatically make a connection with physical ports, we connect as many
 * ports as there are physical ports, leaving leftover ports unconnected.
 *
 * It is assumed that samplerate and number of channels are acceptable since our
 * getcaps method will always provide correct values. If unacceptable caps are
 * received for some reason, we fail here.
 */
static gboolean
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
  GstJackAudioSrc *src;
  GstJackRingBuffer *abuf;
  const char **ports;
  gint sample_rate, buffer_size;
  gint i, channels, res;
  jack_client_t *client;

  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
  abuf = GST_JACK_RING_BUFFER_CAST (buf);

  GST_DEBUG_OBJECT (src, "acquire");

  client = gst_jack_audio_client_get_client (src->client);

  /* sample rate must be that of the server */
  sample_rate = jack_get_sample_rate (client);
  if (sample_rate != spec->rate)
    goto wrong_samplerate;

  channels = spec->channels;

  if (!gst_jack_audio_src_allocate_channels (src, channels))
    goto out_of_ports;

  buffer_size = jack_get_buffer_size (client);

  /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
   * for all channels  */
  spec->segsize = buffer_size * sizeof (gfloat) * channels;
  spec->latency_time = gst_util_uint64_scale (spec->segsize,
      (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
  /* segtotal based on buffer-time latency */
  spec->segtotal = spec->buffer_time / spec->latency_time;

  GST_DEBUG_OBJECT (src, "segsize %d, segtotal %d", spec->segsize,
      spec->segtotal);

  /* allocate the ringbuffer memory now */
  buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
  memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));

  if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
    goto could_not_activate;

  /* if we need to automatically connect the ports, do so now. We must do this
   * after activating the client. */
  if (src->connect == GST_JACK_CONNECT_AUTO) {
    /* find all the physical output ports. A physical output port is a port
     * associated with a hardware device. Someone needs connect to a physical
     * port in order to capture something. */
    ports =
        jack_get_ports (client, NULL, NULL,
        JackPortIsPhysical | JackPortIsOutput);
    if (ports == NULL) {
      /* no ports? fine then we don't do anything except for posting a warning
       * message. */
      GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
          ("No physical output ports found, leaving ports unconnected"));
      goto done;
    }

    for (i = 0; i < channels; i++) {
      /* stop when all output ports are exhausted */
      if (ports[i] == NULL) {
        /* post a warning that we could not connect all ports */
        GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
            ("No more physical ports, leaving some ports unconnected"));
        break;
      }
      GST_DEBUG_OBJECT (src, "try connecting to %s",
          jack_port_name (src->ports[i]));
      /* connect the physical port to a port */

      res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
      g_print ("connecting to %s\n", jack_port_name (src->ports[i]));
      if (res != 0 && res != EEXIST)
        goto cannot_connect;
    }
    free (ports);
  }
done:

  abuf->sample_rate = sample_rate;
  abuf->buffer_size = buffer_size;
  abuf->channels = spec->channels;

  return TRUE;

  /* ERRORS */
wrong_samplerate:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
        ("Wrong samplerate, server is running at %d and we received %d",
            sample_rate, spec->rate));
    return FALSE;
  }
out_of_ports:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
        ("Cannot allocate more Jack ports"));
    return FALSE;
  }
could_not_activate:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
        ("Could not activate client (%d:%s)", res, g_strerror (res)));
    return FALSE;
  }
cannot_connect:
  {
    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
        ("Could not connect input ports to physical ports (%d:%s)",
            res, g_strerror (res)));
    free (ports);
    return FALSE;
  }
}