示例#1
0
static void
gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
{

  gst_pulsesrc_destroy_stream (pulsesrc);

  if (pulsesrc->context) {
    pa_context_disconnect (pulsesrc->context);
    pa_context_unref (pulsesrc->context);
    pulsesrc->context = NULL;
  }
}
示例#2
0
static gboolean
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);

  pa_threaded_mainloop_lock (pulsesrc->mainloop);
  gst_pulsesrc_destroy_stream (pulsesrc);

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  pulsesrc->read_buffer = NULL;
  pulsesrc->read_buffer_length = 0;

  return TRUE;
}
示例#3
0
static void
gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
{

  gst_pulsesrc_destroy_stream (pulsesrc);

  if (pulsesrc->context) {
    pa_context_disconnect (pulsesrc->context);

    /* Make sure we don't get any further callbacks */
    pa_context_set_state_callback (pulsesrc->context, NULL, NULL);
    pa_context_set_subscribe_callback (pulsesrc->context, NULL, NULL);

    pa_context_unref (pulsesrc->context);

    pulsesrc->context = NULL;
  }
}
示例#4
0
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
  pa_buffer_attr wanted;
  const pa_buffer_attr *actual;
  GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  wanted.maxlength = -1;
  wanted.tlength = -1;
  wanted.prebuf = 0;
  wanted.minreq = -1;
  wanted.fragsize = spec->segsize;

  GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
  GST_INFO_OBJECT (pulsesrc, "tlength:   %d", wanted.tlength);
  GST_INFO_OBJECT (pulsesrc, "prebuf:    %d", wanted.prebuf);
  GST_INFO_OBJECT (pulsesrc, "minreq:    %d", wanted.minreq);
  GST_INFO_OBJECT (pulsesrc, "fragsize:  %d", wanted.fragsize);

  if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
          PA_STREAM_INTERPOLATE_TIMING |
          PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_NOT_MONOTONOUS |
#ifdef HAVE_PULSE_0_9_11
          PA_STREAM_ADJUST_LATENCY |
#endif
          PA_STREAM_START_CORKED) < 0) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to connect stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  pulsesrc->corked = TRUE;

  for (;;) {
    pa_stream_state_t state;

    state = pa_stream_get_state (pulsesrc->stream);

    if (!PA_STREAM_IS_GOOD (state)) {
      GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
          ("Failed to connect stream: %s",
              pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
      goto unlock_and_fail;
    }

    if (state == PA_STREAM_READY)
      break;

    /* Wait until the stream is ready */
    pa_threaded_mainloop_wait (pulsesrc->mainloop);
  }

  /* get the actual buffering properties now */
  actual = pa_stream_get_buffer_attr (pulsesrc->stream);

  GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
  GST_INFO_OBJECT (pulsesrc, "tlength:   %d (wanted: %d)",
      actual->tlength, wanted.tlength);
  GST_INFO_OBJECT (pulsesrc, "prebuf:    %d", actual->prebuf);
  GST_INFO_OBJECT (pulsesrc, "minreq:    %d (wanted %d)", actual->minreq,
      wanted.minreq);
  GST_INFO_OBJECT (pulsesrc, "fragsize:  %d (wanted %d)",
      actual->fragsize, wanted.fragsize);

  if (actual->fragsize >= wanted.fragsize) {
    spec->segsize = actual->fragsize;
  } else {
    spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
  }
  spec->segtotal = actual->maxlength / spec->segsize;

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return TRUE;

unlock_and_fail:
  {
    gst_pulsesrc_destroy_stream (pulsesrc);

    pa_threaded_mainloop_unlock (pulsesrc->mainloop);
    return FALSE;
  }
}
示例#5
0
static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps * caps)
{
  pa_channel_map channel_map;
  GstStructure *s;
  gboolean need_channel_layout = FALSE;
  GstRingBufferSpec spec;
  const gchar *name;

  memset (&spec, 0, sizeof (GstRingBufferSpec));
  spec.latency_time = GST_SECOND;
  if (!gst_ring_buffer_parse_caps (&spec, caps)) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
        ("Can't parse caps."), (NULL));
    goto fail;
  }
  /* Keep the refcount of the caps at 1 to make them writable */
  gst_caps_unref (spec.caps);

  if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec)) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
        ("Invalid sample specification."), (NULL));
    goto fail;
  }

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  if (!pulsesrc->context) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
    goto unlock_and_fail;
  }

  s = gst_caps_get_structure (caps, 0);
  if (!gst_structure_has_field (s, "channel-layout") ||
      !gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
    if (spec.channels == 1)
      pa_channel_map_init_mono (&channel_map);
    else if (spec.channels == 2)
      pa_channel_map_init_stereo (&channel_map);
    else
      need_channel_layout = TRUE;
  }

  name = "Record Stream";
  if (pulsesrc->proplist) {
    if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
                name, &pulsesrc->sample_spec,
                (need_channel_layout) ? NULL : &channel_map,
                pulsesrc->proplist))) {
      GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
          ("Failed to create stream: %s",
              pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
      goto unlock_and_fail;
    }
  } else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
              name, &pulsesrc->sample_spec,
              (need_channel_layout) ? NULL : &channel_map))) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to create stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  if (need_channel_layout) {
    const pa_channel_map *m = pa_stream_get_channel_map (pulsesrc->stream);

    gst_pulse_channel_map_to_gst (m, &spec);
    caps = spec.caps;
  }

  GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, caps);

  pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
      pulsesrc);
  pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
      pulsesrc);
  pa_stream_set_underflow_callback (pulsesrc->stream,
      gst_pulsesrc_stream_underflow_cb, pulsesrc);
  pa_stream_set_overflow_callback (pulsesrc->stream,
      gst_pulsesrc_stream_overflow_cb, pulsesrc);
  pa_stream_set_latency_update_callback (pulsesrc->stream,
      gst_pulsesrc_stream_latency_update_cb, pulsesrc);

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return TRUE;

unlock_and_fail:
  gst_pulsesrc_destroy_stream (pulsesrc);

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

fail:
  return FALSE;
}
示例#6
0
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
  pa_buffer_attr wanted;
  const pa_buffer_attr *actual;
  GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
  pa_stream_flags_t flags;
  pa_operation *o;
  GstAudioClock *clock;

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  {
    GstAudioRingBufferSpec s = *spec;
    const pa_channel_map *m;

    m = pa_stream_get_channel_map (pulsesrc->stream);
    gst_pulse_channel_map_to_gst (m, &s);
    gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
        (pulsesrc)->ringbuffer, s.info.position);
  }

  /* enable event notifications */
  GST_LOG_OBJECT (pulsesrc, "subscribing to context events");
  if (!(o = pa_context_subscribe (pulsesrc->context,
              PA_SUBSCRIPTION_MASK_SOURCE_OUTPUT, NULL, NULL))) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("pa_context_subscribe() failed: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  pa_operation_unref (o);

  wanted.maxlength = -1;
  wanted.tlength = -1;
  wanted.prebuf = 0;
  wanted.minreq = -1;
  wanted.fragsize = spec->segsize;

  GST_INFO_OBJECT (pulsesrc, "maxlength: %d", wanted.maxlength);
  GST_INFO_OBJECT (pulsesrc, "tlength:   %d", wanted.tlength);
  GST_INFO_OBJECT (pulsesrc, "prebuf:    %d", wanted.prebuf);
  GST_INFO_OBJECT (pulsesrc, "minreq:    %d", wanted.minreq);
  GST_INFO_OBJECT (pulsesrc, "fragsize:  %d", wanted.fragsize);

  flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
      PA_STREAM_NOT_MONOTONIC | PA_STREAM_ADJUST_LATENCY |
      PA_STREAM_START_CORKED;

  if (pulsesrc->mute_set && pulsesrc->mute)
    flags |= PA_STREAM_START_MUTED;

  if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &wanted,
          flags) < 0) {
    goto connect_failed;
  }

  /* our clock will now start from 0 again */
  clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SRC (pulsesrc)->clock);
  gst_audio_clock_reset (clock, 0);

  pulsesrc->corked = TRUE;

  for (;;) {
    pa_stream_state_t state;

    state = pa_stream_get_state (pulsesrc->stream);

    if (!PA_STREAM_IS_GOOD (state))
      goto stream_is_bad;

    if (state == PA_STREAM_READY)
      break;

    /* Wait until the stream is ready */
    pa_threaded_mainloop_wait (pulsesrc->mainloop);
  }
  pulsesrc->stream_connected = TRUE;

  /* store the source output index so it can be accessed via a property */
  pulsesrc->source_output_idx = pa_stream_get_index (pulsesrc->stream);
  g_object_notify (G_OBJECT (pulsesrc), "source-output-index");

  if (pulsesrc->volume_set) {
    gst_pulsesrc_set_stream_volume (pulsesrc, pulsesrc->volume);
    pulsesrc->volume_set = FALSE;
  }

  /* get the actual buffering properties now */
  actual = pa_stream_get_buffer_attr (pulsesrc->stream);

  GST_INFO_OBJECT (pulsesrc, "maxlength: %d", actual->maxlength);
  GST_INFO_OBJECT (pulsesrc, "tlength:   %d (wanted: %d)",
      actual->tlength, wanted.tlength);
  GST_INFO_OBJECT (pulsesrc, "prebuf:    %d", actual->prebuf);
  GST_INFO_OBJECT (pulsesrc, "minreq:    %d (wanted %d)", actual->minreq,
      wanted.minreq);
  GST_INFO_OBJECT (pulsesrc, "fragsize:  %d (wanted %d)",
      actual->fragsize, wanted.fragsize);

  if (actual->fragsize >= wanted.fragsize) {
    spec->segsize = actual->fragsize;
  } else {
    spec->segsize = actual->fragsize * (wanted.fragsize / actual->fragsize);
  }
  spec->segtotal = actual->maxlength / spec->segsize;

  if (!pulsesrc->paused) {
    GST_DEBUG_OBJECT (pulsesrc, "uncorking because we are playing");
    gst_pulsesrc_set_corked (pulsesrc, FALSE, FALSE);
  }
  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return TRUE;

  /* ERRORS */
connect_failed:
  {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to connect stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }
stream_is_bad:
  {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to connect stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }
unlock_and_fail:
  {
    gst_pulsesrc_destroy_stream (pulsesrc);

    pa_threaded_mainloop_unlock (pulsesrc->mainloop);
    return FALSE;
  }
}
示例#7
0
static gboolean
gst_pulsesrc_create_stream (GstPulseSrc * pulsesrc, GstCaps ** caps)
{
  pa_channel_map channel_map;
  const pa_channel_map *m;
  GstStructure *s;
  gboolean need_channel_layout = FALSE;
  GstAudioRingBufferSpec spec;
  const gchar *name;

  s = gst_caps_get_structure (*caps, 0);
  gst_structure_get_int (s, "channels", &spec.info.channels);
  if (!gst_structure_has_field (s, "channel-mask")) {
    if (spec.info.channels == 1) {
      pa_channel_map_init_mono (&channel_map);
    } else if (spec.info.channels == 2) {
      gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
          GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_LEFT) |
          GST_AUDIO_CHANNEL_POSITION_MASK (FRONT_RIGHT), NULL);
      pa_channel_map_init_stereo (&channel_map);
    } else {
      need_channel_layout = TRUE;
      gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
          G_GUINT64_CONSTANT (0), NULL);
    }
  }

  memset (&spec, 0, sizeof (GstAudioRingBufferSpec));
  spec.latency_time = GST_SECOND;
  if (!gst_audio_ring_buffer_parse_caps (&spec, *caps))
    goto invalid_caps;

  /* Keep the refcount of the caps at 1 to make them writable */
  gst_caps_unref (spec.caps);

  if (!need_channel_layout
      && !gst_pulse_gst_to_channel_map (&channel_map, &spec)) {
    need_channel_layout = TRUE;
    gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK,
        G_GUINT64_CONSTANT (0), NULL);
    memset (spec.info.position, 0xff, sizeof (spec.info.position));
  }

  if (!gst_pulse_fill_sample_spec (&spec, &pulsesrc->sample_spec))
    goto invalid_spec;

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  if (!pulsesrc->context)
    goto bad_context;

  name = "Record Stream";
  if (pulsesrc->proplist) {
    if (!(pulsesrc->stream = pa_stream_new_with_proplist (pulsesrc->context,
                name, &pulsesrc->sample_spec,
                (need_channel_layout) ? NULL : &channel_map,
                pulsesrc->proplist)))
      goto create_failed;

  } else if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
              name, &pulsesrc->sample_spec,
              (need_channel_layout) ? NULL : &channel_map)))
    goto create_failed;

  m = pa_stream_get_channel_map (pulsesrc->stream);
  gst_pulse_channel_map_to_gst (m, &spec);
  gst_audio_channel_positions_to_valid_order (spec.info.position,
      spec.info.channels);
  gst_caps_unref (*caps);
  *caps = gst_audio_info_to_caps (&spec.info);

  GST_DEBUG_OBJECT (pulsesrc, "Caps are %" GST_PTR_FORMAT, *caps);

  pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
      pulsesrc);
  pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
      pulsesrc);
  pa_stream_set_underflow_callback (pulsesrc->stream,
      gst_pulsesrc_stream_underflow_cb, pulsesrc);
  pa_stream_set_overflow_callback (pulsesrc->stream,
      gst_pulsesrc_stream_overflow_cb, pulsesrc);
  pa_stream_set_latency_update_callback (pulsesrc->stream,
      gst_pulsesrc_stream_latency_update_cb, pulsesrc);

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  return TRUE;

  /* ERRORS */
invalid_caps:
  {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
        ("Can't parse caps."), (NULL));
    goto fail;
  }
invalid_spec:
  {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
        ("Invalid sample specification."), (NULL));
    goto fail;
  }
bad_context:
  {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context"), (NULL));
    goto unlock_and_fail;
  }
create_failed:
  {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
        ("Failed to create stream: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }
unlock_and_fail:
  {
    gst_pulsesrc_destroy_stream (pulsesrc);

    pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  fail:
    return FALSE;
  }
}