C++ GST_STR_NULL函数代码示例

static GstJackAudioConnection *
gst_jack_audio_get_connection (const gchar * id, const gchar * server,
    jack_client_t * jclient, jack_status_t * status)
{
  GstJackAudioConnection *conn;
  GList *found;
  FindData data;

  GST_DEBUG ("getting connection for id %s, server %s", id,
      GST_STR_NULL (server));

  data.id = id;
  data.server = server;

  G_LOCK (connections_lock);
  found =
      g_list_find_custom (connections, &data, (GCompareFunc) connection_find);
  if (found != NULL && jclient != NULL) {
    /* we found it, increase refcount and return it */
    conn = (GstJackAudioConnection *) found->data;
    conn->refcount++;

    GST_DEBUG ("found connection %p", conn);
  } else {
    /* make new connection */
    conn = gst_jack_audio_make_connection (id, server, jclient, status);
    if (conn != NULL) {
      GST_DEBUG ("created connection %p", conn);
      /* add to list on success */
      connections = g_list_prepend (connections, conn);
    } else {
      GST_WARNING ("could not create connection");
    }
  }
  G_UNLOCK (connections_lock);

  return conn;
}

static void
print_plugin (const gchar * marker, GstRegistry * registry, GstPlugin * plugin)
{
  const gchar *name;
  GList *features, *f;

  name = gst_plugin_get_name (plugin);

  GST_DEBUG ("%s: plugin %p %d %s file: %s", marker, plugin,
      GST_OBJECT_REFCOUNT (plugin), name,
      GST_STR_NULL (gst_plugin_get_filename (plugin)));

  features = gst_registry_get_feature_list_by_plugin (registry, name);
  for (f = features; f != NULL; f = f->next) {
    GstPluginFeature *feature;

    feature = GST_PLUGIN_FEATURE (f->data);

    GST_LOG ("%s:    feature: %p %s", marker, feature,
        GST_OBJECT_NAME (feature));
  }
  gst_plugin_feature_list_free (features);
}

/**
 * gst_element_factory_make:
 * @factoryname: a named factory to instantiate
 * @name: (allow-none): name of new element, or NULL to automatically create
 *    a unique name
 *
 * Create a new element of the type defined by the given element factory.
 * If name is NULL, then the element will receive a guaranteed unique name,
 * consisting of the element factory name and a number.
 * If name is given, it will be given the name supplied.
 *
 * Returns: (transfer full): new #GstElement or NULL if unable to create element
 */
GstElement *
gst_element_factory_make (const gchar * factoryname, const gchar * name)
{
  GstElementFactory *factory;
  GstElement *element;

  g_return_val_if_fail (factoryname != NULL, NULL);
  g_return_val_if_fail (gst_is_initialized (), NULL);

  GST_LOG ("gstelementfactory: make \"%s\" \"%s\"",
      factoryname, GST_STR_NULL (name));

  factory = gst_element_factory_find (factoryname);
  if (factory == NULL)
    goto no_factory;

  GST_LOG_OBJECT (factory, "found factory %p", factory);
  element = gst_element_factory_create (factory, name);
  if (element == NULL)
    goto create_failed;

  gst_object_unref (factory);
  return element;

  /* ERRORS */
no_factory:
  {
    GST_INFO ("no such element factory \"%s\"!", factoryname);
    return NULL;
  }
create_failed:
  {
    GST_INFO_OBJECT (factory, "couldn't create instance!");
    gst_object_unref (factory);
    return NULL;
  }
}

static void
jack_shutdown_cb (void *arg)
{
  GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
  GList *walk;

  GST_DEBUG ("disconnect client %s from server %s", conn->id,
      GST_STR_NULL (conn->server));

  g_mutex_lock (conn->lock);
  for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
    GstJackAudioClient *client = (GstJackAudioClient *) walk->data;

    if (client->shutdown)
      client->shutdown (client->user_data);
  }
  for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
    GstJackAudioClient *client = (GstJackAudioClient *) walk->data;

    if (client->shutdown)
      client->shutdown (client->user_data);
  }
  g_mutex_unlock (conn->lock);
}

static gboolean
gst_mms_do_seek (GstBaseSrc * src, GstSegment * segment)
{
  mms_off_t start;
  GstMMS *mmssrc = GST_MMS (src);

  if (segment->format == GST_FORMAT_TIME) {
    if (!mmsx_time_seek (NULL, mmssrc->connection,
            (double) segment->start / GST_SECOND)) {
      GST_LOG_OBJECT (mmssrc, "mmsx_time_seek() failed");
      return FALSE;
    }
    start = mmsx_get_current_pos (mmssrc->connection);
    GST_INFO_OBJECT (mmssrc, "sought to %" GST_TIME_FORMAT ", offset after "
        "seek: %" G_GINT64_FORMAT, GST_TIME_ARGS (segment->start), start);
  } else if (segment->format == GST_FORMAT_BYTES) {
    start = mmsx_seek (NULL, mmssrc->connection, segment->start, SEEK_SET);
    /* mmsx_seek will close and reopen the connection when seeking with the
       mmsh protocol, if the reopening fails this is indicated with -1 */
    if (start == -1) {
      GST_DEBUG_OBJECT (mmssrc, "connection broken during seek");
      return FALSE;
    }
    GST_INFO_OBJECT (mmssrc, "sought to: %" G_GINT64_FORMAT " bytes, "
        "result: %" G_GINT64_FORMAT, segment->start, start);
  } else {
    GST_DEBUG_OBJECT (mmssrc, "unsupported seek segment format: %s",
        GST_STR_NULL (gst_format_get_name (segment->format)));
    return FALSE;
  }
  gst_segment_init (segment, GST_FORMAT_BYTES);
  gst_segment_set_seek (segment, segment->rate, GST_FORMAT_BYTES,
      segment->flags, GST_SEEK_TYPE_SET, start, GST_SEEK_TYPE_NONE,
      segment->stop, NULL);
  return TRUE;
}

void
totem_gst_message_print (GstMessage *msg,
			 GstElement *play,
			 const char *filename)
{
  GError *err = NULL;
  char *dbg = NULL;

  g_return_if_fail (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR);

  if (play != NULL) {
    g_return_if_fail (filename != NULL);

    GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN_CAST (play),
			       GST_DEBUG_GRAPH_SHOW_ALL ^ GST_DEBUG_GRAPH_SHOW_NON_DEFAULT_PARAMS,
			       filename);
  }

  gst_message_parse_error (msg, &err, &dbg);
  if (err) {
    char *uri;

    g_object_get (play, "uri", &uri, NULL);
    GST_ERROR ("message = %s", GST_STR_NULL (err->message));
    GST_ERROR ("domain  = %d (%s)", err->domain,
        GST_STR_NULL (g_quark_to_string (err->domain)));
    GST_ERROR ("code    = %d", err->code);
    GST_ERROR ("debug   = %s", GST_STR_NULL (dbg));
    GST_ERROR ("source  = %" GST_PTR_FORMAT, msg->src);
    GST_ERROR ("uri     = %s", GST_STR_NULL (uri));
    g_free (uri);

    g_message ("Error: %s\n%s\n", GST_STR_NULL (err->message),
        GST_STR_NULL (dbg));

    g_error_free (err);
  }
  g_free (dbg);
}

static gboolean
gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
  GstStructure *structure;
  GstRtpG722Depay *rtpg722depay;
  gint clock_rate, payload, samplerate;
  gint channels;
  GstCaps *srccaps;
  gboolean res;
  const gchar *channel_order;
  const GstRTPChannelOrder *order;

  rtpg722depay = GST_RTP_G722_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  payload = 96;
  gst_structure_get_int (structure, "payload", &payload);
  switch (payload) {
    case GST_RTP_PAYLOAD_G722:
      channels = 1;
      clock_rate = 8000;
      samplerate = 16000;
      break;
    default:
      /* no fixed mapping, we need clock-rate */
      channels = 0;
      clock_rate = 0;
      samplerate = 0;
      break;
  }

  /* caps can overwrite defaults */
  clock_rate =
      gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
  if (clock_rate == 0)
    goto no_clockrate;

  if (clock_rate == 8000)
    samplerate = 16000;

  if (samplerate == 0)
    samplerate = clock_rate;

  channels =
      gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
  if (channels == 0) {
    channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
    if (channels == 0) {
      /* channels defaults to 1 otherwise */
      channels = 1;
    }
  }

  depayload->clock_rate = clock_rate;
  rtpg722depay->rate = samplerate;
  rtpg722depay->channels = channels;

  srccaps = gst_caps_new_simple ("audio/G722",
      "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);

  /* add channel positions */
  channel_order = gst_structure_get_string (structure, "channel-order");

  order = gst_rtp_channels_get_by_order (channels, channel_order);
  if (order) {
    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
        order->pos);
  } else {
    GstAudioChannelPosition *pos;

    GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
        (NULL), ("Unknown channel order '%s' for %d channels",
            GST_STR_NULL (channel_order), channels));
    /* create default NONE layout */
    pos = gst_rtp_channels_create_default (channels);
    gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
    g_free (pos);
  }

  res = gst_pad_set_caps (depayload->srcpad, srccaps);
  gst_caps_unref (srccaps);

  return res;

  /* ERRORS */
no_clockrate:
  {
    GST_ERROR_OBJECT (depayload, "no clock-rate specified");
    return FALSE;
  }
}

static gboolean
gst_gsettings_audio_sink_change_child (GstGSettingsAudioSink * sink)
{
  const gchar *key = NULL;
  gchar *new_string;
  GError *err = NULL;
  GstElement *new_kid;

  GST_OBJECT_LOCK (sink);
  switch (sink->profile) {
    case GST_GSETTINGS_AUDIOSINK_PROFILE_SOUNDS:
      key = GST_GSETTINGS_KEY_SOUNDS_AUDIOSINK;
      break;
    case GST_GSETTINGS_AUDIOSINK_PROFILE_MUSIC:
      key = GST_GSETTINGS_KEY_MUSIC_AUDIOSINK;
      break;
    case GST_GSETTINGS_AUDIOSINK_PROFILE_CHAT:
      key = GST_GSETTINGS_KEY_CHAT_AUDIOSINK;
      break;
    default:
      break;
  }

  new_string = g_settings_get_string (sink->settings, key);

  if (new_string != NULL && sink->gsettings_str != NULL &&
      (strlen (new_string) == 0 ||
          strcmp (sink->gsettings_str, new_string) == 0)) {
    g_free (new_string);
    GST_DEBUG_OBJECT (sink,
        "GSettings key was updated, but it didn't change. Ignoring");
    GST_OBJECT_UNLOCK (sink);
    return TRUE;
  }
  GST_OBJECT_UNLOCK (sink);

  GST_DEBUG_OBJECT (sink, "GSettings key changed from '%s' to '%s'",
      GST_STR_NULL (sink->gsettings_str), GST_STR_NULL (new_string));

  if (new_string) {
    new_kid = gst_parse_bin_from_description (new_string, TRUE, &err);
    if (err) {
      GST_ERROR_OBJECT (sink, "error creating bin '%s': %s", new_string,
          err->message);
      g_error_free (err);
    }
  } else {
    new_kid = NULL;
  }

  if (new_kid == NULL) {
    GST_ELEMENT_ERROR (sink, LIBRARY, SETTINGS, (NULL),
        ("Failed to render audio sink from GSettings"));
    goto fail;
  }

  if (!gst_switch_sink_set_child (GST_SWITCH_SINK (sink), new_kid)) {
    GST_WARNING_OBJECT (sink, "Failed to update child element");
    goto fail;
  }

  g_free (sink->gsettings_str);
  sink->gsettings_str = new_string;

  return TRUE;

fail:
  g_free (new_string);
  return FALSE;
}

static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
  GstPulseSrc *pulsesrc = GST_PULSESRC_CAST (asrc);
  gchar *name = gst_pulse_client_name ();

  pa_threaded_mainloop_lock (pulsesrc->mainloop);

  g_assert (!pulsesrc->context);
  g_assert (!pulsesrc->stream);

  GST_DEBUG_OBJECT (pulsesrc, "opening device");

  if (!(pulsesrc->context =
          pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop),
              name))) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"),
        (NULL));
    goto unlock_and_fail;
  }

  pa_context_set_state_callback (pulsesrc->context,
      gst_pulsesrc_context_state_cb, pulsesrc);

  GST_DEBUG_OBJECT (pulsesrc, "connect to server %s",
      GST_STR_NULL (pulsesrc->server));

  if (pa_context_connect (pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
    GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
            pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
    goto unlock_and_fail;
  }

  for (;;) {
    pa_context_state_t state;

    state = pa_context_get_state (pulsesrc->context);

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

    if (state == PA_CONTEXT_READY)
      break;

    /* Wait until the context is ready */
    pa_threaded_mainloop_wait (pulsesrc->mainloop);
  }
  GST_DEBUG_OBJECT (pulsesrc, "connected");

  pa_threaded_mainloop_unlock (pulsesrc->mainloop);

  g_free (name);
  return TRUE;

  /* ERRORS */
unlock_and_fail:
  {
    gst_pulsesrc_destroy_context (pulsesrc);

    pa_threaded_mainloop_unlock (pulsesrc->mainloop);

    g_free (name);
    return FALSE;
  }
}

/* make a connection with @id and @server. Returns NULL on failure with the
 * status set. */
static GstJackAudioConnection *
gst_jack_audio_make_connection (const gchar * id, const gchar * server,
    jack_client_t * jclient, jack_status_t * status)
{
  GstJackAudioConnection *conn;
  jack_options_t options;
  gint res;

  *status = 0;

  GST_DEBUG ("new client %s, connecting to server %s", id,
      GST_STR_NULL (server));

  /* never start a server */
  options = JackNoStartServer;
  /* if we have a servername, use it */
  if (server != NULL)
    options |= JackServerName;
  /* open the client */
  if (jclient == NULL)
    jclient = jack_client_open (id, options, status, server);
  if (jclient == NULL)
    goto could_not_open;

  /* now create object */
  conn = g_new (GstJackAudioConnection, 1);
  conn->refcount = 1;
  g_mutex_init (&conn->lock);
  g_cond_init (&conn->flush_cond);
  conn->id = g_strdup (id);
  conn->server = g_strdup (server);
  conn->client = jclient;
  conn->n_clients = 0;
  conn->src_clients = NULL;
  conn->sink_clients = NULL;
  conn->cur_ts = -1;
  conn->transport_state = GST_STATE_VOID_PENDING;

  /* set our callbacks  */
  jack_set_process_callback (jclient, jack_process_cb, conn);
  /* these callbacks cause us to error */
  jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn);
  jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn);
  jack_on_shutdown (jclient, jack_shutdown_cb, conn);

  /* all callbacks are set, activate the client */
  GST_INFO ("activate jack_client %p", jclient);
  if ((res = jack_activate (jclient)))
    goto could_not_activate;

  GST_DEBUG ("opened connection %p", conn);

  return conn;

  /* ERRORS */
could_not_open:
  {
    GST_DEBUG ("failed to open jack client, %d", *status);
    return NULL;
  }
could_not_activate:
  {
    GST_ERROR ("Could not activate client (%d)", res);
    *status = JackFailure;
    g_mutex_clear (&conn->lock);
    g_free (conn->id);
    g_free (conn->server);
    g_free (conn);
    return NULL;
  }
}

int
main (int argc, char **argv)
{
  GstBus *bus;
  GOptionContext *ctx;
  GIOChannel *io_stdin;
  GError *err = NULL;
  gboolean res;
  GOptionEntry options[] = {
    {NULL}
  };
  GThread *rthread;

  /* Clear application state */
  memset (state, 0, sizeof (*state));
  state->animate = TRUE;

  /* must initialise the threading system before using any other GLib funtion */
  if (!g_thread_supported ())
    g_thread_init (NULL);

  ctx = g_option_context_new ("[ADDITIONAL ARGUMENTS]");
  g_option_context_add_main_entries (ctx, options, NULL);
  g_option_context_add_group (ctx, gst_init_get_option_group ());
  if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
    g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
    exit (1);
  }
  g_option_context_free (ctx);

  if (argc != 2) {
    g_print ("Usage: %s <URI> or <PIPELINE-DESCRIPTION>\n", argv[0]);
    exit (1);
  }

  /* Initialize GStreamer */
  gst_init (&argc, &argv);

  /* initialize inter thread comunnication */
  init_intercom (state);

  TRACE_VC_MEMORY ("state 0");

  if (!(rthread = g_thread_new ("render", (GThreadFunc) render_func, NULL))) {
    g_print ("Render thread create failed\n");
    exit (1);
  }

  /* Initialize player */
  if (gst_uri_is_valid (argv[1])) {
    res = init_playbin_player (state, argv[1]);
  } else {
    res = init_parse_launch_player (state, argv[1]);
  }

  if (!res)
    goto done;

  /* Create a GLib Main Loop and set it to run */
  state->main_loop = g_main_loop_new (NULL, FALSE);

  /* Add a keyboard watch so we get notified of keystrokes */
  io_stdin = g_io_channel_unix_new (fileno (stdin));
  g_io_add_watch (io_stdin, G_IO_IN, (GIOFunc) handle_keyboard, state);
  g_io_channel_unref (io_stdin);

  /* *INDENT-OFF* */
  g_print ("Available commands: \n"
      "  a - Toggle animation \n"
      "  p - Pause playback \n"
      "  r - Resume playback \n"
      "  l - Query position/duration\n"
      "  f - Seek 30 seconds forward \n"
      "  b - Seek 30 seconds backward \n"
      "  q - Quit \n");
  /* *INDENT-ON* */

  /* Connect the bus handlers */
  bus = gst_element_get_bus (state->pipeline);

  gst_bus_set_sync_handler (bus, (GstBusSyncHandler) bus_sync_handler, state,
      NULL);

  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
  gst_bus_enable_sync_message_emission (bus);

  g_signal_connect (G_OBJECT (bus), "message::error", (GCallback) error_cb,
      state);
  g_signal_connect (G_OBJECT (bus), "message::buffering",
      (GCallback) buffering_cb, state);
  g_signal_connect (G_OBJECT (bus), "message::eos", (GCallback) eos_cb, state);
  g_signal_connect (G_OBJECT (bus), "message::qos", (GCallback) qos_cb, state);
  g_signal_connect (G_OBJECT (bus), "message::state-changed",
      (GCallback) state_changed_cb, state);
  gst_object_unref (bus);

  /* Make player start playing */
  gst_element_set_state (state->pipeline, GST_STATE_PLAYING);

  /* Start the mainloop */
//.........这里部分代码省略.........

static gboolean
gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
  GstStructure *structure;
  GstRtpL16Depay *rtpL16depay;
  gint clock_rate, payload;
  gint channels;
  GstCaps *srccaps;
  gboolean res;
  const gchar *channel_order;
  const GstRTPChannelOrder *order;
  GstAudioInfo *info;

  rtpL16depay = GST_RTP_L16_DEPAY (depayload);

  structure = gst_caps_get_structure (caps, 0);

  payload = 96;
  gst_structure_get_int (structure, "payload", &payload);
  switch (payload) {
    case GST_RTP_PAYLOAD_L16_STEREO:
      channels = 2;
      clock_rate = 44100;
      break;
    case GST_RTP_PAYLOAD_L16_MONO:
      channels = 1;
      clock_rate = 44100;
      break;
    default:
      /* no fixed mapping, we need clock-rate */
      channels = 0;
      clock_rate = 0;
      break;
  }

  /* caps can overwrite defaults */
  clock_rate =
      gst_rtp_L16_depay_parse_int (structure, "clock-rate", clock_rate);
  if (clock_rate == 0)
    goto no_clockrate;

  channels =
      gst_rtp_L16_depay_parse_int (structure, "encoding-params", channels);
  if (channels == 0) {
    channels = gst_rtp_L16_depay_parse_int (structure, "channels", channels);
    if (channels == 0) {
      /* channels defaults to 1 otherwise */
      channels = 1;
    }
  }

  depayload->clock_rate = clock_rate;

  info = &rtpL16depay->info;
  gst_audio_info_init (info);
  info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
  info->rate = clock_rate;
  info->channels = channels;
  info->bpf = (info->finfo->width / 8) * channels;

  /* add channel positions */
  channel_order = gst_structure_get_string (structure, "channel-order");

  order = gst_rtp_channels_get_by_order (channels, channel_order);
  rtpL16depay->order = order;
  if (order) {
    memcpy (info->position, order->pos,
        sizeof (GstAudioChannelPosition) * channels);
    gst_audio_channel_positions_to_valid_order (info->position, info->channels);
  } else {
    GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
        (NULL), ("Unknown channel order '%s' for %d channels",
            GST_STR_NULL (channel_order), channels));
    /* create default NONE layout */
    gst_rtp_channels_create_default (channels, info->position);
  }

  srccaps = gst_audio_info_to_caps (info);
  res = gst_pad_set_caps (depayload->srcpad, srccaps);
  gst_caps_unref (srccaps);

  return res;

  /* ERRORS */
no_clockrate:
  {
    GST_ERROR_OBJECT (depayload, "no clock-rate specified");
    return FALSE;
  }
}

//.........这里部分代码省略.........
    GstBuffer *codec_data = gst_value_get_buffer (h);
    GstMapInfo minfo;
    guint8 *data;

    gst_buffer_map (codec_data, &minfo, GST_MAP_READ);
    data = g_memdup (minfo.data, minfo.size);
    self->codec_datas = g_list_prepend (self->codec_datas, data);
    gst_amc_format_set_buffer (format, "csd-0", data, minfo.size, &err);
    if (err)
      GST_ELEMENT_WARNING_FROM_ERROR (self, err);
    gst_buffer_unmap (codec_data, &minfo);
  } else if (gst_structure_has_field (s, "streamheader")) {
    const GValue *sh = gst_structure_get_value (s, "streamheader");
    gint nsheaders = gst_value_array_get_size (sh);
    GstBuffer *buf;
    const GValue *h;
    gint i, j;
    gchar *fname;
    GstMapInfo minfo;
    guint8 *data;

    for (i = 0, j = 0; i < nsheaders; i++) {
      h = gst_value_array_get_value (sh, i);
      buf = gst_value_get_buffer (h);

      if (strcmp (mime, "audio/vorbis") == 0) {
        guint8 header_type;

        gst_buffer_extract (buf, 0, &header_type, 1);

        /* Only use the identification and setup packets */
        if (header_type != 0x01 && header_type != 0x05)
          continue;
      }

      fname = g_strdup_printf ("csd-%d", j);
      gst_buffer_map (buf, &minfo, GST_MAP_READ);
      data = g_memdup (minfo.data, minfo.size);
      self->codec_datas = g_list_prepend (self->codec_datas, data);
      gst_amc_format_set_buffer (format, fname, data, minfo.size, &err);
      if (err)
        GST_ELEMENT_WARNING_FROM_ERROR (self, err);
      gst_buffer_unmap (buf, &minfo);
      g_free (fname);
      j++;
    }
  }

  format_string = gst_amc_format_to_string (format, &err);
  if (err)
    GST_ELEMENT_WARNING_FROM_ERROR (self, err);
  GST_DEBUG_OBJECT (self, "Configuring codec with format: %s",
      GST_STR_NULL (format_string));
  g_free (format_string);

  if (!gst_amc_codec_configure (self->codec, format, 0, &err)) {
    GST_ERROR_OBJECT (self, "Failed to configure codec");
    GST_ELEMENT_ERROR_FROM_ERROR (self, err);
    return FALSE;
  }

  gst_amc_format_free (format);

  if (!gst_amc_codec_start (self->codec, &err)) {
    GST_ERROR_OBJECT (self, "Failed to start codec");
    GST_ELEMENT_ERROR_FROM_ERROR (self, err);
    return FALSE;
  }

  self->spf = -1;
  /* TODO: Implement for other codecs too */
  if (gst_structure_has_name (s, "audio/mpeg")) {
    gint mpegversion = -1;

    gst_structure_get_int (s, "mpegversion", &mpegversion);
    if (mpegversion == 1) {
      gint layer = -1, mpegaudioversion = -1;

      gst_structure_get_int (s, "layer", &layer);
      gst_structure_get_int (s, "mpegaudioversion", &mpegaudioversion);
      if (layer == 1)
        self->spf = 384;
      else if (layer == 2)
        self->spf = 1152;
      else if (layer == 3 && mpegaudioversion != -1)
        self->spf = (mpegaudioversion == 1 ? 1152 : 576);
    }
  }

  self->started = TRUE;
  self->input_caps_changed = TRUE;

  /* Start the srcpad loop again */
  self->flushing = FALSE;
  self->downstream_flow_ret = GST_FLOW_OK;
  gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
      (GstTaskFunction) gst_amc_audio_dec_loop, decoder, NULL);

  return TRUE;
}

GstMixerOptions *
gst_sunaudiomixer_options_new (GstSunAudioMixerCtrl * mixer, gint track_num)
{
    GstMixerOptions *opts;
    GstSunAudioMixerOptions *sun_opts;
    GstMixerTrack *track;
    const gchar *label;
    gint i;
    struct audio_info audioinfo;

    if ((mixer == NULL) || (mixer->mixer_fd == -1)) {
        g_warning ("mixer not initialized");
        return NULL;
    }

    if (track_num != GST_SUNAUDIO_TRACK_RECSRC) {
        g_warning ("invalid options track");
        return (NULL);
    }

    label = N_("Record Source");

    opts = g_object_new (GST_TYPE_SUNAUDIO_MIXER_OPTIONS,
                         "untranslated-label", label, NULL);
    sun_opts = GST_SUNAUDIO_MIXER_OPTIONS (opts);
    track = GST_MIXER_TRACK (opts);

    GST_DEBUG_OBJECT (opts, "New mixer options, track %d: %s",
                      track_num, GST_STR_NULL (label));

    /* save off names for the record sources */
    sun_opts->names[0] = g_quark_from_string (_("Microphone"));
    sun_opts->names[1] = g_quark_from_string (_("Line In"));
    sun_opts->names[2] = g_quark_from_string (_("Internal CD"));
    sun_opts->names[3] = g_quark_from_string (_("SPDIF In"));
    sun_opts->names[4] = g_quark_from_string (_("AUX 1 In"));
    sun_opts->names[5] = g_quark_from_string (_("AUX 2 In"));
    sun_opts->names[6] = g_quark_from_string (_("Codec Loopback"));
    sun_opts->names[7] = g_quark_from_string (_("SunVTS Loopback"));

    /* set basic information */
    track->label = g_strdup (_(label));
    track->num_channels = 0;
    track->min_volume = 0;
    track->max_volume = 0;
    track->flags =
        GST_MIXER_TRACK_INPUT | GST_MIXER_TRACK_WHITELIST |
        GST_MIXER_TRACK_NO_RECORD;

    if (ioctl (mixer->mixer_fd, AUDIO_GETINFO, &audioinfo) < 0) {
        g_warning ("Error getting audio device settings");
        g_object_unref (G_OBJECT (sun_opts));
        return NULL;
    }

    sun_opts->avail = audioinfo.record.avail_ports;
    sun_opts->track_num = track_num;

    for (i = 0; i < 8; i++) {
        if ((1 << i) & audioinfo.record.avail_ports) {
            const char *s = g_quark_to_string (sun_opts->names[i]);
            opts->values = g_list_append (opts->values, g_strdup (s));
            GST_DEBUG_OBJECT (opts, "option for track %d: %s",
                              track_num, GST_STR_NULL (s));
        }
    }

    return opts;
}

/* returns TRUE if there was an error or we caught a keyboard interrupt. */
static gboolean
event_loop (GstElement * pipeline, gboolean blocking, GstState target_state)
{
  GstBus *bus;
  GstMessage *message = NULL;
  gboolean res = FALSE;
  gboolean buffering = FALSE;

  bus = gst_element_get_bus (GST_ELEMENT (pipeline));

#ifndef DISABLE_FAULT_HANDLER
  g_timeout_add (50, (GSourceFunc) check_intr, pipeline);
#endif

  while (TRUE) {
    message = gst_bus_poll (bus, GST_MESSAGE_ANY, blocking ? -1 : 0);

    /* if the poll timed out, only when !blocking */
    if (message == NULL)
      goto exit;

    switch (GST_MESSAGE_TYPE (message)) {
      case GST_MESSAGE_NEW_CLOCK:
      {
        GstClock *clock;

        gst_message_parse_new_clock (message, &clock);

        g_print ("New clock: %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
        break;
      }
      case GST_MESSAGE_EOS:
        g_print ("Got EOS from element \"%s\".\n",
            GST_STR_NULL (GST_ELEMENT_NAME (GST_MESSAGE_SRC (message))));
        goto exit;
      case GST_MESSAGE_TAG:
        if (tags) {
          GstTagList *tags;

          gst_message_parse_tag (message, &tags);
          g_print ("FOUND TAG      : found by element \"%s\".\n",
              GST_STR_NULL (GST_ELEMENT_NAME (GST_MESSAGE_SRC (message))));
          gst_tag_list_foreach (tags, print_tag, NULL);
          gst_tag_list_free (tags);
        }
        break;
      case GST_MESSAGE_INFO:{
        GError *gerror;
        gchar *debug;
        gchar *name = gst_object_get_path_string (GST_MESSAGE_SRC (message));

        gst_message_parse_info (message, &gerror, &debug);
        if (debug) {
          g_print ("INFO:\n%s\n", debug);
        }
        g_error_free (gerror);
        g_free (debug);
        g_free (name);
        break;
      }
      case GST_MESSAGE_WARNING:{
        GError *gerror;
        gchar *debug;
        gchar *name = gst_object_get_path_string (GST_MESSAGE_SRC (message));

        /* dump graph on warning */
        GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
            GST_DEBUG_GRAPH_SHOW_ALL, "gst-launch.warning");

        gst_message_parse_warning (message, &gerror, &debug);
        g_print ("WARNING: from element %s: %s\n", name, gerror->message);
        if (debug) {
          g_print ("Additional debug info:\n%s\n", debug);
        }
        g_error_free (gerror);
        g_free (debug);
        g_free (name);
        break;
      }
      case GST_MESSAGE_ERROR:{
        GError *gerror;
        gchar *debug;

        /* dump graph on error */
        GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
            GST_DEBUG_GRAPH_SHOW_ALL, "gst-launch.error");

        gst_message_parse_error (message, &gerror, &debug);
        gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
        g_error_free (gerror);
        g_free (debug);
        /* we have an error */
        res = TRUE;
        goto exit;
      }
      case GST_MESSAGE_STATE_CHANGED:{
        GstState old, newX, pending;

        gst_message_parse_state_changed (message, &old, &newX, &pending);
//.........这里部分代码省略.........