C++ (Cpp) GST_AUDIO_ENCODER_SINK_PADの例

コード例 #1

ファイル: gstfdkaacenc.c プロジェクト: MaZderMind/gst-plugins-bad

static GstCaps *
gst_fdkaacenc_get_caps (GstAudioEncoder * enc, GstCaps * filter)
{
  GstCaps *res, *caps;
  gint i;

  caps = gst_caps_new_empty ();

  for (i = 0; i < G_N_ELEMENTS (channel_layouts); i++) {
    guint64 channel_mask;
    GstCaps *tmp =
        gst_caps_make_writable (gst_pad_get_pad_template_caps
        (GST_AUDIO_ENCODER_SINK_PAD (enc)));

    if (channel_layouts[i].channels == 1) {
      gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
          channel_layouts[i].channels, NULL);
    } else {
      gst_audio_channel_positions_to_mask (channel_layouts[i].positions,
          channel_layouts[i].channels, FALSE, &channel_mask);
      gst_caps_set_simple (tmp, "channels", G_TYPE_INT,
          channel_layouts[i].channels, "channel-mask", GST_TYPE_BITMASK,
          channel_mask, NULL);
    }

    gst_caps_append (caps, tmp);
  }

  res = gst_audio_encoder_proxy_getcaps (enc, caps, filter);
  gst_caps_unref (caps);

  return res;
}

コード例 #2

ファイル: gstsbcenc.c プロジェクト: asrashley/gst-plugins-bad

static void
gst_sbc_enc_init (GstSbcEnc * self)
{
  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (self));
  self->subbands = 0;
  self->blocks = 0;
  self->rate = 0;
  self->channels = 0;
  self->bitpool = 0;
}

コード例 #3

ファイル: gstavaudenc.c プロジェクト: GStreamer/gst-libav

static void
gst_ffmpegaudenc_init (GstFFMpegAudEnc * ffmpegaudenc)
{
    GstFFMpegAudEncClass *klass =
        (GstFFMpegAudEncClass *) G_OBJECT_GET_CLASS (ffmpegaudenc);

    GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (ffmpegaudenc));

    /* ffmpeg objects */
    ffmpegaudenc->context = avcodec_alloc_context3 (klass->in_plugin);
    ffmpegaudenc->opened = FALSE;
    ffmpegaudenc->frame = av_frame_alloc ();

    ffmpegaudenc->compliance = FFMPEG_DEFAULT_COMPLIANCE;

    gst_audio_encoder_set_drainable (GST_AUDIO_ENCODER (ffmpegaudenc), TRUE);
}

コード例 #4

ファイル: gstvorbisenc.c プロジェクト: ConfusedReality/pkg_multimedia_gst-plugins-base

static void
gst_vorbis_enc_init (GstVorbisEnc * vorbisenc)
{
  GstAudioEncoder *enc = GST_AUDIO_ENCODER (vorbisenc);

  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));

  vorbisenc->channels = -1;
  vorbisenc->frequency = -1;

  vorbisenc->managed = FALSE;
  vorbisenc->max_bitrate = MAX_BITRATE_DEFAULT;
  vorbisenc->bitrate = BITRATE_DEFAULT;
  vorbisenc->min_bitrate = MIN_BITRATE_DEFAULT;
  vorbisenc->quality = QUALITY_DEFAULT;
  vorbisenc->quality_set = FALSE;
  vorbisenc->last_message = NULL;

  /* arrange granulepos marking (and required perfect ts) */
  gst_audio_encoder_set_mark_granule (enc, TRUE);
  gst_audio_encoder_set_perfect_timestamp (enc, TRUE);
}

コード例 #5

ファイル: gstopusenc.c プロジェクト: kanongil/gst-plugins-bad

static GstCaps *
gst_opus_enc_sink_getcaps (GstAudioEncoder * benc)
{
  GstOpusEnc *enc;
  GstCaps *caps;
  GstCaps *peercaps = NULL;
  GstCaps *intersect = NULL;
  guint i;
  gboolean allow_multistream;

  enc = GST_OPUS_ENC (benc);

  GST_DEBUG_OBJECT (enc, "sink getcaps");

  peercaps = gst_pad_peer_get_caps (GST_AUDIO_ENCODER_SRC_PAD (benc));
  if (!peercaps) {
    GST_DEBUG_OBJECT (benc, "No peercaps, returning template sink caps");
    return
        gst_caps_copy (gst_pad_get_pad_template_caps
        (GST_AUDIO_ENCODER_SINK_PAD (benc)));
  }

  intersect = gst_caps_intersect (peercaps,
      gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (benc)));
  gst_caps_unref (peercaps);

  if (gst_caps_is_empty (intersect))
    return intersect;

  allow_multistream = FALSE;
  for (i = 0; i < gst_caps_get_size (intersect); i++) {
    GstStructure *s = gst_caps_get_structure (intersect, i);
    gboolean multistream;
    if (gst_structure_get_boolean (s, "multistream", &multistream)) {
      if (multistream) {
        allow_multistream = TRUE;
      }
    } else {
      allow_multistream = TRUE;
    }
  }

  gst_caps_unref (intersect);

  caps =
      gst_caps_copy (gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SINK_PAD
          (benc)));
  if (!allow_multistream) {
    GValue range = { 0 };
    g_value_init (&range, GST_TYPE_INT_RANGE);
    gst_value_set_int_range (&range, 1, 2);
    for (i = 0; i < gst_caps_get_size (caps); i++) {
      GstStructure *s = gst_caps_get_structure (caps, i);
      gst_structure_set_value (s, "channels", &range);
    }
    g_value_unset (&range);
  }

  GST_DEBUG_OBJECT (enc, "Returning caps: %" GST_PTR_FORMAT, caps);
  return caps;
}

コード例 #6

ファイル: mulaw-encode.c プロジェクト: ConfusedReality/pkg_multimedia_gst-plugins-good

static void
gst_mulawenc_init (GstMuLawEnc * mulawenc)
{
  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (mulawenc));
}

コード例 #7

ファイル: gstsirenenc.c プロジェクト: GrokImageCompression/gst-plugins-bad

static void
gst_siren_enc_init (GstSirenEnc * enc)
{
  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
}