static void
gst_audio_decoder_tester_class_init (GstAudioDecoderTesterClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audiosink_class = GST_AUDIO_DECODER_CLASS (klass);
static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-test-custom"));
static GstStaticPadTemplate src_templ = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw"));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_templ));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_templ));
gst_element_class_set_metadata (element_class,
"AudioDecoderTester", "Decoder/Audio", "yep", "me");
audiosink_class->start = gst_audio_decoder_tester_start;
audiosink_class->stop = gst_audio_decoder_tester_stop;
audiosink_class->flush = gst_audio_decoder_tester_flush;
audiosink_class->handle_frame = gst_audio_decoder_tester_handle_frame;
audiosink_class->set_format = gst_audio_decoder_tester_set_format;
}
static void
gst_vorbis_dec_class_init (GstVorbisDecClass * klass)
{
GstPadTemplate *src_template, *sink_template;
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
gobject_class->finalize = vorbis_dec_finalize;
src_template = gst_static_pad_template_get (&vorbis_dec_src_factory);
gst_element_class_add_pad_template (element_class, src_template);
sink_template = gst_static_pad_template_get (&vorbis_dec_sink_factory);
gst_element_class_add_pad_template (element_class, sink_template);
gst_element_class_set_static_metadata (element_class,
"Vorbis audio decoder", "Codec/Decoder/Audio",
GST_VORBIS_DEC_DESCRIPTION,
"Benjamin Otte <*****@*****.**>, Chris Lord <*****@*****.**>");
base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop);
base_class->handle_frame = GST_DEBUG_FUNCPTR (vorbis_dec_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush);
}
static void
gst_siren_dec_class_init (GstSirenDecClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (sirendec_debug, "sirendec", 0, "sirendec");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&srctemplate));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sinktemplate));
gst_element_class_set_metadata (element_class, "Siren Decoder element",
"Codec/Decoder/Audio ",
"Decode streams encoded with the Siren7 codec into 16bit PCM",
"Youness Alaoui <*****@*****.**>");
base_class->start = GST_DEBUG_FUNCPTR (gst_siren_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_dec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_dec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_siren_dec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_dec_handle_frame);
GST_DEBUG ("Class Init done");
}
static void
gst_omx_audio_dec_class_init (GstOMXAudioDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
gobject_class->finalize = gst_omx_audio_dec_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_omx_audio_dec_change_state);
audio_decoder_class->open = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_open);
audio_decoder_class->close = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_close);
audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_start);
audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_stop);
audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_omx_audio_dec_flush);
audio_decoder_class->set_format =
GST_DEBUG_FUNCPTR (gst_omx_audio_dec_set_format);
audio_decoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_omx_audio_dec_handle_frame);
klass->cdata.type = GST_OMX_COMPONENT_TYPE_FILTER;
klass->cdata.default_src_template_caps =
"audio/x-raw, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, " G_STRINGIFY (OMX_AUDIO_MAXCHANNELS) " ], "
"format = (string) " GST_AUDIO_FORMATS_ALL;
}
static void
gst_atdec_class_init (GstATDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioDecoderClass *audio_decoder_class = GST_AUDIO_DECODER_CLASS (klass);
gst_element_class_add_pad_template (GST_ELEMENT_CLASS (klass),
gst_static_pad_template_get (&gst_atdec_src_template));
gst_element_class_add_pad_template (GST_ELEMENT_CLASS (klass),
gst_static_pad_template_get (&gst_atdec_sink_template));
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass),
"AudioToolbox based audio decoder",
"Codec/Decoder/Audio",
"AudioToolbox based audio decoder",
"Alessandro Decina <*****@*****.**>");
gobject_class->set_property = gst_atdec_set_property;
gobject_class->get_property = gst_atdec_get_property;
gobject_class->finalize = gst_atdec_finalize;
audio_decoder_class->start = GST_DEBUG_FUNCPTR (gst_atdec_start);
audio_decoder_class->stop = GST_DEBUG_FUNCPTR (gst_atdec_stop);
audio_decoder_class->set_format = GST_DEBUG_FUNCPTR (gst_atdec_set_format);
audio_decoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_atdec_handle_frame);
audio_decoder_class->flush = GST_DEBUG_FUNCPTR (gst_atdec_flush);
}
static void
gst_vorbis_dec_class_init (GstVorbisDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
gobject_class->finalize = vorbis_dec_finalize;
base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop);
base_class->handle_frame = GST_DEBUG_FUNCPTR (vorbis_dec_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush);
}
static void
gst_ffmpegauddec_class_init (GstFFMpegAudDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioDecoderClass *gstaudiodecoder_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_ffmpegauddec_finalize;
gstaudiodecoder_class->stop = GST_DEBUG_FUNCPTR (gst_ffmpegauddec_stop);
gstaudiodecoder_class->set_format =
GST_DEBUG_FUNCPTR (gst_ffmpegauddec_set_format);
gstaudiodecoder_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_ffmpegauddec_handle_frame);
gstaudiodecoder_class->flush = GST_DEBUG_FUNCPTR (gst_ffmpegauddec_flush);
}
static gboolean
gst_ffmpegauddec_propose_allocation (GstAudioDecoder * decoder,
GstQuery * query)
{
GstAllocationParams params;
gst_allocation_params_init (¶ms);
params.flags = GST_MEMORY_FLAG_ZERO_PADDED;
params.align = 15;
params.padding = FF_INPUT_BUFFER_PADDING_SIZE;
/* we would like to have some padding so that we don't have to
* memcpy. We don't suggest an allocator. */
gst_query_add_allocation_param (query, NULL, ¶ms);
return GST_AUDIO_DECODER_CLASS (parent_class)->propose_allocation (decoder,
query);
}
static void
gst_mulawdec_class_init (GstMuLawDecClass * klass)
{
GstElementClass *element_class = (GstElementClass *) klass;
GstAudioDecoderClass *audiodec_class = GST_AUDIO_DECODER_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&mulaw_dec_src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&mulaw_dec_sink_factory));
audiodec_class->start = GST_DEBUG_FUNCPTR (gst_mulawdec_start);
audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_mulawdec_set_format);
audiodec_class->handle_frame = GST_DEBUG_FUNCPTR (gst_mulawdec_handle_frame);
gst_element_class_set_static_metadata (element_class, "Mu Law audio decoder",
"Codec/Decoder/Audio",
"Convert 8bit mu law to 16bit PCM",
"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
}
static void gst_imx_audio_uniaudio_dec_class_init(GstImxAudioUniaudioDecClass *klass)
{
GObjectClass *object_class;
GstAudioDecoderClass *base_class;
GstElementClass *element_class;
GstPadTemplate *sink_template;
GstCaps *sink_template_caps;
GST_DEBUG_CATEGORY_INIT(imx_audio_uniaudio_dec_debug, "imxuniaudiodec", 0, "Freescale i.MX uniaudio decoder");
object_class = G_OBJECT_CLASS(klass);
base_class = GST_AUDIO_DECODER_CLASS(klass);
element_class = GST_ELEMENT_CLASS(klass);
gst_imx_audio_uniaudio_codec_table_init();
sink_template_caps = gst_imx_audio_uniaudio_codec_table_get_caps();
sink_template = gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sink_template_caps);
gst_element_class_add_pad_template(element_class, sink_template);
gst_element_class_add_pad_template(element_class, gst_static_pad_template_get(&static_src_template));
GST_DEBUG("decoder sink caps: %" GST_PTR_FORMAT, (gpointer)sink_template_caps);
object_class->finalize = GST_DEBUG_FUNCPTR(gst_imx_audio_uniaudio_dec_finalize);
base_class->start = GST_DEBUG_FUNCPTR(gst_imx_audio_uniaudio_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR(gst_imx_audio_uniaudio_dec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR(gst_imx_audio_uniaudio_dec_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR(gst_imx_audio_uniaudio_dec_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR(gst_imx_audio_uniaudio_dec_flush);
gst_element_class_set_static_metadata(
element_class,
"Freescale i.MX uniaudio decoder",
"Codec/Decoder/Audio",
"audio decoding using the Freescale i.MX uniaudio codecs",
"Carlos Rafael Giani <*****@*****.**>"
);
}
static void
gst_amc_audio_dec_class_init (GstAmcAudioDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *audiodec_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_amc_audio_dec_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_amc_audio_dec_change_state);
audiodec_class->start = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_start);
audiodec_class->stop = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_stop);
audiodec_class->open = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_open);
audiodec_class->close = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_close);
audiodec_class->flush = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_flush);
audiodec_class->set_format = GST_DEBUG_FUNCPTR (gst_amc_audio_dec_set_format);
audiodec_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_amc_audio_dec_handle_frame);
}
static void
gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
{
GstAudioDecoderClass *base_class;
GstElementClass *element_class;
GstPadTemplate *src_template, *sink_template;
int error;
GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
base_class = GST_AUDIO_DECODER_CLASS (klass);
element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_set_static_metadata (element_class,
"mpg123 mp3 decoder",
"Codec/Decoder/Audio",
"Decodes mp3 streams using the mpg123 library",
"Carlos Rafael Giani <*****@*****.**>");
/* Not using static pad template for srccaps, since the comma-separated list
* of formats needs to be created depending on whatever mpg123 supports */
{
const int *format_list;
const long *rates_list;
size_t num, i;
GString *s;
GstCaps *src_template_caps;
s = g_string_new ("audio/x-raw, ");
mpg123_encodings (&format_list, &num);
g_string_append (s, "format = { ");
for (i = 0; i < num; ++i) {
switch (format_list[i]) {
case MPG123_ENC_SIGNED_16:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (S16));
break;
case MPG123_ENC_UNSIGNED_16:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (U16));
break;
case MPG123_ENC_SIGNED_24:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (S24));
break;
case MPG123_ENC_UNSIGNED_24:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (U24));
break;
case MPG123_ENC_SIGNED_32:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (S32));
break;
case MPG123_ENC_UNSIGNED_32:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (U32));
break;
case MPG123_ENC_FLOAT_32:
g_string_append (s, (i > 0) ? ", " : "");
g_string_append (s, GST_AUDIO_NE (F32));
break;
default:
GST_DEBUG ("Ignoring mpg123 format %d", format_list[i]);
break;
}
}
g_string_append (s, " }, ");
mpg123_rates (&rates_list, &num);
g_string_append (s, "rate = (int) { ");
for (i = 0; i < num; ++i) {
g_string_append_printf (s, "%s%lu", (i > 0) ? ", " : "", rates_list[i]);
}
g_string_append (s, "}, ");
g_string_append (s, "channels = (int) [ 1, 2 ], ");
g_string_append (s, "layout = (string) interleaved");
src_template_caps = gst_caps_from_string (s->str);
src_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
src_template_caps);
g_string_free (s, TRUE);
}
sink_template = gst_static_pad_template_get (&static_sink_template);
gst_element_class_add_pad_template (element_class, sink_template);
gst_element_class_add_pad_template (element_class, src_template);
base_class->start = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_stop);
base_class->handle_frame =
GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_handle_frame);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_set_format);
base_class->flush = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_flush);
error = mpg123_init ();
if (G_UNLIKELY (error != MPG123_OK))
GST_ERROR ("Could not initialize mpg123 library: %s",
mpg123_plain_strerror (error));
else
GST_INFO ("mpg123 library initialized");
}
