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faac: port to 0.11

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This commit is contained in:
Wim Taymans 2011-10-16 16:45:24 +02:00
parent 0806e5cbb9
commit e856f6f475
2 changed files with 41 additions and 42 deletions
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2
configure.ac
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@ -305,7 +305,7 @@ GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
videomaxrate videomeasure videoparsers videosignal vmnc y4m \
decklink fbdev linsys shm vcd \
voaacenc apexsink bz2 camerabin2 cdaudio celt cog curl dc1394 dirac directfb dts resindvd \
faac gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
gsettings gsm jp2k ladspa modplug mpeg2enc mplex mimic \
musepack musicbrainz nas neon ofa openal rsvg schro sdl sndfile soundtouch timidity \
vp8 wildmidi xvid "
AC_SUBST(GST_PLUGINS_NONPORTED)

81
ext/faac/gstfaac.c
View file

@ -56,11 +56,8 @@
"88200, " \
"96000"
#define SINK_CAPS \
"audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE (S16) ", " \
"rate = (int) {" SAMPLE_RATES "}, " \
"channels = (int) [ 1, 6 ] "
@ -125,7 +122,7 @@ static void gst_faac_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_faac_configure_source_pad (GstFaac * faac);
static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc);
static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter);
static gboolean gst_faac_start (GstAudioEncoder * enc);
static gboolean gst_faac_stop (GstAudioEncoder * enc);
@ -144,25 +141,8 @@ GST_DEBUG_CATEGORY_STATIC (faac_debug);
#define FAAC_DEFAULT_MIDSIDE TRUE
#define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL
GST_BOILERPLATE (GstFaac, gst_faac, GstAudioEncoder, GST_TYPE_AUDIO_ENCODER);
static void
gst_faac_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class, "AAC audio encoder",
"Codec/Encoder/Audio",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
}
#define gst_faac_parent_class parent_class
G_DEFINE_TYPE (GstFaac, gst_faac, GST_TYPE_AUDIO_ENCODER);
#define GST_TYPE_FAAC_RATE_CONTROL (gst_faac_brtype_get_type ())
static GType
@ -209,13 +189,22 @@ static void
gst_faac_class_init (GstFaacClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_faac_set_property;
gobject_class->get_property = gst_faac_get_property;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (gstelement_class, "AAC audio encoder",
"Codec/Encoder/Audio",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
base_class->start = GST_DEBUG_FUNCPTR (gst_faac_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format);
@ -251,10 +240,12 @@ gst_faac_class_init (GstFaacClass * klass)
"Block type encorcing",
GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
}
static void
gst_faac_init (GstFaac * faac, GstFaacClass * klass)
gst_faac_init (GstFaac * faac)
{
}
@ -315,7 +306,7 @@ static const GstAudioChannelPosition aac_channel_positions[][8] = {
};
static GstCaps *
gst_faac_getcaps (GstAudioEncoder * enc)
gst_faac_getcaps (GstAudioEncoder * enc, GstCaps * filter)
{
static volatile gsize sinkcaps = 0;
@ -338,10 +329,8 @@ gst_faac_getcaps (GstAudioEncoder * enc)
}
g_value_unset (&tmp_v);
s = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
s = gst_structure_new ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16), NULL);
gst_structure_set_value (s, "rate", &rates_arr);
for (i = 1; i <= 6; i++) {
@ -588,7 +577,7 @@ gst_faac_configure_source_pad (GstFaac * faac)
/* copy it into a buffer */
codec_data = gst_buffer_new_and_alloc (config_len);
memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
gst_buffer_fill (codec_data, 0, config, config_len);
/* add to caps */
gst_caps_set_simple (srccaps,
@ -647,28 +636,35 @@ gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
GstFaac *faac = GST_FAAC (enc);
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
gint size, ret_size;
const guint8 *data;
gsize size, ret_size;
guint8 *data;
guint8 *out_data;
gsize out_size;
out_buf = gst_buffer_new_and_alloc (faac->bytes);
out_data = gst_buffer_map (out_buf, &out_size, NULL, GST_MAP_WRITE);
if (G_LIKELY (in_buf)) {
data = GST_BUFFER_DATA (in_buf);
size = GST_BUFFER_SIZE (in_buf);
data = gst_buffer_map (in_buf, &size, NULL, GST_MAP_READ);
} else {
data = NULL;
size = 0;
}
if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
size / faac->bps, GST_BUFFER_DATA (out_buf),
GST_BUFFER_SIZE (out_buf))) < 0))
size / faac->bps, out_data, out_size)) < 0))
goto encode_failed;
gst_buffer_unmap (in_buf, data, size);
GST_LOG_OBJECT (faac, "encoder return: %d", ret_size);
if (ret_size > 0) {
GST_BUFFER_SIZE (out_buf) = ret_size;
gst_buffer_unmap (out_buf, out_data, ret_size);
ret = gst_audio_encoder_finish_frame (enc, out_buf, faac->samples);
} else {
gst_buffer_unmap (out_buf, out_data, 0);
gst_buffer_unref (out_buf);
}
return ret;
@ -677,6 +673,9 @@ gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
encode_failed:
{
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
gst_buffer_unmap (in_buf, data, size);
gst_buffer_unmap (out_buf, out_data, 0);
gst_buffer_unref (out_buf);
return GST_FLOW_ERROR;
}
}