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adpcmenc: port to 0.11

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This commit is contained in:
Mark Nauwelaerts 2012-01-26 23:28:20 +01:00
parent de606f64eb
commit 936bee00f4
3 changed files with 46 additions and 52 deletions
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2
configure.ac
View file

@ -313,7 +313,7 @@ AG_GST_DEFAULT_ELEMENTS
dnl *** plug-ins to include ***
dnl Non ported plugins (non-dependant, then dependant)
dnl Make sure you have a space before and after all plugins
GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
GST_PLUGINS_NONPORTED=" adpcmdec aiff asfmux \
camerabin cdxaparse coloreffects \
dccp debugutils faceoverlay festival \
fieldanalysis freeverb freeze frei0r gaudieffects geometrictransform h264parse \

44
gst/adpcmdec/adpcmdec.c
View file

@ -53,12 +53,10 @@ static GstStaticPadTemplate adpcmdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"depth = (int)16, "
"width = (int)16, "
"endianness = (int)" G_STRINGIFY (G_BYTE_ORDER) ", "
"signed = (boolean)TRUE, "
"channels = (int) [1,2], " "rate = (int)[1, MAX]")
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) [1, MAX], channels = (int) [1,2]")
);
enum adpcm_layout
@ -83,7 +81,7 @@ typedef struct _ADPCMDec
} ADPCMDec;
GType adpcmdec_get_type (void);
GST_BOILERPLATE (ADPCMDec, adpcmdec, GstAudioDecoder, GST_TYPE_AUDIO_DECODER);
G_DEFINE_TYPE (ADPCMDec, adpcmdec, GST_TYPE_AUDIO_DECODER);
static gboolean
adpcmdec_set_format (GstAudioDecoder * bdec, GstCaps * in_caps)
@ -112,15 +110,13 @@ adpcmdec_set_format (GstAudioDecoder * bdec, GstCaps * in_caps)
if (!gst_structure_get_int (structure, "channels", &dec->channels))
return FALSE;
caps = gst_caps_new_simple ("audio/x-raw-int",
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved",
"rate", G_TYPE_INT, dec->rate,
"channels", G_TYPE_INT, dec->channels,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
"channels", G_TYPE_INT, dec->channels, NULL);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (bdec), caps);
gst_audio_decoder_set_outcaps (bdec, caps);
gst_caps_unref (caps);
return TRUE;
@ -457,26 +453,16 @@ adpcmdec_stop (GstAudioDecoder * dec)
}
static void
adpcmdec_init (ADPCMDec * dec, ADPCMDecClass * klass)
adpcmdec_init (ADPCMDec * dec)
{
}
static void
adpcmdec_class_init (ADPCMDecClass * klass)
{
GstElementClass *element_class = (GstElementClass *) klass;
GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) klass;
base_class->start = GST_DEBUG_FUNCPTR (adpcmdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (adpcmdec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (adpcmdec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (adpcmdec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmdec_handle_frame);
}
static void
adpcmdec_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&adpcmdec_sink_template));
gst_element_class_add_pad_template (element_class,
@ -485,6 +471,12 @@ adpcmdec_base_init (gpointer klass)
"Codec/Decoder/Audio",
"Decode MS and IMA ADPCM audio",
"Pioneers of the Inevitable <songbird@songbirdnest.com>");
base_class->start = GST_DEBUG_FUNCPTR (adpcmdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (adpcmdec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (adpcmdec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (adpcmdec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmdec_handle_frame);
}
static gboolean

52
gst/adpcmenc/adpcmenc.c
View file

@ -46,9 +46,10 @@ static GstStaticPadTemplate adpcmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"depth = (int)16, "
"width = (int)16, " "channels = (int) [1,2], " "rate = (int)[1, MAX]")
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) [1, MAX], channels = (int) [1,2]")
);
static GstStaticPadTemplate adpcmenc_src_template =
@ -131,7 +132,7 @@ typedef struct _ADPCMEnc
} ADPCMEnc;
GType adpcmenc_get_type (void);
GST_BOILERPLATE (ADPCMEnc, adpcmenc, GstAudioEncoder, GST_TYPE_AUDIO_ENCODER);
G_DEFINE_TYPE (ADPCMEnc, adpcmenc, GST_TYPE_AUDIO_ENCODER);
static gboolean
adpcmenc_setup (ADPCMEnc * enc)
@ -338,10 +339,13 @@ adpcmenc_encode_block (ADPCMEnc * enc, const gint16 * samples, int blocksize)
{
gboolean res = FALSE;
GstBuffer *outbuf = NULL;
GstMapInfo omap;
if (enc->layout == LAYOUT_ADPCM_DVI) {
outbuf = gst_buffer_new_and_alloc (enc->blocksize);
res = adpcmenc_encode_ima_block (enc, samples, GST_BUFFER_DATA (outbuf));
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
res = adpcmenc_encode_ima_block (enc, samples, omap.data);
gst_buffer_unmap (outbuf, &omap);
} else {
/* should not happen afaics */
g_assert_not_reached ();
@ -368,6 +372,7 @@ adpcmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
GstBuffer *outbuf;
int input_bytes_per_block;
const int BYTES_PER_SAMPLE = 2;
GstMapInfo map;
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
@ -378,15 +383,17 @@ adpcmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
input_bytes_per_block =
enc->samples_per_block * BYTES_PER_SAMPLE * enc->channels;
if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < input_bytes_per_block)) {
GST_DEBUG_OBJECT (enc, "discarding trailing data %d",
GST_BUFFER_SIZE (buffer));
gst_buffer_map (buffer, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < input_bytes_per_block)) {
GST_DEBUG_OBJECT (enc, "discarding trailing data %d", (gint) map.size);
gst_buffer_unmap (buffer, &map);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
samples = (gint16 *) GST_BUFFER_DATA (buffer);
samples = (gint16 *) map.data;
outbuf = adpcmenc_encode_block (enc, samples, enc->blocksize);
gst_buffer_unmap (buffer, &map);
ret = gst_audio_encoder_finish_frame (benc, outbuf, enc->samples_per_block);
@ -411,7 +418,7 @@ adpcmenc_stop (GstAudioEncoder * enc)
}
static void
adpcmenc_init (ADPCMEnc * enc, ADPCMEncClass * klass)
adpcmenc_init (ADPCMEnc * enc)
{
/* Set defaults. */
enc->blocksize = DEFAULT_ADPCM_BLOCK_SIZE;
@ -422,11 +429,21 @@ static void
adpcmenc_class_init (ADPCMEncClass * klass)
{
GObjectClass *gobjectclass = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) klass;
GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) klass;
gobjectclass->set_property = adpcmenc_set_property;
gobjectclass->get_property = adpcmenc_get_property;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&adpcmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&adpcmenc_src_template));
gst_element_class_set_details_simple (element_class, "ADPCM encoder",
"Codec/Encoder/Audio",
"Encode ADPCM audio",
"Pioneers of the Inevitable <songbird@songbirdnest.com>");
base_class->start = GST_DEBUG_FUNCPTR (adpcmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (adpcmenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (adpcmenc_set_format);
@ -444,21 +461,6 @@ adpcmenc_class_init (ADPCMEncClass * klass)
MIN_ADPCM_BLOCK_SIZE, MAX_ADPCM_BLOCK_SIZE,
DEFAULT_ADPCM_BLOCK_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
adpcmenc_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&adpcmenc_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&adpcmenc_src_template));
gst_element_class_set_details_simple (element_class, "ADPCM encoder",
"Codec/Encoder/Audio",
"Encode ADPCM audio",
"Pioneers of the Inevitable <songbird@songbirdnest.com>");
}
static gboolean