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faad: port to 0.11

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This commit is contained in:
Wim Taymans 2011-09-27 13:22:31 +02:00
parent 7f4cf50496
commit f03b320c8d
1 changed files with 35 additions and 51 deletions
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86
ext/faad/gstfaad.c
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@ -95,24 +95,12 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }") GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) { 2, 4 }")
); );
#define STATIC_INT_CAPS(bpp) \ #define STATIC_RAW_CAPS(format) \
"audio/x-raw-int, " \ "audio/x-raw, " \
"endianness = (int) BYTE_ORDER, " \ "format = (string) "GST_AUDIO_NE(format)", " \
"signed = (bool) TRUE, " \
"width = (int) " G_STRINGIFY (bpp) ", " \
"depth = (int) " G_STRINGIFY (bpp) ", " \
"rate = (int) [ 8000, 96000 ], " \ "rate = (int) [ 8000, 96000 ], " \
"channels = (int) [ 1, 8 ]" "channels = (int) [ 1, 8 ]"
#if 0
#define STATIC_FLOAT_CAPS(bpp) \
"audio/x-raw-float, " \
"endianness = (int) BYTE_ORDER, " \
"depth = (int) " G_STRINGIFY (bpp) ", " \
"rate = (int) [ 8000, 96000 ], " \
"channels = (int) [ 1, 8 ]"
#endif
/* /*
* All except 16-bit integer are disabled until someone fixes FAAD. * All except 16-bit integer are disabled until someone fixes FAAD.
* FAAD allocates approximately 8*1024*2 bytes bytes, which is enough * FAAD allocates approximately 8*1024*2 bytes bytes, which is enough
@ -122,16 +110,16 @@ static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
*/ */
#define STATIC_CAPS \ #define STATIC_CAPS \
STATIC_INT_CAPS (16) STATIC_RAW_CAPS (S16)
#if 0 #if 0
#define NOTUSED "; " \ #define NOTUSED "; " \
STATIC_INT_CAPS (24) \ STATIC_RAW_CAPS (S24) \
"; " \ "; " \
STATIC_INT_CAPS (32) \ STATIC_RAW_CAPS (S32) \
"; " \ "; " \
STATIC_FLOAT_CAPS (32) \ STATIC_RAW_CAPS (F32) \
"; " \ "; " \
STATIC_FLOAT_CAPS (64) STATIC_RAW_CAPS (F64)
#endif #endif
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
@ -154,12 +142,14 @@ static void gst_faad_flush (GstAudioDecoder * dec, gboolean hard);
static gboolean gst_faad_open_decoder (GstFaad * faad); static gboolean gst_faad_open_decoder (GstFaad * faad);
static void gst_faad_close_decoder (GstFaad * faad); static void gst_faad_close_decoder (GstFaad * faad);
GST_BOILERPLATE (GstFaad, gst_faad, GstAudioDecoder, GST_TYPE_AUDIO_DECODER); #define gst_faad_parent_class parent_class
G_DEFINE_TYPE (GstFaad, gst_faad, GST_TYPE_AUDIO_DECODER);
static void static void
gst_faad_base_init (gpointer klass) gst_faad_class_init (GstFaadClass * klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template)); gst_static_pad_template_get (&src_template));
@ -171,26 +161,18 @@ gst_faad_base_init (gpointer klass)
"Free MPEG-2/4 AAC decoder", "Free MPEG-2/4 AAC decoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>"); "Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
}
static void
gst_faad_class_init (GstFaadClass * klass)
{
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start); base_class->start = GST_DEBUG_FUNCPTR (gst_faad_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop); base_class->stop = GST_DEBUG_FUNCPTR (gst_faad_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format); base_class->set_format = GST_DEBUG_FUNCPTR (gst_faad_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse); base_class->parse = GST_DEBUG_FUNCPTR (gst_faad_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faad_handle_frame);
base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush); base_class->flush = GST_DEBUG_FUNCPTR (gst_faad_flush);
GST_DEBUG_CATEGORY_INIT (faad_debug, "faad", 0, "AAC decoding");
} }
static void static void
gst_faad_init (GstFaad * faad, GstFaadClass * klass) gst_faad_init (GstFaad * faad)
{ {
gst_faad_reset (faad); gst_faad_reset (faad);
} }
@ -280,6 +262,8 @@ gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
GstStructure *str = gst_caps_get_structure (caps, 0); GstStructure *str = gst_caps_get_structure (caps, 0);
GstBuffer *buf; GstBuffer *buf;
const GValue *value; const GValue *value;
guint8 *cdata;
gsize csize;
/* clean up current decoder, rather than trying to reconfigure */ /* clean up current decoder, rather than trying to reconfigure */
gst_faad_close_decoder (faad); gst_faad_close_decoder (faad);
@ -294,8 +278,6 @@ gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
guint32 samplerate; guint32 samplerate;
#endif #endif
guint8 channels; guint8 channels;
guint8 *cdata;
guint csize;
/* We have codec data, means packetised stream */ /* We have codec data, means packetised stream */
faad->packetised = TRUE; faad->packetised = TRUE;
@ -303,8 +285,7 @@ gst_faad_set_format (GstAudioDecoder * dec, GstCaps * caps)
buf = gst_value_get_buffer (value); buf = gst_value_get_buffer (value);
g_return_val_if_fail (buf != NULL, FALSE); g_return_val_if_fail (buf != NULL, FALSE);
cdata = GST_BUFFER_DATA (buf); cdata = gst_buffer_map (buf, &csize, NULL, GST_MAP_READ);
csize = GST_BUFFER_SIZE (buf);
if (csize < 2) if (csize < 2)
goto wrong_length; goto wrong_length;
@ -375,18 +356,21 @@ wrong_length:
{ {
GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long"); GST_DEBUG_OBJECT (faad, "codec_data less than 2 bytes long");
gst_object_unref (faad); gst_object_unref (faad);
gst_buffer_unmap (buf, cdata, csize);
return FALSE; return FALSE;
} }
open_failed: open_failed:
{ {
GST_DEBUG_OBJECT (faad, "failed to create decoder"); GST_DEBUG_OBJECT (faad, "failed to create decoder");
gst_object_unref (faad); gst_object_unref (faad);
gst_buffer_unmap (buf, cdata, csize);
return FALSE; return FALSE;
} }
init_failed: init_failed:
{ {
GST_DEBUG_OBJECT (faad, "faacDecInit2() failed"); GST_DEBUG_OBJECT (faad, "faacDecInit2() failed");
gst_object_unref (faad); gst_object_unref (faad);
gst_buffer_unmap (buf, cdata, csize);
return FALSE; return FALSE;
} }
} }
@ -662,9 +646,13 @@ gst_faad_parse (GstAudioDecoder * dec, GstAdapter * adapter,
*length = size; *length = size;
return GST_FLOW_OK; return GST_FLOW_OK;
} else { } else {
data = gst_adapter_peek (adapter, size); gboolean ret;
return gst_faad_sync (faad, data, size, !eos, offset, length) ?
GST_FLOW_OK : GST_FLOW_UNEXPECTED; data = gst_adapter_map (adapter, size);
ret = gst_faad_sync (faad, data, size, !eos, offset, length);
gst_adapter_unmap (adapter, 0);
return (ret ? GST_FLOW_OK : GST_FLOW_UNEXPECTED);
} }
} }
@ -673,7 +661,7 @@ gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{ {
GstFaad *faad; GstFaad *faad;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
guint input_size; gsize input_size;
guchar *input_data; guchar *input_data;
GstBuffer *outbuf; GstBuffer *outbuf;
faacDecFrameInfo info; faacDecFrameInfo info;
@ -685,8 +673,7 @@ gst_faad_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
if (G_UNLIKELY (!buffer)) if (G_UNLIKELY (!buffer))
return GST_FLOW_OK; return GST_FLOW_OK;
input_data = GST_BUFFER_DATA (buffer); input_data = gst_buffer_map (buffer, &input_size, NULL, GST_MAP_READ);
input_size = GST_BUFFER_SIZE (buffer);
init: init:
/* init if not already done during capsnego */ /* init if not already done during capsnego */
@ -764,20 +751,17 @@ init:
goto sample_overflow; goto sample_overflow;
/* note: info.samples is total samples, not per channel */ /* note: info.samples is total samples, not per channel */
ret = /* FIXME, add bufferpool and allocator support to the base class */
gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD outbuf = gst_buffer_new_allocate (NULL, info.samples * faad->bps, 0);
(faad), 0, info.samples * faad->bps, gst_buffer_fill (outbuf, 0, out, info.samples * faad->bps);
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (faad)), &outbuf);
if (ret != GST_FLOW_OK)
goto out;
memcpy (GST_BUFFER_DATA (outbuf), out, GST_BUFFER_SIZE (outbuf));
ret = gst_audio_decoder_finish_frame (dec, outbuf, 1); ret = gst_audio_decoder_finish_frame (dec, outbuf, 1);
} }
} while (FALSE); } while (FALSE);
out: out:
gst_buffer_unmap (buffer, input_data, input_size);
return ret; return ret;
/* ERRORS */ /* ERRORS */