gstreamer/ext/wavpack/gstwavpackdec.c
Tim-Philipp Müller f54198bd02 ext/wavpack/gstwavpackdec.c: Add debug category, use boilerplate macros, fix handling of widths of 32 bits.
Original commit message from CVS:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_setcaps),
(gst_wavpack_dec_base_init), (gst_wavpack_dec_dispose),
(gst_wavpack_dec_class_init), (gst_wavpack_dec_sink_event),
(gst_wavpack_dec_init), (gst_wavpack_dec_format_samples),
(gst_wavpack_dec_chain), (gst_wavpack_dec_plugin_init):
Add debug category, use boilerplate macros, fix handling
of widths of 32 bits.
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init),
(gst_wavpack_parse_dispose), (gst_wavpack_parse_class_init),
(gst_wavpack_parse_index_get_last_entry),
(gst_wavpack_parse_index_get_entry_from_sample),
(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
(gst_wavpack_parse_src_query),
(gst_wavpack_parse_scan_to_find_sample),
(gst_wavpack_parse_send_newsegment),
(gst_wavpack_parse_handle_seek_event),
(gst_wavpack_parse_src_event), (gst_wavpack_parse_init),
(gst_wavpack_parse_get_upstream_length),
(gst_wavpack_parse_pull_buffer),
(gst_wavpack_parse_create_src_pad), (gst_wavpack_parse_loop),
(gst_wavpack_parse_change_state),
(gst_wavepack_parse_sink_activate),
(gst_wavepack_parse_sink_activate_pull),
(gst_wavpack_parse_plugin_init):
* ext/wavpack/gstwavpackparse.h:
Rewrite a bit, mostly to fix flow logic and to make seeking work.
Fix buffer/event refcounting. Add some debug statements. Add
width of 32 to source pad template caps. Use boilerplate macros.
2006-01-29 20:07:49 +00:00

389 lines
11 KiB
C

/* GStreamer Wavpack plugin
* (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
*
* gstwavpackdec.c: raw Wavpack bitstream decoder
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
#include <math.h>
#include <string.h>
#include <wavpack/wavpack.h>
#include "gstwavpackdec.h"
#include "gstwavpackcommon.h"
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_dec_debug);
#define GST_CAT_DEFAULT gst_wavpack_dec_debug
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"width = (int) { 8, 16, 24, 32 }, "
"channels = (int) { 1, 2 }, "
"rate = (int) [ 6000, 192000 ], " "framed = (boolean) true")
);
static GstStaticPadTemplate wvc_sink_factory =
GST_STATIC_PAD_TEMPLATE ("wvcsink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"width = (int) { 8, 16, 24, 32 }, "
"depth = (int) { 8, 16, 24, 32 }, "
"channels = (int) { 1, 2 }, "
"rate = (int) [ 6000, 192000 ], "
"endianness = (int) LITTLE_ENDIAN, " "signed = (boolean) true")
/*
"audio/x-raw-float, "
"width = (int) 32, "
"channels = (int) { 1, 2 }, "
"rate = (int) [ 6000, 192000 ], " "endianness = (int) LITTLE_ENDIAN"
*/
);
static GstFlowReturn gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buffer);
static gboolean gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event);
GST_BOILERPLATE (GstWavpackDec, gst_wavpack_dec, GstElement, GST_TYPE_ELEMENT)
static gboolean gst_wavpack_dec_setcaps (GstPad * pad, GstCaps * caps)
{
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstStructure *structure;
GstCaps *srccaps;
gint bits, rate, channels;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "rate", &rate) ||
!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "width", &bits)) {
return FALSE;
}
wavpackdec->samplerate = rate;
wavpackdec->channels = channels;
wavpackdec->width = bits;
/* 32-bit output seems to be in fact 32 bit int (e.g. Prod_Girls.wv) */
/* if (bits != 32) { */
srccaps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, wavpackdec->samplerate,
"channels", G_TYPE_INT, wavpackdec->channels,
"depth", G_TYPE_INT, bits,
"width", G_TYPE_INT, bits,
"endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
"signed", G_TYPE_BOOLEAN, TRUE, NULL);
/*
} else {
srccaps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, wavpackdec->samplerate,
"channels", G_TYPE_INT, wavpackdec->channels,
"width", G_TYPE_INT, 32,
"endianness", G_TYPE_INT, G_LITTLE_ENDIAN, NULL);
}
*/
/* gst_pad_set_caps (wavpackdec->sinkpad, caps); */
gst_pad_set_caps (wavpackdec->srcpad, srccaps);
gst_pad_use_fixed_caps (wavpackdec->srcpad);
return TRUE;
}
#if 0
static GstPadLinkReturn
gst_wavpack_dec_wvclink (GstPad * pad, GstPad * peer)
{
if (!gst_caps_is_fixed (GST_PAD_CAPS (peer)))
return GST_PAD_LINK_REFUSED;
return GST_PAD_LINK_OK;
}
#endif
static void
gst_wavpack_dec_base_init (gpointer klass)
{
static GstElementDetails plugin_details = {
"WAVPACK decoder",
"Codec/Decoder/Audio",
"Decode Wavpack audio data",
"Arwed v. Merkatz <v.merkatz@gmx.net>"
};
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&wvc_sink_factory));
gst_element_class_set_details (element_class, &plugin_details);
}
static void
gst_wavpack_dec_dispose (GObject * object)
{
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (object);
g_free (wavpackdec->decodebuf);
wavpackdec->decodebuf = NULL;
/* FIXME: what about wavpackdec->stream and wavpackdec->context? (tpm) */
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wavpack_dec_class_init (GstWavpackDecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->dispose = gst_wavpack_dec_dispose;
}
static gboolean
gst_wavpack_dec_src_query (GstPad * pad, GstQuery * query)
{
return gst_pad_query_default (pad, query);
}
static gboolean
gst_wavpack_dec_sink_event (GstPad * pad, GstEvent * event)
{
GstWavpackDec *dec;
dec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "Received %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
/* TODO: save current segment so we can do clipping, for now
* we'll just leave the clipping to the audio sink */
break;
}
default:
break;
}
gst_object_unref (dec);
return gst_pad_event_default (pad, event);
}
static void
gst_wavpack_dec_init (GstWavpackDec * wavpackdec, GstWavpackDecClass * gklass)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackdec);
wavpackdec->sinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"sink"), "sink");
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->sinkpad);
gst_pad_set_chain_function (wavpackdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_chain));
gst_pad_set_setcaps_function (wavpackdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_setcaps));
gst_pad_set_event_function (wavpackdec->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_sink_event));
#if 0
wavpackdec->wvcsinkpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"wvcsink"), "wvcsink");
gst_pad_set_link_function (wavpackdec->wvcsinkpad, gst_wavpack_dec_wvclink);
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->wvcsinkpad);
#endif
wavpackdec->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"src"), "src");
gst_pad_use_fixed_caps (wavpackdec->srcpad);
gst_pad_set_query_function (wavpackdec->srcpad,
GST_DEBUG_FUNCPTR (gst_wavpack_dec_src_query));
gst_element_add_pad (GST_ELEMENT (wavpackdec), wavpackdec->srcpad);
wavpackdec->decodebuf = NULL;
wavpackdec->decodebuf_size = 0;
wavpackdec->stream = (WavpackStream *) g_malloc0 (sizeof (WavpackStream));
wavpackdec->context = (WavpackContext *) g_malloc0 (sizeof (WavpackContext));
}
static void
gst_wavpack_dec_setup_context (GstWavpackDec * wavpackdec, guchar * data,
guchar * cdata)
{
WavpackContext *context = wavpackdec->context;
WavpackStream *stream = wavpackdec->stream;
guint buffer_size;
memset (context, 0, sizeof (context));
context->open_flags = 0;
context->current_stream = 0;
context->num_streams = 1;
memset (stream, 0, sizeof (stream));
context->streams[0] = stream;
gst_wavpack_read_header (&stream->wphdr, data);
stream->blockbuff = data;
if (cdata) {
context->wvc_flag = TRUE;
gst_wavpack_read_header (&stream->wphdr, cdata);
stream->block2buff = cdata;
} else {
context->wvc_flag = FALSE;
}
buffer_size =
stream->wphdr.block_samples * wavpackdec->channels * sizeof (int32_t);
if (wavpackdec->decodebuf_size < buffer_size) {
wavpackdec->decodebuf =
(int32_t *) g_realloc (wavpackdec->decodebuf, buffer_size);
wavpackdec->decodebuf_size = buffer_size;
}
unpack_init (context);
}
static GstBuffer *
gst_wavpack_dec_format_samples (GstWavpackDec * wavpackdec, int32_t * samples,
guint num_samples)
{
GstBuffer *buf;
gint i;
guint8 *dst;
int32_t temp;
buf =
gst_buffer_new_and_alloc (num_samples * wavpackdec->width / 8 *
wavpackdec->channels);
dst = (guint8 *) GST_BUFFER_DATA (buf);
switch (wavpackdec->width) {
case 8:
for (i = 0; i < num_samples * wavpackdec->channels; ++i)
*dst++ = *samples++ + 128;
break;
case 16:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
*dst++ = (guint8) (temp = *samples++);
*dst++ = (guint8) (temp >> 8);
}
break;
case 24:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
*dst++ = (guint8) (temp = *samples++);
*dst++ = (guint8) (temp >> 8);
*dst++ = (guint8) (temp >> 16);
}
break;
case 32:
for (i = 0; i < num_samples * wavpackdec->channels; ++i) {
*dst++ = (guint8) (temp = *samples++);
*dst++ = (guint8) (temp >> 8);
*dst++ = (guint8) (temp >> 16);
*dst++ = (guint8) (temp >> 24);
}
break;
default:
break;
}
return buf;
}
static GstFlowReturn
gst_wavpack_dec_chain (GstPad * pad, GstBuffer * buf)
{
GstWavpackDec *wavpackdec = GST_WAVPACK_DEC (gst_pad_get_parent (pad));
GstBuffer *outbuf, *cbuf = NULL;
GstFlowReturn ret = GST_FLOW_OK;
#if 0
if (gst_pad_is_linked (wavpackdec->wvcsinkpad)) {
if (GST_FLOW_OK != gst_pad_pull_range (wavpackdec->wvcsinkpad,
wavpackdec->wvcflushed_bytes, -1, &cbuf)) {
cbuf = NULL;
} else {
wavpackdec->wvcflushed_bytes += GST_BUFFER_SIZE (cbuf);
}
}
#endif
gst_wavpack_dec_setup_context (wavpackdec, GST_BUFFER_DATA (buf),
cbuf ? GST_BUFFER_DATA (cbuf) : NULL);
unpack_samples (wavpackdec->context, wavpackdec->decodebuf,
wavpackdec->context->streams[0]->wphdr.block_samples);
outbuf =
gst_wavpack_dec_format_samples (wavpackdec, wavpackdec->decodebuf,
wavpackdec->context->streams[0]->wphdr.block_samples);
gst_buffer_stamp (outbuf, buf);
gst_buffer_unref (buf);
if (cbuf) {
gst_buffer_unref (cbuf);
}
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (wavpackdec->srcpad));
GST_LOG_OBJECT (wavpackdec, "pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
ret = gst_pad_push (wavpackdec->srcpad, outbuf);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (wavpackdec, "pad_push: %s", gst_flow_get_name (ret));
}
return ret;
}
gboolean
gst_wavpack_dec_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackdec",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_DEC))
return FALSE;
GST_DEBUG_CATEGORY_INIT (gst_wavpack_dec_debug, "wavpackdec", 0,
"wavpack decoder");
return TRUE;
}