gstreamer/ext/wavpack/gstwavpackparse.c
Stefan Kost bf3fd4f95d Define GstElementDetails as const and also static (when defined as global)
Original commit message from CVS:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/arts/gst_arts.c:
* ext/artsd/gstartsdsink.c:
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/audioresample/gstaudioresample.c:
* ext/bz2/gstbz2dec.c:
* ext/bz2/gstbz2enc.c:
* ext/cdaudio/gstcdaudio.c:
* ext/directfb/dfbvideosink.c:
* ext/divx/gstdivxdec.c:
* ext/divx/gstdivxenc.c:
* ext/dts/gstdtsdec.c: (gst_dtsdec_base_init):
* ext/faac/gstfaac.c: (gst_faac_base_init):
* ext/faad/gstfaad.c:
* ext/gsm/gstgsmdec.c:
* ext/gsm/gstgsmenc.c:
* ext/hermes/gsthermescolorspace.c:
* ext/ivorbis/vorbisfile.c:
* ext/lcs/gstcolorspace.c:
* ext/libfame/gstlibfame.c:
* ext/libmms/gstmms.c: (gst_mms_base_init):
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_base_init):
* ext/nas/nassink.c: (gst_nassink_base_init):
* ext/neon/gstneonhttpsrc.c:
* ext/sdl/sdlaudiosink.c:
* ext/sdl/sdlvideosink.c:
* ext/shout/gstshout.c:
* ext/snapshot/gstsnapshot.c:
* ext/sndfile/gstsf.c:
* ext/swfdec/gstswfdec.c:
* ext/tarkin/gsttarkindec.c:
* ext/tarkin/gsttarkinenc.c:
* ext/theora/theoradec.c:
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_base_init):
* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_base_init):
* ext/xvid/gstxviddec.c:
* ext/xvid/gstxvidenc.c:
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxa_parse_base_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_base_init):
* gst/chart/gstchart.c:
* gst/colorspace/gstcolorspace.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init):
* gst/festival/gstfestival.c:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstiir.c:
* gst/filter/gstlpwsinc.c:
* gst/freeze/gstfreeze.c:
* gst/games/gstpuzzle.c: (gst_puzzle_base_init):
* gst/librfb/gstrfbsrc.c:
* gst/mixmatrix/mixmatrix.c:
* gst/mpeg1sys/gstmpeg1systemencode.c:
* gst/mpeg1videoparse/gstmp1videoparse.c:
* gst/mpeg2sub/gstmpeg2subt.c:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/multifilesink/gstmultifilesink.c:
* gst/overlay/gstoverlay.c:
* gst/passthrough/gstpassthrough.c:
* gst/playondemand/gstplayondemand.c:
* gst/qtdemux/qtdemux.c:
* gst/rtjpeg/gstrtjpegdec.c:
* gst/rtjpeg/gstrtjpegenc.c:
* gst/smooth/gstsmooth.c:
* gst/smoothwave/gstsmoothwave.c:
* gst/spectrum/gstspectrum.c:
* gst/speed/gstspeed.c:
* gst/stereo/gststereo.c:
* gst/switch/gstswitch.c:
* gst/tta/gstttadec.c: (gst_tta_dec_base_init):
* gst/tta/gstttaparse.c: (gst_tta_parse_base_init):
* gst/vbidec/gstvbidec.c:
* gst/videocrop/gstvideocrop.c:
* gst/videodrop/gstvideodrop.c:
* gst/virtualdub/gstxsharpen.c:
* gst/xingheader/gstxingmux.c: (gst_xing_mux_base_init):
* gst/y4m/gsty4mencode.c:
* sys/cdrom/gstcdplayer.c:
* sys/directdraw/gstdirectdrawsink.c:
* sys/directsound/gstdirectsoundsink.c:
* sys/glsink/glimagesink.c:
* sys/qcam/gstqcamsrc.c:
* sys/v4l2/gstv4l2src.c:
* sys/vcd/vcdsrc.c: (gst_vcdsrc_base_init):
* sys/ximagesrc/ximagesrc.c:
Define GstElementDetails as const and also static (when defined as
global)
2006-04-25 21:56:38 +00:00

852 lines
25 KiB
C

/* GStreamer wavpack plugin
* (c) 2005 Arwed v. Merkatz <v.merkatz@gmx.net>
* (c) 2006 Tim-Philipp Müller <tim centricular net>
*
* gstwavpackparse.c: wavpack file parser
*
* 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 "gstwavpackparse.h"
#include "gstwavpackcommon.h"
GST_DEBUG_CATEGORY_STATIC (gst_wavpack_parse_debug);
#define GST_CAT_DEFAULT gst_wavpack_parse_debug
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-wavpack, "
"framed = (boolean) false; "
"audio/x-wavpack-correction, " "framed = (boolean) false")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
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_src_factory = GST_STATIC_PAD_TEMPLATE ("wvcsrc",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("audio/x-wavpack-correction, " "framed = (boolean) true")
);
static gboolean gst_wavepack_parse_sink_activate (GstPad * sinkpad);
static gboolean
gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active);
static void gst_wavpack_parse_loop (GstElement * element);
static GstStateChangeReturn gst_wavpack_parse_change_state (GstElement *
element, GstStateChange transition);
static void gst_wavpack_parse_reset (GstWavpackParse * wavpackparse);
static gint64 gst_wavpack_parse_get_upstream_length (GstWavpackParse * wvparse);
static GstBuffer *gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse,
gint64 offset, guint size, GstFlowReturn * flow);
GST_BOILERPLATE (GstWavpackParse, gst_wavpack_parse, GstElement,
GST_TYPE_ELEMENT);
static void
gst_wavpack_parse_base_init (gpointer klass)
{
static const GstElementDetails plugin_details =
GST_ELEMENT_DETAILS ("WavePack parser",
"Codec/Demuxer/Audio",
"Parses Wavpack files",
"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 (&wvc_src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details (element_class, &plugin_details);
}
static void
gst_wavpack_parse_dispose (GObject * object)
{
gst_wavpack_parse_reset (GST_WAVPACK_PARSE (object));
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wavpack_parse_class_init (GstWavpackParseClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->dispose = gst_wavpack_parse_dispose;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_wavpack_parse_change_state);
}
static GstWavpackParseIndexEntry *
gst_wavpack_parse_index_get_last_entry (GstWavpackParse * wvparse)
{
gint last;
g_assert (wvparse->entries != NULL);
g_assert (wvparse->entries->len > 0);
last = wvparse->entries->len - 1;
return &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, last);
}
static GstWavpackParseIndexEntry *
gst_wavpack_parse_index_get_entry_from_sample (GstWavpackParse * wvparse,
gint64 sample_offset)
{
gint i;
if (wvparse->entries == NULL || wvparse->entries->len == 0)
return NULL;
for (i = wvparse->entries->len - 1; i >= 0; --i) {
GstWavpackParseIndexEntry *entry;
entry = &g_array_index (wvparse->entries, GstWavpackParseIndexEntry, i);
GST_LOG_OBJECT (wvparse, "Index entry %03u: sample %" G_GINT64_FORMAT " @"
" byte %" G_GINT64_FORMAT, entry->sample_offset, entry->byte_offset);
if (entry->sample_offset <= sample_offset &&
sample_offset < entry->sample_offset_end) {
GST_LOG_OBJECT (wvparse, "found match");
return entry;
}
}
GST_LOG_OBJECT (wvparse, "no match in index");
return NULL;
}
static void
gst_wavpack_parse_index_append_entry (GstWavpackParse * wvparse,
gint64 byte_offset, gint64 sample_offset, gint64 num_samples)
{
GstWavpackParseIndexEntry entry;
if (wvparse->entries == NULL) {
wvparse->entries = g_array_new (FALSE, TRUE,
sizeof (GstWavpackParseIndexEntry));
} else {
/* do we have this one already? */
entry = *gst_wavpack_parse_index_get_last_entry (wvparse);
if (entry.byte_offset >= byte_offset)
return;
}
GST_LOG_OBJECT (wvparse, "Adding index entry %8" G_GINT64_FORMAT " - %"
GST_TIME_FORMAT " @ offset 0x%08" G_GINT64_MODIFIER "x", sample_offset,
GST_TIME_ARGS (gst_util_uint64_scale_int (sample_offset,
GST_SECOND, wvparse->samplerate)), byte_offset);
entry.byte_offset = byte_offset;
entry.sample_offset = sample_offset;
entry.sample_offset_end = sample_offset + num_samples;
g_array_append_val (wvparse->entries, entry);
}
static void
gst_wavpack_parse_reset (GstWavpackParse * wavpackparse)
{
wavpackparse->total_samples = 0;
wavpackparse->samplerate = 0;
wavpackparse->channels = 0;
gst_segment_init (&wavpackparse->segment, GST_FORMAT_UNDEFINED);
wavpackparse->current_offset = 0;
wavpackparse->need_newsegment = TRUE;
wavpackparse->upstream_length = -1;
if (wavpackparse->entries) {
g_array_free (wavpackparse->entries, TRUE);
wavpackparse->entries = NULL;
}
if (wavpackparse->srcpad != NULL) {
gboolean res;
GST_DEBUG_OBJECT (wavpackparse, "Removing src pad");
res = gst_element_remove_pad (GST_ELEMENT (wavpackparse),
wavpackparse->srcpad);
g_return_if_fail (res != FALSE);
gst_object_unref (wavpackparse->srcpad);
wavpackparse->srcpad = NULL;
}
}
static gboolean
gst_wavpack_parse_src_query (GstPad * pad, GstQuery * query)
{
GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
GstFormat format;
gboolean ret = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:{
gint64 cur, len;
guint rate;
GST_OBJECT_LOCK (wavpackparse);
cur = wavpackparse->segment.last_stop;
len = wavpackparse->total_samples;
rate = wavpackparse->samplerate;
GST_OBJECT_UNLOCK (wavpackparse);
if (len <= 0 || rate == 0) {
GST_DEBUG_OBJECT (wavpackparse, "haven't read header yet");
break;
}
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
cur = gst_util_uint64_scale_int (cur, GST_SECOND, rate);
gst_query_set_position (query, GST_FORMAT_TIME, cur);
ret = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, GST_FORMAT_DEFAULT, cur);
ret = TRUE;
break;
default:
GST_DEBUG_OBJECT (wavpackparse, "cannot handle position query in "
"%s format", gst_format_get_name (format));
break;
}
break;
}
case GST_QUERY_DURATION:{
gint64 len;
guint rate;
GST_OBJECT_LOCK (wavpackparse);
rate = wavpackparse->samplerate;
len = wavpackparse->total_samples;
GST_OBJECT_UNLOCK (wavpackparse);
if (len <= 0 || rate == 0) {
GST_DEBUG_OBJECT (wavpackparse, "haven't read header yet");
break;
}
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
len = gst_util_uint64_scale_int (len, GST_SECOND, rate);
gst_query_set_duration (query, GST_FORMAT_TIME, len);
ret = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_duration (query, GST_FORMAT_DEFAULT, len);
ret = TRUE;
break;
default:
GST_DEBUG_OBJECT (wavpackparse, "cannot handle duration query in "
"%s format", gst_format_get_name (format));
break;
}
break;
}
default:{
ret = gst_pad_query_default (pad, query);
break;
}
}
gst_object_unref (wavpackparse);
return ret;
}
/* returns TRUE on success, with byte_offset set to the offset of the
* wavpack chunk containing the sample requested. start_sample will be
* set to the first sample in the chunk starting at byte_offset.
* Scanning from the last known header offset to the wanted position
* when seeking forward isn't very clever, but seems fast enough in
* practice and has the nice side effect of populating our index
* table */
static gboolean
gst_wavpack_parse_scan_to_find_sample (GstWavpackParse * parse,
gint64 sample, gint64 * byte_offset, gint64 * start_sample)
{
GstWavpackParseIndexEntry *entry;
GstFlowReturn ret;
gint64 off = 0;
/* first, check if we have to scan at all */
entry = gst_wavpack_parse_index_get_entry_from_sample (parse, sample);
if (entry) {
*byte_offset = entry->byte_offset;
*start_sample = entry->sample_offset;
GST_LOG_OBJECT (parse, "Found index entry: sample %" G_GINT64_FORMAT
" @ offset %" G_GINT64_FORMAT, entry->sample_offset,
entry->byte_offset);
return TRUE;
}
GST_LOG_OBJECT (parse, "No matching entry in index, scanning file ...");
/* if we have an index, we can start scanning from the last known offset
* in there, after all we know our wanted sample is not in the index */
if (parse->entries && parse->entries->len > 0) {
GstWavpackParseIndexEntry *entry;
entry = gst_wavpack_parse_index_get_last_entry (parse);
off = entry->byte_offset;
}
/* now scan forward until we find the chunk we're looking for or hit EOS */
do {
WavpackHeader header = { {0,}
, 0,
};
GstBuffer *buf;
buf = gst_wavpack_parse_pull_buffer (parse, off, sizeof (WavpackHeader),
&ret);
if (buf == NULL)
break;
gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf));
gst_buffer_unref (buf);
gst_wavpack_parse_index_append_entry (parse, off, header.block_index,
header.block_samples);
if (header.block_index <= sample &&
sample < (header.block_index + header.block_samples)) {
*byte_offset = off;
*start_sample = header.block_index;
return TRUE;
}
off += header.ckSize + 8;
} while (1);
GST_DEBUG_OBJECT (parse, "scan failed: %s (off=0x%08" G_GINT64_MODIFIER "x)",
gst_flow_get_name (ret), off);
return FALSE;
}
static gboolean
gst_wavpack_parse_send_newsegment (GstWavpackParse * wvparse, gboolean update)
{
GstSegment *s = &wvparse->segment;
gboolean ret;
gint64 stop_time = -1;
gint64 start_time = 0;
gint64 cur_pos_time;
gint64 diff;
/* segment is in DEFAULT format, but we want to send a TIME newsegment */
start_time = gst_util_uint64_scale_int (s->start, GST_SECOND,
wvparse->samplerate);
if (s->stop != -1) {
stop_time = gst_util_uint64_scale_int (s->stop, GST_SECOND,
wvparse->samplerate);
}
GST_DEBUG_OBJECT (wvparse, "sending newsegment from %" GST_TIME_FORMAT
" to %" GST_TIME_FORMAT, GST_TIME_ARGS (start_time),
GST_TIME_ARGS (stop_time));
/* after a seek, s->last_stop will point to a chunk boundary, ie. from
* which sample we will start sending data again, while s->start will
* point to the sample we actually want to seek to and want to start
* playing right after the seek. Adjust clock-time for the difference
* so we start playing from start_time */
cur_pos_time = gst_util_uint64_scale_int (s->last_stop, GST_SECOND,
wvparse->samplerate);
diff = start_time - cur_pos_time;
ret = gst_pad_push_event (wvparse->srcpad,
gst_event_new_new_segment (update, s->rate, GST_FORMAT_TIME,
start_time, stop_time, start_time - diff));
return ret;
}
static gboolean
gst_wavpack_parse_handle_seek_event (GstWavpackParse * wvparse,
GstEvent * event)
{
GstSeekFlags seek_flags;
GstSeekType start_type;
GstSeekType stop_type;
GstSegment segment;
GstFormat format;
gboolean only_update;
gboolean flush, ret;
gdouble speed;
gint64 stop;
gint64 start; /* sample we want to seek to */
gint64 byte_offset; /* byte offset the chunk we seek to starts at */
gint64 chunk_start; /* first sample in chunk we seek to */
guint rate;
gst_event_parse_seek (event, &speed, &format, &seek_flags, &start_type,
&start, &stop_type, &stop);
if (format != GST_FORMAT_DEFAULT && format != GST_FORMAT_TIME) {
GST_DEBUG ("seeking is only supported in TIME or DEFAULT format");
return FALSE;
}
if (speed < 0.0) {
GST_DEBUG ("only forward playback supported, rate %f not allowed", speed);
return FALSE;
}
GST_OBJECT_LOCK (wvparse);
rate = wvparse->samplerate;
if (rate == 0) {
GST_OBJECT_UNLOCK (wvparse);
GST_DEBUG ("haven't read header yet");
return FALSE;
}
/* convert from time to samples if necessary */
if (format == GST_FORMAT_TIME) {
if (start_type != GST_SEEK_TYPE_NONE)
start = gst_util_uint64_scale_int (start, rate, GST_SECOND);
if (stop_type != GST_SEEK_TYPE_NONE)
stop = gst_util_uint64_scale_int (stop, rate, GST_SECOND);
}
flush = ((seek_flags & GST_SEEK_FLAG_FLUSH) != 0);
if (start < 0) {
GST_OBJECT_UNLOCK (wvparse);
GST_DEBUG_OBJECT (wvparse, "Invalid start sample %" G_GINT64_FORMAT, start);
return FALSE;
}
/* operate on segment copy until we know the seek worked */
segment = wvparse->segment;
gst_segment_set_seek (&segment, speed, GST_FORMAT_DEFAULT,
seek_flags, start_type, start, stop_type, stop, &only_update);
#if 0
if (only_update) {
wvparse->segment = segment;
gst_wavpack_parse_send_newsegment (wvparse, TRUE);
goto done;
}
#endif
gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_start ());
if (flush) {
gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_start ());
} else {
gst_pad_stop_task (wvparse->sinkpad);
}
GST_PAD_STREAM_LOCK (wvparse->sinkpad);
gst_pad_push_event (wvparse->sinkpad, gst_event_new_flush_stop ());
if (flush) {
gst_pad_push_event (wvparse->srcpad, gst_event_new_flush_stop ());
}
GST_DEBUG_OBJECT (wvparse, "Performing seek to %" GST_TIME_FORMAT " sample %"
G_GINT64_FORMAT, GST_TIME_ARGS (segment.start * GST_SECOND / rate),
start);
ret = gst_wavpack_parse_scan_to_find_sample (wvparse, segment.start,
&byte_offset, &chunk_start);
if (ret) {
GST_DEBUG_OBJECT (wvparse, "new offset: %" G_GINT64_FORMAT, byte_offset);
wvparse->current_offset = byte_offset;
/* we want to send a newsegment event with the actual seek position
* as start, even though our first buffer might start before the
* configured segment. We leave it up to the decoder or sink to crop
* the output buffers accordingly */
wvparse->segment = segment;
wvparse->segment.last_stop = chunk_start;
gst_wavpack_parse_send_newsegment (wvparse, FALSE);
} else {
GST_DEBUG_OBJECT (wvparse, "seek failed: don't know where to seek to");
}
GST_PAD_STREAM_UNLOCK (wvparse->sinkpad);
GST_OBJECT_UNLOCK (wvparse);
gst_pad_start_task (wvparse->sinkpad,
(GstTaskFunction) gst_wavpack_parse_loop, wvparse);
return ret;
}
static gboolean
gst_wavpack_parse_src_event (GstPad * pad, GstEvent * event)
{
GstWavpackParse *wavpackparse;
gboolean ret;
wavpackparse = GST_WAVPACK_PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
ret = gst_wavpack_parse_handle_seek_event (wavpackparse, event);
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (wavpackparse);
return ret;
}
static void
gst_wavpack_parse_init (GstWavpackParse * wavpackparse,
GstWavpackParseClass * gclass)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (wavpackparse);
GstPadTemplate *tmpl;
tmpl = gst_element_class_get_pad_template (klass, "sink");
wavpackparse->sinkpad = gst_pad_new_from_template (tmpl, "sink");
gst_pad_set_activate_function (wavpackparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavepack_parse_sink_activate));
gst_pad_set_activatepull_function (wavpackparse->sinkpad,
GST_DEBUG_FUNCPTR (gst_wavepack_parse_sink_activate_pull));
gst_element_add_pad (GST_ELEMENT (wavpackparse), wavpackparse->sinkpad);
wavpackparse->srcpad = NULL;
gst_wavpack_parse_reset (wavpackparse);
}
static gint64
gst_wavpack_parse_get_upstream_length (GstWavpackParse * wavpackparse)
{
GstPad *peer;
gint64 length = -1;
peer = gst_pad_get_peer (wavpackparse->sinkpad);
if (peer) {
GstFormat format = GST_FORMAT_BYTES;
if (!gst_pad_query_duration (peer, &format, &length)) {
length = -1;
} else {
GST_DEBUG ("upstream length: %" G_GINT64_FORMAT, length);
}
gst_object_unref (peer);
} else {
GST_DEBUG ("no peer!");
}
return length;
}
static GstBuffer *
gst_wavpack_parse_pull_buffer (GstWavpackParse * wvparse, gint64 offset,
guint size, GstFlowReturn * flow)
{
GstFlowReturn flow_ret;
GstBuffer *buf = NULL;
if (offset + size >= wvparse->upstream_length) {
wvparse->upstream_length = gst_wavpack_parse_get_upstream_length (wvparse);
if (offset + size >= wvparse->upstream_length) {
GST_DEBUG_OBJECT (wvparse, "EOS: %" G_GINT64_FORMAT " + %u > %"
G_GINT64_FORMAT, offset, size, wvparse->upstream_length);
flow_ret = GST_FLOW_UNEXPECTED;
goto done;
}
}
flow_ret = gst_pad_pull_range (wvparse->sinkpad, offset, size, &buf);
if (flow_ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (wvparse, "pull_range (%" G_GINT64_FORMAT ", %u) "
"failed, flow: %s", offset, size, gst_flow_get_name (flow_ret));
return NULL;
}
if (GST_BUFFER_SIZE (buf) < size) {
GST_DEBUG_OBJECT (wvparse, "Short read at offset %" G_GINT64_FORMAT
", got only %u of %u bytes", offset, GST_BUFFER_SIZE (buf), size);
gst_buffer_unref (buf);
buf = NULL;
flow_ret = GST_FLOW_UNEXPECTED;
}
done:
if (flow)
*flow = flow_ret;
return buf;
}
static gboolean
gst_wavpack_parse_create_src_pad (GstWavpackParse * wvparse, GstBuffer * buf,
WavpackHeader * header)
{
WavpackMetadata meta;
GstCaps *caps = NULL;
guchar *bufptr;
g_assert (wvparse->srcpad == NULL);
bufptr = GST_BUFFER_DATA (buf) + sizeof (WavpackHeader);
while (read_metadata_buff (&meta, GST_BUFFER_DATA (buf), &bufptr)) {
switch (meta.id) {
case ID_WVC_BITSTREAM:{
caps = gst_caps_new_simple ("audio/x-wavpack-correction",
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
wvparse->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wvparse), "wvcsrc"), "wvcsrc");
break;
}
case ID_RIFF_HEADER:{
WaveHeader wheader;
/* skip RiffChunkHeader and ChunkHeader */
g_memmove (&wheader, meta.data + 20, sizeof (WaveHeader));
little_endian_to_native (&wheader, WaveHeaderFormat);
wvparse->samplerate = wheader.SampleRate;
wvparse->channels = wheader.NumChannels;
wvparse->total_samples = header->total_samples;
caps = gst_caps_new_simple ("audio/x-wavpack",
"width", G_TYPE_INT, wheader.BitsPerSample,
"channels", G_TYPE_INT, wvparse->channels,
"rate", G_TYPE_INT, wvparse->samplerate,
"framed", G_TYPE_BOOLEAN, TRUE, NULL);
wvparse->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (wvparse), "src"), "src");
break;
}
default:{
GST_WARNING_OBJECT (wvparse, "unhandled ID: 0x%02x", meta.id);
break;
}
}
if (caps != NULL)
break;
}
if (caps == NULL || wvparse->srcpad == NULL)
return FALSE;
GST_DEBUG_OBJECT (wvparse, "Added src pad with caps %" GST_PTR_FORMAT, caps);
gst_pad_set_query_function (wvparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_query));
gst_pad_set_event_function (wvparse->srcpad,
GST_DEBUG_FUNCPTR (gst_wavpack_parse_src_event));
gst_pad_set_caps (wvparse->srcpad, caps);
gst_pad_use_fixed_caps (wvparse->srcpad);
gst_object_ref (wvparse->srcpad);
gst_element_add_pad (GST_ELEMENT (wvparse), wvparse->srcpad);
gst_element_no_more_pads (GST_ELEMENT (wvparse));
return TRUE;
}
static void
gst_wavpack_parse_loop (GstElement * element)
{
GstWavpackParse *wavpackparse = GST_WAVPACK_PARSE (element);
GstFlowReturn flow_ret;
WavpackHeader header = { {0,}, 0, };
GstBuffer *buf = NULL;
GST_LOG_OBJECT (wavpackparse, "Current offset: %" G_GINT64_FORMAT,
wavpackparse->current_offset);
buf = gst_wavpack_parse_pull_buffer (wavpackparse,
wavpackparse->current_offset, sizeof (WavpackHeader), &flow_ret);
if (buf == NULL && flow_ret == GST_FLOW_UNEXPECTED) {
goto eos;
} else if (buf == NULL) {
goto pause;
}
gst_wavpack_read_header (&header, GST_BUFFER_DATA (buf));
gst_buffer_unref (buf);
GST_LOG_OBJECT (wavpackparse, "Read header at offset %" G_GINT64_FORMAT
": chunk size = %u+8", wavpackparse->current_offset, header.ckSize);
buf = gst_wavpack_parse_pull_buffer (wavpackparse,
wavpackparse->current_offset, header.ckSize + 8, &flow_ret);
if (buf == NULL && flow_ret == GST_FLOW_UNEXPECTED) {
goto eos;
} else if (buf == NULL) {
goto pause;
}
if (wavpackparse->srcpad == NULL) {
if (!gst_wavpack_parse_create_src_pad (wavpackparse, buf, &header)) {
GST_ELEMENT_ERROR (wavpackparse, STREAM, DECODE, (NULL), (NULL));
goto pause;
}
}
gst_wavpack_parse_index_append_entry (wavpackparse,
wavpackparse->current_offset, header.block_index, header.block_samples);
wavpackparse->current_offset += header.ckSize + 8;
wavpackparse->segment.last_stop = header.block_index;
if (wavpackparse->need_newsegment) {
if (gst_wavpack_parse_send_newsegment (wavpackparse, FALSE))
wavpackparse->need_newsegment = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (header.block_index,
GST_SECOND, wavpackparse->samplerate);
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (header.block_samples,
GST_SECOND, wavpackparse->samplerate);
GST_BUFFER_OFFSET (buf) = header.block_index;
gst_buffer_set_caps (buf, GST_PAD_CAPS (wavpackparse->srcpad));
GST_LOG_OBJECT (wavpackparse, "Pushing buffer with time %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
flow_ret = gst_pad_push (wavpackparse->srcpad, buf);
if (flow_ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (wavpackparse, "Push failed, flow: %s",
gst_flow_get_name (flow_ret));
goto pause;
}
return;
eos:
{
GST_DEBUG_OBJECT (wavpackparse, "sending EOS");
if (wavpackparse->srcpad) {
gst_pad_push_event (wavpackparse->srcpad, gst_event_new_eos ());
}
/* fall through and pause task */
}
pause:
{
GST_DEBUG_OBJECT (wavpackparse, "Pausing task");
gst_pad_pause_task (wavpackparse->sinkpad);
return;
}
}
static GstStateChangeReturn
gst_wavpack_parse_change_state (GstElement * element, GstStateChange transition)
{
GstWavpackParse *wvparse = GST_WAVPACK_PARSE (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&wvparse->segment, GST_FORMAT_DEFAULT);
wvparse->segment.last_stop = 0;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_wavpack_parse_reset (wvparse);
break;
default:
break;
}
return ret;
}
static gboolean
gst_wavepack_parse_sink_activate (GstPad * sinkpad)
{
if (gst_pad_check_pull_range (sinkpad)) {
return gst_pad_activate_pull (sinkpad, TRUE);
} else {
return FALSE;
}
}
static gboolean
gst_wavepack_parse_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
gboolean result;
if (active) {
result = gst_pad_start_task (sinkpad,
(GstTaskFunction) gst_wavpack_parse_loop, GST_PAD_PARENT (sinkpad));
} else {
result = gst_pad_stop_task (sinkpad);
}
return result;
}
gboolean
gst_wavpack_parse_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "wavpackparse",
GST_RANK_PRIMARY, GST_TYPE_WAVPACK_PARSE)) {
return FALSE;
}
GST_DEBUG_CATEGORY_INIT (gst_wavpack_parse_debug, "wavpackparse", 0,
"wavpack file parser");
return TRUE;
}