gstreamer/ext/vorbis/vorbisdec.c
Stefan Kost 2d826700fa Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
2006-03-24 10:42:11 +00:00

989 lines
26 KiB
C

/* GStreamer
* Copyright (C) 2004 Benjamin Otte <in7y118@public.uni-hamburg.de>
*
* 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.
*/
/**
* SECTION:element-vorbisdec
* @short_description: a decoder that decodes Vorbis to raw audio
* @see_also: vorbisenc, oggdemux
*
* <refsect2>
* <para>
* This element decodes a Vorbis stream to raw float audio.
* <ulink url="http://www.vorbis.com/">Vorbis</ulink> is a royalty-free
* audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
* Foundation</ulink>.
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
* </programlisting>
* Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
* </para>
* </refsect2>
*
* Last reviewed on 2006-03-01 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "vorbisdec.h"
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/tag/tag.h>
#include <gst/audio/multichannel.h>
GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug);
#define GST_CAT_DEFAULT vorbisdec_debug
static GstElementDetails vorbis_dec_details = GST_ELEMENT_DETAILS ("VorbisDec",
"Codec/Decoder/Audio",
"decode raw vorbis streams to float audio",
"Benjamin Otte <in7y118@public.uni-hamburg.de>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
};
static GstStaticPadTemplate vorbis_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 8000, 50000 ], "
"channels = (int) [ 1, 6 ], " "endianness = (int) BYTE_ORDER, "
/* no ifdef in macros, please
#ifdef GST_VORBIS_DEC_SEQUENTIAL
"layout = \"sequential\", "
#endif
*/
"width = (int) 32")
);
static GstStaticPadTemplate vorbis_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-vorbis")
);
GST_BOILERPLATE (GstVorbisDec, gst_vorbis_dec, GstElement, GST_TYPE_ELEMENT);
static void vorbisdec_finalize (GObject * object);
static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer);
static GstStateChangeReturn vorbis_dec_change_state (GstElement * element,
GstStateChange transition);
#if 0
static const GstFormat *vorbis_dec_get_formats (GstPad * pad);
#endif
static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event);
static gboolean vorbis_dec_src_query (GstPad * pad, GstQuery * query);
static gboolean vorbis_dec_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value);
static gboolean vorbis_dec_sink_query (GstPad * pad, GstQuery * query);
static void
gst_vorbis_dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstPadTemplate *src_template, *sink_template;
src_template = gst_static_pad_template_get (&vorbis_dec_src_factory);
gst_element_class_add_pad_template (element_class, src_template);
sink_template = gst_static_pad_template_get (&vorbis_dec_sink_factory);
gst_element_class_add_pad_template (element_class, sink_template);
gst_element_class_set_details (element_class, &vorbis_dec_details);
}
static void
gst_vorbis_dec_class_init (GstVorbisDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
gobject_class->finalize = vorbisdec_finalize;
gstelement_class->change_state = vorbis_dec_change_state;
}
#if 0
static const GstFormat *
vorbis_dec_get_formats (GstPad * pad)
{
static GstFormat src_formats[] = {
GST_FORMAT_BYTES,
GST_FORMAT_DEFAULT, /* samples in the audio case */
GST_FORMAT_TIME,
0
};
static GstFormat sink_formats[] = {
/*GST_FORMAT_BYTES, */
GST_FORMAT_TIME,
GST_FORMAT_DEFAULT, /* granulepos or samples */
0
};
return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats);
}
#endif
#if 0
static const GstEventMask *
vorbis_get_event_masks (GstPad * pad)
{
static const GstEventMask vorbis_dec_src_event_masks[] = {
{GST_EVENT_SEEK, GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH},
{0,}
};
return vorbis_dec_src_event_masks;
}
#endif
static const GstQueryType *
vorbis_get_query_types (GstPad * pad)
{
static const GstQueryType vorbis_dec_src_query_types[] = {
GST_QUERY_POSITION,
0
};
return vorbis_dec_src_query_types;
}
static void
gst_vorbis_dec_init (GstVorbisDec * dec, GstVorbisDecClass * g_class)
{
dec->sinkpad = gst_pad_new_from_static_template (&vorbis_dec_sink_factory,
"sink");
gst_pad_set_event_function (dec->sinkpad, vorbis_dec_sink_event);
gst_pad_set_chain_function (dec->sinkpad, vorbis_dec_chain);
gst_pad_set_query_function (dec->sinkpad, vorbis_dec_sink_query);
gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad);
dec->srcpad = gst_pad_new_from_static_template (&vorbis_dec_src_factory,
"src");
gst_pad_set_event_function (dec->srcpad, vorbis_dec_src_event);
gst_pad_set_query_type_function (dec->srcpad, vorbis_get_query_types);
gst_pad_set_query_function (dec->srcpad, vorbis_dec_src_query);
gst_pad_use_fixed_caps (dec->srcpad);
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->queued = NULL;
}
static void
vorbisdec_finalize (GObject * object)
{
/* Release any possibly allocated libvorbis data.
* _clear functions can safely be called multiple times
*/
GstVorbisDec *vd = GST_VORBIS_DEC (object);
vorbis_block_clear (&vd->vb);
vorbis_dsp_clear (&vd->vd);
vorbis_comment_clear (&vd->vc);
vorbis_info_clear (&vd->vi);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
vorbis_dec_convert (GstPad * pad,
GstFormat src_format, gint64 src_value,
GstFormat * dest_format, gint64 * dest_value)
{
gboolean res = TRUE;
GstVorbisDec *dec;
guint64 scale = 1;
dec = GST_VORBIS_DEC (GST_PAD_PARENT (pad));
if (dec->packetno < 1)
return FALSE;
if (src_format == *dest_format) {
*dest_value = src_value;
return TRUE;
}
if (dec->sinkpad == pad &&
(src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES))
return FALSE;
switch (src_format) {
case GST_FORMAT_TIME:
switch (*dest_format) {
case GST_FORMAT_BYTES:
scale = sizeof (float) * dec->vi.channels;
case GST_FORMAT_DEFAULT:
*dest_value =
scale * gst_util_uint64_scale_int (src_value, dec->vi.rate,
GST_SECOND);
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_DEFAULT:
switch (*dest_format) {
case GST_FORMAT_BYTES:
*dest_value = src_value * sizeof (float) * dec->vi.channels;
break;
case GST_FORMAT_TIME:
*dest_value =
gst_util_uint64_scale_int (src_value, GST_SECOND, dec->vi.rate);
break;
default:
res = FALSE;
}
break;
case GST_FORMAT_BYTES:
switch (*dest_format) {
case GST_FORMAT_DEFAULT:
*dest_value = src_value / (sizeof (float) * dec->vi.channels);
break;
case GST_FORMAT_TIME:
*dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
dec->vi.rate * sizeof (float) * dec->vi.channels);
break;
default:
res = FALSE;
}
break;
default:
res = FALSE;
}
return res;
}
static gboolean
vorbis_dec_src_query (GstPad * pad, GstQuery * query)
{
gint64 granulepos;
GstVorbisDec *dec;
gboolean res = FALSE;
dec = GST_VORBIS_DEC (GST_PAD_PARENT (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gint64 value;
granulepos = dec->granulepos;
gst_query_parse_position (query, &format, NULL);
/* and convert to the final format */
if (!(res =
vorbis_dec_convert (pad, GST_FORMAT_DEFAULT, granulepos, &format,
&value)))
goto error;
value = (value - dec->segment_start) + dec->segment_time;
gst_query_set_position (query, format, value);
GST_LOG_OBJECT (dec,
"query %u: peer returned granulepos: %llu - we return %llu (format %u)",
query, granulepos, value, format);
break;
}
case GST_QUERY_DURATION:
{
/* query peer for total length */
if (!gst_pad_is_linked (dec->sinkpad)) {
GST_WARNING_OBJECT (dec, "sink pad %" GST_PTR_FORMAT " is not linked",
dec->sinkpad);
goto error;
}
if (!(res = gst_pad_query (GST_PAD_PEER (dec->sinkpad), query)))
goto error;
break;
}
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res =
vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
return res;
error:
{
GST_WARNING_OBJECT (dec, "error handling query");
return res;
}
}
static gboolean
vorbis_dec_sink_query (GstPad * pad, GstQuery * query)
{
GstVorbisDec *dec;
gboolean res;
dec = GST_VORBIS_DEC (GST_PAD_PARENT (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CONVERT:
{
GstFormat src_fmt, dest_fmt;
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res =
vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val)))
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;
}
default:
res = gst_pad_query_default (pad, query);
break;
}
error:
return res;
}
static gboolean
vorbis_dec_src_event (GstPad * pad, GstEvent * event)
{
gboolean res = TRUE;
GstVorbisDec *dec = GST_VORBIS_DEC (GST_PAD_PARENT (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:{
GstFormat format, tformat;
gdouble rate;
GstEvent *real_seek;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gint64 tcur, tstop;
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
&stop_type, &stop);
/* we have to ask our peer to seek to time here as we know
* nothing about how to generate a granulepos from the src
* formats or anything.
*
* First bring the requested format to time
*/
tformat = GST_FORMAT_TIME;
if (!(res = vorbis_dec_convert (pad, format, cur, &tformat, &tcur)))
goto error;
if (!(res = vorbis_dec_convert (pad, format, stop, &tformat, &tstop)))
goto error;
/* then seek with time on the peer */
real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME,
flags, cur_type, tcur, stop_type, tstop);
res = gst_pad_push_event (dec->sinkpad, real_seek);
gst_event_unref (event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
return res;
error:
gst_event_unref (event);
return res;
}
static gboolean
vorbis_dec_sink_event (GstPad * pad, GstEvent * event)
{
gboolean ret = FALSE;
GstVorbisDec *dec;
dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (dec, "handling event");
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
ret = gst_pad_push_event (dec->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment (event, &update, &rate, &format, &start,
&stop, &time);
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
if (rate <= 0.0)
goto newseg_wrong_rate;
/* now copy over the values */
dec->segment_rate = rate;
dec->segment_start = start;
dec->segment_stop = stop;
dec->segment_time = time;
dec->granulepos = -1;
dec->cur_timestamp = GST_CLOCK_TIME_NONE;
dec->prev_timestamp = GST_CLOCK_TIME_NONE;
#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
vorbis_synthesis_restart (&dec->vd);
#endif
ret = gst_pad_push_event (dec->srcpad, event);
break;
}
default:
ret = gst_pad_push_event (dec->srcpad, event);
break;
}
done:
gst_object_unref (dec);
return ret;
/* ERRORS */
newseg_wrong_format:
{
GST_DEBUG ("received non TIME newsegment");
goto done;
}
newseg_wrong_rate:
{
GST_DEBUG ("negative rates not supported yet");
goto done;
}
}
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
caps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, vd->vi.rate,
"channels", G_TYPE_INT, vd->vi.channels,
"endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
switch (vd->vi.channels) {
case 1:
case 2:
/* nothing */
break;
case 3:{
static GstAudioChannelPosition pos3[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
};
pos = pos3;
break;
}
case 4:{
static GstAudioChannelPosition pos4[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
};
pos = pos4;
break;
}
case 5:{
static GstAudioChannelPosition pos5[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT
};
pos = pos5;
break;
}
case 6:{
static GstAudioChannelPosition pos6[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE
};
pos = pos6;
break;
}
default:
goto channel_count_error;
}
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
gst_pad_set_caps (vd->srcpad, caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
/* ERROR */
channel_count_error:
{
gst_caps_unref (caps);
GST_ELEMENT_ERROR (vd, STREAM, NOT_IMPLEMENTED, (NULL),
("Unsupported channel count %d", vd->vi.channels));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet)
{
guint bitrate = 0;
gchar *encoder = NULL;
GstTagList *list;
GstBuffer *buf;
GST_DEBUG_OBJECT (vd, "parsing comment packet");
buf = gst_buffer_new_and_alloc (packet->bytes);
GST_BUFFER_DATA (buf) = packet->packet;
list =
gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7,
&encoder);
gst_buffer_unref (buf);
if (!list) {
GST_ERROR_OBJECT (vd, "couldn't decode comments");
list = gst_tag_list_new ();
}
if (encoder) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
g_free (encoder);
}
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER_VERSION, vd->vi.version,
GST_TAG_AUDIO_CODEC, "Vorbis", NULL);
if (vd->vi.bitrate_nominal > 0) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_NOMINAL_BITRATE, (guint) vd->vi.bitrate_nominal, NULL);
bitrate = vd->vi.bitrate_nominal;
}
if (vd->vi.bitrate_upper > 0) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, NULL);
if (!bitrate)
bitrate = vd->vi.bitrate_upper;
}
if (vd->vi.bitrate_lower > 0) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL);
if (!bitrate)
bitrate = vd->vi.bitrate_lower;
}
if (bitrate) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, (guint) bitrate, NULL);
}
gst_element_found_tags_for_pad (GST_ELEMENT (vd), vd->srcpad, list);
return GST_FLOW_OK;
}
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd)
{
g_assert (vd->initialized == FALSE);
vorbis_synthesis_init (&vd->vd, &vd->vi);
vorbis_block_init (&vd->vd, &vd->vb);
vd->initialized = TRUE;
return GST_FLOW_OK;
}
static GstFlowReturn
vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet)
{
GstFlowReturn res;
GST_DEBUG_OBJECT (vd, "parsing header packet");
/* Packetno = 0 if the first byte is exactly 0x01 */
packet->b_o_s = (packet->packet[0] == 0x1) ? 1 : 0;
if (vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet))
goto header_read_error;
switch (packet->packet[0]) {
case 0x01:
res = vorbis_handle_identification_packet (vd);
break;
case 0x03:
res = vorbis_handle_comment_packet (vd, packet);
break;
case 0x05:
res = vorbis_handle_type_packet (vd);
break;
default:
/* ignore */
g_warning ("unknown vorbis header packet found");
res = GST_FLOW_OK;
break;
}
return res;
/* ERRORS */
header_read_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't read header packet"));
return GST_FLOW_ERROR;
}
}
static void
copy_samples (float *out, float **in, guint samples, gint channels)
{
gint i, j;
#ifdef GST_VORBIS_DEC_SEQUENTIAL
for (i = 0; i < channels; i++) {
memcpy (out, in[i], samples * sizeof (float));
out += samples;
}
#else
for (j = 0; j < samples; j++) {
for (i = 0; i < channels; i++) {
*out++ = in[i][j];
}
}
#endif
}
static GstFlowReturn
vorbis_dec_push (GstVorbisDec * dec, GstBuffer * buf)
{
GstFlowReturn result;
gint64 outoffset = GST_BUFFER_OFFSET (buf);
if (outoffset == -1) {
dec->queued = g_list_append (dec->queued, buf);
GST_DEBUG_OBJECT (dec, "queued buffer");
result = GST_FLOW_OK;
} else {
if (dec->queued) {
gint64 size;
GList *walk;
GST_DEBUG_OBJECT (dec, "first buffer with offset %lld", outoffset);
size = g_list_length (dec->queued);
for (walk = g_list_last (dec->queued); walk;
walk = g_list_previous (walk)) {
GstBuffer *buffer = GST_BUFFER (walk->data);
outoffset -=
GST_BUFFER_SIZE (buffer) / (sizeof (float) * dec->vi.channels);
GST_BUFFER_OFFSET (buffer) = outoffset;
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (outoffset, GST_SECOND, dec->vi.rate);
GST_DEBUG_OBJECT (dec, "patch buffer %" G_GUINT64_FORMAT
" offset %" G_GUINT64_FORMAT, size, outoffset);
size--;
}
for (walk = dec->queued; walk; walk = g_list_next (walk)) {
GstBuffer *buffer = GST_BUFFER (walk->data);
/* ignore the result */
gst_pad_push (dec->srcpad, buffer);
}
g_list_free (dec->queued);
dec->queued = NULL;
}
result = gst_pad_push (dec->srcpad, buf);
}
return result;
}
static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet)
{
float **pcm;
guint sample_count;
GstBuffer *out;
GstFlowReturn result;
gint size;
if (!vd->initialized)
goto not_initialized;
/* normal data packet */
if (vorbis_synthesis (&vd->vb, packet))
goto could_not_read;
if (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0)
goto not_accepted;
/* assume all goes well here */
result = GST_FLOW_OK;
/* count samples ready for reading */
if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
goto done;
size = sample_count * vd->vi.channels * sizeof (float);
/* alloc buffer for it */
result =
gst_pad_alloc_buffer_and_set_caps (vd->srcpad, GST_BUFFER_OFFSET_NONE,
size, GST_PAD_CAPS (vd->srcpad), &out);
if (result != GST_FLOW_OK)
goto done;
/* get samples ready for reading now, should be sample_count */
if ((vorbis_synthesis_pcmout (&vd->vd, &pcm)) != sample_count)
goto wrong_samples;
/* copy samples in buffer */
copy_samples ((float *) GST_BUFFER_DATA (out), pcm, sample_count,
vd->vi.channels);
GST_BUFFER_SIZE (out) = size;
GST_BUFFER_OFFSET (out) = vd->granulepos;
if (vd->granulepos != -1) {
GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count;
GST_BUFFER_TIMESTAMP (out) =
gst_util_uint64_scale_int (vd->granulepos, GST_SECOND, vd->vi.rate);
} else {
GST_BUFFER_TIMESTAMP (out) = -1;
}
/* this should not overflow */
GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate;
if (vd->cur_timestamp != GST_CLOCK_TIME_NONE) {
GST_BUFFER_TIMESTAMP (out) = vd->cur_timestamp;
GST_DEBUG ("cur_timestamp: %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT " = % "
GST_TIME_FORMAT, GST_TIME_ARGS (vd->cur_timestamp),
GST_TIME_ARGS (GST_BUFFER_DURATION (out)),
GST_TIME_ARGS (vd->cur_timestamp + GST_BUFFER_DURATION (out)));
vd->cur_timestamp += GST_BUFFER_DURATION (out);
GST_BUFFER_OFFSET (out) = GST_CLOCK_TIME_TO_FRAMES (vd->cur_timestamp,
vd->vi.rate);
GST_BUFFER_OFFSET_END (out) = GST_BUFFER_OFFSET (out) + sample_count;
}
if (vd->granulepos != -1)
vd->granulepos += sample_count;
result = vorbis_dec_push (vd, out);
done:
vorbis_synthesis_read (&vd->vd, sample_count);
/* granulepos is the last sample in the packet */
if (packet->granulepos != -1)
vd->granulepos = packet->granulepos;
return result;
/* ERRORS */
not_initialized:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("no header sent yet (packet no is %d)", packet->packetno));
return GST_FLOW_ERROR;
}
could_not_read:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't read data packet"));
return GST_FLOW_ERROR;
}
not_accepted:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder did not accept data packet"));
return GST_FLOW_ERROR;
}
wrong_samples:
{
gst_buffer_unref (out);
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder reported wrong number of samples"));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
vorbis_dec_chain (GstPad * pad, GstBuffer * buffer)
{
GstVorbisDec *vd;
ogg_packet packet;
GstFlowReturn result = GST_FLOW_OK;
vd = GST_VORBIS_DEC (GST_PAD_PARENT (pad));
if (GST_BUFFER_SIZE (buffer) == 0) {
gst_buffer_unref (buffer);
GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty buffer received"));
return GST_FLOW_ERROR;
}
/* only ogg has granulepos, demuxers of other container formats
* might provide us with timestamps instead (e.g. matroskademux) */
if (GST_BUFFER_OFFSET_END (buffer) == GST_BUFFER_OFFSET_NONE &&
GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE) {
/* we might get multiple consecutive buffers with the same timestamp */
if (GST_BUFFER_TIMESTAMP (buffer) != vd->prev_timestamp) {
vd->cur_timestamp = GST_BUFFER_TIMESTAMP (buffer);
vd->prev_timestamp = GST_BUFFER_TIMESTAMP (buffer);
}
} else {
vd->cur_timestamp = GST_CLOCK_TIME_NONE;
vd->prev_timestamp = GST_CLOCK_TIME_NONE;
}
/* make ogg_packet out of the buffer */
packet.packet = GST_BUFFER_DATA (buffer);
packet.bytes = GST_BUFFER_SIZE (buffer);
packet.granulepos = GST_BUFFER_OFFSET_END (buffer);
packet.packetno = vd->packetno++;
/*
* FIXME. Is there anyway to know that this is the last packet and
* set e_o_s??
* Yes there is, keep one packet at all times and only push out when
* you receive a new one. Implement this.
*/
packet.e_o_s = 0;
GST_DEBUG_OBJECT (vd, "vorbis granule: %" G_GINT64_FORMAT,
(gint64) packet.granulepos);
/* switch depending on packet type */
if (packet.packet[0] & 1) {
if (vd->initialized) {
GST_WARNING_OBJECT (vd, "Ignoring header");
goto done;
}
result = vorbis_handle_header_packet (vd, &packet);
} else {
result = vorbis_handle_data_packet (vd, &packet);
}
GST_DEBUG_OBJECT (vd, "offset end: %" G_GINT64_FORMAT,
(gint64) GST_BUFFER_OFFSET_END (buffer));
done:
gst_buffer_unref (buffer);
return result;
}
static GstStateChangeReturn
vorbis_dec_change_state (GstElement * element, GstStateChange transition)
{
GstVorbisDec *vd = GST_VORBIS_DEC (element);
GstStateChangeReturn res;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
vorbis_info_init (&vd->vi);
vorbis_comment_init (&vd->vc);
vd->initialized = FALSE;
vd->cur_timestamp = GST_CLOCK_TIME_NONE;
vd->prev_timestamp = GST_CLOCK_TIME_NONE;
vd->granulepos = -1;
vd->packetno = 0;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
res = parent_class->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (vd, "PAUSED -> READY, clearing vorbis structures");
vorbis_block_clear (&vd->vb);
vorbis_dsp_clear (&vd->vd);
vorbis_comment_clear (&vd->vc);
vorbis_info_clear (&vd->vi);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}