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gstreamer/ext/vorbis/vorbisdec.c

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/* 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
* @see_also: vorbisenc, oggdemux
*
* 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>.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
* ]| Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
* </refsect2>
*
* Last reviewed on 2006-03-01 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstvorbisdec.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 const GstElementDetails vorbis_dec_details =
GST_ELEMENT_DETAILS ("Vorbis audio decoder",
"Codec/Decoder/Audio",
"decode raw vorbis streams to float audio",
"Benjamin Otte <in7y118@public.uni-hamburg.de>");
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) [ 1, MAX ], "
"channels = (int) [ 1, 256 ], " "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 vorbis_dec_finalize (GObject * object);
static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer);
static GstFlowReturn vorbis_dec_chain_forward (GstVorbisDec * vd,
gboolean discont, GstBuffer * buffer);
static GstStateChangeReturn vorbis_dec_change_state (GstElement * element,
GstStateChange transition);
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 = vorbis_dec_finalize;
gstelement_class->change_state = GST_DEBUG_FUNCPTR (vorbis_dec_change_state);
}
static const GstQueryType *
vorbis_get_query_types (GstPad * pad)
{
static const GstQueryType vorbis_dec_src_query_types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
GST_QUERY_CONVERT,
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,
GST_DEBUG_FUNCPTR (vorbis_dec_sink_event));
gst_pad_set_chain_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (vorbis_dec_chain));
gst_pad_set_query_function (dec->sinkpad,
GST_DEBUG_FUNCPTR (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,
GST_DEBUG_FUNCPTR (vorbis_dec_src_event));
gst_pad_set_query_type_function (dec->srcpad,
GST_DEBUG_FUNCPTR (vorbis_get_query_types));
gst_pad_set_query_function (dec->srcpad,
GST_DEBUG_FUNCPTR (vorbis_dec_src_query));
gst_pad_use_fixed_caps (dec->srcpad);
gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad);
dec->queued = NULL;
dec->pendingevents = NULL;
dec->taglist = NULL;
}
static void
vorbis_dec_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 void
gst_vorbis_dec_reset (GstVorbisDec * dec)
{
dec->cur_timestamp = GST_CLOCK_TIME_NONE;
dec->prev_timestamp = GST_CLOCK_TIME_NONE;
dec->granulepos = -1;
dec->discont = TRUE;
dec->seqnum = gst_util_seqnum_next ();
gst_segment_init (&dec->segment, GST_FORMAT_TIME);
g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->queued);
dec->queued = NULL;
g_list_foreach (dec->gather, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->gather);
dec->gather = NULL;
g_list_foreach (dec->decode, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->decode);
dec->decode = NULL;
g_list_foreach (dec->pendingevents, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->pendingevents);
dec->pendingevents = NULL;
if (dec->taglist)
gst_tag_list_free (dec->taglist);
dec->taglist = NULL;
}
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;
if (src_format == *dest_format) {
*dest_value = src_value;
return TRUE;
}
dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
if (!dec->initialized)
goto no_header;
if (dec->sinkpad == pad &&
(src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES))
goto no_format;
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;
}
done:
gst_object_unref (dec);
return res;
/* ERRORS */
no_header:
{
GST_DEBUG_OBJECT (dec, "no header packets received");
res = FALSE;
goto done;
}
no_format:
{
GST_DEBUG_OBJECT (dec, "formats unsupported");
res = FALSE;
goto done;
}
}
static gboolean
vorbis_dec_src_query (GstPad * pad, GstQuery * query)
{
GstVorbisDec *dec;
gboolean res = FALSE;
dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
gint64 granulepos, value;
GstFormat my_format, format;
gint64 time;
/* we start from the last seen granulepos */
granulepos = dec->granulepos;
gst_query_parse_position (query, &format, NULL);
/* and convert to the final format in two steps with time as the
* intermediate step */
my_format = GST_FORMAT_TIME;
if (!(res =
vorbis_dec_convert (pad, GST_FORMAT_DEFAULT, granulepos,
&my_format, &time)))
goto error;
/* correct for the segment values */
time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time);
GST_LOG_OBJECT (dec,
"query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time));
/* and convert to the final format */
if (!(res = vorbis_dec_convert (pad, my_format, time, &format, &value)))
goto error;
gst_query_set_position (query, format, value);
GST_LOG_OBJECT (dec,
"query %p: we return %lld (format %u)", query, value, format);
break;
}
case GST_QUERY_DURATION:
{
GstPad *peer;
if (!(peer = gst_pad_get_peer (dec->sinkpad))) {
GST_WARNING_OBJECT (dec, "sink pad %" GST_PTR_FORMAT " is not linked",
dec->sinkpad);
goto error;
}
res = gst_pad_query (peer, query);
gst_object_unref (peer);
if (!res)
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;
}
done:
gst_object_unref (dec);
return res;
/* ERRORS */
error:
{
GST_WARNING_OBJECT (dec, "error handling query");
goto done;
}
}
static gboolean
vorbis_dec_sink_query (GstPad * pad, GstQuery * query)
{
GstVorbisDec *dec;
gboolean res;
dec = GST_VORBIS_DEC (gst_pad_get_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;
}
done:
gst_object_unref (dec);
return res;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (dec, "error converting value");
goto done;
}
}
static gboolean
vorbis_dec_src_event (GstPad * pad, GstEvent * event)
{
gboolean res = TRUE;
GstVorbisDec *dec;
dec = GST_VORBIS_DEC (gst_pad_get_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;
guint32 seqnum;
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
&stop_type, &stop);
seqnum = gst_event_get_seqnum (event);
gst_event_unref (event);
/* 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 convert_error;
if (!(res = vorbis_dec_convert (pad, format, stop, &tformat, &tstop)))
goto convert_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);
gst_event_set_seqnum (real_seek, seqnum);
res = gst_pad_push_event (dec->sinkpad, real_seek);
break;
}
default:
res = gst_pad_push_event (dec->sinkpad, event);
break;
}
done:
gst_object_unref (dec);
return res;
/* ERRORS */
convert_error:
{
GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek");
goto done;
}
}
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_FLUSH_START:
ret = gst_pad_push_event (dec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
/* here we must clean any state in the decoder */
#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
vorbis_synthesis_restart (&dec->vd);
#endif
gst_vorbis_dec_reset (dec);
ret = gst_pad_push_event (dec->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* we need time for now */
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
GST_DEBUG_OBJECT (dec,
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (time));
/* now configure the values */
gst_segment_set_newsegment_full (&dec->segment, update,
rate, arate, format, start, stop, time);
dec->seqnum = gst_event_get_seqnum (event);
if (dec->initialized)
/* and forward */
ret = gst_pad_push_event (dec->srcpad, event);
else {
/* store it to send once we're initialized */
dec->pendingevents = g_list_append (dec->pendingevents, event);
ret = TRUE;
}
break;
}
case GST_EVENT_TAG:
{
if (dec->initialized)
/* and forward */
ret = gst_pad_push_event (dec->srcpad, event);
else {
/* store it to send once we're initialized */
dec->pendingevents = g_list_append (dec->pendingevents, event);
ret = TRUE;
}
break;
}
default:
ret = gst_pad_push_event (dec->srcpad, event);
break;
}
done:
gst_object_unref (dec);
return ret;
/* ERRORS */
newseg_wrong_format:
{
GST_DEBUG_OBJECT (dec, "received non TIME newsegment");
goto done;
}
}
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
switch (vd->vi.channels) {
case 1:
case 2:
/* nothing */
break;
case 3:{
static const 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 const 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 const 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 const 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;
}
/* FIXME: for >6 channels the layout is not defined by the Vorbis
* spec. These are the gstreamer "defaults" for 7/8 channels and
* NONE layouts for more channels
*/
case 7:{
static const GstAudioChannelPosition pos7[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
};
pos = pos7;
/* fallthrough */
}
case 8:{
static const GstAudioChannelPosition pos8[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
};
pos = pos8;
/* fallthrough */
}
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, vd->vi.channels);
GST_ELEMENT_WARNING (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < vd->vi.channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
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);
if (pos) {
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (vd->vi.channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
gst_pad_set_caps (vd->srcpad, caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
static GstFlowReturn
vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet)
{
guint bitrate = 0;
gchar *encoder = NULL;
GstTagList *list, *old_list;
GstBuffer *buf;
GST_DEBUG_OBJECT (vd, "parsing comment packet");
buf = gst_buffer_new ();
GST_BUFFER_DATA (buf) = packet->packet;
GST_BUFFER_SIZE (buf) = packet->bytes;
list =
gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7,
&encoder);
old_list = vd->taglist;
vd->taglist = gst_tag_list_merge (vd->taglist, list, GST_TAG_MERGE_REPLACE);
if (old_list)
gst_tag_list_free (old_list);
gst_tag_list_free (list);
gst_buffer_unref (buf);
if (!vd->taglist) {
GST_ERROR_OBJECT (vd, "couldn't decode comments");
vd->taglist = gst_tag_list_new ();
}
if (encoder) {
if (encoder[0])
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
g_free (encoder);
}
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER_VERSION, vd->vi.version,
GST_TAG_AUDIO_CODEC, "Vorbis", NULL);
if (vd->vi.bitrate_nominal > 0 && vd->vi.bitrate_nominal <= 0x7FFFFFFF) {
gst_tag_list_add (vd->taglist, 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 && vd->vi.bitrate_upper <= 0x7FFFFFFF) {
gst_tag_list_add (vd->taglist, 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 && vd->vi.bitrate_lower <= 0x7FFFFFFF) {
gst_tag_list_add (vd->taglist, 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 (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, (guint) bitrate, NULL);
}
if (vd->initialized) {
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad,
vd->taglist);
vd->taglist = NULL;
} else {
/* Only post them as messages for the time being. *
* They will be pushed on the pad once the decoder is initialized */
gst_element_post_message (GST_ELEMENT_CAST (vd),
gst_message_new_tag (GST_OBJECT (vd), gst_tag_list_copy (vd->taglist)));
}
return GST_FLOW_OK;
}
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd)
{
GList *walk;
gint res;
g_assert (vd->initialized == FALSE);
if (G_UNLIKELY ((res = vorbis_synthesis_init (&vd->vd, &vd->vi))))
goto synthesis_init_error;
if (G_UNLIKELY ((res = vorbis_block_init (&vd->vd, &vd->vb))))
goto block_init_error;
vd->initialized = TRUE;
if (vd->pendingevents) {
for (walk = vd->pendingevents; walk; walk = g_list_next (walk))
gst_pad_push_event (vd->srcpad, GST_EVENT_CAST (walk->data));
g_list_free (vd->pendingevents);
vd->pendingevents = NULL;
}
if (vd->taglist) {
/* The tags have already been sent on the bus as messages. */
gst_pad_push_event (vd->srcpad, gst_event_new_tag (vd->taglist));
vd->taglist = NULL;
}
return GST_FLOW_OK;
/* ERRORS */
synthesis_init_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't initialize synthesis (%d)", res));
return GST_FLOW_ERROR;
}
block_init_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't initialize block (%d)", res));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet)
{
GstFlowReturn res;
gint ret;
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 ((ret = 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 (%d)", ret));
return GST_FLOW_ERROR;
}
}
/* These samples can be outside of the float -1.0 -- 1.0 range, this
* is allowed, downstream elements are supposed to clip */
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_forward (GstVorbisDec * dec, GstBuffer * buf)
{
GstFlowReturn result;
gint64 outoffset, origoffset;
origoffset = GST_BUFFER_OFFSET (buf);
again:
outoffset = origoffset;
if (outoffset == -1) {
dec->queued = g_list_append (dec->queued, buf);
GST_DEBUG_OBJECT (dec, "queued buffer");
result = GST_FLOW_OK;
} else {
if (G_UNLIKELY (dec->queued)) {
guint size;
GstClockTime ts;
GList *walk;
GST_DEBUG_OBJECT (dec, "first buffer with offset %lld", outoffset);
ts = gst_util_uint64_scale_int (outoffset, GST_SECOND, dec->vi.rate);
size = g_list_length (dec->queued);
/* we walk the queued up list in reverse, and set the buffer fields
* calculating backwards */
for (walk = g_list_last (dec->queued); walk;
walk = g_list_previous (walk)) {
GstBuffer *buffer = GST_BUFFER (walk->data);
guint offset;
offset = GST_BUFFER_SIZE (buffer) / (sizeof (float) * dec->vi.channels);
if (outoffset >= offset)
outoffset -= offset;
else {
/* we can't go below 0, this means this first offset was at the eos
* page and we need to clip to it instead */
GST_DEBUG_OBJECT (dec, "clipping %" G_GINT64_FORMAT,
offset - outoffset);
origoffset += (offset - outoffset);
goto again;
}
GST_BUFFER_OFFSET (buffer) = outoffset;
GST_BUFFER_TIMESTAMP (buffer) =
gst_util_uint64_scale_int (outoffset, GST_SECOND, dec->vi.rate);
GST_BUFFER_DURATION (buffer) = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP
(buffer), ts);
ts = GST_BUFFER_TIMESTAMP (buffer);
GST_DEBUG_OBJECT (dec, "patch buffer %u, offset %" G_GUINT64_FORMAT
", timestamp %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
size, outoffset,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
size--;
}
for (walk = dec->queued; walk; walk = g_list_next (walk)) {
GstBuffer *buffer = GST_BUFFER (walk->data);
/* clips to the configured segment, or returns NULL with buffer
* unreffed when the input buffer is completely outside the segment */
if (!(buffer = gst_audio_buffer_clip (buffer, &dec->segment,
dec->vi.rate, dec->vi.channels * sizeof (float))))
continue;
if (dec->discont) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
dec->discont = FALSE;
}
/* ignore the result */
gst_pad_push (dec->srcpad, buffer);
}
g_list_free (dec->queued);
dec->queued = NULL;
}
/* clip */
if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate,
dec->vi.channels * sizeof (float))))
return GST_FLOW_OK;
if (dec->discont) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
dec->discont = FALSE;
}
result = gst_pad_push (dec->srcpad, buf);
}
return result;
}
static GstFlowReturn
vorbis_dec_push_reverse (GstVorbisDec * dec, GstBuffer * buf)
{
GstFlowReturn result = GST_FLOW_OK;
dec->queued = g_list_prepend (dec->queued, buf);
return result;
}
static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet)
{
float **pcm;
guint sample_count;
GstBuffer *out;
GstFlowReturn result;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, nextts;
gint size;
if (!vd->initialized)
goto not_initialized;
/* FIXME, we should queue undecoded packets here until we get
* a timestamp, then we reverse timestamp the queued packets and
* clip them, then we decode only the ones we want and don't
* keep decoded data in memory.
* Ideally, of course, the demuxer gives us a valid timestamp on
* the first packet.
*/
/* normal data packet */
/* FIXME, we can skip decoding if the packet is outside of the
* segment, this is however not very trivial as we need a previous
* packet to decode the current one so we must be carefull not to
* throw away too much. For now we decode everything and clip right
* before pushing data. */
if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet)))
goto could_not_read;
if (G_UNLIKELY (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;
GST_LOG_OBJECT (vd, "%d samples ready for reading", sample_count);
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 (G_UNLIKELY (result != GST_FLOW_OK))
goto done;
/* get samples ready for reading now, should be sample_count */
if (G_UNLIKELY ((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;
/* this should not overflow */
GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate;
if (packet->granulepos != -1)
vd->granulepos = packet->granulepos - sample_count;
if (vd->cur_timestamp != GST_CLOCK_TIME_NONE) {
/* we have incoming timestamps */
timestamp = vd->cur_timestamp;
GST_DEBUG_OBJECT (vd,
"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 (timestamp, vd->vi.rate);
GST_BUFFER_OFFSET_END (out) = GST_BUFFER_OFFSET (out) + sample_count;
} else {
/* we have incoming granulepos */
GST_BUFFER_OFFSET (out) = vd->granulepos;
if (vd->granulepos != -1) {
GST_DEBUG_OBJECT (vd, "granulepos: %" G_GINT64_FORMAT, vd->granulepos);
GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count;
timestamp =
gst_util_uint64_scale_int (vd->granulepos, GST_SECOND, vd->vi.rate);
nextts =
gst_util_uint64_scale_int (vd->granulepos + sample_count,
GST_SECOND, vd->vi.rate);
GST_DEBUG_OBJECT (vd, "corresponding timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
/* calculate a nano-second accurate duration */
GST_BUFFER_DURATION (out) = GST_CLOCK_DIFF (timestamp, nextts);
GST_DEBUG_OBJECT (vd, "set duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_DURATION (out)));
} else {
timestamp = -1;
}
}
GST_BUFFER_TIMESTAMP (out) = timestamp;
if (vd->granulepos != -1)
vd->granulepos += sample_count;
if (vd->segment.rate >= 0.0)
result = vorbis_dec_push_forward (vd, out);
else
result = vorbis_dec_push_reverse (vd, out);
done:
vorbis_synthesis_read (&vd->vd, sample_count);
GST_DEBUG_OBJECT (vd,
"decoded %ld bytes into %d samples, ts %" GST_TIME_FORMAT, packet->bytes,
sample_count, GST_TIME_ARGS (timestamp));
/* 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"));
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_decode_buffer (GstVorbisDec * vd, GstBuffer * buffer)
{
ogg_packet packet;
GstFlowReturn result = GST_FLOW_OK;
GstClockTime timestamp;
guint64 offset_end;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
offset_end = GST_BUFFER_OFFSET_END (buffer);
/* only ogg has granulepos, demuxers of other container formats
* might provide us with timestamps instead (e.g. matroskademux) */
if (offset_end == GST_BUFFER_OFFSET_NONE && timestamp != GST_CLOCK_TIME_NONE) {
/* we might get multiple consecutive buffers with the same timestamp */
if (timestamp != vd->prev_timestamp) {
vd->cur_timestamp = timestamp;
vd->prev_timestamp = timestamp;
}
} 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 = offset_end;
packet.packetno = 0; /* we don't care */
/*
* 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;
/* error out on empty header packets, but just skip empty data packets */
if (G_UNLIKELY (packet.bytes == 0)) {
if (vd->initialized)
goto empty_buffer;
else
goto empty_header;
}
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, "Already initialized, so ignoring header packet");
goto done;
}
result = vorbis_handle_header_packet (vd, &packet);
} else {
result = vorbis_handle_data_packet (vd, &packet);
}
done:
return result;
empty_buffer:
{
/* don't error out here, just ignore the buffer, it's invalid for vorbis
* but not fatal. */
GST_WARNING_OBJECT (vd, "empty buffer received, ignoring");
if (packet.granulepos != -1)
vd->granulepos = packet.granulepos;
result = GST_FLOW_OK;
goto done;
}
/* ERRORS */
empty_header:
{
GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty header received"));
result = GST_FLOW_ERROR;
vd->discont = TRUE;
goto done;
}
}
/*
* Input:
* Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS
* Discont flag: D D D D
*
* - Each Discont marks a discont in the decoding order.
*
* for vorbis, each buffer is a keyframe when we have the previous
* buffer. This means that to decode buffer 7, we need buffer 6, which
* arrives out of order.
*
* we first gather buffers in the gather queue until we get a DISCONT. We
* prepend each incomming buffer so that they are in reversed order.
*
* gather queue: 9 8 7
* decode queue:
* output queue:
*
* When a DISCONT is received (buffer 4), we move the gather queue to the
* decode queue. This is simply done be taking the head of the gather queue
* and prepending it to the decode queue. This yields:
*
* gather queue:
* decode queue: 7 8 9
* output queue:
*
* Then we decode each buffer in the decode queue in order and put the output
* buffer in the output queue. The first buffer (7) will not produce any output
* because it needs the previous buffer (6) which did not arrive yet. This
* yields:
*
* gather queue:
* decode queue: 7 8 9
* output queue: 9 8
*
* Then we remove the consumed buffers from the decode queue. Buffer 7 is not
* completely consumed, we need to keep it around for when we receive buffer
* 6. This yields:
*
* gather queue:
* decode queue: 7
* output queue: 9 8
*
* Then we accumulate more buffers:
*
* gather queue: 6 5 4
* decode queue: 7
* output queue:
*
* prepending to the decode queue on DISCONT yields:
*
* gather queue:
* decode queue: 4 5 6 7
* output queue:
*
* after decoding and keeping buffer 4:
*
* gather queue:
* decode queue: 4
* output queue: 7 6 5
*
* Etc..
*/
static GstFlowReturn
vorbis_dec_flush_decode (GstVorbisDec * dec)
{
GstFlowReturn res = GST_FLOW_OK;
GList *walk;
walk = dec->decode;
GST_DEBUG_OBJECT (dec, "flushing buffers to decoder");
while (walk) {
GList *next;
GstBuffer *buf = GST_BUFFER_CAST (walk->data);
GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT,
buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
next = g_list_next (walk);
/* decode buffer, prepend to output queue */
res = vorbis_dec_decode_buffer (dec, buf);
/* if we generated output, we can discard the buffer, else we
* keep it in the queue */
if (dec->queued) {
GST_DEBUG_OBJECT (dec, "decoded buffer to %p", dec->queued->data);
dec->decode = g_list_delete_link (dec->decode, walk);
gst_buffer_unref (buf);
} else {
GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping");
}
walk = next;
}
if (dec->granulepos != -1) {
GstClockTime endts;
endts =
gst_util_uint64_scale_int (dec->granulepos, GST_SECOND, dec->vi.rate);
GST_DEBUG_OBJECT (dec, "we have granulepos %" G_GUINT64_FORMAT ", ts %"
GST_TIME_FORMAT, dec->granulepos, GST_TIME_ARGS (endts));
while (dec->queued) {
GstBuffer *buf;
guint sample_count;
buf = GST_BUFFER_CAST (dec->queued->data);
sample_count =
GST_BUFFER_SIZE (buf) / (dec->vi.channels * sizeof (float));
GST_BUFFER_OFFSET_END (buf) = dec->granulepos;
endts =
gst_util_uint64_scale_int (dec->granulepos, GST_SECOND, dec->vi.rate);
dec->granulepos -= sample_count;
GST_BUFFER_OFFSET (buf) = dec->granulepos;
GST_BUFFER_TIMESTAMP (buf) =
gst_util_uint64_scale_int (dec->granulepos, GST_SECOND, dec->vi.rate);
GST_BUFFER_DURATION (buf) = endts - GST_BUFFER_TIMESTAMP (buf);
/* clip, this will unref the buffer in case of clipping */
if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate,
dec->vi.channels * sizeof (float)))) {
GST_DEBUG_OBJECT (dec, "clipped buffer %p", buf);
goto next;
}
if (dec->discont) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
dec->discont = FALSE;
}
GST_DEBUG_OBJECT (dec, "pushing buffer %p, samples %u, "
"ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT,
buf, sample_count, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
res = gst_pad_push (dec->srcpad, buf);
next:
dec->queued = g_list_delete_link (dec->queued, dec->queued);
}
} else {
GST_DEBUG_OBJECT (dec, "we don't have a granulepos yet, delayed push");
}
return res;
}
static GstFlowReturn
vorbis_dec_chain_reverse (GstVorbisDec * vd, gboolean discont, GstBuffer * buf)
{
GstFlowReturn result = GST_FLOW_OK;
/* if we have a discont, move buffers to the decode list */
if (G_UNLIKELY (discont)) {
GST_DEBUG_OBJECT (vd, "received discont");
while (vd->gather) {
GstBuffer *gbuf;
gbuf = GST_BUFFER_CAST (vd->gather->data);
/* remove from the gather list */
vd->gather = g_list_delete_link (vd->gather, vd->gather);
/* copy to decode queue */
vd->decode = g_list_prepend (vd->decode, gbuf);
}
/* flush and decode the decode queue */
result = vorbis_dec_flush_decode (vd);
}
GST_DEBUG_OBJECT (vd, "gathering buffer %p, size %u", buf,
GST_BUFFER_SIZE (buf));
/* add buffer to gather queue */
vd->gather = g_list_prepend (vd->gather, buf);
return result;
}
static GstFlowReturn
vorbis_dec_chain_forward (GstVorbisDec * vd, gboolean discont,
GstBuffer * buffer)
{
GstFlowReturn result;
result = vorbis_dec_decode_buffer (vd, buffer);
gst_buffer_unref (buffer);
return result;
}
static GstFlowReturn
vorbis_dec_chain (GstPad * pad, GstBuffer * buffer)
{
GstVorbisDec *vd;
GstFlowReturn result = GST_FLOW_OK;
gboolean discont;
vd = GST_VORBIS_DEC (gst_pad_get_parent (pad));
discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT);
/* resync on DISCONT */
if (G_UNLIKELY (discont)) {
GST_DEBUG_OBJECT (vd, "received DISCONT buffer");
vd->granulepos = -1;
vd->cur_timestamp = GST_CLOCK_TIME_NONE;
vd->prev_timestamp = GST_CLOCK_TIME_NONE;
#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
vorbis_synthesis_restart (&vd->vd);
#endif
vd->discont = TRUE;
}
if (vd->segment.rate >= 0.0)
result = vorbis_dec_chain_forward (vd, discont, buffer);
else
result = vorbis_dec_chain_reverse (vd, discont, buffer);
gst_object_unref (vd);
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;
gst_vorbis_dec_reset (vd);
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");
vd->initialized = FALSE;
vorbis_block_clear (&vd->vb);
vorbis_dsp_clear (&vd->vd);
vorbis_comment_clear (&vd->vc);
vorbis_info_clear (&vd->vi);
gst_vorbis_dec_reset (vd);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
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