Commit cf53cbd8 authored by Wim Taymans's avatar Wim Taymans
Browse files

ext/vorbis/vorbisdec.*: Refactor, use STREAM_LOCK.

Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
(vorbis_handle_type_packet), (vorbis_handle_header_packet),
(copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain),
(vorbis_dec_change_state):
* ext/vorbis/vorbisdec.h:
Refactor, use STREAM_LOCK.
parent 3e0bc017
2005-04-28 Wim Taymans <wim@fluendo.com>
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
(vorbis_handle_type_packet), (vorbis_handle_header_packet),
(copy_samples), (vorbis_handle_data_packet), (vorbis_dec_chain),
(vorbis_dec_change_state):
* ext/vorbis/vorbisdec.h:
Refactor, use STREAM_LOCK.
2005-04-28 Wim Taymans <wim@fluendo.com>
 
* ext/theora/theoradec.c: (_inc_granulepos),
......
......@@ -252,7 +252,7 @@ vorbis_dec_src_query (GstPad * pad, GstQueryType query, GstFormat * format,
return FALSE;
GST_LOG_OBJECT (dec,
"query %u: peer returned granulepos: %llu - we return %llu (format %u)\n",
"query %u: peer returned granulepos: %llu - we return %llu (format %u)",
query, granulepos, *value, *format);
return TRUE;
}
......@@ -308,9 +308,16 @@ vorbis_dec_sink_event (GstPad * pad, GstEvent * event)
"setting granuleposition to %" G_GUINT64_FORMAT " after discont",
start_value);
} else {
GST_WARNING_OBJECT (dec,
"discont event didn't include offset, we might set it wrong now");
if (gst_event_discont_get_value (event, GST_FORMAT_TIME,
(gint64 *) & start_value, &end_value)) {
dec->granulepos = start_value * dec->vi.rate / GST_SECOND;
} else {
GST_WARNING_OBJECT (dec,
"discont event didn't include offset, we might set it wrong now");
}
}
if (dec->packetno < 3) {
if (dec->granulepos != 0)
GST_ELEMENT_ERROR (dec, STREAM, DECODE, (NULL),
......@@ -353,207 +360,315 @@ vorbis_dec_sink_event (GstPad * pad, GstEvent * event)
return ret;
}
static GstFlowReturn
vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet)
{
gchar *encoder = NULL;
GstTagList *list;
GstBuffer *buf;
GST_DEBUG ("parsing comment packet");
buf = gst_buffer_new_and_alloc (packet->bytes);
GST_BUFFER_DATA (buf) = packet->packet;
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_DONTFREE);
list = gst_tag_list_from_vorbiscomment_buffer (buf, "\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_upper > 0)
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, 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);
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);
//gst_element_found_tags_for_pad (GST_ELEMENT (vd), vd->srcpad, 0, list);
return GST_FLOW_OK;
}
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd, ogg_packet * packet)
{
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
/* done */
vorbis_synthesis_init (&vd->vd, &vd->vi);
vorbis_block_init (&vd->vd, &vd->vb);
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, "buffer-frames", G_TYPE_INT, 0, 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);
vd->initialized = TRUE;
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_header_packet (GstVorbisDec * vd, ogg_packet * packet)
{
GstFlowReturn res;
if (packet->packet[0] / 2 != packet->packetno)
goto unexpected_packet;
GST_DEBUG ("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->packetno) {
case 1:
res = vorbis_handle_comment_packet (vd, packet);
break;
case 2:
res = vorbis_handle_type_packet (vd, packet);
break;
default:
/* ignore */
res = GST_FLOW_OK;
break;
}
return res;
/* ERRORS */
unexpected_packet:
{
/* FIXME: just skip? */
GST_WARNING_OBJECT (GST_ELEMENT (vd),
"unexpected packet type %d, expected %d",
(gint) packet->packet[0], (gint) packet->packetno);
return GST_FLOW_UNEXPECTED;
}
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_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet)
{
float **pcm;
guint sample_count;
GstFlowReturn result;
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;
sample_count = vorbis_synthesis_pcmout (&vd->vd, &pcm);
if (sample_count > 0) {
GstBuffer *out;
out = gst_pad_alloc_buffer (vd->srcpad, GST_BUFFER_OFFSET_NONE,
sample_count * vd->vi.channels * sizeof (float),
GST_PAD_CAPS (vd->srcpad));
if (out != NULL) {
float *out_data = (float *) GST_BUFFER_DATA (out);
copy_samples (out_data, pcm, sample_count, vd->vi.channels);
GST_BUFFER_OFFSET (out) = vd->granulepos;
GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count;
GST_BUFFER_TIMESTAMP (out) = vd->granulepos * GST_SECOND / vd->vi.rate;
GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate;
result = gst_pad_push (vd->srcpad, out);
vd->granulepos += sample_count;
} else {
/* no buffer.. */
result = GST_FLOW_OK;
}
vorbis_synthesis_read (&vd->vd, sample_count);
} else {
/* no samples.. */
result = GST_FLOW_OK;
}
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;
}
}
static GstFlowReturn
vorbis_dec_chain (GstPad * pad, GstBuffer * buffer)
{
GstBuffer *buf = GST_BUFFER (buffer);
GstVorbisDec *vd;
ogg_packet packet;
GstFlowReturn result = GST_FLOW_OK;
GST_STREAM_LOCK (pad);
vd = GST_VORBIS_DEC (GST_PAD_PARENT (pad));
/* make ogg_packet out of the buffer */
packet.packet = GST_BUFFER_DATA (buf);
packet.bytes = GST_BUFFER_SIZE (buf);
packet.granulepos = GST_BUFFER_OFFSET_END (buf);
packet.packet = GST_BUFFER_DATA (buffer);
packet.bytes = GST_BUFFER_SIZE (buffer);
packet.granulepos = GST_BUFFER_OFFSET_END (buffer);
packet.packetno = vd->packetno++;
if (GST_BUFFER_OFFSET_END_IS_VALID (buf))
vd->granulepos = GST_BUFFER_OFFSET_END (buf);;
/*
* FIXME. Is there anyway to know that this is the last packet and
* set e_o_s??
*/
packet.e_o_s = 0;
GST_DEBUG ("vorbis granule: %lld", packet.granulepos);
/* switch depending on packet type */
if (packet.packet[0] & 1) {
/* header packet */
if (packet.packet[0] / 2 != packet.packetno) {
/* FIXME: just skip? */
GST_WARNING_OBJECT (GST_ELEMENT (vd),
"unexpected packet type %d, expected %d",
(gint) packet.packet[0], (gint) packet.packetno);
gst_buffer_unref (buffer);
return GST_FLOW_UNEXPECTED;
}
/* 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)) {
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't read header packet"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (packet.packetno == 1) {
gchar *encoder = NULL;
GstTagList *list =
gst_tag_list_from_vorbiscomment_buffer (buf, "\003vorbis", 7,
&encoder);
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_upper > 0)
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, 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);
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);
//gst_element_found_tags_for_pad (GST_ELEMENT (vd), vd->srcpad, 0, list);
} else if (packet.packetno == 2) {
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
/* done */
vorbis_synthesis_init (&vd->vd, &vd->vi);
vorbis_block_init (&vd->vd, &vd->vb);
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, "buffer-frames", G_TYPE_INT, 0, 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:
gst_buffer_unref (buffer);
gst_caps_unref (caps);
GST_ELEMENT_ERROR (vd, STREAM, NOT_IMPLEMENTED, (NULL),
("Unsupported channel count %d", vd->vi.channels));
return GST_FLOW_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);
if (packet.packetno > 3) {
GST_WARNING_OBJECT (vd, "Ignoring header");
goto done;
}
result = vorbis_handle_header_packet (vd, &packet);
} else {
float **pcm;
guint sample_count;
if (packet.packetno < 3) {
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("no header sent yet (packet no is %d)", packet.packetno));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
/* normal data packet */
if (vorbis_synthesis (&vd->vb, &packet)) {
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("couldn't read data packet"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0) {
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("vorbis decoder did not accept data packet"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
sample_count = vorbis_synthesis_pcmout (&vd->vd, &pcm);
if (sample_count > 0) {
int i, j;
GstBuffer *out = gst_pad_alloc_buffer (vd->srcpad, GST_BUFFER_OFFSET_NONE,
sample_count * vd->vi.channels * sizeof (float),
GST_PAD_CAPS (vd->srcpad));
float *out_data;
result = vorbis_handle_data_packet (vd, &packet);
}
if (out != NULL) {
out_data = (float *) GST_BUFFER_DATA (out);
/* granulepos is the last sample in the packet */
if (GST_BUFFER_OFFSET_END_IS_VALID (buffer))
vd->granulepos = GST_BUFFER_OFFSET_END (buffer);;
done:
GST_STREAM_UNLOCK (pad);
#ifdef GST_VORBIS_DEC_SEQUENTIAL
for (i = 0; i < vd->vi.channels; i++) {
memcpy (out_data, pcm[i], sample_count * sizeof (float));
out_data += sample_count;
}
#else
for (j = 0; j < sample_count; j++) {
for (i = 0; i < vd->vi.channels; i++) {
*out_data = pcm[i][j];
out_data++;
}
}
#endif
GST_BUFFER_OFFSET (out) = vd->granulepos;
GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count;
GST_BUFFER_TIMESTAMP (out) = vd->granulepos * GST_SECOND / vd->vi.rate;
GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate;
result = gst_pad_push (vd->srcpad, out);
vd->granulepos += sample_count;
}
vorbis_synthesis_read (&vd->vd, sample_count);
}
}
gst_buffer_unref (buffer);
return result;
......@@ -574,6 +689,9 @@ vorbis_dec_change_state (GstElement * element)
case GST_STATE_READY_TO_PAUSED:
vorbis_info_init (&vd->vi);
vorbis_comment_init (&vd->vc);
vd->initialized = FALSE;
vd->granulepos = 0;
vd->packetno = 0;
break;
case GST_STATE_PAUSED_TO_PLAYING:
break;
......@@ -587,12 +705,14 @@ vorbis_dec_change_state (GstElement * element)
case GST_STATE_PLAYING_TO_PAUSED:
break;
case GST_STATE_PAUSED_TO_READY:
GST_STREAM_LOCK (vd->sinkpad);
vorbis_block_clear (&vd->vb);
vorbis_dsp_clear (&vd->vd);
vorbis_comment_clear (&vd->vc);
vorbis_info_clear (&vd->vi);
vd->packetno = 0;
vd->granulepos = 0;
GST_STREAM_UNLOCK (vd->sinkpad);
break;
case GST_STATE_READY_TO_NULL:
break;
......
......@@ -57,6 +57,8 @@ struct _GstVorbisDec {
vorbis_block vb;
guint packetno;
guint64 granulepos;
gboolean initialized;
};
struct _GstVorbisDecClass {
......
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