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;
}
"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),
}
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;
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;
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;
projects / platform / upstream / gst-plugins-base.git / commitdiff