);
#define gst_vorbis_dec_parent_class parent_class
-G_DEFINE_TYPE (GST_VORBIS_DEC_GLIB_TYPE_NAME, gst_vorbis_dec, GST_TYPE_ELEMENT);
+G_DEFINE_TYPE (GstVorbisDec, gst_vorbis_dec, GST_TYPE_AUDIO_DECODER);
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 GstFlowReturn vorbis_dec_chain_reverse (GstVorbisDec * vd,
- gboolean discont, GstBuffer * buf);
-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 gboolean vorbis_dec_start (GstAudioDecoder * dec);
+static gboolean vorbis_dec_stop (GstAudioDecoder * dec);
+static GstFlowReturn vorbis_dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
+static void vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard);
static void
gst_vorbis_dec_class_init (GstVorbisDecClass * klass)
{
+ GstPadTemplate *src_template, *sink_template;
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
- GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
gobject_class->finalize = vorbis_dec_finalize;
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&vorbis_dec_src_factory));
+ src_template = gst_static_pad_template_get (&vorbis_dec_src_factory);
+ gst_element_class_add_pad_template (element_class, src_template);
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&vorbis_dec_sink_factory));
+ 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_simple (gstelement_class,
+ gst_element_class_set_details_simple (element_class,
"Vorbis audio decoder", "Codec/Decoder/Audio",
GST_VORBIS_DEC_DESCRIPTION,
"Benjamin Otte <otte@gnome.org>, Chris Lord <chris@openedhand.com>");
- 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;
+ base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (vorbis_dec_handle_frame);
+ base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush);
}
static void
gst_vorbis_dec_init (GstVorbisDec * dec)
{
- 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
#ifndef USE_TREMOLO
vorbis_block_clear (&vd->vb);
#endif
-
vorbis_dsp_clear (&vd->vd);
vorbis_comment_clear (&vd->vc);
vorbis_info_clear (&vd->vi);
static void
gst_vorbis_dec_reset (GstVorbisDec * dec)
{
- dec->last_timestamp = GST_CLOCK_TIME_NONE;
- 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)
+vorbis_dec_start (GstAudioDecoder * dec)
{
- gboolean res = TRUE;
- GstVorbisDec *dec;
- guint64 scale = 1;
-
- if (src_format == *dest_format) {
- *dest_value = src_value;
- return TRUE;
- }
+ GstVorbisDec *vd = GST_VORBIS_DEC (dec);
- 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 = dec->width * 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 * dec->width * 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 / (dec->width * dec->vi.channels);
- break;
- case GST_FORMAT_TIME:
- *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND,
- dec->vi.rate * dec->width * dec->vi.channels);
- break;
- default:
- res = FALSE;
- }
- break;
- default:
- res = FALSE;
- }
-done:
- gst_object_unref (dec);
-
- return res;
+ GST_DEBUG_OBJECT (dec, "start");
+ vorbis_info_init (&vd->vi);
+ vorbis_comment_init (&vd->vc);
+ vd->initialized = FALSE;
+ gst_vorbis_dec_reset (vd);
- /* 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;
- }
+ return TRUE;
}
static gboolean
-vorbis_dec_src_query (GstPad * pad, GstQuery * query)
+vorbis_dec_stop (GstAudioDecoder * dec)
{
- GstVorbisDec *dec;
- gboolean res = FALSE;
-
- dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
- if (G_UNLIKELY (dec == NULL))
- return FALSE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_POSITION:
- {
- gint64 value;
- GstFormat format;
- gint64 time;
-
- gst_query_parse_position (query, &format, NULL);
-
- /* we start from the last seen time */
- time = dec->last_timestamp;
- /* 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, GST_FORMAT_TIME, time, &format, &value)))
- goto error;
-
- gst_query_set_position (query, format, value);
-
- GST_LOG_OBJECT (dec,
- "query %p: we return %" G_GINT64_FORMAT " (format %u)", query, value,
- format);
-
- break;
- }
- case GST_QUERY_DURATION:
- {
- res = gst_pad_peer_query (dec->sinkpad, query);
- if (!res)
- goto error;
+ GstVorbisDec *vd = GST_VORBIS_DEC (dec);
- 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;
+ GST_DEBUG_OBJECT (dec, "stop");
+ vd->initialized = FALSE;
+#ifndef USE_TREMOLO
+ vorbis_block_clear (&vd->vb);
+#endif
+ vorbis_dsp_clear (&vd->vd);
+ vorbis_comment_clear (&vd->vc);
+ vorbis_info_clear (&vd->vi);
+ gst_vorbis_dec_reset (vd);
- /* 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;
- }
+ return TRUE;
}
+#if 0
static gboolean
vorbis_dec_src_event (GstPad * pad, GstEvent * event)
{
GstVorbisDec *dec;
dec = GST_VORBIS_DEC (gst_pad_get_parent (pad));
- if (G_UNLIKELY (dec == NULL)) {
- gst_event_unref (event);
- return FALSE;
- }
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_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:
- if (dec->segment.rate < 0.0)
- vorbis_dec_chain_reverse (dec, TRUE, NULL);
- 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_SEGMENT:
- {
- const GstSegment *segment;
-
- gst_event_parse_segment (event, &segment);
-
- /* we need time for now */
- if (segment->format != GST_FORMAT_TIME)
- goto newseg_wrong_format;
-
- GST_DEBUG_OBJECT (dec, "segment: %" GST_SEGMENT_FORMAT, segment);
-
- /* now configure the values */
- gst_segment_copy_into (segment, &dec->segment);
- 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_event_default (pad, 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;
- gint width = GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH;
+ GstAudioInfo info;
+
+ gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
+ vd->vi.channels);
- switch (vd->vi.channels) {
+ switch (info.channels) {
case 1:
case 2:
/* nothing */
case 6:
case 7:
case 8:
- pos = gst_vorbis_channel_positions[vd->vi.channels - 1];
+ {
+ const GstAudioChannelPosition *pos;
+ gint i;
+
+ pos = gst_vorbis_channel_positions[info.channels - 1];
+ for (i = 0; i < info.channels; i++)
+ info.position[i] = pos[i];
break;
+ }
default:{
- gint i;
- GstAudioChannelPosition *posn =
- g_new (GstAudioChannelPosition, vd->vi.channels);
+ gint i, max_pos = MAX (info.channels, 64);
- GST_ELEMENT_WARNING (GST_ELEMENT (vd), STREAM, DECODE,
+ GST_ELEMENT_WARNING (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;
+ for (i = 0; i < max_pos; i++)
+ info.position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+ break;
}
}
- /* negotiate width with downstream */
- caps = gst_pad_get_allowed_caps (vd->srcpad);
- if (caps) {
- if (!gst_caps_is_empty (caps)) {
- GstStructure *s;
-
- s = gst_caps_get_structure (caps, 0);
- /* template ensures 16 or 32 */
- gst_structure_get_int (s, "width", &width);
-
- GST_INFO_OBJECT (vd, "using %s with %d channels and %d bit audio depth",
- gst_structure_get_name (s), vd->vi.channels, width);
- }
- gst_caps_unref (caps);
- }
- vd->width = width >> 3;
+ caps = gst_audio_info_to_caps (&info);
+ gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (vd), caps);
+ gst_caps_unref (caps);
+ vd->info = info;
/* select a copy_samples function, this way we can have specialized versions
* for mono/stereo and avoid the depth switch in tremor case */
- vd->copy_samples = get_copy_sample_func (vd->vi.channels, vd->width);
-
- caps = gst_caps_make_writable (gst_pad_get_pad_template_caps (vd->srcpad));
- gst_caps_set_simple (caps, "rate", G_TYPE_INT, vd->vi.rate,
- "channels", G_TYPE_INT, vd->vi.channels,
- "width", G_TYPE_INT, width, 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);
+ vd->copy_samples = get_copy_sample_func (info.channels);
return GST_FLOW_OK;
}
}
if (vd->initialized) {
- gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad,
- vd->taglist);
+ gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd),
+ GST_AUDIO_DECODER_SRC_PAD (vd), vd->taglist);
vd->taglist = NULL;
} else {
/* Only post them as messages for the time being. *
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd)
{
- GList *walk;
gint res;
g_assert (vd->initialized == FALSE);
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));
+ gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (vd),
+ gst_event_new_tag (vd->taglist));
vd->taglist = NULL;
}
return GST_FLOW_OK;
/* Packetno = 0 if the first byte is exactly 0x01 */
packet->b_o_s = ((gst_ogg_packet_data (packet))[0] == 0x1) ? 1 : 0;
-#ifdef USE_TREMOLO
+#ifdef USE_TREMELO
if ((ret = vorbis_dsp_headerin (&vd->vi, &vd->vc, packet)))
#else
if ((ret = vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet)))
res = GST_FLOW_OK;
break;
}
+
+ /* consumer header packet/frame */
+ gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), NULL, 1);
+
return res;
/* ERRORS */
}
static GstFlowReturn
-vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf)
+vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer)
{
- GstFlowReturn result;
-
- /* clip */
- if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate,
- dec->vi.channels * dec->width))) {
- GST_LOG_OBJECT (dec, "clipped buffer");
- return GST_FLOW_OK;
- }
+ ogg_packet *packet;
+ ogg_packet_wrapper packet_wrapper;
+ GstFlowReturn ret;
- if (dec->discont) {
- GST_LOG_OBJECT (dec, "setting DISCONT");
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
- dec->discont = FALSE;
- }
+ gst_ogg_packet_wrapper_map (&packet_wrapper, buffer);
+ packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
- GST_DEBUG_OBJECT (dec,
- "pushing time %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ ret = vorbis_handle_header_packet (vd, packet);
- result = gst_pad_push (dec->srcpad, buf);
+ gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer);
- return result;
+ return ret;
}
+#define MIN_NUM_HEADERS 3
static GstFlowReturn
-vorbis_dec_push_reverse (GstVorbisDec * dec, GstBuffer * buf)
+vorbis_dec_handle_header_caps (GstVorbisDec * vd)
{
GstFlowReturn result = GST_FLOW_OK;
+ GstCaps *caps;
+ GstStructure *s = NULL;
+ const GValue *array = NULL;
- dec->queued = g_list_prepend (dec->queued, buf);
+ caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (vd));
+ if (caps)
+ s = gst_caps_get_structure (caps, 0);
+ if (s)
+ array = gst_structure_get_value (s, "streamheader");
- return result;
-}
+ if (caps)
+ gst_caps_unref (caps);
-static void
-vorbis_do_timestamps (GstVorbisDec * vd, GstBuffer * buf, gboolean reverse,
- GstClockTime timestamp, GstClockTime duration)
-{
- /* interpolate reverse */
- if (vd->last_timestamp != -1 && duration != -1 && reverse)
- vd->last_timestamp -= duration;
-
- /* take buffer timestamp, use interpolated timestamp otherwise */
- if (timestamp != -1)
- vd->last_timestamp = timestamp;
- else
- timestamp = vd->last_timestamp;
-
- /* interpolate forwards */
- if (vd->last_timestamp != -1 && duration != -1 && !reverse)
- vd->last_timestamp += duration;
-
- GST_LOG_OBJECT (vd,
- "keeping timestamp %" GST_TIME_FORMAT " ts %" GST_TIME_FORMAT " dur %"
- GST_TIME_FORMAT, GST_TIME_ARGS (vd->last_timestamp),
- GST_TIME_ARGS (timestamp), GST_TIME_ARGS (duration));
-
- if (buf) {
- GST_BUFFER_TIMESTAMP (buf) = timestamp;
- GST_BUFFER_DURATION (buf) = duration;
+ if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) {
+ const GValue *value = NULL;
+ GstBuffer *buf = NULL;
+ gint i = 0;
+
+ while (result == GST_FLOW_OK) {
+ value = gst_value_array_get_value (array, i);
+ buf = gst_value_get_buffer (value);
+ if (!buf)
+ goto null_buffer;
+ result = vorbis_dec_handle_header_buffer (vd, buf);
+ i++;
+ }
+ } else
+ goto array_error;
+
+done:
+ return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK);
+
+ /* ERRORS */
+array_error:
+ {
+ GST_WARNING_OBJECT (vd, "streamheader array not found");
+ result = GST_FLOW_ERROR;
+ goto done;
+ }
+null_buffer:
+ {
+ GST_WARNING_OBJECT (vd, "streamheader with null buffer received");
+ result = GST_FLOW_ERROR;
+ goto done;
}
}
+
static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
GstClockTime timestamp, GstClockTime duration)
{
-#ifdef USE_TREMOLO
+#ifdef USE_TREMELO
vorbis_sample_t *pcm;
#else
vorbis_sample_t **pcm;
guint8 *data;
gsize size;
- if (G_UNLIKELY (!vd->initialized))
- goto not_initialized;
+ if (G_UNLIKELY (!vd->initialized)) {
+ result = vorbis_dec_handle_header_caps (vd);
+ if (result != GST_FLOW_OK)
+ goto not_initialized;
+ }
/* normal data packet */
/* FIXME, we can skip decoding if the packet is outside of the
* throw away too much. For now we decode everything and clip right
* before pushing data. */
-#ifdef USE_TREMOLO
- if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vd, packet, 1)))
+#ifdef USE_TREMELO
+ if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vb, packet, 1)))
goto could_not_read;
#else
if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet)))
if ((sample_count = vorbis_dsp_pcmout (&vd->vd, NULL, 0)) == 0)
#else
if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0)
-#endif
goto done;
+#endif
- size = sample_count * vd->vi.channels * vd->width;
+ size = sample_count * vd->info.bpf;
GST_LOG_OBJECT (vd, "%d samples ready for reading, size %d", sample_count,
size);
/* alloc buffer for it */
- out = gst_buffer_new_and_alloc (size);
+ out = gst_buffer_new_allocate (NULL, size, 0);
+ data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
/* get samples ready for reading now, should be sample_count */
#ifdef USE_TREMOLO
- pcm = GST_BUFFER_DATA (out);
- if (G_UNLIKELY ((vorbis_dsp_pcmout (&vd->vd, pcm,
- sample_count)) != sample_count))
+ if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, data, sample_count) !=
+ sample_count))
#else
- if (G_UNLIKELY ((vorbis_synthesis_pcmout (&vd->vd, &pcm)) != sample_count))
+ if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count))
#endif
goto wrong_samples;
#ifndef USE_TREMOLO
/* copy samples in buffer */
- data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE);
vd->copy_samples ((vorbis_sample_t *) data, pcm,
- sample_count, vd->vi.channels, vd->width);
+ sample_count, vd->info.channels);
#endif
GST_LOG_OBJECT (vd, "setting output size to %d", size);
gst_buffer_unmap (out, data, size);
- /* this should not overflow */
- if (duration == -1)
- duration = sample_count * GST_SECOND / vd->vi.rate;
-
- vorbis_do_timestamps (vd, out, FALSE, timestamp, duration);
-
- if (vd->segment.rate >= 0.0)
- result = vorbis_dec_push_forward (vd, out);
- else
- result = vorbis_dec_push_reverse (vd, out);
-
done:
- if (out == NULL) {
- /* no output, still keep track of timestamps */
- vorbis_do_timestamps (vd, NULL, FALSE, timestamp, duration);
- }
+ /* whether or not data produced, consume one frame and advance time */
+ result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), out, 1);
+
#ifdef USE_TREMOLO
vorbis_dsp_read (&vd->vd, sample_count);
#else
{
GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
(NULL), ("no header sent yet"));
- return GST_FLOW_ERROR;
+ return GST_FLOW_NOT_NEGOTIATED;
}
could_not_read:
{
}
static GstFlowReturn
-vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer)
+vorbis_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
ogg_packet *packet;
ogg_packet_wrapper packet_wrapper;
- GstFlowReturn ret;
-
- gst_ogg_packet_wrapper_map (&packet_wrapper, buffer);
- packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
-
- ret = vorbis_handle_header_packet (vd, packet);
-
- gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer);
-
- return ret;
-}
-
-
-#define MIN_NUM_HEADERS 3
-static GstFlowReturn
-vorbis_dec_handle_header_caps (GstVorbisDec * vd, GstBuffer * buffer)
-{
GstFlowReturn result = GST_FLOW_OK;
- GstCaps *caps;
- GstStructure *s;
- const GValue *array;
- const GValue *value = NULL;
- GstBuffer *buf = NULL;
-
- if ((caps = gst_pad_get_current_caps (vd->sinkpad)) == NULL)
- goto no_caps;
-
- if ((s = gst_caps_get_structure (caps, 0)) == NULL)
- goto no_caps;
-
- array = gst_structure_get_value (s, "streamheader");
-
- if (array == NULL || (gst_value_array_get_size (array) < MIN_NUM_HEADERS))
- goto array_error;
-
- /* initial header */
- value = gst_value_array_get_value (array, 0);
- buf = gst_value_get_buffer (value);
- if (!buf)
- goto null_buffer;
- result = vorbis_dec_handle_header_buffer (vd, buf);
- if (result != GST_FLOW_OK)
- goto buffer_error;
-
- /* comment header */
- value = gst_value_array_get_value (array, 1);
- buf = gst_value_get_buffer (value);
- if (!buf)
- goto null_buffer;
- result = vorbis_dec_handle_header_buffer (vd, buf);
- if (result != GST_FLOW_OK)
- goto buffer_error;
-
- /* bitstream codebook header */
- value = gst_value_array_get_value (array, 2);
- buf = gst_value_get_buffer (value);
- if (!buf)
- goto null_buffer;
- result = vorbis_dec_handle_header_buffer (vd, buf);
- if (result != GST_FLOW_OK)
- goto buffer_error;
-
- return result;
+ GstVorbisDec *vd = GST_VORBIS_DEC (dec);
-no_caps:
- {
- GST_WARNING_OBJECT (vd, "no caps negotiated");
- return GST_FLOW_NOT_NEGOTIATED;
- }
-array_error:
- {
- GST_WARNING_OBJECT (vd, "streamheader array not found");
- return GST_FLOW_NOT_NEGOTIATED;
- }
-null_buffer:
- {
- GST_WARNING_OBJECT (vd, "streamheader with null buffer received");
- return GST_FLOW_NOT_NEGOTIATED;
- }
-buffer_error:
- {
- GST_WARNING_OBJECT (vd, "error handling buffer");
- return GST_FLOW_NOT_NEGOTIATED;
- }
-}
-
-static GstFlowReturn
-vorbis_dec_decode_buffer (GstVorbisDec * vd, GstBuffer * buffer)
-{
- ogg_packet *packet;
- ogg_packet_wrapper packet_wrapper;
- GstFlowReturn result = GST_FLOW_OK;
+ /* no draining etc */
+ if (G_UNLIKELY (!buffer))
+ return GST_FLOW_OK;
+ GST_LOG_OBJECT (vd, "got buffer %p", buffer);
/* make ogg_packet out of the buffer */
gst_ogg_packet_wrapper_map (&packet_wrapper, buffer);
packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
} else {
GstClockTime timestamp, duration;
- /* try to find header in caps so we can initialize the decoder */
- if (!vd->initialized) {
- result = vorbis_dec_handle_header_caps (vd, buffer);
- if (result != GST_FLOW_OK)
- goto invalid_caps_header;
- }
-
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
}
done:
+ GST_LOG_OBJECT (vd, "unmap buffer %p", buffer);
gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer);
return result;
{
GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty header received"));
result = GST_FLOW_ERROR;
- vd->discont = TRUE;
- goto done;
- }
-
-invalid_caps_header:
- {
- GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL),
- ("invalid streamheader in caps"));
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;
- }
- while (dec->queued) {
- GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
- GstClockTime timestamp, duration;
-
- timestamp = GST_BUFFER_TIMESTAMP (buf);
- duration = GST_BUFFER_DURATION (buf);
-
- vorbis_do_timestamps (dec, buf, TRUE, timestamp, duration);
- res = vorbis_dec_push_forward (dec, buf);
-
- dec->queued = g_list_delete_link (dec->queued, dec->queued);
- }
- 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);
- }
-
- if (G_LIKELY (buf)) {
- GST_DEBUG_OBJECT (vd,
- "gathering buffer %p of size %u, time %" GST_TIME_FORMAT
- ", dur %" GST_TIME_FORMAT, buf, gst_buffer_get_size (buf),
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (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)
+static void
+vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard)
{
- GstVorbisDec *vd;
- GstFlowReturn result = GST_FLOW_OK;
- gboolean discont;
+ GstVorbisDec *vd = GST_VORBIS_DEC (dec);
- 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->last_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 = GST_ELEMENT_CLASS (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;
-
-#ifndef USE_TREMOLO
- vorbis_block_clear (&vd->vb);
+ vorbis_synthesis_restart (&vd->vd);
#endif
- 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;
+ if (hard)
+ gst_vorbis_dec_reset (vd);
}
projects / platform / upstream / gstreamer.git / blobdiff