"rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
);
-GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT);
+GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstAudioDecoder,
+ GST_TYPE_AUDIO_DECODER);
+
+static gboolean gst_a52dec_start (GstAudioDecoder * dec);
+static gboolean gst_a52dec_stop (GstAudioDecoder * dec);
+static gboolean gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps);
+static gboolean gst_a52dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
+ gint * offset, gint * length);
+static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * dec,
+ GstBuffer * buffer);
+static GstFlowReturn gst_a52dec_pre_push (GstAudioDecoder * bdec,
+ GstBuffer ** buffer);
static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer);
-static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf);
-static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps);
-static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event);
-static GstStateChangeReturn gst_a52dec_change_state (GstElement * element,
- GstStateChange transition);
static void gst_a52dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
gst_a52dec_class_init (GstA52DecClass * klass)
{
GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
+ GstAudioDecoderClass *gstbase_class;
guint cpuflags;
gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
+ gstbase_class = (GstAudioDecoderClass *) klass;
gobject_class->set_property = gst_a52dec_set_property;
gobject_class->get_property = gst_a52dec_get_property;
- gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state);
+ gstbase_class->start = GST_DEBUG_FUNCPTR (gst_a52dec_start);
+ gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_a52dec_stop);
+ gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_a52dec_set_format);
+ gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_a52dec_parse);
+ gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_a52dec_handle_frame);
+ gstbase_class->pre_push = GST_DEBUG_FUNCPTR (gst_a52dec_pre_push);
/**
* GstA52Dec::drc
static void
gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class)
{
- /* create the sink and src pads */
- a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_pad_set_setcaps_function (a52dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps));
- gst_pad_set_chain_function (a52dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_a52dec_chain));
- gst_pad_set_event_function (a52dec->sinkpad,
- GST_DEBUG_FUNCPTR (gst_a52dec_sink_event));
- gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
-
- a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_pad_use_fixed_caps (a52dec->srcpad);
- gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
-
a52dec->request_channels = A52_CHANNEL;
a52dec->dynamic_range_compression = FALSE;
a52dec->state = NULL;
a52dec->samples = NULL;
- gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
+ /* retrieve and intercept base class chain.
+ * Quite HACKish, but that's dvd specs/caps for you,
+ * since one buffer needs to be split into 2 frames */
+ a52dec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (a52dec));
+ gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (a52dec),
+ GST_DEBUG_FUNCPTR (gst_a52dec_chain));
+}
+
+static gboolean
+gst_a52dec_start (GstAudioDecoder * dec)
+{
+ GstA52Dec *a52dec = GST_A52DEC (dec);
+ GstA52DecClass *klass;
+
+ GST_DEBUG_OBJECT (dec, "start");
+
+ klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
+ a52dec->state = a52_init (klass->a52_cpuflags);
+ a52dec->samples = a52_samples (a52dec->state);
+ a52dec->bit_rate = -1;
+ a52dec->sample_rate = -1;
+ a52dec->stream_channels = A52_CHANNEL;
+ a52dec->using_channels = A52_CHANNEL;
+ a52dec->level = 1;
+ a52dec->bias = 0;
+ a52dec->flag_update = TRUE;
+
+ /* call upon legacy upstream byte support (e.g. seeking) */
+ gst_audio_decoder_set_byte_time (dec, TRUE);
+
+ return TRUE;
+}
+
+static gboolean
+gst_a52dec_stop (GstAudioDecoder * dec)
+{
+ GstA52Dec *a52dec = GST_A52DEC (dec);
+
+ GST_DEBUG_OBJECT (dec, "stop");
+
+ a52dec->samples = NULL;
+ if (a52dec->state) {
+ a52_free (a52dec->state);
+ a52dec->state = NULL;
+ }
+ if (a52dec->pending_tags) {
+ gst_tag_list_free (a52dec->pending_tags);
+ a52dec->pending_tags = NULL;
+ }
+
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
+ gint * _offset, gint * len)
+{
+ GstA52Dec *a52dec;
+ guint8 *data;
+ gint av, size;
+ gint length = 0, flags, sample_rate, bit_rate;
+ GstFlowReturn result = GST_FLOW_UNEXPECTED;
+
+ a52dec = GST_A52DEC (bdec);
+
+ size = av = gst_adapter_available (adapter);
+ data = (guint8 *) gst_adapter_peek (adapter, av);
+
+ /* find and read header */
+ bit_rate = a52dec->bit_rate;
+ sample_rate = a52dec->sample_rate;
+ flags = 0;
+ while (av >= 7) {
+ length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
+
+ if (length == 0) {
+ /* shift window to re-find sync */
+ data++;
+ size--;
+ } else if (length <= size) {
+ GST_LOG_OBJECT (a52dec, "Sync: frame size %d", length);
+ result = GST_FLOW_OK;
+ break;
+ } else {
+ GST_LOG_OBJECT (a52dec, "Not enough data available (needed %d had %d)",
+ length, size);
+ break;
+ }
+ }
+
+ *_offset = av - size;
+ *len = length;
+
+ return result;
}
static gint
return chans;
}
-static void
-clear_queued (GstA52Dec * dec)
-{
- g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (dec->queued);
- dec->queued = NULL;
-}
-
-static GstFlowReturn
-flush_queued (GstA52Dec * dec)
-{
- GstFlowReturn ret = GST_FLOW_OK;
-
- while (dec->queued) {
- GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
-
- GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf,
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-
- /* iterate ouput queue an push downstream */
- ret = gst_pad_push (dec->srcpad, buf);
-
- dec->queued = g_list_delete_link (dec->queued, dec->queued);
- }
- return ret;
-}
-
-static GstFlowReturn
-gst_a52dec_drain (GstA52Dec * dec)
-{
- GstFlowReturn ret = GST_FLOW_OK;
-
- if (dec->segment.rate < 0.0) {
- /* if we have some queued frames for reverse playback, flush
- * them now */
- ret = flush_queued (dec);
- }
- return ret;
-}
-
-static GstFlowReturn
-gst_a52dec_push (GstA52Dec * a52dec,
- GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
-{
- GstBuffer *buf;
- int chans, n, c;
- GstFlowReturn result;
-
- flags &= (A52_CHANNEL_MASK | A52_LFE);
- chans = gst_a52dec_channels (flags, NULL);
- if (!chans) {
- GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
- ("invalid channel flags: %d", flags));
- return GST_FLOW_ERROR;
- }
-
- result =
- gst_pad_alloc_buffer_and_set_caps (srcpad, 0,
- 256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf);
- if (result != GST_FLOW_OK)
- return result;
-
- for (n = 0; n < 256; n++) {
- for (c = 0; c < chans; c++) {
- ((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
- samples[c * 256 + n];
- }
- }
-
- GST_BUFFER_TIMESTAMP (buf) = timestamp;
- GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
-
- result = GST_FLOW_OK;
- if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment,
- a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) {
- /* set discont when needed */
- if (a52dec->discont) {
- GST_LOG_OBJECT (a52dec, "marking DISCONT");
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
- a52dec->discont = FALSE;
- }
-
- if (a52dec->segment.rate > 0.0) {
- GST_DEBUG_OBJECT (a52dec,
- "Pushing buffer with ts %" GST_TIME_FORMAT " duration %"
- GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-
- result = gst_pad_push (srcpad, buf);
- } else {
- /* reverse playback, queue frame till later when we get a discont. */
- GST_DEBUG_OBJECT (a52dec, "queued frame");
- a52dec->queued = g_list_prepend (a52dec->queued, buf);
- }
- }
- return result;
-}
-
static gboolean
gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad)
{
return result;
}
-static gboolean
-gst_a52dec_sink_event (GstPad * pad, GstEvent * event)
+static void
+gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
{
- GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
- gboolean ret = FALSE;
-
- GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_NEWSEGMENT:
- {
- GstFormat fmt;
- gboolean update;
- gint64 start, end, pos;
- gdouble rate, arate;
-
- gst_event_parse_new_segment_full (event, &update, &rate, &arate, &fmt,
- &start, &end, &pos);
-
- /* drain queued buffers before activating the segment so that we can clip
- * against the old segment first */
- gst_a52dec_drain (a52dec);
-
- if (fmt != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) {
- GST_WARNING ("No time in newsegment event %p (format is %s)",
- event, gst_format_get_name (fmt));
- gst_event_unref (event);
- a52dec->sent_segment = FALSE;
- /* set some dummy values, FIXME: do proper conversion */
- a52dec->time = start = pos = 0;
- fmt = GST_FORMAT_TIME;
- end = -1;
- } else {
- a52dec->time = start;
- a52dec->sent_segment = TRUE;
- GST_DEBUG_OBJECT (a52dec,
- "Pushing newseg rate %g, applied rate %g, format %d, start %"
- G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT ", pos %"
- G_GINT64_FORMAT, rate, arate, fmt, start, end, pos);
-
- ret = gst_pad_push_event (a52dec->srcpad, event);
- }
+ GstTagList *taglist;
- gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start,
- end, pos);
- break;
- }
- case GST_EVENT_TAG:
- ret = gst_pad_push_event (a52dec->srcpad, event);
- break;
- case GST_EVENT_EOS:
- gst_a52dec_drain (a52dec);
- ret = gst_pad_push_event (a52dec->srcpad, event);
- break;
- case GST_EVENT_FLUSH_START:
- ret = gst_pad_push_event (a52dec->srcpad, event);
- break;
- case GST_EVENT_FLUSH_STOP:
- if (a52dec->cache) {
- gst_buffer_unref (a52dec->cache);
- a52dec->cache = NULL;
- }
- clear_queued (a52dec);
- gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
- ret = gst_pad_push_event (a52dec->srcpad, event);
- break;
- default:
- ret = gst_pad_push_event (a52dec->srcpad, event);
- break;
+ taglist = gst_tag_list_new ();
+ gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE,
+ (guint) a52dec->bit_rate, NULL);
+
+ if (a52dec->pending_tags) {
+ gst_tag_list_free (a52dec->pending_tags);
+ a52dec->pending_tags = NULL;
}
- gst_object_unref (a52dec);
- return ret;
+ a52dec->pending_tags = taglist;
}
-static void
-gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
+static GstFlowReturn
+gst_a52dec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer)
{
- GstTagList *taglist;
+ GstA52Dec *a52dec = GST_A52DEC (bdec);
- taglist = gst_tag_list_new ();
-
- gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
- GST_TAG_AUDIO_CODEC, "Dolby Digital (AC-3)",
- GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
+ if (G_UNLIKELY (a52dec->pending_tags)) {
+ gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
+ GST_AUDIO_DECODER_SRC_PAD (a52dec), a52dec->pending_tags);
+ a52dec->pending_tags = NULL;
+ }
- gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
- GST_PAD (a52dec->srcpad), taglist);
+ return GST_FLOW_OK;
}
static GstFlowReturn
-gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
- guint length, gint flags, gint sample_rate, gint bit_rate)
+gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
+ GstA52Dec *a52dec;
gint channels, i;
gboolean need_reneg = FALSE;
+ gint size, chans;
+ gint length = 0, flags, sample_rate, bit_rate;
+ guint8 *data;
+ GstFlowReturn result = GST_FLOW_OK;
+ GstBuffer *outbuf;
+ const gint num_blocks = 6;
+
+ a52dec = GST_A52DEC (bdec);
+
+ /* no fancy draining */
+ if (G_UNLIKELY (!buffer))
+ return GST_FLOW_OK;
+
+ /* parsed stuff already, so this should work out fine */
+ data = GST_BUFFER_DATA (buffer);
+ size = GST_BUFFER_SIZE (buffer);
+ g_assert (size >= 7);
+
+ /* re-obtain some sync header info,
+ * should be same as during _parse and could also be cached there,
+ * but anyway ... */
+ bit_rate = a52dec->bit_rate;
+ sample_rate = a52dec->sample_rate;
+ flags = 0;
+ length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
+ g_assert (length == size);
/* update stream information, renegotiate or re-streaminfo if needed */
need_reneg = FALSE;
a52dec->flag_update = FALSE;
- caps = gst_pad_get_allowed_caps (a52dec->srcpad);
+ caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec));
if (caps && gst_caps_get_size (caps) > 0) {
GstCaps *copy = gst_caps_copy_nth (caps, 0);
GstStructure *structure = gst_caps_get_structure (copy, 0);
} else {
flags = a52dec->using_channels;
}
+
/* process */
flags |= A52_ADJUST_LEVEL;
a52dec->level = 1;
if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
- GST_WARNING ("a52_frame error");
- a52dec->discont = TRUE;
- return GST_FLOW_OK;
+ GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
+ ("a52_frame error"), result);
+ goto exit;
}
+
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
if (a52dec->using_channels != channels) {
need_reneg = TRUE;
/* negotiate if required */
if (need_reneg) {
- GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
+ GST_DEBUG_OBJECT (a52dec,
+ "a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d",
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
- if (!gst_a52dec_reneg (a52dec, a52dec->srcpad)) {
- GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
- return GST_FLOW_ERROR;
- }
+ if (!gst_a52dec_reneg (a52dec, GST_AUDIO_DECODER_SRC_PAD (a52dec)))
+ goto failed_negotiation;
}
if (a52dec->dynamic_range_compression == FALSE) {
a52_dynrng (a52dec->state, NULL, NULL);
}
- /* each frame consists of 6 blocks */
- for (i = 0; i < 6; i++) {
+ flags &= (A52_CHANNEL_MASK | A52_LFE);
+ chans = gst_a52dec_channels (flags, NULL);
+ if (!chans)
+ goto invalid_flags;
+
+ /* handle decoded data;
+ * each frame has 6 blocks, one block is 256 samples, ea */
+ result =
+ gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (a52dec), 0,
+ 256 * chans * (SAMPLE_WIDTH / 8) * num_blocks,
+ GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (a52dec)), &outbuf);
+ if (result != GST_FLOW_OK)
+ goto exit;
+
+ data = GST_BUFFER_DATA (outbuf);
+ for (i = 0; i < num_blocks; i++) {
if (a52_block (a52dec->state)) {
- /* ignore errors but mark a discont */
- GST_WARNING ("a52_block error %d", i);
- a52dec->discont = TRUE;
+ /* also marks discont */
+ GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL),
+ ("error decoding block %d", i), result);
+ if (result != GST_FLOW_OK)
+ goto exit;
} else {
- GstFlowReturn ret;
+ gint n, c;
- /* push on */
- ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
- a52dec->samples, a52dec->time);
- if (ret != GST_FLOW_OK)
- return ret;
+ for (n = 0; n < 256; n++) {
+ for (c = 0; c < chans; c++) {
+ ((sample_t *) data)[n * chans + c] = a52dec->samples[c * 256 + n];
+ }
+ }
}
- a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
+ data += 256 * chans * (SAMPLE_WIDTH / 8);
}
- return GST_FLOW_OK;
+ result = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
+
+exit:
+ return result;
+
+ /* ERRORS */
+failed_negotiation:
+ {
+ GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
+ return GST_FLOW_ERROR;
+ }
+invalid_flags:
+ {
+ GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
+ ("Invalid channel flags: %d", flags));
+ return GST_FLOW_ERROR;
+ }
}
static gboolean
-gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_a52dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
- GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
+ GstA52Dec *a52dec = GST_A52DEC (bdec);
GstStructure *structure;
structure = gst_caps_get_structure (caps, 0);
else
a52dec->dvdmode = FALSE;
- gst_object_unref (a52dec);
-
return TRUE;
}
gst_a52dec_chain (GstPad * pad, GstBuffer * buf)
{
GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
- GstFlowReturn ret;
+ GstFlowReturn ret = GST_FLOW_OK;
gint first_access;
- if (GST_BUFFER_IS_DISCONT (buf)) {
- GST_LOG_OBJECT (a52dec, "received DISCONT");
- gst_a52dec_drain (a52dec);
- /* clear cache on discont and mark a discont in the element */
- if (a52dec->cache) {
- gst_buffer_unref (a52dec->cache);
- a52dec->cache = NULL;
- }
- a52dec->discont = TRUE;
- }
-
if (a52dec->dvdmode) {
gint size = GST_BUFFER_SIZE (buf);
guchar *data = GST_BUFFER_DATA (buf);
goto bad_first_access_parameter;
subbuf = gst_buffer_create_sub (buf, offset, len);
+ gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE;
- ret = gst_a52dec_chain_raw (pad, subbuf);
- if (ret != GST_FLOW_OK)
+ ret = a52dec->base_chain (pad, subbuf);
+ if (ret != GST_FLOW_OK) {
+ gst_buffer_unref (buf);
goto done;
+ }
offset += len;
len = size - offset;
if (len > 0) {
subbuf = gst_buffer_create_sub (buf, offset, len);
+ gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
- ret = gst_a52dec_chain_raw (pad, subbuf);
+ ret = a52dec->base_chain (pad, subbuf);
}
+ gst_buffer_unref (buf);
} else {
/* first_access = 0 or 1, so if there's a timestamp it applies to the first byte */
subbuf = gst_buffer_create_sub (buf, offset, size - offset);
- GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf);
- ret = gst_a52dec_chain_raw (pad, subbuf);
+ gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL);
+ ret = a52dec->base_chain (pad, subbuf);
}
} else {
- gst_buffer_ref (buf);
- ret = gst_a52dec_chain_raw (pad, buf);
+ ret = a52dec->base_chain (pad, buf);
}
done:
- gst_buffer_unref (buf);
return ret;
/* ERRORS */
}
}
-static GstFlowReturn
-gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf)
-{
- GstA52Dec *a52dec;
- guint8 *data;
- guint size;
- gint length = 0, flags, sample_rate, bit_rate;
- GstFlowReturn result = GST_FLOW_OK;
-
- a52dec = GST_A52DEC (GST_PAD_PARENT (pad));
-
- if (!a52dec->sent_segment) {
- GstSegment segment;
-
- /* Create a basic segment. Usually, we'll get a new-segment sent by
- * another element that will know more information (a demuxer). If we're
- * just looking at a raw AC3 stream, we won't - so we need to send one
- * here, but we don't know much info, so just send a minimal TIME
- * new-segment event
- */
- gst_segment_init (&segment, GST_FORMAT_TIME);
- gst_pad_push_event (a52dec->srcpad, gst_event_new_new_segment (FALSE,
- segment.rate, segment.format, segment.start,
- segment.duration, segment.start));
- a52dec->sent_segment = TRUE;
- }
-
- /* merge with cache, if any. Also make sure timestamps match */
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- a52dec->time = GST_BUFFER_TIMESTAMP (buf);
- GST_DEBUG_OBJECT (a52dec,
- "Received buffer with ts %" GST_TIME_FORMAT " duration %"
- GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
- }
-
- if (a52dec->cache) {
- buf = gst_buffer_join (a52dec->cache, buf);
- a52dec->cache = NULL;
- }
- data = GST_BUFFER_DATA (buf);
- size = GST_BUFFER_SIZE (buf);
-
- /* find and read header */
- bit_rate = a52dec->bit_rate;
- sample_rate = a52dec->sample_rate;
- flags = 0;
- while (size >= 7) {
- length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
-
- if (length == 0) {
- /* no sync */
- data++;
- size--;
- } else if (length <= size) {
- GST_DEBUG ("Sync: %d", length);
-
- if (flags != a52dec->prev_flags)
- a52dec->flag_update = TRUE;
- a52dec->prev_flags = flags;
-
- result = gst_a52dec_handle_frame (a52dec, data,
- length, flags, sample_rate, bit_rate);
- if (result != GST_FLOW_OK) {
- size = 0;
- break;
- }
- size -= length;
- data += length;
- } else {
- /* not enough data */
- GST_LOG ("Not enough data available");
- break;
- }
- }
-
- /* keep cache */
- if (length == 0) {
- GST_LOG ("No sync found");
- }
-
- if (size > 0) {
- a52dec->cache = gst_buffer_create_sub (buf,
- GST_BUFFER_SIZE (buf) - size, size);
- }
-
- gst_buffer_unref (buf);
-
- return result;
-}
-
-static GstStateChangeReturn
-gst_a52dec_change_state (GstElement * element, GstStateChange transition)
-{
- GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
- GstA52Dec *a52dec = GST_A52DEC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:{
- GstA52DecClass *klass;
-
- klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec));
- a52dec->state = a52_init (klass->a52_cpuflags);
-
- if (!a52dec->state) {
- GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL),
- ("Failed to initialize a52 state"));
- ret = GST_STATE_CHANGE_FAILURE;
- }
- break;
- }
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- a52dec->samples = a52_samples (a52dec->state);
- a52dec->bit_rate = -1;
- a52dec->sample_rate = -1;
- a52dec->stream_channels = A52_CHANNEL;
- a52dec->using_channels = A52_CHANNEL;
- a52dec->level = 1;
- a52dec->bias = 0;
- a52dec->time = 0;
- a52dec->sent_segment = FALSE;
- a52dec->flag_update = TRUE;
- gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED);
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- break;
- default:
- break;
- }
-
- ret = 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:
- a52dec->samples = NULL;
- if (a52dec->cache) {
- gst_buffer_unref (a52dec->cache);
- a52dec->cache = NULL;
- }
- clear_queued (a52dec);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- if (a52dec->state) {
- a52_free (a52dec->state);
- a52dec->state = NULL;
- }
- break;
- default:
- break;
- }
-
- return ret;
-}
-
static void
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
projects / platform / upstream / gstreamer.git / commitdiff