--- /dev/null
+/* GStreamer DTS decoder plugin based on libdtsdec
+ * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include "_stdint.h"
+#include <stdlib.h>
+
+#include <gst/gst.h>
+#include <dts.h>
+
+#include "gstdtsdec.h"
+
+GST_DEBUG_CATEGORY_STATIC (dtsdec_debug);
+#define GST_CAT_DEFAULT (dtsdec_debug)
+
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0,
+ ARG_DRC
+ /* FILL ME */
+};
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-dts")
+ );
+
+#if defined(LIBDTS_FIXED)
+#define DTS_CAPS "audio/x-raw-int, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "signed = (boolean) true, " \
+ "width = (int) 16, " \
+ "depth = (int) 16"
+#define SAMPLE_WIDTH 16
+#elif defined(LIBDTS_DOUBLE)
+#define DTS_CAPS "audio/x-raw-float, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 64"
+#define SAMPLE_WIDTH 64
+#else
+#define DTS_CAPS "audio/x-raw-float, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 32"
+#define SAMPLE_WIDTH 32
+#endif
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS (DTS_CAPS ", "
+ "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]")
+ );
+
+static void gst_dtsdec_base_init (GstDtsDecClass * klass);
+static void gst_dtsdec_class_init (GstDtsDecClass * klass);
+static void gst_dtsdec_init (GstDtsDec * dtsdec);
+
+static void gst_dtsdec_loop (GstElement * element);
+static GstElementStateReturn gst_dtsdec_change_state (GstElement * element);
+
+static void gst_dtsdec_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_dtsdec_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static GstElementClass *parent_class = NULL;
+
+/* static guint gst_dtsdec_signals[LAST_SIGNAL] = { 0 }; */
+
+GType
+gst_dtsdec_get_type (void)
+{
+ static GType dtsdec_type = 0;
+
+ if (!dtsdec_type) {
+ static const GTypeInfo dtsdec_info = {
+ sizeof (GstDtsDecClass),
+ (GBaseInitFunc) gst_dtsdec_base_init,
+ NULL, (GClassInitFunc) gst_dtsdec_class_init,
+ NULL,
+ NULL,
+ sizeof (GstDtsDec),
+ 0,
+ (GInstanceInitFunc) gst_dtsdec_init,
+ };
+
+ dtsdec_type =
+ g_type_register_static (GST_TYPE_ELEMENT, "GstDtsDec", &dtsdec_info, 0);
+
+ GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder");
+ }
+ return dtsdec_type;
+}
+
+static void
+gst_dtsdec_base_init (GstDtsDecClass * klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ static GstElementDetails gst_dtsdec_details = {
+ "DTS audio decoder",
+ "Codec/Audio/Decoder",
+ "Decodes DTS audio streams",
+ "Ronald Bultje <rbultje@ronald.bitfreak.net>"
+ };
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_factory));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&src_factory));
+ gst_element_class_set_details (element_class, &gst_dtsdec_details);
+}
+
+static void
+gst_dtsdec_class_init (GstDtsDecClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
+ g_param_spec_boolean ("drc", "Dynamic Range Compression",
+ "Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
+
+ gobject_class->set_property = gst_dtsdec_set_property;
+ gobject_class->get_property = gst_dtsdec_get_property;
+
+ gstelement_class->change_state = gst_dtsdec_change_state;
+}
+
+static void
+gst_dtsdec_init (GstDtsDec * dtsdec)
+{
+ GstElement *element = GST_ELEMENT (dtsdec);
+
+ /* create the sink and src pads */
+ dtsdec->sinkpad =
+ gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
+ (dtsdec), "sink"), "sink");
+ gst_element_add_pad (element, dtsdec->sinkpad);
+ gst_element_set_loop_function (element, gst_dtsdec_loop);
+
+ dtsdec->srcpad =
+ gst_pad_new_from_template (gst_element_get_pad_template (element,
+ "src"), "src");
+ gst_pad_use_explicit_caps (dtsdec->srcpad);
+ gst_element_add_pad (element, dtsdec->srcpad);
+
+ GST_FLAG_SET (element, GST_ELEMENT_EVENT_AWARE);
+ dtsdec->dynamic_range_compression = FALSE;
+}
+
+static gint
+gst_dtsdec_channels (uint32_t flags)
+{
+ gint chans = 0;
+
+ switch (flags & DTS_CHANNEL_MASK) {
+ case DTS_MONO:
+ chans = 1;
+ break;
+ case DTS_CHANNEL:
+ case DTS_STEREO:
+ case DTS_STEREO_SUMDIFF:
+ case DTS_STEREO_TOTAL:
+ case DTS_DOLBY:
+ chans = 2;
+ break;
+ case DTS_3F:
+ case DTS_2F1R:
+ chans = 3;
+ break;
+ case DTS_3F1R:
+ case DTS_2F2R:
+ chans = 4;
+ break;
+ case DTS_3F2R:
+ chans = 5;
+ break;
+ case DTS_4F2R:
+ chans = 6;
+ break;
+ default:
+ /* error */
+ g_warning ("dtsdec: invalid flags 0x%x", flags);
+ return 0;
+ }
+ if (flags & DTS_LFE)
+ chans += 1;
+
+ return chans;
+}
+
+static gboolean
+gst_dtsdec_renegotiate (GstDtsDec * dts)
+{
+ GstCaps *caps = gst_caps_from_string (DTS_CAPS);
+ gint channels = gst_dtsdec_channels (dts->using_channels);
+
+ GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d",
+ channels, dts->sample_rate);
+
+ gst_caps_set_simple (caps,
+ "channels", G_TYPE_INT, channels,
+ "rate", G_TYPE_INT, (gint) dts->sample_rate, NULL);
+
+ return gst_pad_set_explicit_caps (dts->srcpad, caps);
+}
+
+static void
+gst_dtsdec_handle_event (GstDtsDec * dts)
+{
+ guint32 remaining;
+ GstEvent *event;
+
+ gst_bytestream_get_status (dts->bs, &remaining, &event);
+
+ if (!event) {
+ GST_ELEMENT_ERROR (dts, RESOURCE, READ, (NULL), (NULL));
+ return;
+ }
+
+ GST_LOG ("Handling event of type %d timestamp %llu", GST_EVENT_TYPE (event),
+ GST_EVENT_TIMESTAMP (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_DISCONTINUOUS:
+ case GST_EVENT_FLUSH:
+ gst_bytestream_flush_fast (dts->bs, remaining);
+ break;
+ default:
+ break;
+ }
+
+ gst_pad_event_default (dts->sinkpad, event);
+}
+
+static void
+gst_dtsdec_update_streaminfo (GstDtsDec * dts)
+{
+ GstTagList *taglist;
+
+ taglist = gst_tag_list_new ();
+
+ gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
+ GST_TAG_BITRATE, (guint) dts->bit_rate, NULL);
+
+ gst_element_found_tags_for_pad (GST_ELEMENT (dts),
+ dts->srcpad, dts->current_ts, taglist);
+}
+
+static void
+gst_dtsdec_loop (GstElement * element)
+{
+ GstDtsDec *dts = GST_DTSDEC (element);
+ guint8 *data;
+ GstBuffer *buf, *out;
+ sample_t *samples;
+ gint i, length, flags, sample_rate, bit_rate, frame_length, s, c, num_c;
+ gint channels, skipped = 0, num_blocks;
+ guint32 got_bytes;
+ gboolean need_renegotiation = FALSE;
+ GstClockTime timestamp = 0;
+
+ /* find sync. Don't know what 3840 is based on... */
+#define MAX_SKIP 3840
+ while (skipped < MAX_SKIP) {
+ got_bytes = gst_bytestream_peek_bytes (dts->bs, &data, 7);
+ if (got_bytes < 7) {
+ gst_dtsdec_handle_event (dts);
+ return;
+ }
+ length = dts_syncinfo (dts->state, data, &flags,
+ &sample_rate, &bit_rate, &frame_length);
+ if (length == 0) {
+ /* shift window to re-find sync */
+ gst_bytestream_flush_fast (dts->bs, 1);
+ skipped++;
+ GST_LOG ("Skipped");
+ } else
+ break;
+ }
+
+ if (skipped >= MAX_SKIP) {
+ GST_ELEMENT_ERROR (dts, RESOURCE, SYNC, (NULL), (NULL));
+ return;
+ }
+
+ /* go over stream properties, update caps/streaminfo if needed */
+ if (dts->sample_rate != sample_rate) {
+ need_renegotiation = TRUE;
+ dts->sample_rate = sample_rate;
+ }
+
+ dts->stream_channels = flags;
+
+ if (bit_rate != dts->bit_rate) {
+ dts->bit_rate = bit_rate;
+ gst_dtsdec_update_streaminfo (dts);
+ }
+
+ /* read the header + rest of frame */
+ got_bytes = gst_bytestream_read (dts->bs, &buf, length);
+ if (got_bytes < length) {
+ gst_dtsdec_handle_event (dts);
+ return;
+ }
+
+ data = GST_BUFFER_DATA (buf);
+ timestamp = gst_bytestream_get_timestamp (dts->bs);
+ if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+ if (timestamp == dts->last_ts) {
+ timestamp = dts->current_ts;
+ } else {
+ dts->last_ts = timestamp;
+ }
+ }
+
+ /* process */
+ flags = dts->request_channels | DTS_ADJUST_LEVEL;
+ dts->level = 1;
+
+ if (dts_frame (dts->state, data, &flags, &dts->level, dts->bias)) {
+ GST_WARNING ("dts_frame error");
+ goto end;
+ }
+
+ channels = flags & (DTS_CHANNEL_MASK | DTS_LFE);
+
+ if (dts->using_channels != channels) {
+ need_renegotiation = TRUE;
+ dts->using_channels = channels;
+ }
+
+ if (need_renegotiation == TRUE) {
+ GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x",
+ dts->sample_rate, dts->stream_channels, dts->using_channels);
+ if (!gst_dtsdec_renegotiate (dts))
+ goto end;
+ }
+
+ if (dts->dynamic_range_compression == FALSE) {
+ dts_dynrng (dts->state, NULL, NULL);
+ }
+
+ /* handle decoded data, one block is 256 samples */
+ num_blocks = dts_blocks_num (dts->state);
+ for (i = 0; i < num_blocks; i++) {
+ if (dts_block (dts->state)) {
+ GST_WARNING ("dts_block error %d", i);
+ continue;
+ }
+
+ samples = dts_samples (dts->state);
+ num_c = gst_dtsdec_channels (dts->using_channels);
+ out = gst_buffer_new_and_alloc ((SAMPLE_WIDTH / 8) * 256 * num_c);
+ GST_BUFFER_TIMESTAMP (out) = timestamp;
+ GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate;
+
+ /* libdts returns buffers in 256-sample-blocks per channel,
+ * we want interleaved. And we need to copy anyway... */
+ data = GST_BUFFER_DATA (out);
+ for (s = 0; s < 256; s++) {
+ for (c = 0; c < num_c; c++) {
+ *(sample_t *) data = samples[s + c * 256];
+ data += (SAMPLE_WIDTH / 8);
+ }
+ }
+
+ /* push on */
+ gst_pad_push (dts->srcpad, GST_DATA (out));
+
+ timestamp += GST_SECOND * 256 / dts->sample_rate;
+ }
+
+ dts->current_ts = timestamp;
+
+end:
+ gst_buffer_unref (buf);
+}
+
+static GstElementStateReturn
+gst_dtsdec_change_state (GstElement * element)
+{
+ GstDtsDec *dts = GST_DTSDEC (element);
+
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:{
+ GstCPUFlags cpuflags;
+ uint32_t mm_accel = 0;
+
+ dts->bs = gst_bytestream_new (dts->sinkpad);
+ cpuflags = gst_cpu_get_flags ();
+ if (cpuflags & GST_CPU_FLAG_MMX)
+ mm_accel |= MM_ACCEL_X86_MMX;
+ if (cpuflags & GST_CPU_FLAG_3DNOW)
+ mm_accel |= MM_ACCEL_X86_3DNOW;
+ if (cpuflags & GST_CPU_FLAG_MMXEXT)
+ mm_accel |= MM_ACCEL_X86_MMXEXT;
+
+ dts->state = dts_init (mm_accel);
+ break;
+ }
+ case GST_STATE_READY_TO_PAUSED:
+ dts->samples = dts_samples (dts->state);
+ dts->bit_rate = -1;
+ dts->sample_rate = -1;
+ dts->stream_channels = 0;
+ /* FIXME force stereo for now */
+ dts->request_channels = DTS_STEREO;
+ dts->using_channels = 0;
+ dts->level = 1;
+ dts->bias = 0;
+ dts->last_ts = 0;
+ dts->current_ts = 0;
+ break;
+ case GST_STATE_PAUSED_TO_READY:
+ dts->samples = NULL;
+ break;
+ case GST_STATE_READY_TO_NULL:
+ gst_bytestream_destroy (dts->bs);
+ dts->bs = NULL;
+ dts_free (dts->state);
+ dts->state = NULL;
+ break;
+ default:
+ break;
+ }
+
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+static void
+gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value,
+ GParamSpec * pspec)
+{
+ GstDtsDec *dts = GST_DTSDEC (object);
+
+ switch (prop_id) {
+ case ARG_DRC:
+ dts->dynamic_range_compression = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
+{
+ GstDtsDec *dts = GST_DTSDEC (object);
+
+ switch (prop_id) {
+ case ARG_DRC:
+ g_value_set_boolean (value, dts->dynamic_range_compression);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ if (!gst_library_load ("gstbytestream"))
+ return FALSE;
+
+ if (!gst_element_register (plugin, "dtsdec", GST_RANK_PRIMARY,
+ GST_TYPE_DTSDEC))
+ return FALSE;
+
+ return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "dtsdec",
+ "Decodes DTS audio streams",
+ plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN);
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