/* GStreamer DTS decoder plugin based on libdtsdec * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net> * Copyright (C) 2009 Jan Schmidt <thaytan@noraisin.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. */ /** * SECTION:element-dtsdec * * Digital Theatre System (DTS) audio decoder * * <refsect2> * <title>Example launch line</title> * |[ * gst-launch dvdreadsrc title=1 ! mpegpsdemux ! dtsdec ! audioresample ! audioconvert ! alsasink * ]| Play a DTS audio track from a dvd. * |[ * gst-launch filesrc location=abc.dts ! dtsdec ! audioresample ! audioconvert ! alsasink * ]| Decode a standalone file and play it. * </refsect2> */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include "_stdint.h" #include <stdlib.h> #include <gst/gst.h> #include <gst/audio/audio.h> #ifndef DTS_OLD #include <dca.h> #else #include <dts.h> typedef struct dts_state_s dca_state_t; #define DCA_MONO DTS_MONO #define DCA_CHANNEL DTS_CHANNEL #define DCA_STEREO DTS_STEREO #define DCA_STEREO_SUMDIFF DTS_STEREO_SUMDIFF #define DCA_STEREO_TOTAL DTS_STEREO_TOTAL #define DCA_3F DTS_3F #define DCA_2F1R DTS_2F1R #define DCA_3F1R DTS_3F1R #define DCA_2F2R DTS_2F2R #define DCA_3F2R DTS_3F2R #define DCA_4F2R DTS_4F2R #define DCA_DOLBY DTS_DOLBY #define DCA_CHANNEL_MAX DTS_CHANNEL_MAX #define DCA_CHANNEL_BITS DTS_CHANNEL_BITS #define DCA_CHANNEL_MASK DTS_CHANNEL_MASK #define DCA_LFE DTS_LFE #define DCA_ADJUST_LEVEL DTS_ADJUST_LEVEL #define dca_init dts_init #define dca_syncinfo dts_syncinfo #define dca_frame dts_frame #define dca_dynrng dts_dynrng #define dca_blocks_num dts_blocks_num #define dca_block dts_block #define dca_samples dts_samples #define dca_free dts_free #endif #include "gstdtsdec.h" #if HAVE_ORC #include <orc/orc.h> #endif #if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED) #define SAMPLE_WIDTH 16 #define SAMPLE_FORMAT GST_AUDIO_NE(S16) #define SAMPLE_TYPE GST_AUDIO_FORMAT_S16 #elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE) #define SAMPLE_WIDTH 64 #define SAMPLE_FORMAT GST_AUDIO_NE(F64) #define SAMPLE_TYPE GST_AUDIO_FORMAT_F64 #else #define SAMPLE_WIDTH 32 #define SAMPLE_FORMAT GST_AUDIO_NE(F32) #define SAMPLE_TYPE GST_AUDIO_FORMAT_F32 #endif GST_DEBUG_CATEGORY_STATIC (dtsdec_debug); #define GST_CAT_DEFAULT (dtsdec_debug) enum { PROP_0, PROP_DRC }; static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-dts; audio/x-private1-dts") ); static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " SAMPLE_FORMAT ", " "layout = (string) interleaved, " "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]") ); G_DEFINE_TYPE (GstDtsDec, gst_dtsdec, GST_TYPE_AUDIO_DECODER); static gboolean gst_dtsdec_start (GstAudioDecoder * dec); static gboolean gst_dtsdec_stop (GstAudioDecoder * dec); static gboolean gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps); static gboolean gst_dtsdec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length); static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer); static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf); 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 void gst_dtsdec_class_init (GstDtsDecClass * 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_dtsdec_set_property; gobject_class->get_property = gst_dtsdec_get_property; gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&sink_factory)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&src_factory)); gst_element_class_set_details_simple (gstelement_class, "DTS audio decoder", "Codec/Decoder/Audio", "Decodes DTS audio streams", "Jan Schmidt <thaytan@noraisin.net>, " "Ronald Bultje <rbultje@ronald.bitfreak.net>"); gstbase_class->start = GST_DEBUG_FUNCPTR (gst_dtsdec_start); gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_dtsdec_stop); gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_dtsdec_set_format); gstbase_class->parse = GST_DEBUG_FUNCPTR (gst_dtsdec_parse); gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_dtsdec_handle_frame); /** * GstDtsDec::drc * * Set to true to apply the recommended DTS dynamic range compression * to the audio stream. Dynamic range compression makes loud sounds * softer and soft sounds louder, so you can more easily listen * to the stream without disturbing other people. */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DRC, g_param_spec_boolean ("drc", "Dynamic Range Compression", "Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); klass->dts_cpuflags = 0; #if HAVE_ORC cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx")); if (cpuflags & ORC_TARGET_MMX_MMX) klass->dts_cpuflags |= MM_ACCEL_X86_MMX; if (cpuflags & ORC_TARGET_MMX_3DNOW) klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW; if (cpuflags & ORC_TARGET_MMX_MMXEXT) klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT; #else cpuflags = 0; klass->dts_cpuflags = 0; #endif GST_LOG ("CPU flags: dts=%08x, liboil=%08x", klass->dts_cpuflags, cpuflags); } static void gst_dtsdec_init (GstDtsDec * dtsdec) { dtsdec->request_channels = DCA_CHANNEL; dtsdec->dynamic_range_compression = FALSE; /* retrieve and intercept base class chain. * Quite HACKish, but that's dvd specs for you, * since one buffer needs to be split into 2 frames */ dtsdec->base_chain = GST_PAD_CHAINFUNC (GST_AUDIO_DECODER_SINK_PAD (dtsdec)); gst_pad_set_chain_function (GST_AUDIO_DECODER_SINK_PAD (dtsdec), GST_DEBUG_FUNCPTR (gst_dtsdec_chain)); } static gboolean gst_dtsdec_start (GstAudioDecoder * dec) { GstDtsDec *dts = GST_DTSDEC (dec); GstDtsDecClass *klass; GST_DEBUG_OBJECT (dec, "start"); klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts)); dts->state = dca_init (klass->dts_cpuflags); dts->samples = dca_samples (dts->state); dts->bit_rate = -1; dts->sample_rate = -1; dts->stream_channels = DCA_CHANNEL; dts->using_channels = DCA_CHANNEL; dts->level = 1; dts->bias = 0; dts->flag_update = TRUE; /* call upon legacy upstream byte support (e.g. seeking) */ gst_audio_decoder_set_estimate_rate (dec, TRUE); return TRUE; } static gboolean gst_dtsdec_stop (GstAudioDecoder * dec) { GstDtsDec *dts = GST_DTSDEC (dec); GST_DEBUG_OBJECT (dec, "stop"); dts->samples = NULL; if (dts->state) { dca_free (dts->state); dts->state = NULL; } return TRUE; } static GstFlowReturn gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter, gint * _offset, gint * len) { GstDtsDec *dts; guint8 *data; gint av, size; gint length = 0, flags, sample_rate, bit_rate, frame_length; GstFlowReturn result = GST_FLOW_EOS; dts = GST_DTSDEC (bdec); size = av = gst_adapter_available (adapter); data = (guint8 *) gst_adapter_map (adapter, av); /* find and read header */ bit_rate = dts->bit_rate; sample_rate = dts->sample_rate; flags = 0; while (size >= 7) { length = dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate, &frame_length); if (length == 0) { /* shift window to re-find sync */ data++; size--; } else if (length <= size) { GST_LOG_OBJECT (dts, "Sync: frame size %d", length); result = GST_FLOW_OK; break; } else { GST_LOG_OBJECT (dts, "Not enough data available (needed %d had %d)", length, size); break; } } gst_adapter_unmap (adapter); *_offset = av - size; *len = length; return result; } static gint gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition * pos) { gint chans = 0; switch (flags & DCA_CHANNEL_MASK) { case DCA_MONO: chans = 1; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; } break; /* case DCA_CHANNEL: */ case DCA_STEREO: case DCA_STEREO_SUMDIFF: case DCA_STEREO_TOTAL: case DCA_DOLBY: chans = 2; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; } break; case DCA_3F: chans = 3; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; } break; case DCA_2F1R: chans = 3; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; } break; case DCA_3F1R: chans = 4; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; } break; case DCA_2F2R: chans = 4; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } break; case DCA_3F2R: chans = 5; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } break; case DCA_4F2R: chans = 6; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER; pos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER; pos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; pos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } break; default: g_warning ("dtsdec: invalid flags 0x%x", flags); return 0; } if (flags & DCA_LFE) { if (pos) { pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE1; } chans += 1; } return chans; } static gboolean gst_dtsdec_renegotiate (GstDtsDec * dts) { gint channels; gboolean result = FALSE; GstAudioChannelPosition from[6], to[6]; GstAudioInfo info; channels = gst_dtsdec_channels (dts->using_channels, from); if (!channels) goto done; GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d", channels, dts->sample_rate); memcpy (to, from, sizeof (GstAudioChannelPosition) * channels); gst_audio_channel_positions_to_valid_order (to, channels); gst_audio_get_channel_reorder_map (channels, from, to, dts->channel_reorder_map); gst_audio_info_init (&info); gst_audio_info_set_format (&info, SAMPLE_TYPE, dts->sample_rate, channels, (channels > 1 ? to : NULL)); if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dts), &info)) goto done; result = TRUE; done: return result; } static void gst_dtsdec_update_streaminfo (GstDtsDec * dts) { GstTagList *taglist; if (dts->bit_rate > 3) { taglist = gst_tag_list_new_empty (); /* 1 => open bitrate, 2 => variable bitrate, 3 => lossless */ gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE, (guint) dts->bit_rate, NULL); gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dts), taglist, GST_TAG_MERGE_REPLACE); } } static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) { GstDtsDec *dts; gint channels, i, num_blocks; gboolean need_renegotiation = FALSE; guint8 *data; gsize size; GstMapInfo map; gint chans; gint length = 0, flags, sample_rate, bit_rate, frame_length; GstFlowReturn result = GST_FLOW_OK; GstBuffer *outbuf; dts = GST_DTSDEC (bdec); /* no fancy draining */ if (G_UNLIKELY (!buffer)) return GST_FLOW_OK; /* parsed stuff already, so this should work out fine */ gst_buffer_map (buffer, &map, GST_MAP_READ); data = map.data; size = map.size; g_assert (size >= 7); bit_rate = dts->bit_rate; sample_rate = dts->sample_rate; flags = 0; length = dca_syncinfo (dts->state, data, &flags, &sample_rate, &bit_rate, &frame_length); g_assert (length == size); if (flags != dts->prev_flags) { dts->prev_flags = flags; dts->flag_update = TRUE; } /* go over stream properties, renegotiate or update streaminfo if needed */ if (dts->sample_rate != sample_rate) { need_renegotiation = TRUE; dts->sample_rate = sample_rate; } if (flags) { dts->stream_channels = flags & (DCA_CHANNEL_MASK | DCA_LFE); } if (bit_rate != dts->bit_rate) { dts->bit_rate = bit_rate; gst_dtsdec_update_streaminfo (dts); } /* If we haven't had an explicit number of channels chosen through properties * at this point, choose what to downmix to now, based on what the peer will * accept - this allows a52dec to do downmixing in preference to a * downstream element such as audioconvert. * FIXME: Add the property back in for forcing output channels. */ if (dts->request_channels != DCA_CHANNEL) { flags = dts->request_channels; } else if (dts->flag_update) { GstCaps *caps; dts->flag_update = FALSE; caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dts)); if (caps && gst_caps_get_size (caps) > 0) { GstCaps *copy = gst_caps_copy_nth (caps, 0); GstStructure *structure = gst_caps_get_structure (copy, 0); gint channels; const int dts_channels[6] = { DCA_MONO, DCA_STEREO, DCA_STEREO | DCA_LFE, DCA_2F2R, DCA_2F2R | DCA_LFE, DCA_3F2R | DCA_LFE, }; /* Prefer the original number of channels, but fixate to something * preferred (first in the caps) downstream if possible. */ gst_structure_fixate_field_nearest_int (structure, "channels", flags ? gst_dtsdec_channels (flags, NULL) : 6); gst_structure_get_int (structure, "channels", &channels); if (channels <= 6) flags = dts_channels[channels - 1]; else flags = dts_channels[5]; gst_caps_unref (copy); } else if (flags) { flags = dts->stream_channels; } else { flags = DCA_3F2R | DCA_LFE; } if (caps) gst_caps_unref (caps); } else { flags = dts->using_channels; } /* process */ flags |= DCA_ADJUST_LEVEL; dts->level = 1; if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) { gst_buffer_unmap (buffer, &map); GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL), ("dts_frame error"), result); goto exit; } gst_buffer_unmap (buffer, &map); channels = flags & (DCA_CHANNEL_MASK | DCA_LFE); if (dts->using_channels != channels) { need_renegotiation = TRUE; dts->using_channels = channels; } /* negotiate if required */ if (need_renegotiation) { GST_DEBUG_OBJECT (dts, "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 failed_negotiation; } if (dts->dynamic_range_compression == FALSE) { dca_dynrng (dts->state, NULL, NULL); } flags &= (DCA_CHANNEL_MASK | DCA_LFE); chans = gst_dtsdec_channels (flags, NULL); if (!chans) goto invalid_flags; /* handle decoded data, one block is 256 samples */ num_blocks = dca_blocks_num (dts->state); outbuf = gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); data = map.data; size = map.size; { guint8 *ptr = data; for (i = 0; i < num_blocks; i++) { if (dca_block (dts->state)) { /* also marks discont */ GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL), ("error decoding block %d", i), result); if (result != GST_FLOW_OK) goto exit; } else { gint n, c; gint *reorder_map = dts->channel_reorder_map; for (n = 0; n < 256; n++) { for (c = 0; c < chans; c++) { ((sample_t *) ptr)[n * chans + reorder_map[c]] = dts->samples[c * 256 + n]; } } } ptr += 256 * chans * (SAMPLE_WIDTH / 8); } } gst_buffer_unmap (outbuf, &map); result = gst_audio_decoder_finish_frame (bdec, outbuf, 1); exit: return result; /* ERRORS */ failed_negotiation: { GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL)); return GST_FLOW_ERROR; } invalid_flags: { GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL), ("Invalid channel flags: %d", flags)); return GST_FLOW_ERROR; } } static gboolean gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps) { GstDtsDec *dts = GST_DTSDEC (bdec); GstStructure *structure; structure = gst_caps_get_structure (caps, 0); if (structure && gst_structure_has_name (structure, "audio/x-private1-dts")) dts->dvdmode = TRUE; else dts->dvdmode = FALSE; return TRUE; } static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; GstDtsDec *dts = GST_DTSDEC (parent); gint first_access; if (dts->dvdmode) { guint8 data[2]; gsize size; gint offset, len; GstBuffer *subbuf; size = gst_buffer_extract (buf, 0, data, 2); if (size < 2) goto not_enough_data; first_access = (data[0] << 8) | data[1]; /* Skip the first_access header */ offset = 2; if (first_access > 1) { /* Length of data before first_access */ len = first_access - 1; if (len <= 0 || offset + len > size) goto bad_first_access_parameter; subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len); GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE; ret = dts->base_chain (pad, parent, subbuf); if (ret != GST_FLOW_OK) { gst_buffer_unref (buf); goto done; } offset += len; len = size - offset; if (len > 0) { subbuf = gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, offset, len); GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf); ret = dts->base_chain (pad, parent, 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_copy_region (buf, GST_BUFFER_COPY_ALL, offset, size - offset); GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf); ret = dts->base_chain (pad, parent, subbuf); gst_buffer_unref (buf); } } else { ret = dts->base_chain (pad, parent, buf); } done: return ret; /* ERRORS */ not_enough_data: { GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL), ("Insufficient data in buffer. Can't determine first_acess")); gst_buffer_unref (buf); return GST_FLOW_ERROR; } bad_first_access_parameter: { GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL), ("Bad first_access parameter (%d) in buffer", first_access)); gst_buffer_unref (buf); return GST_FLOW_ERROR; } } 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 PROP_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 PROP_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) { GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder"); #if HAVE_ORC orc_init (); #endif 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_NAME, GST_PACKAGE_ORIGIN);