/* GStreamer DTS decoder plugin based on libdtsdec * Copyright (C) 2004 Ronald Bultje * * 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. */ /* TODO: - Port to libdca API instead of relying on the compat header. * libdca is the successor of libdts: * http://www.videolan.org/developers/libdca.html */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "_stdint.h" #include #include #include #include #include "gstdtsdec.h" #include #include #include GST_DEBUG_CATEGORY_STATIC (dtsdec_debug); #define GST_CAT_DEFAULT (dtsdec_debug) static const GstElementDetails gst_dtsdec_details = GST_ELEMENT_DETAILS ("DTS audio decoder", "Codec/Decoder/Audio", "Decodes DTS audio streams", "Ronald Bultje "); enum { ARG_0, ARG_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") ); #if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED) #define DTS_CAPS "audio/x-raw-int, " \ "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \ "signed = (boolean) true, " \ "width = (int) 16, " \ "depth = (int) 16" #define SAMPLE_WIDTH 16 #elif defined(LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE) #define DTS_CAPS "audio/x-raw-float, " \ "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \ "width = (int) 64" #define SAMPLE_WIDTH 64 #else #define DTS_CAPS "audio/x-raw-float, " \ "endianness = (int) " G_STRINGIFY (G_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 ]") ); GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstElement, GST_TYPE_ELEMENT); static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps); static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event); static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf); static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element, GstStateChange transition); 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_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); 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); GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS audio decoder"); } static void gst_dtsdec_class_init (GstDtsDecClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; guint cpuflags; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_dtsdec_set_property; gobject_class->get_property = gst_dtsdec_get_property; gstelement_class->change_state = gst_dtsdec_change_state; 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)); oil_init (); klass->dts_cpuflags = 0; cpuflags = oil_cpu_get_flags (); if (cpuflags & OIL_IMPL_FLAG_MMX) klass->dts_cpuflags |= MM_ACCEL_X86_MMX; if (cpuflags & OIL_IMPL_FLAG_3DNOW) klass->dts_cpuflags |= MM_ACCEL_X86_3DNOW; if (cpuflags & OIL_IMPL_FLAG_MMXEXT) klass->dts_cpuflags |= MM_ACCEL_X86_MMXEXT; GST_LOG ("CPU flags: dts=%08x, liboil=%08x", klass->dts_cpuflags, cpuflags); } static void gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class) { dtsdec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); gst_pad_set_setcaps_function (dtsdec->sinkpad, GST_DEBUG_FUNCPTR (gst_dtsdec_sink_setcaps)); gst_pad_set_chain_function (dtsdec->sinkpad, GST_DEBUG_FUNCPTR (gst_dtsdec_chain)); gst_pad_set_event_function (dtsdec->sinkpad, GST_DEBUG_FUNCPTR (gst_dtsdec_sink_event)); gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->sinkpad); dtsdec->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); gst_pad_use_fixed_caps (dtsdec->srcpad); gst_element_add_pad (GST_ELEMENT (dtsdec), dtsdec->srcpad); dtsdec->dynamic_range_compression = FALSE; } static gint gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos) { gint chans = 0; GstAudioChannelPosition *tpos = NULL; if (pos) { /* Allocate the maximum, for ease */ tpos = *pos = g_new (GstAudioChannelPosition, 7); if (!tpos) return 0; } switch (flags & DTS_CHANNEL_MASK) { case DTS_MONO: chans = 1; if (tpos) tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; break; /* case DTS_CHANNEL: */ case DTS_STEREO: case DTS_STEREO_SUMDIFF: case DTS_STEREO_TOTAL: case DTS_DOLBY: chans = 2; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; } break; case DTS_3F: chans = 3; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; } break; case DTS_2F1R: chans = 3; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; } break; case DTS_3F1R: chans = 4; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; } break; case DTS_2F2R: chans = 4; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; tpos[2] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } break; case DTS_3F2R: chans = 5; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; tpos[3] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } break; case DTS_4F2R: chans = 6; if (tpos) { tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER; tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER; tpos[2] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; tpos[3] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; tpos[4] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; tpos[5] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } break; default: g_warning ("dtsdec: invalid flags 0x%x", flags); return 0; } if (flags & DTS_LFE) { if (tpos) { tpos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE; } chans += 1; } return chans; } static gboolean gst_dtsdec_renegotiate (GstDtsDec * dts) { GstAudioChannelPosition *pos; GstCaps *caps = gst_caps_from_string (DTS_CAPS); gint channels = gst_dtsdec_channels (dts->using_channels, &pos); gboolean result = FALSE; if (!channels) goto done; 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); gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); g_free (pos); if (!gst_pad_set_caps (dts->srcpad, caps)) goto done; result = TRUE; done: if (caps) { gst_caps_unref (caps); } return result; } static gboolean gst_dtsdec_sink_event (GstPad * pad, GstEvent * event) { GstDtsDec *dtsdec = GST_DTSDEC (gst_pad_get_parent (pad)); gboolean ret = FALSE; GST_LOG_OBJECT (dtsdec, "%s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_NEWSEGMENT:{ GstFormat format; gint64 val; gst_event_parse_new_segment (event, NULL, NULL, &format, &val, NULL, NULL); if (format != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (val)) { GST_WARNING ("No time in newsegment event %p", event); } else { dtsdec->current_ts = val; } if (dtsdec->cache) { gst_buffer_unref (dtsdec->cache); dtsdec->cache = NULL; } ret = gst_pad_event_default (pad, event); break; } case GST_EVENT_FLUSH_STOP: if (dtsdec->cache) { gst_buffer_unref (dtsdec->cache); dtsdec->cache = NULL; } ret = gst_pad_event_default (pad, event); break; default: ret = gst_pad_event_default (pad, event); break; } gst_object_unref (dtsdec); return ret; } static gboolean gst_dtsdec_sink_setcaps (GstPad * pad, GstCaps * caps) { GstDtsDec *dts = GST_DTSDEC (gst_pad_get_parent (pad)); 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; gst_object_unref (dts); return TRUE; } 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, taglist); } static GstFlowReturn gst_dtsdec_handle_frame (GstDtsDec * dts, guint8 * data, guint length, gint flags, gint sample_rate, gint bit_rate) { gboolean need_renegotiation = FALSE; gint channels, num_blocks; GstBuffer *out; gint i, s, c, num_c; sample_t *samples; GstFlowReturn result = GST_FLOW_OK; /* 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); } if (dts->request_channels == DTS_CHANNEL) { GstCaps *caps; caps = gst_pad_get_allowed_caps (dts->srcpad); 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] = { DTS_MONO, DTS_STEREO, DTS_STEREO | DTS_LFE, DTS_2F2R, DTS_2F2R | DTS_LFE, DTS_3F2R | DTS_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) dts->request_channels = dts_channels[channels - 1]; else dts->request_channels = dts_channels[5]; gst_caps_unref (copy); } else if (flags) { dts->request_channels = dts->stream_channels; } else { dts->request_channels = DTS_3F2R | DTS_LFE; } if (caps) gst_caps_unref (caps); } /* 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"); return GST_FLOW_OK; } 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)) { GST_ELEMENT_ERROR (dts, CORE, NEGOTIATION, (NULL), (NULL)); return GST_FLOW_ERROR; } } 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, NULL); result = gst_pad_alloc_buffer_and_set_caps (dts->srcpad, 0, (SAMPLE_WIDTH / 8) * 256 * num_c, GST_PAD_CAPS (dts->srcpad), &out); if (result != GST_FLOW_OK) break; GST_BUFFER_TIMESTAMP (out) = dts->current_ts; GST_BUFFER_DURATION (out) = GST_SECOND * 256 / dts->sample_rate; dts->current_ts += GST_BUFFER_DURATION (out); /* 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 */ result = gst_pad_push (dts->srcpad, out); if (result != GST_FLOW_OK) break; } return result; } static GstFlowReturn gst_dtsdec_chain_raw (GstPad * pad, GstBuffer * buf) { GstDtsDec *dts; guint8 *data; gint size; gint length, flags, sample_rate, bit_rate, frame_length; GstFlowReturn result = GST_FLOW_OK; dts = GST_DTSDEC (GST_PAD_PARENT (pad)); if (dts->cache) { buf = gst_buffer_join (dts->cache, buf); dts->cache = NULL; } data = GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); length = 0; while (size >= 7) { length = dts_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_DEBUG ("Sync: frame size %d", length); result = gst_dtsdec_handle_frame (dts, data, length, flags, sample_rate, bit_rate); if (result != GST_FLOW_OK) { size = 0; break; } size -= length; data += length; } else { GST_LOG ("Not enough data available (needed %d had %d)", length, size); break; } } /* keep cache */ if (length == 0) { GST_LOG ("No sync found"); } if (size > 0) { dts->cache = gst_buffer_create_sub (buf, GST_BUFFER_SIZE (buf) - size, size); } gst_buffer_unref (buf); return result; } static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf) { GstFlowReturn res = GST_FLOW_OK; GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad)); gint first_access; if (dts->dvdmode) { gint size = GST_BUFFER_SIZE (buf); guint8 *data = GST_BUFFER_DATA (buf); gint offset, len; GstBuffer *subbuf; 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_create_sub (buf, offset, len); GST_BUFFER_TIMESTAMP (subbuf) = GST_CLOCK_TIME_NONE; res = gst_dtsdec_chain_raw (pad, subbuf); if (res != GST_FLOW_OK) goto done; offset += len; len = size - offset; if (len > 0) { subbuf = gst_buffer_create_sub (buf, offset, len); GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buf); res = gst_dtsdec_chain_raw (pad, subbuf); } } 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); res = gst_dtsdec_chain_raw (pad, subbuf); } } else { res = gst_dtsdec_chain_raw (pad, buf); } done: return res; /* ERRORS */ not_enough_data: { GST_ELEMENT_ERROR (GST_ELEMENT (dts), STREAM, DECODE, (NULL), ("Insufficient data in buffer. Can't determine first_acess")); 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)); return GST_FLOW_ERROR; } } static GstStateChangeReturn gst_dtsdec_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstDtsDec *dts = GST_DTSDEC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY:{ GstDtsDecClass *klass; klass = GST_DTSDEC_CLASS (G_OBJECT_GET_CLASS (dts)); dts->state = dts_init (klass->dts_cpuflags); break; } case GST_STATE_CHANGE_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_CHANNEL; dts->using_channels = 0; dts->level = 1; dts->bias = 0; dts->current_ts = 0; 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: dts->samples = NULL; if (dts->cache) { gst_buffer_unref (dts->cache); dts->cache = NULL; } break; case GST_STATE_CHANGE_READY_TO_NULL: dts_free (dts->state); dts->state = NULL; break; default: break; } return ret; } 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_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);