From b7c53b95c1f155fedde73925e571b6b69af8dfbd Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Tue, 10 Jan 2012 16:59:18 +0100 Subject: dtsdec: Port to 0.11 --- ext/dts/gstdtsdec.c | 283 ++++++++++++++++++++++++++-------------------------- ext/dts/gstdtsdec.h | 2 + 2 files changed, 144 insertions(+), 141 deletions(-) (limited to 'ext/dts') diff --git a/ext/dts/gstdtsdec.c b/ext/dts/gstdtsdec.c index f71219478..74ef14c01 100644 --- a/ext/dts/gstdtsdec.c +++ b/ext/dts/gstdtsdec.c @@ -43,7 +43,7 @@ #include #include -#include +#include #ifndef DTS_OLD #include @@ -87,10 +87,13 @@ typedef struct dts_state_s dca_state_t; #if defined(LIBDTS_FIXED) || defined(LIBDCA_FIXED) #define SAMPLE_WIDTH 16 +#define SAMPLE_FORMAT GST_AUDIO_NE(S16) #elif defined (LIBDTS_DOUBLE) || defined(LIBDCA_DOUBLE) #define SAMPLE_WIDTH 64 +#define SAMPLE_FORMAT GST_AUDIO_NE(F64) #else #define SAMPLE_WIDTH 32 +#define SAMPLE_FORMAT GST_AUDIO_NE(F32) #endif GST_DEBUG_CATEGORY_STATIC (dtsdec_debug); @@ -98,8 +101,8 @@ GST_DEBUG_CATEGORY_STATIC (dtsdec_debug); enum { - ARG_0, - ARG_DRC + PROP_0, + PROP_DRC }; static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", @@ -108,27 +111,16 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", 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) " G_STRINGIFY (SAMPLE_WIDTH) ", " \ - "depth = (int) 16" -#else -#define DTS_CAPS "audio/x-raw-float, " \ - "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " \ - "width = (int) " G_STRINGIFY (SAMPLE_WIDTH) -#endif - static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, - GST_STATIC_CAPS (DTS_CAPS ", " + GST_STATIC_CAPS ("audio/x-raw, " + "format = (string) " SAMPLE_FORMAT ", " + "layout = (string) interleaved, " "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]") ); -GST_BOILERPLATE (GstDtsDec, gst_dtsdec, GstAudioDecoder, - GST_TYPE_AUDIO_DECODER); +G_DEFINE_TYPE (GstDtsDec, gst_dtsdec, GST_TYPE_AUDIO_DECODER); static gboolean gst_dtsdec_start (GstAudioDecoder * dec); static gboolean gst_dtsdec_stop (GstAudioDecoder * dec); @@ -140,45 +132,39 @@ static GstFlowReturn gst_dtsdec_handle_frame (GstAudioDecoder * dec, static GstFlowReturn gst_dtsdec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer); -static GstFlowReturn gst_dtsdec_chain (GstPad * pad, GstBuffer * buf); +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_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_simple (element_class, "DTS audio decoder", - "Codec/Decoder/Audio", - "Decodes DTS audio streams", - "Jan Schmidt , " - "Ronald Bultje "); - - GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder"); -} - 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 , " + "Ronald Bultje "); + 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); @@ -194,7 +180,7 @@ gst_dtsdec_class_init (GstDtsDecClass * klass) * 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), ARG_DRC, + 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)); @@ -218,7 +204,7 @@ gst_dtsdec_class_init (GstDtsDecClass * klass) } static void -gst_dtsdec_init (GstDtsDec * dtsdec, GstDtsDecClass * g_class) +gst_dtsdec_init (GstDtsDec * dtsdec) { dtsdec->request_channels = DCA_CHANNEL; dtsdec->dynamic_range_compression = FALSE; @@ -284,12 +270,12 @@ gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter, guint8 *data; gint av, size; gint length = 0, flags, sample_rate, bit_rate, frame_length; - GstFlowReturn result = GST_FLOW_UNEXPECTED; + GstFlowReturn result = GST_FLOW_EOS; dts = GST_DTSDEC (bdec); size = av = gst_adapter_available (adapter); - data = (guint8 *) gst_adapter_peek (adapter, av); + data = (guint8 *) gst_adapter_map (adapter, av); /* find and read header */ bit_rate = dts->bit_rate; @@ -313,6 +299,7 @@ gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter, break; } } + gst_adapter_unmap (adapter); *_offset = av - size; *len = length; @@ -321,23 +308,16 @@ gst_dtsdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter, } static gint -gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos) +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 & DCA_CHANNEL_MASK) { case DCA_MONO: chans = 1; - if (tpos) - tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_MONO; + if (pos) { + pos[0] = GST_AUDIO_CHANNEL_POSITION_MONO; + } break; /* case DCA_CHANNEL: */ case DCA_STEREO: @@ -345,64 +325,64 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos) case DCA_STEREO_TOTAL: case DCA_DOLBY: chans = 2; - if (tpos) { - tpos[0] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; - tpos[1] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; + 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 (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; + 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 (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; + 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 (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; + 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 (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; + 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 (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; + 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 (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; + 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: @@ -410,8 +390,8 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos) return 0; } if (flags & DCA_LFE) { - if (tpos) { - tpos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE; + if (pos) { + pos[chans] = GST_AUDIO_CHANNEL_POSITION_LFE1; } chans += 1; } @@ -422,24 +402,39 @@ gst_dtsdec_channels (uint32_t flags, GstAudioChannelPosition ** pos) 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); + gint channels; + GstCaps *caps = NULL; gboolean result = FALSE; + GstAudioChannelPosition from[6], to[6]; + + channels = gst_dtsdec_channels (dts->using_channels, from); if (!channels) goto done; - GST_INFO ("dtsdec renegotiate, channels=%d, rate=%d", + GST_INFO_OBJECT (dts, "dtsdec renegotiate, channels=%d, rate=%d", channels, dts->sample_rate); - gst_caps_set_simple (caps, + 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); + + caps = gst_caps_new_simple ("audio/x-raw", + "format", G_TYPE_STRING, SAMPLE_FORMAT, + "layout", G_TYPE_STRING, "interleaved", "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); + "rate", G_TYPE_INT, dts->sample_rate, NULL); + + if (channels > 1) { + guint64 channel_mask = 0; - if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dts), caps)) + gst_audio_channel_positions_to_mask (to, channels, &channel_mask); + gst_caps_set_simple (caps, "channel-mask", GST_TYPE_BITMASK, channel_mask, + NULL); + } + + if (!gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (dts), caps)) goto done; result = TRUE; @@ -457,7 +452,7 @@ gst_dtsdec_update_streaminfo (GstDtsDec * dts) GstTagList *taglist; if (dts->bit_rate > 3) { - taglist = gst_tag_list_new (); + 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); @@ -477,8 +472,8 @@ gst_dtsdec_pre_push (GstAudioDecoder * bdec, GstBuffer ** buffer) GstDtsDec *dts = GST_DTSDEC (bdec); if (G_UNLIKELY (dts->pending_tags)) { - gst_element_found_tags_for_pad (GST_ELEMENT (dts), - GST_AUDIO_DECODER_SRC_PAD (dts), dts->pending_tags); + gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (dts), + gst_event_new_tag (dts->pending_tags)); dts->pending_tags = NULL; } @@ -492,7 +487,8 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) gint channels, i, num_blocks; gboolean need_renegotiation = FALSE; guint8 *data; - gint size, chans; + gsize size; + gint chans; gint length = 0, flags, sample_rate, bit_rate, frame_length; GstFlowReturn result = GST_FLOW_OK; GstBuffer *outbuf; @@ -504,8 +500,7 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) return GST_FLOW_OK; /* parsed stuff already, so this should work out fine */ - data = GST_BUFFER_DATA (buffer); - size = GST_BUFFER_SIZE (buffer); + data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ); g_assert (size >= 7); bit_rate = dts->bit_rate; @@ -590,10 +585,12 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) flags |= DCA_ADJUST_LEVEL; dts->level = 1; if (dca_frame (dts->state, data, &flags, &dts->level, dts->bias)) { + gst_buffer_unmap (buffer, data, size); GST_AUDIO_DECODER_ERROR (dts, 1, STREAM, DECODE, (NULL), ("dts_frame error"), result); goto exit; } + gst_buffer_unmap (buffer, data, size); channels = flags & (DCA_CHANNEL_MASK | DCA_LFE); if (dts->using_channels != channels) { @@ -621,32 +618,34 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) /* handle decoded data, one block is 256 samples */ num_blocks = dca_blocks_num (dts->state); - result = - gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dts), 0, - 256 * chans * (SAMPLE_WIDTH / 8) * num_blocks, - GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dts)), &outbuf); - if (result != GST_FLOW_OK) - goto exit; + outbuf = + gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks); - data = GST_BUFFER_DATA (outbuf); - 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; - - for (n = 0; n < 256; n++) { - for (c = 0; c < chans; c++) { - ((sample_t *) data)[n * chans + c] = dts->samples[c * 256 + n]; + data = gst_buffer_map (outbuf, &size, NULL, GST_MAP_WRITE); + { + 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); } - data += 256 * chans * (SAMPLE_WIDTH / 8); } + gst_buffer_unmap (outbuf, data, size); result = gst_audio_decoder_finish_frame (bdec, outbuf, 1); @@ -684,18 +683,19 @@ gst_dtsdec_set_format (GstAudioDecoder * bdec, GstCaps * caps) } static GstFlowReturn -gst_dtsdec_chain (GstPad * pad, GstBuffer * buf) +gst_dtsdec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstFlowReturn ret = GST_FLOW_OK; - GstDtsDec *dts = GST_DTSDEC (GST_PAD_PARENT (pad)); + GstDtsDec *dts = GST_DTSDEC (parent); gint first_access; if (dts->dvdmode) { - gint size = GST_BUFFER_SIZE (buf); - guint8 *data = GST_BUFFER_DATA (buf); + 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; @@ -711,10 +711,9 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf) if (len <= 0 || offset + len > size) goto bad_first_access_parameter; - subbuf = gst_buffer_create_sub (buf, offset, len); - gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL); + 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, subbuf); + ret = dts->base_chain (pad, parent, subbuf); if (ret != GST_FLOW_OK) { gst_buffer_unref (buf); goto done; @@ -724,23 +723,23 @@ gst_dtsdec_chain (GstPad * pad, GstBuffer * buf) len = size - offset; if (len > 0) { - subbuf = gst_buffer_create_sub (buf, offset, len); - gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL); + 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, subbuf); + 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_create_sub (buf, offset, size - offset); - gst_buffer_copy_metadata (subbuf, buf, GST_BUFFER_COPY_ALL); + 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, subbuf); + ret = dts->base_chain (pad, parent, subbuf); gst_buffer_unref (buf); } } else { - ret = dts->base_chain (pad, buf); + ret = dts->base_chain (pad, parent, buf); } done: @@ -770,7 +769,7 @@ gst_dtsdec_set_property (GObject * object, guint prop_id, const GValue * value, GstDtsDec *dts = GST_DTSDEC (object); switch (prop_id) { - case ARG_DRC: + case PROP_DRC: dts->dynamic_range_compression = g_value_get_boolean (value); break; default: @@ -786,7 +785,7 @@ gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value, GstDtsDec *dts = GST_DTSDEC (object); switch (prop_id) { - case ARG_DRC: + case PROP_DRC: g_value_set_boolean (value, dts->dynamic_range_compression); break; default: @@ -798,6 +797,8 @@ gst_dtsdec_get_property (GObject * object, guint prop_id, GValue * value, static gboolean plugin_init (GstPlugin * plugin) { + GST_DEBUG_CATEGORY_INIT (dtsdec_debug, "dtsdec", 0, "DTS/DCA audio decoder"); + #if HAVE_ORC orc_init (); #endif diff --git a/ext/dts/gstdtsdec.h b/ext/dts/gstdtsdec.h index be6005a0a..16b7e91ea 100644 --- a/ext/dts/gstdtsdec.h +++ b/ext/dts/gstdtsdec.h @@ -56,6 +56,8 @@ struct _GstDtsDec { gint request_channels; gint using_channels; + gint channel_reorder_map[6]; + /* decoding properties */ sample_t level; sample_t bias; -- cgit v1.2.1