/* GStreamer * Copyright (C) <2001> David I. Lehn <dlehn@users.sourceforge.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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-a52dec * @title: a52dec * * Dolby Digital (AC-3) audio decoder. * * ## Example launch line * |[ * gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink * ]| Play audio part of a dvd title. * |[ * gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink * ]| Decode and play a stand alone AC-3 file. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include <stdlib.h> #ifdef HAVE_STDINT_H #include <stdint.h> #endif #include <gst/gst.h> #include <a52dec/a52.h> #if !defined(A52_ACCEL_DETECT) # include <a52dec/mm_accel.h> #endif #include "gsta52dec.h" #if HAVE_ORC #include <orc/orc.h> #endif #ifdef LIBA52_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 (a52dec_debug); #define GST_CAT_DEFAULT (a52dec_debug) /* A52Dec args */ enum { ARG_0, ARG_DRC, ARG_MODE, ARG_LFE, }; static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-ac3; audio/ac3; audio/x-private1-ac3") ); 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 ]") ); static gboolean a52_element_init (GstPlugin * plugin); #define gst_a52dec_parent_class parent_class G_DEFINE_TYPE (GstA52Dec, gst_a52dec, GST_TYPE_AUDIO_DECODER); GST_ELEMENT_REGISTER_DEFINE_CUSTOM (a52dec, a52_element_init); 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 GstFlowReturn 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_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static void gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); #define GST_TYPE_A52DEC_MODE (gst_a52dec_mode_get_type()) static GType gst_a52dec_mode_get_type (void) { static GType a52dec_mode_type = 0; static const GEnumValue a52dec_modes[] = { {A52_MONO, "Mono", "mono"}, {A52_STEREO, "Stereo", "stereo"}, {A52_3F, "3 Front", "3f"}, {A52_2F1R, "2 Front, 1 Rear", "2f1r"}, {A52_3F1R, "3 Front, 1 Rear", "3f1r"}, {A52_2F2R, "2 Front, 2 Rear", "2f2r"}, {A52_3F2R, "3 Front, 2 Rear", "3f2r"}, {A52_DOLBY, "Dolby", "dolby"}, {0, NULL, NULL}, }; if (!a52dec_mode_type) { a52dec_mode_type = g_enum_register_static ("GstA52DecMode", a52dec_modes); } return a52dec_mode_type; } static void gst_a52dec_class_init (GstA52DecClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstAudioDecoderClass *gstbase_class; guint cpuflags = 0; 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; 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); /** * GstA52Dec::drc * * Set to true to apply the recommended Dolby Digital 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), ARG_DRC, g_param_spec_boolean ("drc", "Dynamic Range Compression", "Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstA52Dec::mode * * Force a particular output channel configuration from the decoder. By default, * the channel downmix (if any) is chosen automatically based on the downstream * capabilities of the pipeline. */ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MODE, g_param_spec_enum ("mode", "Decoder Mode", "Decoding Mode (default 3f2r)", GST_TYPE_A52DEC_MODE, A52_3F2R, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstA52Dec::lfe * * Whether to output the LFE (Low Frequency Emitter) channel of the audio stream. */ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_LFE, g_param_spec_boolean ("lfe", "LFE", "LFE", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &sink_factory); gst_element_class_add_static_pad_template (gstelement_class, &src_factory); gst_element_class_set_static_metadata (gstelement_class, "ATSC A/52 audio decoder", "Codec/Decoder/Audio/Converter", "Decodes ATSC A/52 encoded audio streams", "David I. Lehn <dlehn@users.sourceforge.net>"); GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0, "AC3/A52 software decoder"); /* If no CPU instruction based acceleration is available, end up using the * generic software djbfft based one when available in the used liba52 */ #ifdef MM_ACCEL_DJBFFT klass->a52_cpuflags = MM_ACCEL_DJBFFT; #elif defined(A52_ACCEL_DETECT) klass->a52_cpuflags = A52_ACCEL_DETECT; #else klass->a52_cpuflags = 0; #endif #if HAVE_ORC && !defined(A52_ACCEL_DETECT) cpuflags = orc_target_get_default_flags (orc_target_get_by_name ("mmx")); if (cpuflags & ORC_TARGET_MMX_MMX) klass->a52_cpuflags |= MM_ACCEL_X86_MMX; if (cpuflags & ORC_TARGET_MMX_3DNOW) klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW; if (cpuflags & ORC_TARGET_MMX_MMXEXT) klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT; #endif GST_LOG ("CPU flags: a52=%08x, orc=%08x", klass->a52_cpuflags, cpuflags); gst_type_mark_as_plugin_api (GST_TYPE_A52DEC_MODE, 0); } static void gst_a52dec_init (GstA52Dec * a52dec) { a52dec->request_channels = A52_CHANNEL; a52dec->dynamic_range_compression = FALSE; a52dec->state = NULL; a52dec->samples = NULL; gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST (a52dec), TRUE); GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (a52dec)); /* 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; static GMutex init_mutex; GST_DEBUG_OBJECT (dec, "start"); klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec)); g_mutex_lock (&init_mutex); #if defined(A52_ACCEL_DETECT) a52dec->state = a52_init (); /* This line is just to avoid being accused of not using klass */ a52_accel (klass->a52_cpuflags & A52_ACCEL_DETECT); #else a52dec->state = a52_init (klass->a52_cpuflags); #endif g_mutex_unlock (&init_mutex); if (!a52dec->state) { GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), LIBRARY, INIT, (NULL), ("failed to initialize a52 state")); return FALSE; } 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_estimate_rate (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; } return TRUE; } static GstFlowReturn gst_a52dec_parse (GstAudioDecoder * bdec, GstAdapter * adapter, gint * _offset, gint * len) { GstA52Dec *a52dec; const guint8 *data; gint av, size; gint length = 0, flags, sample_rate, bit_rate; GstFlowReturn result = GST_FLOW_EOS; a52dec = GST_A52DEC (bdec); size = av = gst_adapter_available (adapter); data = (const guint8 *) gst_adapter_map (adapter, av); /* find and read header */ bit_rate = a52dec->bit_rate; sample_rate = a52dec->sample_rate; flags = 0; while (size >= 7) { length = a52_syncinfo ((guint8 *) 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; } } gst_adapter_unmap (adapter); *_offset = av - size; *len = length; return result; } static gint gst_a52dec_channels (int flags, GstAudioChannelPosition * pos) { gint chans = 0; if (flags & A52_LFE) { chans += 1; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE1; } } flags &= A52_CHANNEL_MASK; switch (flags) { case A52_3F2R: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } chans += 5; break; case A52_2F2R: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT; pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT; } chans += 4; break; case A52_3F1R: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; } chans += 4; break; case A52_2F1R: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER; } chans += 3; break; case A52_3F: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER; pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; } chans += 3; break; case A52_CHANNEL: /* Dual mono. Should really be handled as 2 src pads */ case A52_STEREO: case A52_DOLBY: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; } chans += 2; break; case A52_MONO: if (pos) { pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_MONO; } chans += 1; break; default: /* error, caller should post error message */ return 0; } return chans; } static gboolean gst_a52dec_reneg (GstA52Dec * a52dec) { gint channels; gboolean result = FALSE; GstAudioChannelPosition from[6], to[6]; GstAudioInfo info; channels = gst_a52dec_channels (a52dec->using_channels, from); if (!channels) goto done; GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d", channels, a52dec->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, a52dec->channel_reorder_map); gst_audio_info_init (&info); gst_audio_info_set_format (&info, SAMPLE_TYPE, a52dec->sample_rate, channels, (channels > 1 ? to : NULL)); if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (a52dec), &info)) goto done; result = TRUE; done: return result; } static void gst_a52dec_update_streaminfo (GstA52Dec * a52dec) { GstTagList *taglist; taglist = gst_tag_list_new_empty (); gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL); gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (a52dec), taglist, GST_TAG_MERGE_REPLACE); gst_tag_list_unref (taglist); } static GstFlowReturn gst_a52dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) { GstA52Dec *a52dec; gint channels, i; gboolean need_reneg = FALSE; gint chans; gint length = 0, flags, sample_rate, bit_rate; GstMapInfo map; 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 */ gst_buffer_map (buffer, &map, GST_MAP_READ); g_assert (map.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 (map.data, &flags, &sample_rate, &bit_rate); g_assert (length == map.size); /* update stream information, renegotiate or re-streaminfo if needed */ need_reneg = FALSE; if (a52dec->sample_rate != sample_rate) { GST_DEBUG_OBJECT (a52dec, "sample rate changed"); need_reneg = TRUE; a52dec->sample_rate = sample_rate; } if (flags) { if (a52dec->stream_channels != (flags & (A52_CHANNEL_MASK | A52_LFE))) { GST_DEBUG_OBJECT (a52dec, "stream channel flags changed, marking update"); a52dec->flag_update = TRUE; } a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE); } if (bit_rate != a52dec->bit_rate) { a52dec->bit_rate = bit_rate; gst_a52dec_update_streaminfo (a52dec); } /* 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. */ if (a52dec->request_channels != A52_CHANNEL) { flags = a52dec->request_channels; } else if (a52dec->flag_update) { GstCaps *caps; a52dec->flag_update = FALSE; 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); gint orig_channels = flags ? gst_a52dec_channels (flags, NULL) : 6; gint fixed_channels = 0; const int a52_channels[6] = { A52_MONO, A52_STEREO, A52_STEREO | A52_LFE, A52_2F2R, A52_2F2R | A52_LFE, A52_3F2R | A52_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", orig_channels); if (gst_structure_get_int (structure, "channels", &fixed_channels) && fixed_channels <= 6) { if (fixed_channels < orig_channels) flags = a52_channels[fixed_channels - 1]; } else { flags = a52_channels[5]; } gst_caps_unref (copy); } else if (flags) flags = a52dec->stream_channels; else flags = A52_3F2R | A52_LFE; if (caps) gst_caps_unref (caps); } else { flags = a52dec->using_channels; } /* process */ flags |= A52_ADJUST_LEVEL; a52dec->level = 1; if (a52_frame (a52dec->state, map.data, &flags, &a52dec->level, a52dec->bias)) { gst_buffer_unmap (buffer, &map); GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL), ("a52_frame error"), result); goto exit; } gst_buffer_unmap (buffer, &map); channels = flags & (A52_CHANNEL_MASK | A52_LFE); if (a52dec->using_channels != channels) { need_reneg = TRUE; a52dec->using_channels = channels; } /* negotiate if required */ if (need_reneg) { 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)) goto failed_negotiation; } if (a52dec->dynamic_range_compression == FALSE) { a52_dynrng (a52dec->state, NULL, NULL); } 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 */ outbuf = gst_buffer_new_and_alloc (256 * chans * (SAMPLE_WIDTH / 8) * num_blocks); gst_buffer_map (outbuf, &map, GST_MAP_WRITE); { guint8 *ptr = map.data; for (i = 0; i < num_blocks; i++) { if (a52_block (a52dec->state)) { /* also marks discont */ GST_AUDIO_DECODER_ERROR (a52dec, 1, STREAM, DECODE, (NULL), ("error decoding block %d", i), result); if (result != GST_FLOW_OK) { gst_buffer_unmap (outbuf, &map); gst_buffer_unref (outbuf); goto exit; } } else { gint n, c; gint *reorder_map = a52dec->channel_reorder_map; for (n = 0; n < 256; n++) { for (c = 0; c < chans; c++) { ((sample_t *) ptr)[n * chans + reorder_map[c]] = a52dec->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 (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_set_format (GstAudioDecoder * bdec, GstCaps * caps) { GstA52Dec *a52dec = GST_A52DEC (bdec); GstStructure *structure; structure = gst_caps_get_structure (caps, 0); if (structure && gst_structure_has_name (structure, "audio/x-private1-ac3")) a52dec->dvdmode = TRUE; else a52dec->dvdmode = FALSE; return TRUE; } static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf) { GstA52Dec *a52dec = GST_A52DEC (parent); GstFlowReturn ret = GST_FLOW_OK; gint first_access; if (a52dec->dvdmode) { gsize size; guint8 data[2]; gint offset; gint len; GstBuffer *subbuf; size = gst_buffer_get_size (buf); if (size < 2) goto not_enough_data; gst_buffer_extract (buf, 0, data, 2); 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 = a52dec->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 = a52dec->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); gst_buffer_unref (buf); ret = a52dec->base_chain (pad, parent, subbuf); } } else { ret = a52dec->base_chain (pad, parent, buf); } done: return ret; /* ERRORS */ not_enough_data: { GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), 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 (a52dec), STREAM, DECODE, (NULL), ("Bad first_access parameter (%d) in buffer", first_access)); gst_buffer_unref (buf); return GST_FLOW_ERROR; } } static void gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstA52Dec *src = GST_A52DEC (object); switch (prop_id) { case ARG_DRC: GST_OBJECT_LOCK (src); src->dynamic_range_compression = g_value_get_boolean (value); GST_OBJECT_UNLOCK (src); break; case ARG_MODE: GST_OBJECT_LOCK (src); src->request_channels &= ~A52_CHANNEL_MASK; src->request_channels |= g_value_get_enum (value); GST_OBJECT_UNLOCK (src); break; case ARG_LFE: GST_OBJECT_LOCK (src); src->request_channels &= ~A52_LFE; src->request_channels |= g_value_get_boolean (value) ? A52_LFE : 0; GST_OBJECT_UNLOCK (src); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstA52Dec *src = GST_A52DEC (object); switch (prop_id) { case ARG_DRC: GST_OBJECT_LOCK (src); g_value_set_boolean (value, src->dynamic_range_compression); GST_OBJECT_UNLOCK (src); break; case ARG_MODE: GST_OBJECT_LOCK (src); g_value_set_enum (value, src->request_channels & A52_CHANNEL_MASK); GST_OBJECT_UNLOCK (src); break; case ARG_LFE: GST_OBJECT_LOCK (src); g_value_set_boolean (value, src->request_channels & A52_LFE); GST_OBJECT_UNLOCK (src); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean a52_element_init (GstPlugin * plugin) { #if HAVE_ORC orc_init (); #endif return gst_element_register (plugin, "a52dec", GST_RANK_SECONDARY, GST_TYPE_A52DEC); } static gboolean plugin_init (GstPlugin * plugin) { return GST_ELEMENT_REGISTER (a52dec, plugin); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, a52dec, "Decodes ATSC A/52 encoded audio streams", plugin_init, VERSION, "GPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);