/* GStreamer * Copyright (C) <2001> David I. Lehn * * 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 #include #include "_stdint.h" #include #include #include #include #include "gsta52dec.h" /* elementfactory information */ static GstElementDetails gst_a52dec_details = { "ATSC A/52 audio decoder", "Codec/Decoder/Audio", "Decodes ATSC A/52 encoded audio streams", "David I. Lehn ", }; #ifdef LIBA52_DOUBLE #define SAMPLE_WIDTH 64 #else #define SAMPLE_WIDTH 32 #endif GST_DEBUG_CATEGORY_STATIC (a52dec_debug); #define GST_CAT_DEFAULT (a52dec_debug) /* A52Dec signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; 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-ac3") ); static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " "endianness = (int) BYTE_ORDER, " "width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", " "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ], " "buffer-frames = (int) 0") ); static void gst_a52dec_base_init (gpointer g_class); static void gst_a52dec_class_init (GstA52DecClass * klass); static void gst_a52dec_init (GstA52Dec * a52dec); static void gst_a52dec_chain (GstPad * pad, GstData * data); static GstElementStateReturn gst_a52dec_change_state (GstElement * element); 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); static GstElementClass *parent_class = NULL; /* static guint gst_a52dec_signals[LAST_SIGNAL] = { 0 }; */ GType gst_a52dec_get_type (void) { static GType a52dec_type = 0; if (!a52dec_type) { static const GTypeInfo a52dec_info = { sizeof (GstA52DecClass), gst_a52dec_base_init, NULL, (GClassInitFunc) gst_a52dec_class_init, NULL, NULL, sizeof (GstA52Dec), 0, (GInstanceInitFunc) gst_a52dec_init, }; a52dec_type = g_type_register_static (GST_TYPE_ELEMENT, "GstA52Dec", &a52dec_info, 0); GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0, "AC3/A52 software decoder"); } return a52dec_type; } static void gst_a52dec_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_a52dec_details); } static void gst_a52dec_class_init (GstA52DecClass * 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_a52dec_set_property; gobject_class->get_property = gst_a52dec_get_property; gstelement_class->change_state = gst_a52dec_change_state; } static void gst_a52dec_init (GstA52Dec * a52dec) { GST_FLAG_SET (GST_ELEMENT (a52dec), GST_ELEMENT_EVENT_AWARE); /* create the sink and src pads */ a52dec->sinkpad = gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT (a52dec), "sink"), "sink"); gst_pad_set_chain_function (a52dec->sinkpad, gst_a52dec_chain); gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad); a52dec->srcpad = gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT (a52dec), "src"), "src"); gst_pad_use_explicit_caps (a52dec->srcpad); gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad); a52dec->dynamic_range_compression = FALSE; a52dec->cache = NULL; } static int gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos) { int chans = 0; GstAudioChannelPosition *pos = NULL; /* allocated just for safety. Number makes no sense */ if (_pos) { pos = g_new (GstAudioChannelPosition, 6); *_pos = pos; } if (flags & A52_LFE) { chans += 1; if (pos) { pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE; } } 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: */ 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; default: /* error */ g_warning ("a52dec invalid flags %d", flags); g_free (pos); return 0; } return chans; } static int gst_a52dec_push (GstA52Dec * a52dec, GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp) { GstBuffer *buf; int chans, n, c; flags &= (A52_CHANNEL_MASK | A52_LFE); chans = gst_a52dec_channels (flags, NULL); if (!chans) { return 1; } buf = gst_buffer_new (); GST_BUFFER_SIZE (buf) = 256 * chans * (SAMPLE_WIDTH / 8); GST_BUFFER_DATA (buf) = g_malloc (GST_BUFFER_SIZE (buf)); for (n = 0; n < 256; n++) { for (c = 0; c < chans; c++) { ((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] = samples[c * 256 + n]; } } GST_BUFFER_TIMESTAMP (buf) = timestamp; GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate; GST_DEBUG_OBJECT (a52dec, "Pushing buffer with ts %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); gst_pad_push (srcpad, GST_DATA (buf)); return 0; } static gboolean gst_a52dec_reneg (GstPad * pad) { GstAudioChannelPosition *pos; GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad)); gint channels = gst_a52dec_channels (a52dec->using_channels, &pos); GstCaps *caps; if (!channels) return FALSE; GST_INFO ("a52dec: reneg channels:%d rate:%d\n", channels, a52dec->sample_rate); caps = gst_caps_new_simple ("audio/x-raw-float", "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, SAMPLE_WIDTH, "channels", G_TYPE_INT, channels, "rate", G_TYPE_INT, a52dec->sample_rate, "buffer-frames", G_TYPE_INT, 0, NULL); gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); g_free (pos); return gst_pad_set_explicit_caps (pad, caps); } static void gst_a52dec_handle_event (GstA52Dec * a52dec, GstEvent * event) { 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:{ gint64 val; if (!gst_event_discont_get_value (event, GST_FORMAT_TIME, &val) || !GST_CLOCK_TIME_IS_VALID (val)) { GST_WARNING ("No time discont value in event %p", event); } else { a52dec->time = val; } } /* fall-through */ case GST_EVENT_FLUSH: if (a52dec->cache) { gst_buffer_unref (a52dec->cache); a52dec->cache = NULL; } break; default: break; } gst_pad_event_default (a52dec->sinkpad, event); } static void gst_a52dec_update_streaminfo (GstA52Dec * a52dec) { GstTagList *taglist; taglist = gst_tag_list_new (); gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND, GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL); gst_element_found_tags_for_pad (GST_ELEMENT (a52dec), GST_PAD (a52dec->srcpad), a52dec->time, taglist); } static gboolean gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data, guint length, gint flags, gint sample_rate, gint bit_rate) { gint channels, i; gboolean need_reneg = FALSE; /* update stream information, renegotiate or re-streaminfo if needed */ need_reneg = FALSE; if (a52dec->sample_rate != sample_rate) { need_reneg = TRUE; a52dec->sample_rate = sample_rate; } if (flags) { 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); } /* process */ flags = a52dec->request_channels; /* | A52_ADJUST_LEVEL; */ a52dec->level = 1; if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) { GST_WARNING ("a52_frame error"); return TRUE; } 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 == TRUE) { GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d\n", a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels); if (!gst_a52dec_reneg (a52dec->srcpad)) { GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL)); return FALSE; } } if (a52dec->dynamic_range_compression == FALSE) { a52_dynrng (a52dec->state, NULL, NULL); } /* each frame consists of 6 blocks */ for (i = 0; i < 6; i++) { if (a52_block (a52dec->state)) { GST_WARNING ("a52_block error %d", i); } else { /* push on */ gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels, a52dec->samples, a52dec->time); } a52dec->time += 256 * GST_SECOND / a52dec->sample_rate; } return TRUE; } static void gst_a52dec_chain (GstPad * pad, GstData * _data) { GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad)); GstBuffer *buf; guint8 *data; guint size; gint length = 0, flags, sample_rate, bit_rate; /* event handling */ if (GST_IS_EVENT (_data)) { gst_a52dec_handle_event (a52dec, GST_EVENT (_data)); return; } /* merge with cache, if any. Also make sure timestamps match */ buf = GST_BUFFER (_data); if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { a52dec->time = GST_BUFFER_TIMESTAMP (buf); GST_DEBUG_OBJECT (a52dec, "Received buffer with ts %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); } if (a52dec->cache) { buf = gst_buffer_join (a52dec->cache, buf); a52dec->cache = NULL; } data = GST_BUFFER_DATA (buf); size = GST_BUFFER_SIZE (buf); /* find and read header */ bit_rate = a52dec->bit_rate; sample_rate = a52dec->sample_rate; flags = 0; while (size >= 7) { length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate); if (length == 0) { /* no sync */ data++; size--; } else if (length <= size) { GST_DEBUG ("Sync: %d", length); if (!gst_a52dec_handle_frame (a52dec, data, length, flags, sample_rate, bit_rate)) { size = 0; break; } size -= length; data += length; } else { /* not enough data */ GST_LOG ("Not enough data available"); break; } } /* keep cache */ if (length == 0) { GST_LOG ("No sync found"); } if (size > 0) { a52dec->cache = gst_buffer_create_sub (buf, GST_BUFFER_SIZE (buf) - size, size); } gst_buffer_unref (buf); } static GstElementStateReturn gst_a52dec_change_state (GstElement * element) { GstA52Dec *a52dec = GST_A52DEC (element); GstCPUFlags cpuflags; uint32_t a52_cpuflags = 0; switch (GST_STATE_TRANSITION (element)) { case GST_STATE_NULL_TO_READY: cpuflags = gst_cpu_get_flags (); if (cpuflags & GST_CPU_FLAG_MMX) a52_cpuflags |= MM_ACCEL_X86_MMX; if (cpuflags & GST_CPU_FLAG_3DNOW) a52_cpuflags |= MM_ACCEL_X86_3DNOW; if (cpuflags & GST_CPU_FLAG_MMXEXT) a52_cpuflags |= MM_ACCEL_X86_MMXEXT; a52dec->state = a52_init (a52_cpuflags); break; case GST_STATE_READY_TO_PAUSED: a52dec->samples = a52_samples (a52dec->state); a52dec->bit_rate = -1; a52dec->sample_rate = -1; a52dec->stream_channels = A52_CHANNEL; a52dec->request_channels = A52_3F2R | A52_LFE; a52dec->using_channels = A52_CHANNEL; a52dec->level = 1; a52dec->bias = 0; a52dec->time = 0; break; case GST_STATE_PAUSED_TO_PLAYING: break; case GST_STATE_PLAYING_TO_PAUSED: break; case GST_STATE_PAUSED_TO_READY: a52dec->samples = NULL; if (a52dec->cache) { gst_buffer_unref (a52dec->cache); a52dec->cache = NULL; } break; case GST_STATE_READY_TO_NULL: a52_free (a52dec->state); a52dec->state = NULL; break; default: break; } GST_ELEMENT_CLASS (parent_class)->change_state (element); return GST_STATE_SUCCESS; } static void gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstA52Dec *src; /* it's not null if we got it, but it might not be ours */ g_return_if_fail (GST_IS_A52DEC (object)); src = GST_A52DEC (object); switch (prop_id) { case ARG_DRC: src->dynamic_range_compression = g_value_get_boolean (value); 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; /* it's not null if we got it, but it might not be ours */ g_return_if_fail (GST_IS_A52DEC (object)); src = GST_A52DEC (object); switch (prop_id) { case ARG_DRC: g_value_set_boolean (value, src->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 ("gstaudio")) return FALSE; if (!gst_element_register (plugin, "a52dec", GST_RANK_PRIMARY, GST_TYPE_A52DEC)) return FALSE; return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "a52dec", "Decodes ATSC A/52 encoded audio streams", plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN);