/* 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. */ /** * SECTION:element-a52dec * * Dolby Digital (AC-3) audio decoder. * * * Example launch line * |[ * gst-launch dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioresample ! audioconvert ! alsasink * ]| Play audio track from a dvd. * |[ * gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink * ]| Decode a stand alone file and play it. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "_stdint.h" #include #include #include #include #include "gsta52dec.h" #include #include #include /* 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 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-float, " "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", " "width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", " "rate = (int) [ 4000, 96000 ], " "channels = (int) [ 1, 6 ]") ); GST_BOILERPLATE (GstA52Dec, gst_a52dec, GstElement, GST_TYPE_ELEMENT); static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buffer); static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf); static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps); static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event); static GstStateChangeReturn gst_a52dec_change_state (GstElement * element, GstStateChange transition); 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_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); GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0, "AC3/A52 software decoder"); } static void gst_a52dec_class_init (GstA52DecClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; guint cpuflags; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->set_property = gst_a52dec_set_property; gobject_class->get_property = gst_a52dec_get_property; gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_a52dec_change_state); /** * 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)); /** * 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)); /** * 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)); oil_init (); klass->a52_cpuflags = 0; cpuflags = oil_cpu_get_flags (); if (cpuflags & OIL_IMPL_FLAG_MMX) klass->a52_cpuflags |= MM_ACCEL_X86_MMX; if (cpuflags & OIL_IMPL_FLAG_3DNOW) klass->a52_cpuflags |= MM_ACCEL_X86_3DNOW; if (cpuflags & OIL_IMPL_FLAG_MMXEXT) klass->a52_cpuflags |= MM_ACCEL_X86_MMXEXT; GST_LOG ("CPU flags: a52=%08x, liboil=%08x", klass->a52_cpuflags, cpuflags); } static void gst_a52dec_init (GstA52Dec * a52dec, GstA52DecClass * g_class) { /* create the sink and src pads */ a52dec->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink"); gst_pad_set_setcaps_function (a52dec->sinkpad, GST_DEBUG_FUNCPTR (gst_a52dec_sink_setcaps)); gst_pad_set_chain_function (a52dec->sinkpad, GST_DEBUG_FUNCPTR (gst_a52dec_chain)); gst_pad_set_event_function (a52dec->sinkpad, GST_DEBUG_FUNCPTR (gst_a52dec_sink_event)); gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad); a52dec->srcpad = gst_pad_new_from_static_template (&src_factory, "src"); gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad); a52dec->request_channels = A52_CHANNEL; a52dec->dynamic_range_compression = FALSE; gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED); } static gint gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos) { gint 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: /* 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_FRONT_MONO; } chans += 1; break; default: /* error, caller should post error message */ g_free (pos); return 0; } return chans; } static void clear_queued (GstA52Dec * dec) { g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL); g_list_free (dec->queued); dec->queued = NULL; } static GstFlowReturn flush_queued (GstA52Dec * dec) { GstFlowReturn ret = GST_FLOW_OK; while (dec->queued) { GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data); GST_LOG_OBJECT (dec, "pushing buffer %p, timestamp %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); /* iterate ouput queue an push downstream */ ret = gst_pad_push (dec->srcpad, buf); dec->queued = g_list_delete_link (dec->queued, dec->queued); } return ret; } static GstFlowReturn gst_a52dec_drain (GstA52Dec * dec) { GstFlowReturn ret = GST_FLOW_OK; if (dec->segment.rate < 0.0) { /* if we have some queued frames for reverse playback, flush * them now */ ret = flush_queued (dec); } return ret; } static GstFlowReturn gst_a52dec_push (GstA52Dec * a52dec, GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp) { GstBuffer *buf; int chans, n, c; GstFlowReturn result; flags &= (A52_CHANNEL_MASK | A52_LFE); chans = gst_a52dec_channels (flags, NULL); if (!chans) { GST_ELEMENT_ERROR (GST_ELEMENT (a52dec), STREAM, DECODE, (NULL), ("invalid channel flags: %d", flags)); return GST_FLOW_ERROR; } result = gst_pad_alloc_buffer_and_set_caps (srcpad, 0, 256 * chans * (SAMPLE_WIDTH / 8), GST_PAD_CAPS (srcpad), &buf); if (result != GST_FLOW_OK) return result; 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; result = GST_FLOW_OK; if ((buf = gst_audio_buffer_clip (buf, &a52dec->segment, a52dec->sample_rate, (SAMPLE_WIDTH / 8) * chans))) { /* set discont when needed */ if (a52dec->discont) { GST_LOG_OBJECT (a52dec, "marking DISCONT"); GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); a52dec->discont = FALSE; } if (a52dec->segment.rate > 0.0) { 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))); result = gst_pad_push (srcpad, buf); } else { /* reverse playback, queue frame till later when we get a discont. */ GST_DEBUG_OBJECT (a52dec, "queued frame"); a52dec->queued = g_list_prepend (a52dec->queued, buf); } } return result; } static gboolean gst_a52dec_reneg (GstA52Dec * a52dec, GstPad * pad) { GstAudioChannelPosition *pos; gint channels = gst_a52dec_channels (a52dec->using_channels, &pos); GstCaps *caps = NULL; gboolean result = FALSE; if (!channels) goto done; GST_INFO_OBJECT (a52dec, "reneg channels:%d rate:%d", 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, NULL); gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); g_free (pos); if (!gst_pad_set_caps (pad, caps)) goto done; result = TRUE; done: if (caps) gst_caps_unref (caps); return result; } static gboolean gst_a52dec_sink_event (GstPad * pad, GstEvent * event) { GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad)); gboolean ret = FALSE; GST_LOG ("Handling %s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_NEWSEGMENT: { GstFormat fmt; gboolean update; gint64 start, end, pos; gdouble rate; gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &end, &pos); /* drain queued buffers before activating the segment so that we can clip * against the old segment first */ gst_a52dec_drain (a52dec); if (fmt != GST_FORMAT_TIME || !GST_CLOCK_TIME_IS_VALID (start)) { GST_WARNING ("No time in newsegment event %p (format is %s)", event, gst_format_get_name (fmt)); gst_event_unref (event); a52dec->sent_segment = FALSE; /* set some dummy values, FIXME: do proper conversion */ a52dec->time = start = pos = 0; fmt = GST_FORMAT_TIME; end = -1; } else { a52dec->time = start; a52dec->sent_segment = TRUE; ret = gst_pad_push_event (a52dec->srcpad, event); } gst_segment_set_newsegment (&a52dec->segment, update, rate, fmt, start, end, pos); break; } case GST_EVENT_TAG: ret = gst_pad_push_event (a52dec->srcpad, event); break; case GST_EVENT_EOS: gst_a52dec_drain (a52dec); ret = gst_pad_push_event (a52dec->srcpad, event); break; case GST_EVENT_FLUSH_START: ret = gst_pad_push_event (a52dec->srcpad, event); break; case GST_EVENT_FLUSH_STOP: if (a52dec->cache) { gst_buffer_unref (a52dec->cache); a52dec->cache = NULL; } clear_queued (a52dec); gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED); ret = gst_pad_push_event (a52dec->srcpad, event); break; default: ret = gst_pad_push_event (a52dec->srcpad, event); break; } gst_object_unref (a52dec); return ret; } 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_AUDIO_CODEC, "Dolby Digital (AC-3)", GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL); gst_element_found_tags_for_pad (GST_ELEMENT (a52dec), GST_PAD (a52dec->srcpad), taglist); } static GstFlowReturn 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); } /* 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 (a52dec->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 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", flags ? gst_a52dec_channels (flags, NULL) : 6); gst_structure_get_int (structure, "channels", &channels); if (channels <= 6) flags = a52_channels[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, data, &flags, &a52dec->level, a52dec->bias)) { GST_WARNING ("a52_frame error"); a52dec->discont = TRUE; return GST_FLOW_OK; } 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 ("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, a52dec->srcpad)) { GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL)); return GST_FLOW_ERROR; } } 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)) { /* ignore errors but mark a discont */ GST_WARNING ("a52_block error %d", i); a52dec->discont = TRUE; } else { GstFlowReturn ret; /* push on */ ret = gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels, a52dec->samples, a52dec->time); if (ret != GST_FLOW_OK) return ret; } a52dec->time += 256 * GST_SECOND / a52dec->sample_rate; } return GST_FLOW_OK; } static gboolean gst_a52dec_sink_setcaps (GstPad * pad, GstCaps * caps) { GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad)); 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; gst_object_unref (a52dec); return TRUE; } static GstFlowReturn gst_a52dec_chain (GstPad * pad, GstBuffer * buf) { GstA52Dec *a52dec = GST_A52DEC (GST_PAD_PARENT (pad)); GstFlowReturn ret; gint first_access; if (GST_BUFFER_IS_DISCONT (buf)) { GST_LOG_OBJECT (a52dec, "received DISCONT"); gst_a52dec_drain (a52dec); /* clear cache on discont and mark a discont in the element */ if (a52dec->cache) { gst_buffer_unref (a52dec->cache); a52dec->cache = NULL; } a52dec->discont = TRUE; } if (a52dec->dvdmode) { gint size = GST_BUFFER_SIZE (buf); guchar *data = GST_BUFFER_DATA (buf); gint offset; gint 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; ret = gst_a52dec_chain_raw (pad, subbuf); if (ret != 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); ret = gst_a52dec_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); ret = gst_a52dec_chain_raw (pad, subbuf); } } else { ret = gst_a52dec_chain_raw (pad, 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")); 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)); return GST_FLOW_ERROR; } } static GstFlowReturn gst_a52dec_chain_raw (GstPad * pad, GstBuffer * buf) { GstA52Dec *a52dec; guint8 *data; guint size; gint length = 0, flags, sample_rate, bit_rate; GstFlowReturn result = GST_FLOW_OK; a52dec = GST_A52DEC (GST_PAD_PARENT (pad)); if (!a52dec->sent_segment) { GstSegment segment; /* Create a basic segment. Usually, we'll get a new-segment sent by * another element that will know more information (a demuxer). If we're * just looking at a raw AC3 stream, we won't - so we need to send one * here, but we don't know much info, so just send a minimal TIME * new-segment event */ gst_segment_init (&segment, GST_FORMAT_TIME); gst_pad_push_event (a52dec->srcpad, gst_event_new_new_segment (FALSE, segment.rate, segment.format, segment.start, segment.duration, segment.start)); a52dec->sent_segment = TRUE; } /* merge with cache, if any. Also make sure timestamps match */ 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 (flags != a52dec->prev_flags) a52dec->flag_update = TRUE; a52dec->prev_flags = flags; result = gst_a52dec_handle_frame (a52dec, data, length, flags, sample_rate, bit_rate); if (result != GST_FLOW_OK) { 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); return result; } static GstStateChangeReturn gst_a52dec_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; GstA52Dec *a52dec = GST_A52DEC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY:{ GstA52DecClass *klass; klass = GST_A52DEC_CLASS (G_OBJECT_GET_CLASS (a52dec)); a52dec->state = a52_init (klass->a52_cpuflags); break; } case GST_STATE_CHANGE_READY_TO_PAUSED: 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->time = 0; a52dec->sent_segment = FALSE; a52dec->flag_update = TRUE; gst_segment_init (&a52dec->segment, GST_FORMAT_UNDEFINED); 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: a52dec->samples = NULL; if (a52dec->cache) { gst_buffer_unref (a52dec->cache); a52dec->cache = NULL; } clear_queued (a52dec); break; case GST_STATE_CHANGE_READY_TO_NULL: a52_free (a52dec->state); a52dec->state = NULL; break; default: break; } return ret; } 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 plugin_init (GstPlugin * plugin) { /* ensure GstAudioChannelPosition type is registered */ if (!gst_audio_channel_position_get_type ()) return FALSE; if (!gst_element_register (plugin, "a52dec", GST_RANK_SECONDARY, 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_NAME, GST_PACKAGE_ORIGIN);