/* GStreamer * Copyright (C) 2004 Benjamin Otte * * 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-vorbisdec * @see_also: vorbisenc, oggdemux * * This element decodes a Vorbis stream to raw float audio. * Vorbis is a royalty-free * audio codec maintained by the Xiph.org * Foundation. * * * Example pipelines * |[ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink * ]| Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc. * * * Last reviewed on 2006-03-01 (0.10.4) */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "gstvorbisdec.h" #include #include #include #include GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug); #define GST_CAT_DEFAULT vorbisdec_debug static const GstElementDetails vorbis_dec_details = GST_ELEMENT_DETAILS ("Vorbis audio decoder", "Codec/Decoder/Audio", "decode raw vorbis streams to float audio", "Benjamin Otte "); static GstStaticPadTemplate vorbis_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 256 ], " "endianness = (int) BYTE_ORDER, " /* no ifdef in macros, please #ifdef GST_VORBIS_DEC_SEQUENTIAL "layout = \"sequential\", " #endif */ "width = (int) 32") ); static GstStaticPadTemplate vorbis_dec_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-vorbis") ); GST_BOILERPLATE (GstVorbisDec, gst_vorbis_dec, GstElement, GST_TYPE_ELEMENT); static void vorbis_dec_finalize (GObject * object); static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event); static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer); static GstFlowReturn vorbis_dec_chain_forward (GstVorbisDec * vd, gboolean discont, GstBuffer * buffer); static GstStateChangeReturn vorbis_dec_change_state (GstElement * element, GstStateChange transition); static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event); static gboolean vorbis_dec_src_query (GstPad * pad, GstQuery * query); static gboolean vorbis_dec_convert (GstPad * pad, GstFormat src_format, gint64 src_value, GstFormat * dest_format, gint64 * dest_value); static gboolean vorbis_dec_sink_query (GstPad * pad, GstQuery * query); static void gst_vorbis_dec_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GstPadTemplate *src_template, *sink_template; src_template = gst_static_pad_template_get (&vorbis_dec_src_factory); gst_element_class_add_pad_template (element_class, src_template); sink_template = gst_static_pad_template_get (&vorbis_dec_sink_factory); gst_element_class_add_pad_template (element_class, sink_template); gst_element_class_set_details (element_class, &vorbis_dec_details); } static void gst_vorbis_dec_class_init (GstVorbisDecClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); gobject_class->finalize = vorbis_dec_finalize; gstelement_class->change_state = GST_DEBUG_FUNCPTR (vorbis_dec_change_state); } static const GstQueryType * vorbis_get_query_types (GstPad * pad) { static const GstQueryType vorbis_dec_src_query_types[] = { GST_QUERY_POSITION, GST_QUERY_DURATION, GST_QUERY_CONVERT, 0 }; return vorbis_dec_src_query_types; } static void gst_vorbis_dec_init (GstVorbisDec * dec, GstVorbisDecClass * g_class) { dec->sinkpad = gst_pad_new_from_static_template (&vorbis_dec_sink_factory, "sink"); gst_pad_set_event_function (dec->sinkpad, GST_DEBUG_FUNCPTR (vorbis_dec_sink_event)); gst_pad_set_chain_function (dec->sinkpad, GST_DEBUG_FUNCPTR (vorbis_dec_chain)); gst_pad_set_query_function (dec->sinkpad, GST_DEBUG_FUNCPTR (vorbis_dec_sink_query)); gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad); dec->srcpad = gst_pad_new_from_static_template (&vorbis_dec_src_factory, "src"); gst_pad_set_event_function (dec->srcpad, GST_DEBUG_FUNCPTR (vorbis_dec_src_event)); gst_pad_set_query_type_function (dec->srcpad, GST_DEBUG_FUNCPTR (vorbis_get_query_types)); gst_pad_set_query_function (dec->srcpad, GST_DEBUG_FUNCPTR (vorbis_dec_src_query)); gst_pad_use_fixed_caps (dec->srcpad); gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad); dec->queued = NULL; dec->pendingevents = NULL; dec->taglist = NULL; } static void vorbis_dec_finalize (GObject * object) { /* Release any possibly allocated libvorbis data. * _clear functions can safely be called multiple times */ GstVorbisDec *vd = GST_VORBIS_DEC (object); vorbis_block_clear (&vd->vb); vorbis_dsp_clear (&vd->vd); vorbis_comment_clear (&vd->vc); vorbis_info_clear (&vd->vi); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_vorbis_dec_reset (GstVorbisDec * dec) { dec->cur_timestamp = GST_CLOCK_TIME_NONE; dec->prev_timestamp = GST_CLOCK_TIME_NONE; dec->granulepos = -1; dec->discont = TRUE; dec->seqnum = gst_util_seqnum_next (); gst_segment_init (&dec->segment, GST_FORMAT_TIME); g_list_foreach (dec->queued, (GFunc) gst_mini_object_unref, NULL); g_list_free (dec->queued); dec->queued = NULL; g_list_foreach (dec->gather, (GFunc) gst_mini_object_unref, NULL); g_list_free (dec->gather); dec->gather = NULL; g_list_foreach (dec->decode, (GFunc) gst_mini_object_unref, NULL); g_list_free (dec->decode); dec->decode = NULL; g_list_foreach (dec->pendingevents, (GFunc) gst_mini_object_unref, NULL); g_list_free (dec->pendingevents); dec->pendingevents = NULL; if (dec->taglist) gst_tag_list_free (dec->taglist); dec->taglist = NULL; } static gboolean vorbis_dec_convert (GstPad * pad, GstFormat src_format, gint64 src_value, GstFormat * dest_format, gint64 * dest_value) { gboolean res = TRUE; GstVorbisDec *dec; guint64 scale = 1; if (src_format == *dest_format) { *dest_value = src_value; return TRUE; } dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); if (!dec->initialized) goto no_header; if (dec->sinkpad == pad && (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) goto no_format; switch (src_format) { case GST_FORMAT_TIME: switch (*dest_format) { case GST_FORMAT_BYTES: scale = sizeof (float) * dec->vi.channels; case GST_FORMAT_DEFAULT: *dest_value = scale * gst_util_uint64_scale_int (src_value, dec->vi.rate, GST_SECOND); break; default: res = FALSE; } break; case GST_FORMAT_DEFAULT: switch (*dest_format) { case GST_FORMAT_BYTES: *dest_value = src_value * sizeof (float) * dec->vi.channels; break; case GST_FORMAT_TIME: *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, dec->vi.rate); break; default: res = FALSE; } break; case GST_FORMAT_BYTES: switch (*dest_format) { case GST_FORMAT_DEFAULT: *dest_value = src_value / (sizeof (float) * dec->vi.channels); break; case GST_FORMAT_TIME: *dest_value = gst_util_uint64_scale_int (src_value, GST_SECOND, dec->vi.rate * sizeof (float) * dec->vi.channels); break; default: res = FALSE; } break; default: res = FALSE; } done: gst_object_unref (dec); return res; /* ERRORS */ no_header: { GST_DEBUG_OBJECT (dec, "no header packets received"); res = FALSE; goto done; } no_format: { GST_DEBUG_OBJECT (dec, "formats unsupported"); res = FALSE; goto done; } } static gboolean vorbis_dec_src_query (GstPad * pad, GstQuery * query) { GstVorbisDec *dec; gboolean res = FALSE; dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { gint64 granulepos, value; GstFormat my_format, format; gint64 time; /* we start from the last seen granulepos */ granulepos = dec->granulepos; gst_query_parse_position (query, &format, NULL); /* and convert to the final format in two steps with time as the * intermediate step */ my_format = GST_FORMAT_TIME; if (!(res = vorbis_dec_convert (pad, GST_FORMAT_DEFAULT, granulepos, &my_format, &time))) goto error; /* correct for the segment values */ time = gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, time); GST_LOG_OBJECT (dec, "query %p: our time: %" GST_TIME_FORMAT, query, GST_TIME_ARGS (time)); /* and convert to the final format */ if (!(res = vorbis_dec_convert (pad, my_format, time, &format, &value))) goto error; gst_query_set_position (query, format, value); GST_LOG_OBJECT (dec, "query %p: we return %lld (format %u)", query, value, format); break; } case GST_QUERY_DURATION: { GstPad *peer; if (!(peer = gst_pad_get_peer (dec->sinkpad))) { GST_WARNING_OBJECT (dec, "sink pad %" GST_PTR_FORMAT " is not linked", dec->sinkpad); goto error; } res = gst_pad_query (peer, query); gst_object_unref (peer); if (!res) goto error; break; } case GST_QUERY_CONVERT: { GstFormat src_fmt, dest_fmt; gint64 src_val, dest_val; gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); if (!(res = vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val))) goto error; gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); break; } default: res = gst_pad_query_default (pad, query); break; } done: gst_object_unref (dec); return res; /* ERRORS */ error: { GST_WARNING_OBJECT (dec, "error handling query"); goto done; } } static gboolean vorbis_dec_sink_query (GstPad * pad, GstQuery * query) { GstVorbisDec *dec; gboolean res; dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CONVERT: { GstFormat src_fmt, dest_fmt; gint64 src_val, dest_val; gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val); if (!(res = vorbis_dec_convert (pad, src_fmt, src_val, &dest_fmt, &dest_val))) goto error; gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val); break; } default: res = gst_pad_query_default (pad, query); break; } done: gst_object_unref (dec); return res; /* ERRORS */ error: { GST_DEBUG_OBJECT (dec, "error converting value"); goto done; } } static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event) { gboolean res = TRUE; GstVorbisDec *dec; dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: { GstFormat format, tformat; gdouble rate; GstEvent *real_seek; GstSeekFlags flags; GstSeekType cur_type, stop_type; gint64 cur, stop; gint64 tcur, tstop; guint32 seqnum; gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); seqnum = gst_event_get_seqnum (event); gst_event_unref (event); /* we have to ask our peer to seek to time here as we know * nothing about how to generate a granulepos from the src * formats or anything. * * First bring the requested format to time */ tformat = GST_FORMAT_TIME; if (!(res = vorbis_dec_convert (pad, format, cur, &tformat, &tcur))) goto convert_error; if (!(res = vorbis_dec_convert (pad, format, stop, &tformat, &tstop))) goto convert_error; /* then seek with time on the peer */ real_seek = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, cur_type, tcur, stop_type, tstop); gst_event_set_seqnum (real_seek, seqnum); res = gst_pad_push_event (dec->sinkpad, real_seek); break; } default: res = gst_pad_push_event (dec->sinkpad, event); break; } done: gst_object_unref (dec); return res; /* ERRORS */ convert_error: { GST_DEBUG_OBJECT (dec, "cannot convert start/stop for seek"); goto done; } } static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event) { gboolean ret = FALSE; GstVorbisDec *dec; dec = GST_VORBIS_DEC (gst_pad_get_parent (pad)); GST_LOG_OBJECT (dec, "handling event"); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: ret = gst_pad_push_event (dec->srcpad, event); break; case GST_EVENT_FLUSH_START: ret = gst_pad_push_event (dec->srcpad, event); break; case GST_EVENT_FLUSH_STOP: /* here we must clean any state in the decoder */ #ifdef HAVE_VORBIS_SYNTHESIS_RESTART vorbis_synthesis_restart (&dec->vd); #endif gst_vorbis_dec_reset (dec); ret = gst_pad_push_event (dec->srcpad, event); break; case GST_EVENT_NEWSEGMENT: { GstFormat format; gdouble rate, arate; gint64 start, stop, time; gboolean update; gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, &start, &stop, &time); /* we need time for now */ if (format != GST_FORMAT_TIME) goto newseg_wrong_format; GST_DEBUG_OBJECT (dec, "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT, update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time)); /* now configure the values */ gst_segment_set_newsegment_full (&dec->segment, update, rate, arate, format, start, stop, time); dec->seqnum = gst_event_get_seqnum (event); if (dec->initialized) /* and forward */ ret = gst_pad_push_event (dec->srcpad, event); else { /* store it to send once we're initialized */ dec->pendingevents = g_list_append (dec->pendingevents, event); ret = TRUE; } break; } case GST_EVENT_TAG: { if (dec->initialized) /* and forward */ ret = gst_pad_push_event (dec->srcpad, event); else { /* store it to send once we're initialized */ dec->pendingevents = g_list_append (dec->pendingevents, event); ret = TRUE; } break; } default: ret = gst_pad_push_event (dec->srcpad, event); break; } done: gst_object_unref (dec); return ret; /* ERRORS */ newseg_wrong_format: { GST_DEBUG_OBJECT (dec, "received non TIME newsegment"); goto done; } } static GstFlowReturn vorbis_handle_identification_packet (GstVorbisDec * vd) { GstCaps *caps; const GstAudioChannelPosition *pos = NULL; switch (vd->vi.channels) { case 1: case 2: /* nothing */ break; case 3:{ static const GstAudioChannelPosition pos3[] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT }; pos = pos3; break; } case 4:{ static const GstAudioChannelPosition pos4[] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT }; pos = pos4; break; } case 5:{ static const GstAudioChannelPosition pos5[] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT }; pos = pos5; break; } case 6:{ static const GstAudioChannelPosition pos6[] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_LFE }; pos = pos6; break; } /* FIXME: for >6 channels the layout is not defined by the Vorbis * spec. These are the gstreamer "defaults" for 7/8 channels and * NONE layouts for more channels */ case 7:{ static const GstAudioChannelPosition pos7[] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER, }; pos = pos7; /* fallthrough */ } case 8:{ static const GstAudioChannelPosition pos8[] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, GST_AUDIO_CHANNEL_POSITION_LFE, GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT, }; pos = pos8; /* fallthrough */ } default:{ gint i; GstAudioChannelPosition *posn = g_new (GstAudioChannelPosition, vd->vi.channels); GST_ELEMENT_WARNING (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("Using NONE channel layout for more than 8 channels")); for (i = 0; i < vd->vi.channels; i++) posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE; pos = posn; } } caps = gst_caps_new_simple ("audio/x-raw-float", "rate", G_TYPE_INT, vd->vi.rate, "channels", G_TYPE_INT, vd->vi.channels, "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); if (pos) { gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); } if (vd->vi.channels > 8) { g_free ((GstAudioChannelPosition *) pos); } gst_pad_set_caps (vd->srcpad, caps); gst_caps_unref (caps); return GST_FLOW_OK; } static GstFlowReturn vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet) { guint bitrate = 0; gchar *encoder = NULL; GstTagList *list, *old_list; GstBuffer *buf; GST_DEBUG_OBJECT (vd, "parsing comment packet"); buf = gst_buffer_new (); GST_BUFFER_DATA (buf) = packet->packet; GST_BUFFER_SIZE (buf) = packet->bytes; list = gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7, &encoder); old_list = vd->taglist; vd->taglist = gst_tag_list_merge (vd->taglist, list, GST_TAG_MERGE_REPLACE); if (old_list) gst_tag_list_free (old_list); gst_tag_list_free (list); gst_buffer_unref (buf); if (!vd->taglist) { GST_ERROR_OBJECT (vd, "couldn't decode comments"); vd->taglist = gst_tag_list_new (); } if (encoder) { gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE, GST_TAG_ENCODER, encoder, NULL); g_free (encoder); } gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE, GST_TAG_ENCODER_VERSION, vd->vi.version, GST_TAG_AUDIO_CODEC, "Vorbis", NULL); if (vd->vi.bitrate_nominal > 0 && vd->vi.bitrate_nominal <= 0x7FFFFFFF) { gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE, GST_TAG_NOMINAL_BITRATE, (guint) vd->vi.bitrate_nominal, NULL); bitrate = vd->vi.bitrate_nominal; } if (vd->vi.bitrate_upper > 0 && vd->vi.bitrate_upper <= 0x7FFFFFFF) { gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, NULL); if (!bitrate) bitrate = vd->vi.bitrate_upper; } if (vd->vi.bitrate_lower > 0 && vd->vi.bitrate_lower <= 0x7FFFFFFF) { gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL); if (!bitrate) bitrate = vd->vi.bitrate_lower; } if (bitrate) { gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, (guint) bitrate, NULL); } if (vd->initialized) { gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad, vd->taglist); vd->taglist = NULL; } else { /* Only post them as messages for the time being. * * They will be pushed on the pad once the decoder is initialized */ gst_element_post_message (GST_ELEMENT_CAST (vd), gst_message_new_tag (GST_OBJECT (vd), gst_tag_list_copy (vd->taglist))); } return GST_FLOW_OK; } static GstFlowReturn vorbis_handle_type_packet (GstVorbisDec * vd) { GList *walk; gint res; g_assert (vd->initialized == FALSE); if (G_UNLIKELY ((res = vorbis_synthesis_init (&vd->vd, &vd->vi)))) goto synthesis_init_error; if (G_UNLIKELY ((res = vorbis_block_init (&vd->vd, &vd->vb)))) goto block_init_error; vd->initialized = TRUE; if (vd->pendingevents) { for (walk = vd->pendingevents; walk; walk = g_list_next (walk)) gst_pad_push_event (vd->srcpad, GST_EVENT_CAST (walk->data)); g_list_free (vd->pendingevents); vd->pendingevents = NULL; } if (vd->taglist) { /* The tags have already been sent on the bus as messages. */ gst_pad_push_event (vd->srcpad, gst_event_new_tag (vd->taglist)); vd->taglist = NULL; } return GST_FLOW_OK; /* ERRORS */ synthesis_init_error: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("couldn't initialize synthesis (%d)", res)); return GST_FLOW_ERROR; } block_init_error: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("couldn't initialize block (%d)", res)); return GST_FLOW_ERROR; } } static GstFlowReturn vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet) { GstFlowReturn res; gint ret; GST_DEBUG_OBJECT (vd, "parsing header packet"); /* Packetno = 0 if the first byte is exactly 0x01 */ packet->b_o_s = (packet->packet[0] == 0x1) ? 1 : 0; if ((ret = vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet))) goto header_read_error; switch (packet->packet[0]) { case 0x01: res = vorbis_handle_identification_packet (vd); break; case 0x03: res = vorbis_handle_comment_packet (vd, packet); break; case 0x05: res = vorbis_handle_type_packet (vd); break; default: /* ignore */ g_warning ("unknown vorbis header packet found"); res = GST_FLOW_OK; break; } return res; /* ERRORS */ header_read_error: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("couldn't read header packet (%d)", ret)); return GST_FLOW_ERROR; } } /* These samples can be outside of the float -1.0 -- 1.0 range, this * is allowed, downstream elements are supposed to clip */ static void copy_samples (float *out, float **in, guint samples, gint channels) { gint i, j; #ifdef GST_VORBIS_DEC_SEQUENTIAL for (i = 0; i < channels; i++) { memcpy (out, in[i], samples * sizeof (float)); out += samples; } #else for (j = 0; j < samples; j++) { for (i = 0; i < channels; i++) { *out++ = in[i][j]; } } #endif } static GstFlowReturn vorbis_dec_push_forward (GstVorbisDec * dec, GstBuffer * buf) { GstFlowReturn result; gint64 outoffset, origoffset; origoffset = GST_BUFFER_OFFSET (buf); again: outoffset = origoffset; if (outoffset == -1) { dec->queued = g_list_append (dec->queued, buf); GST_DEBUG_OBJECT (dec, "queued buffer"); result = GST_FLOW_OK; } else { if (G_UNLIKELY (dec->queued)) { guint size; GstClockTime ts; GList *walk; GST_DEBUG_OBJECT (dec, "first buffer with offset %lld", outoffset); ts = gst_util_uint64_scale_int (outoffset, GST_SECOND, dec->vi.rate); size = g_list_length (dec->queued); /* we walk the queued up list in reverse, and set the buffer fields * calculating backwards */ for (walk = g_list_last (dec->queued); walk; walk = g_list_previous (walk)) { GstBuffer *buffer = GST_BUFFER (walk->data); guint offset; offset = GST_BUFFER_SIZE (buffer) / (sizeof (float) * dec->vi.channels); if (outoffset >= offset) outoffset -= offset; else { /* we can't go below 0, this means this first offset was at the eos * page and we need to clip to it instead */ GST_DEBUG_OBJECT (dec, "clipping %" G_GINT64_FORMAT, offset - outoffset); origoffset += (offset - outoffset); goto again; } GST_BUFFER_OFFSET (buffer) = outoffset; GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale_int (outoffset, GST_SECOND, dec->vi.rate); GST_BUFFER_DURATION (buffer) = GST_CLOCK_DIFF (GST_BUFFER_TIMESTAMP (buffer), ts); ts = GST_BUFFER_TIMESTAMP (buffer); GST_DEBUG_OBJECT (dec, "patch buffer %u, offset %" G_GUINT64_FORMAT ", timestamp %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT, size, outoffset, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), GST_TIME_ARGS (GST_BUFFER_DURATION (buffer))); size--; } for (walk = dec->queued; walk; walk = g_list_next (walk)) { GstBuffer *buffer = GST_BUFFER (walk->data); /* clips to the configured segment, or returns NULL with buffer * unreffed when the input buffer is completely outside the segment */ if (!(buffer = gst_audio_buffer_clip (buffer, &dec->segment, dec->vi.rate, dec->vi.channels * sizeof (float)))) continue; if (dec->discont) { GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT); dec->discont = FALSE; } /* ignore the result */ gst_pad_push (dec->srcpad, buffer); } g_list_free (dec->queued); dec->queued = NULL; } /* clip */ if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate, dec->vi.channels * sizeof (float)))) return GST_FLOW_OK; if (dec->discont) { GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); dec->discont = FALSE; } result = gst_pad_push (dec->srcpad, buf); } return result; } static GstFlowReturn vorbis_dec_push_reverse (GstVorbisDec * dec, GstBuffer * buf) { GstFlowReturn result = GST_FLOW_OK; dec->queued = g_list_prepend (dec->queued, buf); return result; } static GstFlowReturn vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet) { float **pcm; guint sample_count; GstBuffer *out; GstFlowReturn result; GstClockTime timestamp = GST_CLOCK_TIME_NONE, nextts; gint size; if (!vd->initialized) goto not_initialized; /* FIXME, we should queue undecoded packets here until we get * a timestamp, then we reverse timestamp the queued packets and * clip them, then we decode only the ones we want and don't * keep decoded data in memory. * Ideally, of course, the demuxer gives us a valid timestamp on * the first packet. */ /* normal data packet */ /* FIXME, we can skip decoding if the packet is outside of the * segment, this is however not very trivial as we need a previous * packet to decode the current one so we must be carefull not to * throw away too much. For now we decode everything and clip right * before pushing data. */ if (G_UNLIKELY (vorbis_synthesis (&vd->vb, packet))) goto could_not_read; if (G_UNLIKELY (vorbis_synthesis_blockin (&vd->vd, &vd->vb) < 0)) goto not_accepted; /* assume all goes well here */ result = GST_FLOW_OK; /* count samples ready for reading */ if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0) goto done; GST_LOG_OBJECT (vd, "%d samples ready for reading", sample_count); size = sample_count * vd->vi.channels * sizeof (float); /* alloc buffer for it */ result = gst_pad_alloc_buffer_and_set_caps (vd->srcpad, GST_BUFFER_OFFSET_NONE, size, GST_PAD_CAPS (vd->srcpad), &out); if (G_UNLIKELY (result != GST_FLOW_OK)) goto done; /* get samples ready for reading now, should be sample_count */ if (G_UNLIKELY ((vorbis_synthesis_pcmout (&vd->vd, &pcm)) != sample_count)) goto wrong_samples; /* copy samples in buffer */ copy_samples ((float *) GST_BUFFER_DATA (out), pcm, sample_count, vd->vi.channels); GST_BUFFER_SIZE (out) = size; /* this should not overflow */ GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate; if (packet->granulepos != -1) vd->granulepos = packet->granulepos - sample_count; if (vd->cur_timestamp != GST_CLOCK_TIME_NONE) { /* we have incoming timestamps */ timestamp = vd->cur_timestamp; GST_DEBUG_OBJECT (vd, "cur_timestamp: %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT " = %" GST_TIME_FORMAT, GST_TIME_ARGS (vd->cur_timestamp), GST_TIME_ARGS (GST_BUFFER_DURATION (out)), GST_TIME_ARGS (vd->cur_timestamp + GST_BUFFER_DURATION (out))); vd->cur_timestamp += GST_BUFFER_DURATION (out); GST_BUFFER_OFFSET (out) = GST_CLOCK_TIME_TO_FRAMES (timestamp, vd->vi.rate); GST_BUFFER_OFFSET_END (out) = GST_BUFFER_OFFSET (out) + sample_count; } else { /* we have incoming granulepos */ GST_BUFFER_OFFSET (out) = vd->granulepos; if (vd->granulepos != -1) { GST_DEBUG_OBJECT (vd, "granulepos: %" G_GINT64_FORMAT, vd->granulepos); GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count; timestamp = gst_util_uint64_scale_int (vd->granulepos, GST_SECOND, vd->vi.rate); nextts = gst_util_uint64_scale_int (vd->granulepos + sample_count, GST_SECOND, vd->vi.rate); GST_DEBUG_OBJECT (vd, "corresponding timestamp %" GST_TIME_FORMAT, GST_TIME_ARGS (timestamp)); /* calculate a nano-second accurate duration */ GST_BUFFER_DURATION (out) = GST_CLOCK_DIFF (timestamp, nextts); GST_DEBUG_OBJECT (vd, "set duration %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_DURATION (out))); } else { timestamp = -1; } } GST_BUFFER_TIMESTAMP (out) = timestamp; if (vd->granulepos != -1) vd->granulepos += sample_count; if (vd->segment.rate >= 0.0) result = vorbis_dec_push_forward (vd, out); else result = vorbis_dec_push_reverse (vd, out); done: vorbis_synthesis_read (&vd->vd, sample_count); GST_DEBUG_OBJECT (vd, "decoded %ld bytes into %d samples, ts %" GST_TIME_FORMAT, packet->bytes, sample_count, GST_TIME_ARGS (timestamp)); /* granulepos is the last sample in the packet */ if (packet->granulepos != -1) vd->granulepos = packet->granulepos; return result; /* ERRORS */ not_initialized: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("no header sent yet")); return GST_FLOW_ERROR; } could_not_read: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("couldn't read data packet")); return GST_FLOW_ERROR; } not_accepted: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("vorbis decoder did not accept data packet")); return GST_FLOW_ERROR; } wrong_samples: { gst_buffer_unref (out); GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("vorbis decoder reported wrong number of samples")); return GST_FLOW_ERROR; } } static GstFlowReturn vorbis_dec_decode_buffer (GstVorbisDec * vd, GstBuffer * buffer) { ogg_packet packet; GstFlowReturn result = GST_FLOW_OK; GstClockTime timestamp; guint64 offset_end; timestamp = GST_BUFFER_TIMESTAMP (buffer); offset_end = GST_BUFFER_OFFSET_END (buffer); /* only ogg has granulepos, demuxers of other container formats * might provide us with timestamps instead (e.g. matroskademux) */ if (offset_end == GST_BUFFER_OFFSET_NONE && timestamp != GST_CLOCK_TIME_NONE) { /* we might get multiple consecutive buffers with the same timestamp */ if (timestamp != vd->prev_timestamp) { vd->cur_timestamp = timestamp; vd->prev_timestamp = timestamp; } } else { vd->cur_timestamp = GST_CLOCK_TIME_NONE; vd->prev_timestamp = GST_CLOCK_TIME_NONE; } /* make ogg_packet out of the buffer */ packet.packet = GST_BUFFER_DATA (buffer); packet.bytes = GST_BUFFER_SIZE (buffer); packet.granulepos = offset_end; packet.packetno = 0; /* we don't care */ /* * FIXME. Is there anyway to know that this is the last packet and * set e_o_s?? * Yes there is, keep one packet at all times and only push out when * you receive a new one. Implement this. */ packet.e_o_s = 0; /* error out on empty header packets, but just skip empty data packets */ if (G_UNLIKELY (packet.bytes == 0)) { if (vd->initialized) goto empty_buffer; else goto empty_header; } GST_DEBUG_OBJECT (vd, "vorbis granule: %" G_GINT64_FORMAT, (gint64) packet.granulepos); /* switch depending on packet type */ if (packet.packet[0] & 1) { if (vd->initialized) { GST_WARNING_OBJECT (vd, "Already initialized, so ignoring header packet"); goto done; } result = vorbis_handle_header_packet (vd, &packet); } else { result = vorbis_handle_data_packet (vd, &packet); } done: return result; empty_buffer: { /* don't error out here, just ignore the buffer, it's invalid for vorbis * but not fatal. */ GST_WARNING_OBJECT (vd, "empty buffer received, ignoring"); if (packet.granulepos != -1) vd->granulepos = packet.granulepos; result = GST_FLOW_OK; goto done; } /* ERRORS */ empty_header: { GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty header received")); result = GST_FLOW_ERROR; vd->discont = TRUE; goto done; } } /* * Input: * Buffer decoding order: 7 8 9 4 5 6 3 1 2 EOS * Discont flag: D D D D * * - Each Discont marks a discont in the decoding order. * * for vorbis, each buffer is a keyframe when we have the previous * buffer. This means that to decode buffer 7, we need buffer 6, which * arrives out of order. * * we first gather buffers in the gather queue until we get a DISCONT. We * prepend each incomming buffer so that they are in reversed order. * * gather queue: 9 8 7 * decode queue: * output queue: * * When a DISCONT is received (buffer 4), we move the gather queue to the * decode queue. This is simply done be taking the head of the gather queue * and prepending it to the decode queue. This yields: * * gather queue: * decode queue: 7 8 9 * output queue: * * Then we decode each buffer in the decode queue in order and put the output * buffer in the output queue. The first buffer (7) will not produce any output * because it needs the previous buffer (6) which did not arrive yet. This * yields: * * gather queue: * decode queue: 7 8 9 * output queue: 9 8 * * Then we remove the consumed buffers from the decode queue. Buffer 7 is not * completely consumed, we need to keep it around for when we receive buffer * 6. This yields: * * gather queue: * decode queue: 7 * output queue: 9 8 * * Then we accumulate more buffers: * * gather queue: 6 5 4 * decode queue: 7 * output queue: * * prepending to the decode queue on DISCONT yields: * * gather queue: * decode queue: 4 5 6 7 * output queue: * * after decoding and keeping buffer 4: * * gather queue: * decode queue: 4 * output queue: 7 6 5 * * Etc.. */ static GstFlowReturn vorbis_dec_flush_decode (GstVorbisDec * dec) { GstFlowReturn res = GST_FLOW_OK; GList *walk; walk = dec->decode; GST_DEBUG_OBJECT (dec, "flushing buffers to decoder"); while (walk) { GList *next; GstBuffer *buf = GST_BUFFER_CAST (walk->data); GST_DEBUG_OBJECT (dec, "decoding buffer %p, ts %" GST_TIME_FORMAT, buf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); next = g_list_next (walk); /* decode buffer, prepend to output queue */ res = vorbis_dec_decode_buffer (dec, buf); /* if we generated output, we can discard the buffer, else we * keep it in the queue */ if (dec->queued) { GST_DEBUG_OBJECT (dec, "decoded buffer to %p", dec->queued->data); dec->decode = g_list_delete_link (dec->decode, walk); gst_buffer_unref (buf); } else { GST_DEBUG_OBJECT (dec, "buffer did not decode, keeping"); } walk = next; } if (dec->granulepos != -1) { GstClockTime endts; endts = gst_util_uint64_scale_int (dec->granulepos, GST_SECOND, dec->vi.rate); GST_DEBUG_OBJECT (dec, "we have granulepos %" G_GUINT64_FORMAT ", ts %" GST_TIME_FORMAT, dec->granulepos, GST_TIME_ARGS (endts)); while (dec->queued) { GstBuffer *buf; guint sample_count; buf = GST_BUFFER_CAST (dec->queued->data); sample_count = GST_BUFFER_SIZE (buf) / (dec->vi.channels * sizeof (float)); GST_BUFFER_OFFSET_END (buf) = dec->granulepos; endts = gst_util_uint64_scale_int (dec->granulepos, GST_SECOND, dec->vi.rate); dec->granulepos -= sample_count; GST_BUFFER_OFFSET (buf) = dec->granulepos; GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (dec->granulepos, GST_SECOND, dec->vi.rate); GST_BUFFER_DURATION (buf) = endts - GST_BUFFER_TIMESTAMP (buf); /* clip, this will unref the buffer in case of clipping */ if (!(buf = gst_audio_buffer_clip (buf, &dec->segment, dec->vi.rate, dec->vi.channels * sizeof (float)))) { GST_DEBUG_OBJECT (dec, "clipped buffer %p", buf); goto next; } if (dec->discont) { GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); dec->discont = FALSE; } GST_DEBUG_OBJECT (dec, "pushing buffer %p, samples %u, " "ts %" GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT, buf, sample_count, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); res = gst_pad_push (dec->srcpad, buf); next: dec->queued = g_list_delete_link (dec->queued, dec->queued); } } else { GST_DEBUG_OBJECT (dec, "we don't have a granulepos yet, delayed push"); } return res; } static GstFlowReturn vorbis_dec_chain_reverse (GstVorbisDec * vd, gboolean discont, GstBuffer * buf) { GstFlowReturn result = GST_FLOW_OK; /* if we have a discont, move buffers to the decode list */ if (G_UNLIKELY (discont)) { GST_DEBUG_OBJECT (vd, "received discont"); while (vd->gather) { GstBuffer *gbuf; gbuf = GST_BUFFER_CAST (vd->gather->data); /* remove from the gather list */ vd->gather = g_list_delete_link (vd->gather, vd->gather); /* copy to decode queue */ vd->decode = g_list_prepend (vd->decode, gbuf); } /* flush and decode the decode queue */ result = vorbis_dec_flush_decode (vd); } GST_DEBUG_OBJECT (vd, "gathering buffer %p, size %u", buf, GST_BUFFER_SIZE (buf)); /* add buffer to gather queue */ vd->gather = g_list_prepend (vd->gather, buf); return result; } static GstFlowReturn vorbis_dec_chain_forward (GstVorbisDec * vd, gboolean discont, GstBuffer * buffer) { GstFlowReturn result; result = vorbis_dec_decode_buffer (vd, buffer); gst_buffer_unref (buffer); return result; } static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer) { GstVorbisDec *vd; GstFlowReturn result = GST_FLOW_OK; gboolean discont; vd = GST_VORBIS_DEC (gst_pad_get_parent (pad)); discont = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT); /* resync on DISCONT */ if (G_UNLIKELY (discont)) { GST_DEBUG_OBJECT (vd, "received DISCONT buffer"); vd->granulepos = -1; vd->cur_timestamp = GST_CLOCK_TIME_NONE; vd->prev_timestamp = GST_CLOCK_TIME_NONE; #ifdef HAVE_VORBIS_SYNTHESIS_RESTART vorbis_synthesis_restart (&vd->vd); #endif vd->discont = TRUE; } if (vd->segment.rate >= 0.0) result = vorbis_dec_chain_forward (vd, discont, buffer); else result = vorbis_dec_chain_reverse (vd, discont, buffer); gst_object_unref (vd); return result; } static GstStateChangeReturn vorbis_dec_change_state (GstElement * element, GstStateChange transition) { GstVorbisDec *vd = GST_VORBIS_DEC (element); GstStateChangeReturn res; switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: vorbis_info_init (&vd->vi); vorbis_comment_init (&vd->vc); vd->initialized = FALSE; gst_vorbis_dec_reset (vd); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; default: break; } res = parent_class->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_DEBUG_OBJECT (vd, "PAUSED -> READY, clearing vorbis structures"); vd->initialized = FALSE; vorbis_block_clear (&vd->vb); vorbis_dsp_clear (&vd->vd); vorbis_comment_clear (&vd->vc); vorbis_info_clear (&vd->vi); gst_vorbis_dec_reset (vd); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return res; }