/* 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. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "vorbisdec.h" #include #include #include #include GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug); #define GST_CAT_DEFAULT vorbisdec_debug static GstElementDetails vorbis_dec_details = { "VorbisDec", "Codec/Decoder/Audio", "decode raw vorbis streams to float audio", "Benjamin Otte ", }; /* Filter signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { ARG_0 }; 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) [ 8000, 50000 ], " "channels = (int) [ 1, 6 ], " "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 vorbisdec_finalize (GObject * object); static gboolean vorbis_dec_sink_event (GstPad * pad, GstEvent * event); static GstFlowReturn vorbis_dec_chain (GstPad * pad, GstBuffer * buffer); static GstStateChangeReturn vorbis_dec_change_state (GstElement * element, GstStateChange transition); #if 0 static const GstFormat *vorbis_dec_get_formats (GstPad * pad); #endif 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); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&vorbis_dec_src_factory)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&vorbis_dec_sink_factory)); 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 = vorbisdec_finalize; gstelement_class->change_state = vorbis_dec_change_state; } #if 0 static const GstFormat * vorbis_dec_get_formats (GstPad * pad) { static GstFormat src_formats[] = { GST_FORMAT_BYTES, GST_FORMAT_DEFAULT, /* samples in the audio case */ GST_FORMAT_TIME, 0 }; static GstFormat sink_formats[] = { /*GST_FORMAT_BYTES, */ GST_FORMAT_TIME, GST_FORMAT_DEFAULT, /* granulepos or samples */ 0 }; return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats); } #endif #if 0 static const GstEventMask * vorbis_get_event_masks (GstPad * pad) { static const GstEventMask vorbis_dec_src_event_masks[] = { {GST_EVENT_SEEK, GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH}, {0,} }; return vorbis_dec_src_event_masks; } #endif static const GstQueryType * vorbis_get_query_types (GstPad * pad) { static const GstQueryType vorbis_dec_src_query_types[] = { GST_QUERY_POSITION, 0 }; return vorbis_dec_src_query_types; } static void gst_vorbis_dec_init (GstVorbisDec * dec, GstVorbisDecClass * g_class) { dec->sinkpad = gst_pad_new_from_template (gst_static_pad_template_get (&vorbis_dec_sink_factory), "sink"); gst_pad_set_event_function (dec->sinkpad, vorbis_dec_sink_event); gst_pad_set_chain_function (dec->sinkpad, vorbis_dec_chain); gst_pad_set_query_function (dec->sinkpad, vorbis_dec_sink_query); gst_element_add_pad (GST_ELEMENT (dec), dec->sinkpad); dec->srcpad = gst_pad_new_from_template (gst_static_pad_template_get (&vorbis_dec_src_factory), "src"); gst_pad_set_event_function (dec->srcpad, vorbis_dec_src_event); gst_pad_set_query_type_function (dec->srcpad, vorbis_get_query_types); gst_pad_set_query_function (dec->srcpad, vorbis_dec_src_query); gst_pad_use_fixed_caps (dec->srcpad); gst_element_add_pad (GST_ELEMENT (dec), dec->srcpad); dec->queued = NULL; } static void vorbisdec_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 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; dec = GST_VORBIS_DEC (GST_PAD_PARENT (pad)); if (dec->packetno < 1) return FALSE; if (src_format == *dest_format) { *dest_value = src_value; return TRUE; } if (dec->sinkpad == pad && (src_format == GST_FORMAT_BYTES || *dest_format == GST_FORMAT_BYTES)) return FALSE; 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 * (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 (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 (src_value, GST_SECOND, dec->vi.rate * sizeof (float) * dec->vi.channels); break; default: res = FALSE; } break; default: res = FALSE; } return res; } static gboolean vorbis_dec_src_query (GstPad * pad, GstQuery * query) { gint64 granulepos; GstVorbisDec *dec; gboolean res; dec = GST_VORBIS_DEC (GST_PAD_PARENT (pad)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_POSITION: { GstFormat format; gint64 value; granulepos = dec->granulepos; gst_query_parse_position (query, &format, NULL); /* and convert to the final format */ if (!(res = vorbis_dec_convert (pad, GST_FORMAT_DEFAULT, granulepos, &format, &value))) goto error; value = (value - dec->segment_start) + dec->segment_time; gst_query_set_position (query, format, value); GST_LOG_OBJECT (dec, "query %u: peer returned granulepos: %llu - we return %llu (format %u)", query, granulepos, value, format); break; } case GST_QUERY_DURATION: { /* query peer for total length */ if (!(res = gst_pad_query (GST_PAD_PEER (dec->sinkpad), query))) 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 = FALSE; break; } return res; error: { GST_WARNING_OBJECT (dec, "error handling query"); return res; } } static gboolean vorbis_dec_sink_query (GstPad * pad, GstQuery * query) { GstVorbisDec *dec; gboolean res; dec = GST_VORBIS_DEC (GST_PAD_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 = FALSE; break; } error: return res; } static gboolean vorbis_dec_src_event (GstPad * pad, GstEvent * event) { gboolean res = TRUE; GstVorbisDec *dec = GST_VORBIS_DEC (GST_PAD_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; gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); /* 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 error; if (!(res = vorbis_dec_convert (pad, format, stop, &tformat, &tstop))) goto 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); res = gst_pad_push_event (dec->sinkpad, real_seek); gst_event_unref (event); break; } default: res = gst_pad_event_default (pad, event); break; } return res; error: gst_event_unref (event); return res; } 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: GST_STREAM_LOCK (pad); ret = gst_pad_push_event (dec->srcpad, event); GST_STREAM_UNLOCK (pad); break; case GST_EVENT_NEWSEGMENT: { GstFormat format; gdouble rate; gint64 start, stop, time; gboolean update; GST_STREAM_LOCK (pad); gst_event_parse_newsegment (event, &update, &rate, &format, &start, &stop, &time); if (format != GST_FORMAT_TIME) goto newseg_wrong_format; if (rate <= 0.0) goto newseg_wrong_rate; /* now copy over the values */ dec->segment_rate = rate; dec->segment_start = start; dec->segment_stop = stop; dec->segment_time = time; dec->granulepos = -1; dec->cur_timestamp = GST_CLOCK_TIME_NONE; dec->prev_timestamp = GST_CLOCK_TIME_NONE; #ifdef HAVE_VORBIS_SYNTHESIS_RESTART vorbis_synthesis_restart (&dec->vd); #endif ret = gst_pad_push_event (dec->srcpad, event); GST_STREAM_UNLOCK (pad); break; } default: ret = gst_pad_push_event (dec->srcpad, event); break; } done: gst_object_unref (dec); return ret; /* ERRORS */ newseg_wrong_format: { GST_STREAM_UNLOCK (pad); GST_DEBUG ("received non TIME newsegment"); goto done; } newseg_wrong_rate: { GST_STREAM_UNLOCK (pad); GST_DEBUG ("negative rates not supported yet"); goto done; } } static GstFlowReturn vorbis_handle_identification_packet (GstVorbisDec * vd) { GstCaps *caps; const GstAudioChannelPosition *pos = NULL; 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); switch (vd->vi.channels) { case 1: case 2: /* nothing */ break; case 3:{ static 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 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 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 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; } default: goto channel_count_error; } if (pos) { gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos); } gst_pad_set_caps (vd->srcpad, caps); gst_caps_unref (caps); return GST_FLOW_OK; /* ERROR */ channel_count_error: { gst_caps_unref (caps); GST_ELEMENT_ERROR (vd, STREAM, NOT_IMPLEMENTED, (NULL), ("Unsupported channel count %d", vd->vi.channels)); return GST_FLOW_ERROR; } } static GstFlowReturn vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet) { guint bitrate = 0; gchar *encoder = NULL; GstMessage *message; GstTagList *list; GstBuffer *buf; GST_DEBUG_OBJECT (vd, "parsing comment packet"); buf = gst_buffer_new_and_alloc (packet->bytes); GST_BUFFER_DATA (buf) = packet->packet; list = gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7, &encoder); gst_buffer_unref (buf); if (!list) { GST_ERROR_OBJECT (vd, "couldn't decode comments"); list = gst_tag_list_new (); } if (encoder) { gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_ENCODER, encoder, NULL); g_free (encoder); } gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_ENCODER_VERSION, vd->vi.version, GST_TAG_AUDIO_CODEC, "Vorbis", NULL); if (vd->vi.bitrate_nominal > 0) { gst_tag_list_add (list, 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) { gst_tag_list_add (list, 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) { gst_tag_list_add (list, 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 (list, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, (guint) bitrate, NULL); } message = gst_message_new_tag ((GstObject *) vd, list); gst_element_post_message (GST_ELEMENT (vd), message); return GST_FLOW_OK; } static GstFlowReturn vorbis_handle_type_packet (GstVorbisDec * vd) { g_assert (vd->initialized == FALSE); vorbis_synthesis_init (&vd->vd, &vd->vi); vorbis_block_init (&vd->vd, &vd->vb); vd->initialized = TRUE; return GST_FLOW_OK; } static GstFlowReturn vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet) { GstFlowReturn res; 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 (vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet)) goto header_read_error; /* FIXME: we should probably double-check if packet[0] is 1/3/5 for each * of these */ switch (packet->packetno) { case 0: res = vorbis_handle_identification_packet (vd); break; case 1: res = vorbis_handle_comment_packet (vd, packet); break; case 2: res = vorbis_handle_type_packet (vd); break; default: /* ignore */ 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")); return GST_FLOW_ERROR; } } 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 (GstVorbisDec * dec, GstBuffer * buf) { GstFlowReturn result; gint64 outoffset = GST_BUFFER_OFFSET (buf); if (outoffset == -1) { dec->queued = g_list_append (dec->queued, buf); GST_DEBUG_OBJECT (dec, "queued buffer"); result = GST_FLOW_OK; } else { if (dec->queued) { gint64 size; GList *walk; GST_DEBUG_OBJECT (dec, "first buffer with offset %lld", outoffset); size = g_list_length (dec->queued); for (walk = g_list_last (dec->queued); walk; walk = g_list_previous (walk)) { GstBuffer *buffer = GST_BUFFER (walk->data); outoffset -= GST_BUFFER_SIZE (buffer) / (sizeof (float) * dec->vi.channels); GST_BUFFER_OFFSET (buffer) = outoffset; GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (outoffset, GST_SECOND, dec->vi.rate); GST_DEBUG_OBJECT (dec, "patch buffer %" G_GUINT64_FORMAT " offset %" G_GUINT64_FORMAT, size, outoffset); size--; } for (walk = dec->queued; walk; walk = g_list_next (walk)) { GstBuffer *buffer = GST_BUFFER (walk->data); /* ignore the result */ gst_pad_push (dec->srcpad, buffer); } g_list_free (dec->queued); dec->queued = NULL; } result = gst_pad_push (dec->srcpad, buf); } return result; } static GstFlowReturn vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet) { float **pcm; guint sample_count; GstBuffer *out; GstFlowReturn result; if (!vd->initialized) goto not_initialized; /* normal data packet */ if (vorbis_synthesis (&vd->vb, packet)) goto could_not_read; if (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; /* alloc buffer for it */ result = gst_pad_alloc_buffer (vd->srcpad, GST_BUFFER_OFFSET_NONE, sample_count * vd->vi.channels * sizeof (float), GST_PAD_CAPS (vd->srcpad), &out); if (result != GST_FLOW_OK) goto done; /* get samples ready for reading now, should be sample_count */ if ((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_OFFSET (out) = vd->granulepos; if (vd->granulepos != -1) { GST_BUFFER_OFFSET_END (out) = vd->granulepos + sample_count; GST_BUFFER_TIMESTAMP (out) = gst_util_uint64_scale (vd->granulepos, GST_SECOND, vd->vi.rate); } else { GST_BUFFER_TIMESTAMP (out) = -1; } GST_BUFFER_DURATION (out) = sample_count * GST_SECOND / vd->vi.rate; if (vd->cur_timestamp != GST_CLOCK_TIME_NONE) { GST_BUFFER_TIMESTAMP (out) = vd->cur_timestamp; GST_DEBUG ("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 (vd->cur_timestamp, vd->vi.rate); GST_BUFFER_OFFSET_END (out) = GST_BUFFER_OFFSET (out) + sample_count; } if (vd->granulepos != -1) vd->granulepos += sample_count; vorbis_synthesis_read (&vd->vd, sample_count); result = vorbis_dec_push (vd, out); done: /* 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 (packet no is %d)", packet->packetno)); 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_chain (GstPad * pad, GstBuffer * buffer) { GstVorbisDec *vd; ogg_packet packet; GstFlowReturn result = GST_FLOW_OK; vd = GST_VORBIS_DEC (GST_PAD_PARENT (pad)); if (GST_BUFFER_SIZE (buffer) == 0) { gst_buffer_unref (buffer); GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty buffer received")); return GST_FLOW_ERROR; } /* only ogg has granulepos, demuxers of other container formats * might provide us with timestamps instead (e.g. matroskademux) */ if (GST_BUFFER_OFFSET_END (buffer) == GST_BUFFER_OFFSET_NONE && GST_BUFFER_TIMESTAMP (buffer) != GST_CLOCK_TIME_NONE) { /* we might get multiple consecutive buffers with the same timestamp */ if (GST_BUFFER_TIMESTAMP (buffer) != vd->prev_timestamp) { vd->cur_timestamp = GST_BUFFER_TIMESTAMP (buffer); vd->prev_timestamp = GST_BUFFER_TIMESTAMP (buffer); } } 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 = GST_BUFFER_OFFSET_END (buffer); packet.packetno = vd->packetno++; /* * 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; 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, "Ignoring header"); goto done; } result = vorbis_handle_header_packet (vd, &packet); } else { result = vorbis_handle_data_packet (vd, &packet); } GST_DEBUG_OBJECT (vd, "offset end: %" G_GINT64_FORMAT, (gint64) GST_BUFFER_OFFSET_END (buffer)); done: gst_buffer_unref (buffer); 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; vd->cur_timestamp = GST_CLOCK_TIME_NONE; vd->prev_timestamp = GST_CLOCK_TIME_NONE; vd->granulepos = -1; vd->packetno = 0; 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"); vorbis_block_clear (&vd->vb); vorbis_dsp_clear (&vd->vd); vorbis_comment_clear (&vd->vc); vorbis_info_clear (&vd->vi); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return res; }