/* 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 * @short_description: a decoder that decodes Vorbis to raw audio * @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 "vorbisdec.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 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; } 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) { GList *walk; dec->cur_timestamp = GST_CLOCK_TIME_NONE; dec->prev_timestamp = GST_CLOCK_TIME_NONE; dec->granulepos = -1; dec->discont = TRUE; gst_segment_init (&dec->segment, GST_FORMAT_TIME); for (walk = dec->queued; walk; walk = g_list_next (walk)) { gst_buffer_unref (GST_BUFFER_CAST (walk->data)); } g_list_free (dec->queued); dec->queued = 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; gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); 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); 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 and a positive rate for now */ if (format != GST_FORMAT_TIME) goto newseg_wrong_format; if (rate <= 0.0) goto newseg_wrong_rate; 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); /* and forward */ ret = gst_pad_push_event (dec->srcpad, event); 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; } newseg_wrong_rate: { GST_DEBUG_OBJECT (dec, "negative rates not supported yet"); 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; } default: goto channel_count_error; } 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); } gst_pad_set_caps (vd->srcpad, caps); gst_caps_unref (caps); return GST_FLOW_OK; /* ERROR */ channel_count_error: { 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; 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); } gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad, list); 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; 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")); 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 } /* clip output samples to the segment boundaries */ static gboolean vorbis_do_clip (GstVorbisDec * dec, GstBuffer * buf) { gint64 start, stop, cstart, cstop, diff; guint size; start = GST_BUFFER_TIMESTAMP (buf); stop = start + GST_BUFFER_DURATION (buf); size = GST_BUFFER_SIZE (buf); if (!gst_segment_clip (&dec->segment, GST_FORMAT_TIME, start, stop, &cstart, &cstop)) goto clipped; /* see if some clipping happened */ diff = cstart - start; if (diff > 0) { GST_BUFFER_TIMESTAMP (buf) = cstart; GST_BUFFER_DURATION (buf) -= diff; /* bring clipped time to samples */ diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND); /* samples to bytes */ diff *= (sizeof (float) * dec->vi.channels); GST_DEBUG_OBJECT (dec, "clipping start to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstart), diff); GST_BUFFER_DATA (buf) += diff; GST_BUFFER_SIZE (buf) -= diff; } diff = stop - cstop; if (diff > 0) { GST_BUFFER_DURATION (buf) -= diff; /* bring clipped time to samples and then to bytes */ diff = gst_util_uint64_scale_int (diff, dec->vi.rate, GST_SECOND); diff *= (sizeof (float) * dec->vi.channels); GST_DEBUG_OBJECT (dec, "clipping stop to %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT " bytes", GST_TIME_ARGS (cstop), diff); GST_BUFFER_SIZE (buf) -= diff; } return FALSE; /* dropped buffer */ clipped: { GST_DEBUG_OBJECT (dec, "clipped buffer"); gst_buffer_unref (buf); return TRUE; } } 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 (G_UNLIKELY (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_int (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); /* clips or returns FALSE with buffer unreffed when completely * clipped */ if (vorbis_do_clip (dec, buffer)) 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 (vorbis_do_clip (dec, buf)) 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_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet) { float **pcm; guint sample_count; GstBuffer *out; GstFlowReturn result; 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; 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; 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_int (vd->granulepos, GST_SECOND, vd->vi.rate); } else { GST_BUFFER_TIMESTAMP (out) = -1; } /* this should not overflow */ 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_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 (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; result = vorbis_dec_push (vd, out); done: vorbis_synthesis_read (&vd->vd, sample_count); /* 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_chain (GstPad * pad, GstBuffer * buffer) { GstVorbisDec *vd; ogg_packet packet; GstFlowReturn result = GST_FLOW_OK; GstClockTime timestamp; guint64 offset_end; vd = GST_VORBIS_DEC (gst_pad_get_parent (pad)); /* resync on DISCONT */ if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_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; } 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; if (G_UNLIKELY (packet.bytes < 1)) goto wrong_size; 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_GUINT64_FORMAT, offset_end); done: gst_buffer_unref (buffer); gst_object_unref (vd); return result; /* ERRORS */ wrong_size: { GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty buffer received")); result = GST_FLOW_ERROR; vd->discont = TRUE; goto done; } } 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"); 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; }