/* 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 "gstvorbiscommon.h" GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug); #define GST_CAT_DEFAULT vorbisdec_debug static GstStaticPadTemplate vorbis_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_VORBIS_DEC_SRC_CAPS); static GstStaticPadTemplate vorbis_dec_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-vorbis") ); #define gst_vorbis_dec_parent_class parent_class G_DEFINE_TYPE (GstVorbisDec, gst_vorbis_dec, GST_TYPE_AUDIO_DECODER); static void vorbis_dec_finalize (GObject * object); static gboolean vorbis_dec_start (GstAudioDecoder * dec); static gboolean vorbis_dec_stop (GstAudioDecoder * dec); static GstFlowReturn vorbis_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer); static void vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard); static void gst_vorbis_dec_class_init (GstVorbisDecClass * klass) { GstPadTemplate *src_template, *sink_template; GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); gobject_class->finalize = vorbis_dec_finalize; 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_simple (element_class, "Vorbis audio decoder", "Codec/Decoder/Audio", GST_VORBIS_DEC_DESCRIPTION, "Benjamin Otte , Chris Lord "); base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start); base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop); base_class->handle_frame = GST_DEBUG_FUNCPTR (vorbis_dec_handle_frame); base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush); } static void gst_vorbis_dec_init (GstVorbisDec * dec) { } 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); #ifndef USE_TREMOLO vorbis_block_clear (&vd->vb); #endif 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) { if (dec->taglist) gst_tag_list_free (dec->taglist); dec->taglist = NULL; } static gboolean vorbis_dec_start (GstAudioDecoder * dec) { GstVorbisDec *vd = GST_VORBIS_DEC (dec); GST_DEBUG_OBJECT (dec, "start"); vorbis_info_init (&vd->vi); vorbis_comment_init (&vd->vc); vd->initialized = FALSE; gst_vorbis_dec_reset (vd); return TRUE; } static gboolean vorbis_dec_stop (GstAudioDecoder * dec) { GstVorbisDec *vd = GST_VORBIS_DEC (dec); GST_DEBUG_OBJECT (dec, "stop"); vd->initialized = FALSE; #ifndef USE_TREMOLO vorbis_block_clear (&vd->vb); #endif vorbis_dsp_clear (&vd->vd); vorbis_comment_clear (&vd->vc); vorbis_info_clear (&vd->vi); gst_vorbis_dec_reset (vd); return TRUE; } #if 0 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); /* 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; } } #endif static GstFlowReturn vorbis_handle_identification_packet (GstVorbisDec * vd) { GstCaps *caps; GstAudioInfo info; switch (info.channels) { case 1: case 2: case 3: case 4: case 5: case 6: case 7: case 8: { const GstAudioChannelPosition *pos; pos = gst_vorbis_default_channel_positions[info.channels - 1]; gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate, vd->vi.channels, pos); break; } default:{ GstAudioChannelPosition position[64]; gint i, max_pos = MAX (info.channels, 64); GST_ELEMENT_WARNING (vd, STREAM, DECODE, (NULL), ("Using NONE channel layout for more than 8 channels")); for (i = 0; i < max_pos; i++) position[i] = GST_AUDIO_CHANNEL_POSITION_NONE; gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate, vd->vi.channels, position); break; } } caps = gst_audio_info_to_caps (&info); gst_audio_decoder_set_outcaps (GST_AUDIO_DECODER (vd), caps); gst_caps_unref (caps); vd->info = info; /* select a copy_samples function, this way we can have specialized versions * for mono/stereo and avoid the depth switch in tremor case */ vd->copy_samples = get_copy_sample_func (info.channels); return GST_FLOW_OK; } /* FIXME 0.11: remove tag handling and let container take care of that? */ static GstFlowReturn vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet) { guint bitrate = 0; gchar *encoder = NULL; GstTagList *list, *old_list; guint8 *data; gsize size; GST_DEBUG_OBJECT (vd, "parsing comment packet"); data = gst_ogg_packet_data (packet); size = gst_ogg_packet_size (packet); list = gst_tag_list_from_vorbiscomment (data, size, (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); if (!vd->taglist) { GST_ERROR_OBJECT (vd, "couldn't decode comments"); vd->taglist = gst_tag_list_new_empty (); } if (encoder) { if (encoder[0]) 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_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (vd), gst_event_new_tag (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) { gint res; g_assert (vd->initialized == FALSE); #ifdef USE_TREMOLO if (G_UNLIKELY ((res = vorbis_dsp_init (&vd->vd, &vd->vi)))) goto synthesis_init_error; #else 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; #endif vd->initialized = TRUE; if (vd->taglist) { /* The tags have already been sent on the bus as messages. */ gst_pad_push_event (GST_AUDIO_DECODER_SRC_PAD (vd), 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 = ((gst_ogg_packet_data (packet))[0] == 0x1) ? 1 : 0; #ifdef USE_TREMELO if ((ret = vorbis_dsp_headerin (&vd->vi, &vd->vc, packet))) #else if ((ret = vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet))) #endif goto header_read_error; switch ((gst_ogg_packet_data (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; } } static GstFlowReturn vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer) { ogg_packet *packet; ogg_packet_wrapper packet_wrapper; GstFlowReturn ret; gst_ogg_packet_wrapper_map (&packet_wrapper, buffer); packet = gst_ogg_packet_from_wrapper (&packet_wrapper); ret = vorbis_handle_header_packet (vd, packet); gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer); return ret; } #define MIN_NUM_HEADERS 3 static GstFlowReturn vorbis_dec_handle_header_caps (GstVorbisDec * vd) { GstFlowReturn result = GST_FLOW_OK; GstCaps *caps; GstStructure *s = NULL; const GValue *array = NULL; caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (vd)); if (caps) s = gst_caps_get_structure (caps, 0); if (s) array = gst_structure_get_value (s, "streamheader"); if (caps) gst_caps_unref (caps); if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) { const GValue *value = NULL; GstBuffer *buf = NULL; gint i = 0; while (result == GST_FLOW_OK && i < gst_value_array_get_size (array)) { value = gst_value_array_get_value (array, i); buf = gst_value_get_buffer (value); if (!buf) goto null_buffer; result = vorbis_dec_handle_header_buffer (vd, buf); i++; } } else goto array_error; done: return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK); /* ERRORS */ array_error: { GST_WARNING_OBJECT (vd, "streamheader array not found"); result = GST_FLOW_ERROR; goto done; } null_buffer: { GST_WARNING_OBJECT (vd, "streamheader with null buffer received"); result = GST_FLOW_ERROR; goto done; } } static GstFlowReturn vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet, GstClockTime timestamp, GstClockTime duration) { #ifdef USE_TREMELO vorbis_sample_t *pcm; #else vorbis_sample_t **pcm; #endif guint sample_count; GstBuffer *out = NULL; GstFlowReturn result; guint8 *data; gsize size; if (G_UNLIKELY (!vd->initialized)) { result = vorbis_dec_handle_header_caps (vd); if (result != GST_FLOW_OK) goto not_initialized; } /* 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 careful not to * throw away too much. For now we decode everything and clip right * before pushing data. */ #ifdef USE_TREMELO if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vb, packet, 1))) goto could_not_read; #else 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; #endif /* assume all goes well here */ result = GST_FLOW_OK; /* count samples ready for reading */ #ifdef USE_TREMOLO if ((sample_count = vorbis_dsp_pcmout (&vd->vd, NULL, 0)) == 0) #else if ((sample_count = vorbis_synthesis_pcmout (&vd->vd, NULL)) == 0) goto done; #endif size = sample_count * vd->info.bpf; GST_LOG_OBJECT (vd, "%d samples ready for reading, size %" G_GSIZE_FORMAT, sample_count, size); /* alloc buffer for it */ out = gst_buffer_new_allocate (NULL, size, 0); data = gst_buffer_map (out, NULL, NULL, GST_MAP_WRITE); /* get samples ready for reading now, should be sample_count */ #ifdef USE_TREMOLO if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, data, sample_count) != sample_count)) #else if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count)) #endif goto wrong_samples; #ifdef USE_TREMOLO if (vd->info.channels < 9) gst_audio_reorder_channels (data, size, GST_VORBIS_AUDIO_FORMAT, vd->info.channels, gst_vorbis_channel_positions[vd->info.channels - 1], gst_vorbis_default_channel_positions[vd->info.channels - 1]); #else /* copy samples in buffer */ vd->copy_samples ((vorbis_sample_t *) data, pcm, sample_count, vd->info.channels); #endif GST_LOG_OBJECT (vd, "setting output size to %" G_GSIZE_FORMAT, size); gst_buffer_unmap (out, data, size); done: /* whether or not data produced, consume one frame and advance time */ result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), out, 1); #ifdef USE_TREMOLO vorbis_dsp_read (&vd->vd, sample_count); #else vorbis_synthesis_read (&vd->vd, sample_count); #endif return result; /* ERRORS */ not_initialized: { GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE, (NULL), ("no header sent yet")); return GST_FLOW_NOT_NEGOTIATED; } 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_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer) { ogg_packet *packet; ogg_packet_wrapper packet_wrapper; GstFlowReturn result = GST_FLOW_OK; GstVorbisDec *vd = GST_VORBIS_DEC (dec); /* no draining etc */ if (G_UNLIKELY (!buffer)) return GST_FLOW_OK; GST_LOG_OBJECT (vd, "got buffer %p", buffer); /* make ogg_packet out of the buffer */ gst_ogg_packet_wrapper_map (&packet_wrapper, buffer); packet = gst_ogg_packet_from_wrapper (&packet_wrapper); /* set some more stuff */ packet->granulepos = -1; packet->packetno = 0; /* we don't care */ /* EOS does not matter, it is used in vorbis to implement clipping the last * block of samples based on the granulepos. We clip based on segments. */ packet->e_o_s = 0; GST_LOG_OBJECT (vd, "decode buffer of size %ld", packet->bytes); /* 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; } /* switch depending on packet type */ if ((gst_ogg_packet_data (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); /* consumer header packet/frame */ gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), NULL, 1); } else { GstClockTime timestamp, duration; timestamp = GST_BUFFER_TIMESTAMP (buffer); duration = GST_BUFFER_DURATION (buffer); result = vorbis_handle_data_packet (vd, packet, timestamp, duration); } done: GST_LOG_OBJECT (vd, "unmap buffer %p", buffer); gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer); 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"); result = GST_FLOW_OK; goto done; } /* ERRORS */ empty_header: { GST_ELEMENT_ERROR (vd, STREAM, DECODE, (NULL), ("empty header received")); result = GST_FLOW_ERROR; goto done; } } static void vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard) { GstVorbisDec *vd = GST_VORBIS_DEC (dec); #ifdef HAVE_VORBIS_SYNTHESIS_RESTART vorbis_synthesis_restart (&vd->vd); #endif if (hard) gst_vorbis_dec_reset (vd); }