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815 lines
23 KiB
C
815 lines
23 KiB
C
/* GStreamer
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* Copyright (C) 2004 Benjamin Otte <in7y118@public.uni-hamburg.de>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-vorbisdec
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* @title: vorbisdec
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* @see_also: vorbisenc, oggdemux
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*
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* This element decodes a Vorbis stream to raw float audio.
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* [Vorbis](http://www.vorbis.com/) is a royalty-free audio codec maintained
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* by the [Xiph.org Foundation](http://www.xiph.org/). As it outputs raw float
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* audio you will often need to put an audioconvert element after it.
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*
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* ## Example pipelines
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* |[
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* gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
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* ]|
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* Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "gstvorbisdec.h"
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#include <string.h>
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#include <gst/audio/audio.h>
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#include <gst/tag/tag.h>
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#include "gstvorbiselements.h"
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#include "gstvorbiscommon.h"
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#ifndef TREMOR
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GST_DEBUG_CATEGORY_STATIC (vorbisdec_debug);
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#define GST_CAT_DEFAULT vorbisdec_debug
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#else
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GST_DEBUG_CATEGORY_STATIC (ivorbisdec_debug);
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#define GST_CAT_DEFAULT ivorbisdec_debug
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#endif
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static GstStaticPadTemplate vorbis_dec_src_factory =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_VORBIS_DEC_SRC_CAPS);
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static GstStaticPadTemplate vorbis_dec_sink_factory =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-vorbis")
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);
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#define gst_vorbis_dec_parent_class parent_class
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G_DEFINE_TYPE (GstVorbisDec, gst_vorbis_dec, GST_TYPE_AUDIO_DECODER);
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#ifndef TREMOR
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (vorbisdec, "vorbisdec",
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GST_RANK_PRIMARY, GST_TYPE_VORBIS_DEC,
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GST_DEBUG_CATEGORY_INIT (vorbisdec_debug, "vorbisdec", 0,
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"vorbis decoding element");
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vorbis_element_init (plugin));
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#else
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (ivorbisdec, "ivorbisdec",
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GST_RANK_SECONDARY, GST_TYPE_VORBIS_DEC,
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GST_DEBUG_CATEGORY_INIT (ivorbisdec_debug, "ivorbisdec", 0,
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"vorbis decoding element (integer decoder)");
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vorbis_element_init (plugin));
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#endif
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static void vorbis_dec_finalize (GObject * object);
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static gboolean vorbis_dec_start (GstAudioDecoder * dec);
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static gboolean vorbis_dec_stop (GstAudioDecoder * dec);
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static GstFlowReturn vorbis_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard);
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static gboolean vorbis_dec_set_format (GstAudioDecoder * dec, GstCaps * caps);
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static void vorbis_dec_reset (GstAudioDecoder * dec);
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static void
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gst_vorbis_dec_class_init (GstVorbisDecClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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gobject_class->finalize = vorbis_dec_finalize;
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gst_element_class_add_static_pad_template (element_class,
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&vorbis_dec_src_factory);
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gst_element_class_add_static_pad_template (element_class,
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&vorbis_dec_sink_factory);
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gst_element_class_set_static_metadata (element_class,
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"Vorbis audio decoder", "Codec/Decoder/Audio",
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GST_VORBIS_DEC_DESCRIPTION,
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"Benjamin Otte <otte@gnome.org>, Chris Lord <chris@openedhand.com>");
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base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (vorbis_dec_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (vorbis_dec_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush);
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}
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static void
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gst_vorbis_dec_init (GstVorbisDec * dec)
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{
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(dec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
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}
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static void
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vorbis_dec_finalize (GObject * object)
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{
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/* Release any possibly allocated libvorbis data.
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* _clear functions can safely be called multiple times
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*/
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GstVorbisDec *vd = GST_VORBIS_DEC (object);
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#ifndef USE_TREMOLO
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vorbis_block_clear (&vd->vb);
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#endif
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vorbis_dsp_clear (&vd->vd);
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vorbis_comment_clear (&vd->vc);
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vorbis_info_clear (&vd->vi);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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vorbis_dec_start (GstAudioDecoder * dec)
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{
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GstVorbisDec *vd = GST_VORBIS_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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vorbis_info_init (&vd->vi);
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vorbis_comment_init (&vd->vc);
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vd->initialized = FALSE;
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return TRUE;
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}
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static gboolean
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vorbis_dec_stop (GstAudioDecoder * dec)
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{
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GstVorbisDec *vd = GST_VORBIS_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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vd->initialized = FALSE;
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#ifndef USE_TREMOLO
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vorbis_block_clear (&vd->vb);
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#endif
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vorbis_dsp_clear (&vd->vd);
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vorbis_comment_clear (&vd->vc);
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vorbis_info_clear (&vd->vi);
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if (vd->pending_headers) {
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g_list_free_full (vd->pending_headers, (GDestroyNotify) gst_buffer_unref);
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vd->pending_headers = NULL;
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}
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return TRUE;
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}
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static GstFlowReturn
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vorbis_handle_identification_packet (GstVorbisDec * vd)
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{
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GstAudioInfo info;
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switch (vd->vi.channels) {
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case 1:
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case 2:
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case 3:
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case 4:
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case 5:
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case 6:
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case 7:
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case 8:
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{
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const GstAudioChannelPosition *pos;
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pos = gst_vorbis_default_channel_positions[vd->vi.channels - 1];
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gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
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vd->vi.channels, pos);
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break;
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}
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default:{
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GstAudioChannelPosition position[64];
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gint i, max_pos = MAX (vd->vi.channels, 64);
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GST_ELEMENT_WARNING (vd, STREAM, DECODE,
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(NULL), ("Using NONE channel layout for more than 8 channels"));
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for (i = 0; i < max_pos; i++)
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position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
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gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
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vd->vi.channels, position);
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break;
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}
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}
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gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (vd), &info);
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vd->info = info;
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/* select a copy_samples function, this way we can have specialized versions
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* for mono/stereo and avoid the depth switch in tremor case */
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vd->copy_samples = gst_vorbis_get_copy_sample_func (info.channels);
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return GST_FLOW_OK;
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}
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/* FIXME 0.11: remove tag handling and let container take care of that? */
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static GstFlowReturn
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vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet)
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{
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guint bitrate = 0;
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gchar *encoder = NULL;
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GstTagList *list;
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guint8 *data;
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gsize size;
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GST_DEBUG_OBJECT (vd, "parsing comment packet");
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data = gst_ogg_packet_data (packet);
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size = gst_ogg_packet_size (packet);
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list =
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gst_tag_list_from_vorbiscomment (data, size, (guint8 *) "\003vorbis", 7,
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&encoder);
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if (!list) {
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GST_ERROR_OBJECT (vd, "couldn't decode comments");
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list = gst_tag_list_new_empty ();
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}
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if (encoder) {
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if (encoder[0])
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_ENCODER, encoder, NULL);
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g_free (encoder);
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}
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_ENCODER_VERSION, vd->vi.version,
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GST_TAG_AUDIO_CODEC, "Vorbis", NULL);
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if (vd->vi.bitrate_nominal > 0 && vd->vi.bitrate_nominal <= 0x7FFFFFFF) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_NOMINAL_BITRATE, (guint) vd->vi.bitrate_nominal, NULL);
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bitrate = vd->vi.bitrate_nominal;
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}
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if (vd->vi.bitrate_upper > 0 && vd->vi.bitrate_upper <= 0x7FFFFFFF) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, NULL);
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if (!bitrate)
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bitrate = vd->vi.bitrate_upper;
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}
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if (vd->vi.bitrate_lower > 0 && vd->vi.bitrate_lower <= 0x7FFFFFFF) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL);
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if (!bitrate)
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bitrate = vd->vi.bitrate_lower;
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}
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if (bitrate) {
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gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
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GST_TAG_BITRATE, (guint) bitrate, NULL);
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}
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gst_audio_decoder_merge_tags (GST_AUDIO_DECODER_CAST (vd), list,
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GST_TAG_MERGE_REPLACE);
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gst_tag_list_unref (list);
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return GST_FLOW_OK;
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}
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static GstFlowReturn
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vorbis_handle_type_packet (GstVorbisDec * vd)
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{
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gint res;
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g_assert (!vd->initialized);
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#ifdef USE_TREMOLO
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if (G_UNLIKELY ((res = vorbis_dsp_init (&vd->vd, &vd->vi))))
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goto synthesis_init_error;
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#else
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if (G_UNLIKELY ((res = vorbis_synthesis_init (&vd->vd, &vd->vi))))
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goto synthesis_init_error;
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if (G_UNLIKELY ((res = vorbis_block_init (&vd->vd, &vd->vb))))
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goto block_init_error;
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#endif
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vd->initialized = TRUE;
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return GST_FLOW_OK;
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/* ERRORS */
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synthesis_init_error:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
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(NULL), ("couldn't initialize synthesis (%d)", res));
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return GST_FLOW_ERROR;
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}
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block_init_error:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
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(NULL), ("couldn't initialize block (%d)", res));
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return GST_FLOW_ERROR;
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}
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}
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static GstFlowReturn
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vorbis_handle_header_packet (GstVorbisDec * vd, ogg_packet * packet)
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{
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GstFlowReturn res;
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gint ret;
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GST_DEBUG_OBJECT (vd, "parsing header packet");
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/* Packetno = 0 if the first byte is exactly 0x01 */
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packet->b_o_s = ((gst_ogg_packet_data (packet))[0] == 0x1) ? 1 : 0;
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#ifdef USE_TREMOLO
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if ((ret = vorbis_dsp_headerin (&vd->vi, &vd->vc, packet)))
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#else
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if ((ret = vorbis_synthesis_headerin (&vd->vi, &vd->vc, packet)))
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#endif
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goto header_read_error;
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switch ((gst_ogg_packet_data (packet))[0]) {
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case 0x01:
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res = vorbis_handle_identification_packet (vd);
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break;
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case 0x03:
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res = vorbis_handle_comment_packet (vd, packet);
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break;
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case 0x05:
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res = vorbis_handle_type_packet (vd);
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break;
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default:
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/* ignore */
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GST_WARNING_OBJECT (vd, "unknown vorbis header packet found");
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res = GST_FLOW_OK;
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break;
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}
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return res;
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/* ERRORS */
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header_read_error:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (vd), STREAM, DECODE,
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(NULL), ("couldn't read header packet (%d)", ret));
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return GST_FLOW_ERROR;
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}
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}
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|
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/* Does not take ownership of buffer */
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static GstFlowReturn
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vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer)
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{
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ogg_packet *packet;
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ogg_packet_wrapper packet_wrapper;
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GstFlowReturn ret;
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GstMapInfo map;
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|
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gst_ogg_packet_wrapper_map (&packet_wrapper, buffer, &map);
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packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
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ret = vorbis_handle_header_packet (vd, packet);
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|
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gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer, &map);
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|
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return ret;
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}
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|
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#define MIN_NUM_HEADERS 3
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static GstFlowReturn
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vorbis_dec_handle_header_caps (GstVorbisDec * vd)
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{
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GstFlowReturn result = GST_FLOW_OK;
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GstCaps *caps;
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GstStructure *s = NULL;
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const GValue *array = NULL;
|
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|
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caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (vd));
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if (caps)
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s = gst_caps_get_structure (caps, 0);
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if (s)
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array = gst_structure_get_value (s, "streamheader");
|
|
|
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if (caps)
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gst_caps_unref (caps);
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|
|
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if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) {
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const GValue *value = NULL;
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GstBuffer *buf = NULL;
|
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gint i = 0;
|
|
|
|
if (vd->pending_headers) {
|
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GST_DEBUG_OBJECT (vd,
|
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"got new headers from caps, discarding old pending headers");
|
|
|
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g_list_free_full (vd->pending_headers, (GDestroyNotify) gst_buffer_unref);
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vd->pending_headers = NULL;
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}
|
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|
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while (result == GST_FLOW_OK && i < gst_value_array_get_size (array)) {
|
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value = gst_value_array_get_value (array, i);
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buf = gst_value_get_buffer (value);
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if (!buf)
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goto null_buffer;
|
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result = vorbis_dec_handle_header_buffer (vd, buf);
|
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i++;
|
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}
|
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} else
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goto array_error;
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|
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done:
|
|
return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK);
|
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|
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/* ERRORS */
|
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array_error:
|
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{
|
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GST_WARNING_OBJECT (vd, "streamheader array not found");
|
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result = GST_FLOW_ERROR;
|
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goto done;
|
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}
|
|
null_buffer:
|
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{
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GST_WARNING_OBJECT (vd, "streamheader with null buffer received");
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result = GST_FLOW_ERROR;
|
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goto done;
|
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}
|
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}
|
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|
|
|
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static GstFlowReturn
|
|
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
|
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GstClockTime timestamp, GstClockTime duration)
|
|
{
|
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#ifdef USE_TREMOLO
|
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vorbis_sample_t *pcm;
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#else
|
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vorbis_sample_t **pcm;
|
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#endif
|
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guint sample_count;
|
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GstBuffer *out = NULL;
|
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GstFlowReturn result;
|
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GstMapInfo map;
|
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gsize size;
|
|
|
|
if (G_UNLIKELY (!vd->initialized)) {
|
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result = vorbis_dec_handle_header_caps (vd);
|
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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_TREMOLO
|
|
if (G_UNLIKELY (vorbis_dsp_synthesis (&vd->vd, 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_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (vd), size);
|
|
|
|
gst_buffer_map (out, &map, GST_MAP_WRITE);
|
|
/* get samples ready for reading now, should be sample_count */
|
|
#ifdef USE_TREMOLO
|
|
if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, map.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 (map.data, map.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 *) map.data, pcm,
|
|
sample_count, vd->info.channels);
|
|
#endif
|
|
|
|
GST_LOG_OBJECT (vd, "have output size of %" G_GSIZE_FORMAT, size);
|
|
gst_buffer_unmap (out, &map);
|
|
|
|
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
|
|
check_pending_headers (GstVorbisDec * vd)
|
|
{
|
|
GstBuffer *buffer1, *buffer3, *buffer5;
|
|
GstMapInfo map;
|
|
gboolean isvalid;
|
|
GList *tmp = vd->pending_headers;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
|
|
if (g_list_length (vd->pending_headers) < MIN_NUM_HEADERS)
|
|
goto not_enough;
|
|
|
|
buffer1 = (GstBuffer *) tmp->data;
|
|
tmp = tmp->next;
|
|
buffer3 = (GstBuffer *) tmp->data;
|
|
tmp = tmp->next;
|
|
buffer5 = (GstBuffer *) tmp->data;
|
|
|
|
/* Start checking the headers */
|
|
gst_buffer_map (buffer1, &map, GST_MAP_READ);
|
|
isvalid = map.size >= 1 && map.data[0] == 0x01;
|
|
gst_buffer_unmap (buffer1, &map);
|
|
if (!isvalid) {
|
|
GST_WARNING_OBJECT (vd, "Pending first header was invalid");
|
|
goto cleanup;
|
|
}
|
|
|
|
gst_buffer_map (buffer3, &map, GST_MAP_READ);
|
|
isvalid = map.size >= 1 && map.data[0] == 0x03;
|
|
gst_buffer_unmap (buffer3, &map);
|
|
if (!isvalid) {
|
|
GST_WARNING_OBJECT (vd, "Pending second header was invalid");
|
|
goto cleanup;
|
|
}
|
|
|
|
gst_buffer_map (buffer5, &map, GST_MAP_READ);
|
|
isvalid = map.size >= 1 && map.data[0] == 0x05;
|
|
gst_buffer_unmap (buffer5, &map);
|
|
if (!isvalid) {
|
|
GST_WARNING_OBJECT (vd, "Pending third header was invalid");
|
|
goto cleanup;
|
|
}
|
|
|
|
/* Discard any other pending headers */
|
|
if (tmp->next) {
|
|
GST_DEBUG_OBJECT (vd, "Discarding extra headers");
|
|
g_list_free_full (tmp->next, (GDestroyNotify) gst_buffer_unref);
|
|
tmp->next = NULL;
|
|
}
|
|
g_list_free (vd->pending_headers);
|
|
vd->pending_headers = NULL;
|
|
|
|
GST_DEBUG_OBJECT (vd, "Resetting and processing new headers");
|
|
|
|
/* All good, let's reset ourselves and process the headers */
|
|
vorbis_dec_reset ((GstAudioDecoder *) vd);
|
|
result = vorbis_dec_handle_header_buffer (vd, buffer1);
|
|
gst_buffer_unref (buffer1);
|
|
if (result != GST_FLOW_OK) {
|
|
gst_buffer_unref (buffer3);
|
|
gst_buffer_unref (buffer5);
|
|
return result;
|
|
}
|
|
result = vorbis_dec_handle_header_buffer (vd, buffer3);
|
|
gst_buffer_unref (buffer3);
|
|
if (result != GST_FLOW_OK) {
|
|
gst_buffer_unref (buffer5);
|
|
return result;
|
|
}
|
|
result = vorbis_dec_handle_header_buffer (vd, buffer5);
|
|
gst_buffer_unref (buffer5);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
cleanup:
|
|
{
|
|
g_list_free_full (vd->pending_headers, (GDestroyNotify) gst_buffer_unref);
|
|
vd->pending_headers = NULL;
|
|
return result;
|
|
}
|
|
not_enough:
|
|
{
|
|
GST_LOG_OBJECT (vd,
|
|
"Not enough pending headers to properly reset, ignoring them");
|
|
goto cleanup;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
vorbis_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
|
|
{
|
|
ogg_packet *packet;
|
|
ogg_packet_wrapper packet_wrapper;
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
GstMapInfo map;
|
|
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, &map);
|
|
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) {
|
|
gboolean have_all_headers;
|
|
|
|
GST_LOG_OBJECT (vd, "storing header for later analyzis");
|
|
|
|
/* An identification packet starts a new set of headers */
|
|
if (vd->pending_headers && (gst_ogg_packet_data (packet))[0] == 0x01) {
|
|
GST_DEBUG_OBJECT (vd,
|
|
"got new identification header packet, discarding old pending headers");
|
|
|
|
g_list_free_full (vd->pending_headers, (GDestroyNotify) gst_buffer_unref);
|
|
vd->pending_headers = NULL;
|
|
}
|
|
|
|
/* if we have more than 3 headers with the new one and the new one is the
|
|
* type header, we can initialize the decoder now */
|
|
have_all_headers = g_list_length (vd->pending_headers) >= 2
|
|
&& (gst_ogg_packet_data (packet))[0] == 0x05;
|
|
|
|
if (!vd->pending_headers && (gst_ogg_packet_data (packet))[0] != 0x01) {
|
|
if (vd->initialized) {
|
|
GST_DEBUG_OBJECT (vd,
|
|
"Got another non-identification header after initialization, ignoring");
|
|
} else {
|
|
GST_WARNING_OBJECT (vd,
|
|
"First header was not a identification header, dropping");
|
|
}
|
|
result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), NULL, 1);
|
|
} else {
|
|
vd->pending_headers =
|
|
g_list_append (vd->pending_headers, gst_buffer_ref (buffer));
|
|
result = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (vd), NULL, 1);
|
|
}
|
|
|
|
if (result == GST_FLOW_OK && have_all_headers) {
|
|
result = check_pending_headers (vd);
|
|
}
|
|
} else {
|
|
GstClockTime timestamp, duration;
|
|
|
|
if (vd->pending_headers)
|
|
result = check_pending_headers (vd);
|
|
if (G_UNLIKELY (result != GST_FLOW_OK))
|
|
goto done;
|
|
|
|
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, &map);
|
|
|
|
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);
|
|
|
|
vorbis_synthesis_restart (&vd->vd);
|
|
}
|
|
|
|
static void
|
|
vorbis_dec_reset (GstAudioDecoder * dec)
|
|
{
|
|
GstVorbisDec *vd = GST_VORBIS_DEC (dec);
|
|
|
|
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);
|
|
vorbis_info_init (&vd->vi);
|
|
vorbis_comment_init (&vd->vc);
|
|
}
|
|
|
|
static gboolean
|
|
vorbis_dec_set_format (GstAudioDecoder * dec, GstCaps * caps)
|
|
{
|
|
GstVorbisDec *vd = GST_VORBIS_DEC (dec);
|
|
|
|
GST_DEBUG_OBJECT (vd, "New caps %" GST_PTR_FORMAT " - resetting", caps);
|
|
|
|
/* A set_format call implies new data with new header packets */
|
|
if (!vd->initialized)
|
|
return TRUE;
|
|
|
|
/* We need to free and re-init libvorbis,
|
|
* or it chokes */
|
|
vorbis_dec_reset (dec);
|
|
|
|
return TRUE;
|
|
}
|