gstreamer/ext/vorbis/gstvorbisdec.c

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/* GStreamer
* Copyright (C) 2004 Benjamin Otte <in7y118@public.uni-hamburg.de>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-vorbisdec
* @title: vorbisdec
* @see_also: vorbisenc, oggdemux
*
* This element decodes a Vorbis stream to raw float audio.
* [Vorbis](http://www.vorbis.com/) is a royalty-free audio codec maintained
* by the [Xiph.org Foundation](http://www.xiph.org/). As it outputs raw float
* audio you will often need to put an audioconvert element after it.
*
* ## Example pipelines
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
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* |[
* gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstvorbisdec.h"
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/tag/tag.h>
#include "gstvorbiselements.h"
#include "gstvorbiscommon.h"
#ifndef TREMOR
GST_DEBUG_CATEGORY_STATIC (vorbisdec_debug);
#define GST_CAT_DEFAULT vorbisdec_debug
#else
GST_DEBUG_CATEGORY_STATIC (ivorbisdec_debug);
#define GST_CAT_DEFAULT ivorbisdec_debug
#endif
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);
#ifndef TREMOR
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (vorbisdec, "vorbisdec",
GST_RANK_PRIMARY, GST_TYPE_VORBIS_DEC,
GST_DEBUG_CATEGORY_INIT (vorbisdec_debug, "vorbisdec", 0,
"vorbis decoding element");
vorbis_element_init (plugin));
#else
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (ivorbisdec, "ivorbisdec",
GST_RANK_SECONDARY, GST_TYPE_VORBIS_DEC,
GST_DEBUG_CATEGORY_INIT (ivorbisdec_debug, "ivorbisdec", 0,
"vorbis decoding element (integer decoder)");
vorbis_element_init (plugin));
#endif
static void vorbis_dec_finalize (GObject * object);
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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 gboolean vorbis_dec_set_format (GstAudioDecoder * dec, GstCaps * caps);
static void vorbis_dec_reset (GstAudioDecoder * dec);
static void
gst_vorbis_dec_class_init (GstVorbisDecClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
gobject_class->finalize = vorbis_dec_finalize;
gst_element_class_add_static_pad_template (element_class,
&vorbis_dec_src_factory);
gst_element_class_add_static_pad_template (element_class,
&vorbis_dec_sink_factory);
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gst_element_class_set_static_metadata (element_class,
"Vorbis audio decoder", "Codec/Decoder/Audio",
GST_VORBIS_DEC_DESCRIPTION,
"Benjamin Otte <otte@gnome.org>, Chris Lord <chris@openedhand.com>");
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base_class->start = GST_DEBUG_FUNCPTR (vorbis_dec_start);
base_class->stop = GST_DEBUG_FUNCPTR (vorbis_dec_stop);
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);
base_class->flush = GST_DEBUG_FUNCPTR (vorbis_dec_flush);
}
static void
gst_vorbis_dec_init (GstVorbisDec * dec)
{
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (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 gboolean
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vorbis_dec_start (GstAudioDecoder * dec)
{
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GstVorbisDec *vd = GST_VORBIS_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
vorbis_info_init (&vd->vi);
vorbis_comment_init (&vd->vc);
vd->initialized = FALSE;
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return TRUE;
}
static gboolean
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vorbis_dec_stop (GstAudioDecoder * dec)
{
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GstVorbisDec *vd = GST_VORBIS_DEC (dec);
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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);
if (vd->pending_headers) {
g_list_free_full (vd->pending_headers, (GDestroyNotify) gst_buffer_unref);
vd->pending_headers = NULL;
}
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return TRUE;
}
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstAudioInfo info;
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switch (vd->vi.channels) {
case 1:
case 2:
case 3:
case 4:
case 5:
case 6:
case 7:
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case 8:
{
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const GstAudioChannelPosition *pos;
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pos = gst_vorbis_default_channel_positions[vd->vi.channels - 1];
gst_audio_info_set_format (&info, GST_VORBIS_AUDIO_FORMAT, vd->vi.rate,
vd->vi.channels, pos);
break;
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}
default:{
GstAudioChannelPosition position[64];
gint i, max_pos = MAX (vd->vi.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);
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break;
}
}
gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (vd), &info);
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 = gst_vorbis_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;
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guint8 *data;
gsize size;
GST_DEBUG_OBJECT (vd, "parsing comment packet");
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data = gst_ogg_packet_data (packet);
size = gst_ogg_packet_size (packet);
list =
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gst_tag_list_from_vorbiscomment (data, size, (guint8 *) "\003vorbis", 7,
&encoder);
if (!list) {
GST_ERROR_OBJECT (vd, "couldn't decode comments");
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list = gst_tag_list_new_empty ();
}
if (encoder) {
if (encoder[0])
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 && vd->vi.bitrate_nominal <= 0x7FFFFFFF) {
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 && vd->vi.bitrate_upper <= 0x7FFFFFFF) {
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 && vd->vi.bitrate_lower <= 0x7FFFFFFF) {
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_audio_decoder_merge_tags (GST_AUDIO_DECODER_CAST (vd), list,
GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (list);
return GST_FLOW_OK;
}
static GstFlowReturn
vorbis_handle_type_packet (GstVorbisDec * vd)
{
gint res;
g_assert (!vd->initialized);
#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;
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;
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#ifdef USE_TREMOLO
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;
}
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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;
}
}
/* Does not take ownership of buffer */
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
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static GstFlowReturn
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vorbis_dec_handle_header_buffer (GstVorbisDec * vd, GstBuffer * buffer)
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
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{
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ogg_packet *packet;
ogg_packet_wrapper packet_wrapper;
GstFlowReturn ret;
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GstMapInfo map;
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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);
ret = vorbis_handle_header_packet (vd, packet);
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gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer, &map);
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
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return ret;
Updated seek example. Original commit message from CVS: * docs/libs/tmpl/gstringbuffer.sgml: * examples/seeking/seek.c: (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (update_scale), (do_seek): Updated seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_activate_chain), (gst_ogg_demux_find_chains), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): Push out correct discont values. * ext/theora/theoradec.c: (theora_dec_src_convert), (theora_dec_sink_convert), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_handle_type_packet), (theora_handle_header_packet), (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain), (theora_dec_change_state): Better timestamping. * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain): * ext/vorbis/vorbisdec.h: Better timestamping. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (gst_base_audio_sink_get_times), (gst_base_audio_sink_event), (gst_base_audio_sink_render): Handle syncing on timestamps instead of sample offsets. Make use of DISCONT values as described in design docs. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_get_time): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_acquire), (gst_ring_buffer_set_sample), (gst_ring_buffer_commit), (gst_ring_buffer_read): * gst-libs/gst/audio/gstringbuffer.h: * sys/ximage/ximagesink.c: (gst_ximagesink_get_times), (gst_ximagesink_show_frame): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_times): Correcly convert buffer timestamp to stream time.
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}
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#define MIN_NUM_HEADERS 3
static GstFlowReturn
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vorbis_dec_handle_header_caps (GstVorbisDec * vd)
{
GstFlowReturn result = GST_FLOW_OK;
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GstCaps *caps;
GstStructure *s = NULL;
const GValue *array = NULL;
caps = gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (vd));
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if (caps)
s = gst_caps_get_structure (caps, 0);
if (s)
array = gst_structure_get_value (s, "streamheader");
if (caps)
gst_caps_unref (caps);
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if (array && (gst_value_array_get_size (array) >= MIN_NUM_HEADERS)) {
const GValue *value = NULL;
GstBuffer *buf = NULL;
gint i = 0;
if (vd->pending_headers) {
GST_DEBUG_OBJECT (vd,
"got new headers from caps, discarding old pending headers");
g_list_free_full (vd->pending_headers, (GDestroyNotify) gst_buffer_unref);
vd->pending_headers = NULL;
}
while (result == GST_FLOW_OK && i < gst_value_array_get_size (array)) {
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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;
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done:
return (result != GST_FLOW_OK ? GST_FLOW_NOT_NEGOTIATED : GST_FLOW_OK);
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/* 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;
}
}
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static GstFlowReturn
vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
GstClockTime timestamp, GstClockTime duration)
{
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#ifdef USE_TREMOLO
vorbis_sample_t *pcm;
#else
vorbis_sample_t **pcm;
#endif
guint sample_count;
GstBuffer *out = NULL;
GstFlowReturn result;
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GstMapInfo map;
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gsize size;
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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. */
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#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;
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#endif
size = sample_count * vd->info.bpf;
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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);
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gst_buffer_map (out, &map, GST_MAP_WRITE);
/* get samples ready for reading now, should be sample_count */
#ifdef USE_TREMOLO
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if (G_UNLIKELY (vorbis_dsp_pcmout (&vd->vd, map.data, sample_count) !=
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sample_count))
#else
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if (G_UNLIKELY (vorbis_synthesis_pcmout (&vd->vd, &pcm) != sample_count))
#endif
goto wrong_samples;
#ifdef USE_TREMOLO
if (vd->info.channels < 9)
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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 */
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vd->copy_samples ((vorbis_sample_t *) map.data, pcm,
sample_count, vd->info.channels);
#endif
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GST_LOG_OBJECT (vd, "have output size of %" G_GSIZE_FORMAT, size);
gst_buffer_unmap (out, &map);
done:
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/* 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"));
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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;
}
}
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static GstFlowReturn
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vorbis_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
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{
ogg_packet *packet;
ogg_packet_wrapper packet_wrapper;
GstFlowReturn result = GST_FLOW_OK;
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GstMapInfo map;
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GstVorbisDec *vd = GST_VORBIS_DEC (dec);
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/* no draining etc */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
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GST_LOG_OBJECT (vd, "got buffer %p", buffer);
/* make ogg_packet out of the buffer */
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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:
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GST_LOG_OBJECT (vd, "unmap buffer %p", buffer);
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gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer, &map);
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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;
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goto done;
}
}
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static void
vorbis_dec_flush (GstAudioDecoder * dec, gboolean hard)
{
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GstVorbisDec *vd = GST_VORBIS_DEC (dec);
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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;
}