gstreamer/ext/vorbis/gstvorbisdec.c
2013-05-03 14:30:03 +02:00

631 lines
17 KiB
C

/* 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
* @see_also: vorbisenc, oggdemux
*
* This element decodes a Vorbis stream to raw float audio.
* <ulink url="http://www.vorbis.com/">Vorbis</ulink> is a royalty-free
* audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
* Foundation</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* 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.
* </refsect2>
*
* Last reviewed on 2006-03-01 (0.10.4)
*/
#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 "gstvorbiscommon.h"
#ifndef TREMOR
GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug);
#define GST_CAT_DEFAULT vorbisdec_debug
#else
GST_DEBUG_CATEGORY_EXTERN (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);
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_static_metadata (element_class,
"Vorbis audio decoder", "Codec/Decoder/Audio",
GST_VORBIS_DEC_DESCRIPTION,
"Benjamin Otte <otte@gnome.org>, Chris Lord <chris@openedhand.com>");
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 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;
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);
return TRUE;
}
static GstFlowReturn
vorbis_handle_identification_packet (GstVorbisDec * vd)
{
GstAudioInfo info;
switch (vd->vi.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[vd->vi.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 (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);
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;
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);
if (!list) {
GST_ERROR_OBJECT (vd, "couldn't decode comments");
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 == 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;
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_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;
}
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;
GstMapInfo map;
gst_ogg_packet_wrapper_map (&packet_wrapper, buffer, &map);
packet = gst_ogg_packet_from_wrapper (&packet_wrapper);
ret = vorbis_handle_header_packet (vd, packet);
gst_ogg_packet_wrapper_unmap (&packet_wrapper, buffer, &map);
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_TREMOLO
vorbis_sample_t *pcm;
#else
vorbis_sample_t **pcm;
#endif
guint sample_count;
GstBuffer *out = NULL;
GstFlowReturn result;
GstMapInfo map;
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_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
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) {
if (vd->initialized) {
GST_WARNING_OBJECT (vd, "Already initialized, so ignoring header packet");
goto done;
}
result = vorbis_handle_header_packet (vd, packet);
if (result != GST_FLOW_OK)
goto done;
/* consumer header packet/frame */
result = 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, &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)
{
#ifdef HAVE_VORBIS_SYNTHESIS_RESTART
GstVorbisDec *vd = GST_VORBIS_DEC (dec);
vorbis_synthesis_restart (&vd->vd);
#endif
}