gstreamer/ext/celt/gstceltdec.c

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/* GStreamer
* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
* Copyright (C) 2008 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Based on the speexdec element.
*/
/**
* SECTION:element-celtdec
* @see_also: celtenc, oggdemux
*
* This element decodes a CELT stream to raw integer audio.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=celt.ogg ! oggdemux ! celtdec ! audioconvert ! audioresample ! alsasink
* ]| Decode an Ogg/Celt file. To create an Ogg/Celt file refer to the documentation of celtenc.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstceltdec.h"
#include <string.h>
#include <gst/tag/tag.h>
GST_DEBUG_CATEGORY_STATIC (celtdec_debug);
#define GST_CAT_DEFAULT celtdec_debug
#define DEC_MAX_FRAME_SIZE 2000
static GstStaticPadTemplate celt_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) [ 32000, 64000 ], "
"channels = (int) [ 1, 2 ], "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, " "width = (int) 16, " "depth = (int) 16")
);
static GstStaticPadTemplate celt_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-celt")
);
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GST_BOILERPLATE (GstCeltDec, gst_celt_dec, GstAudioDecoder,
GST_TYPE_AUDIO_DECODER);
static gboolean gst_celt_dec_start (GstAudioDecoder * dec);
static gboolean gst_celt_dec_stop (GstAudioDecoder * dec);
static gboolean gst_celt_dec_set_format (GstAudioDecoder * bdec,
GstCaps * caps);
static GstFlowReturn gst_celt_dec_handle_frame (GstAudioDecoder * dec,
GstBuffer * buffer);
static void
gst_celt_dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_static_pad_template (element_class,
&celt_dec_src_factory);
gst_element_class_add_static_pad_template (element_class,
&celt_dec_sink_factory);
gst_element_class_set_details_simple (element_class, "Celt audio decoder",
"Codec/Decoder/Audio",
"decode celt streams to audio",
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
gst_celt_dec_class_init (GstCeltDecClass * klass)
{
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GstAudioDecoderClass *gstbase_class;
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gstbase_class = (GstAudioDecoderClass *) klass;
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gstbase_class->start = GST_DEBUG_FUNCPTR (gst_celt_dec_start);
gstbase_class->stop = GST_DEBUG_FUNCPTR (gst_celt_dec_stop);
gstbase_class->set_format = GST_DEBUG_FUNCPTR (gst_celt_dec_set_format);
gstbase_class->handle_frame = GST_DEBUG_FUNCPTR (gst_celt_dec_handle_frame);
GST_DEBUG_CATEGORY_INIT (celtdec_debug, "celtdec", 0,
"celt decoding element");
}
static void
gst_celt_dec_reset (GstCeltDec * dec)
{
dec->packetno = 0;
dec->frame_size = 0;
if (dec->state) {
celt_decoder_destroy (dec->state);
dec->state = NULL;
}
if (dec->mode) {
celt_mode_destroy (dec->mode);
dec->mode = NULL;
}
gst_buffer_replace (&dec->streamheader, NULL);
gst_buffer_replace (&dec->vorbiscomment, NULL);
g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->extra_headers);
dec->extra_headers = NULL;
memset (&dec->header, 0, sizeof (dec->header));
}
static void
gst_celt_dec_init (GstCeltDec * dec, GstCeltDecClass * g_class)
{
gst_celt_dec_reset (dec);
}
static gboolean
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gst_celt_dec_start (GstAudioDecoder * dec)
{
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GstCeltDec *cd = GST_CELT_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
gst_celt_dec_reset (cd);
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/* we know about concealment */
gst_audio_decoder_set_plc_aware (dec, TRUE);
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return TRUE;
}
static gboolean
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gst_celt_dec_stop (GstAudioDecoder * dec)
{
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GstCeltDec *cd = GST_CELT_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
gst_celt_dec_reset (cd);
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return TRUE;
}
static GstFlowReturn
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gst_celt_dec_parse_header (GstCeltDec * dec, GstBuffer * buf)
{
GstCaps *caps;
gint error = CELT_OK;
/* get the header */
error =
celt_header_from_packet ((const unsigned char *) GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf), &dec->header);
if (error < 0)
goto invalid_header;
if (memcmp (dec->header.codec_id, "CELT ", 8) != 0)
goto invalid_header;
#ifdef HAVE_CELT_0_7
dec->mode =
celt_mode_create (dec->header.sample_rate,
dec->header.frame_size, &error);
#else
dec->mode =
celt_mode_create (dec->header.sample_rate, dec->header.nb_channels,
dec->header.frame_size, &error);
#endif
if (!dec->mode)
goto mode_init_failed;
/* initialize the decoder */
#ifdef HAVE_CELT_0_11
dec->state =
celt_decoder_create_custom (dec->mode, dec->header.nb_channels, &error);
#else
#ifdef HAVE_CELT_0_7
dec->state = celt_decoder_create (dec->mode, dec->header.nb_channels, &error);
#else
dec->state = celt_decoder_create (dec->mode);
#endif
#endif
if (!dec->state)
goto init_failed;
#ifdef HAVE_CELT_0_8
dec->frame_size = dec->header.frame_size;
#else
celt_mode_info (dec->mode, CELT_GET_FRAME_SIZE, &dec->frame_size);
#endif
/* set caps */
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, dec->header.sample_rate,
"channels", G_TYPE_INT, dec->header.nb_channels,
"signed", G_TYPE_BOOLEAN, TRUE,
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
GST_DEBUG_OBJECT (dec, "rate=%d channels=%d frame-size=%d",
dec->header.sample_rate, dec->header.nb_channels, dec->frame_size);
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if (!gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps))
goto nego_failed;
gst_caps_unref (caps);
return GST_FLOW_OK;
/* ERRORS */
invalid_header:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Invalid header"));
return GST_FLOW_ERROR;
}
mode_init_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Mode initialization failed: %d", error));
return GST_FLOW_ERROR;
}
init_failed:
{
#ifdef HAVE_CELT_0_7
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't initialize decoder: %d", error));
#else
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't initialize decoder"));
#endif
return GST_FLOW_ERROR;
}
nego_failed:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't negotiate format"));
gst_caps_unref (caps);
return GST_FLOW_NOT_NEGOTIATED;
}
}
static GstFlowReturn
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gst_celt_dec_parse_comments (GstCeltDec * dec, GstBuffer * buf)
{
GstTagList *list;
gchar *ver, *encoder = NULL;
list = gst_tag_list_from_vorbiscomment_buffer (buf, NULL, 0, &encoder);
if (!list) {
GST_WARNING_OBJECT (dec, "couldn't decode comments");
list = gst_tag_list_new ();
}
if (encoder) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
}
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, "Celt", NULL);
ver = g_strndup (dec->header.codec_version, 20);
g_strstrip (ver);
if (ver != NULL && *ver != '\0') {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER_VERSION, ver, NULL);
}
if (dec->header.bytes_per_packet > 0) {
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, (guint) dec->header.bytes_per_packet * 8, NULL);
}
GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
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gst_element_found_tags_for_pad (GST_ELEMENT (dec),
GST_AUDIO_DECODER_SRC_PAD (dec), list);
g_free (encoder);
g_free (ver);
return GST_FLOW_OK;
}
static GstFlowReturn
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gst_celt_dec_parse_data (GstCeltDec * dec, GstBuffer * buf)
{
GstFlowReturn res = GST_FLOW_OK;
gint size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
gint error = CELT_OK;
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int skip = 0;
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if (!dec->frame_size)
goto not_negotiated;
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if (G_LIKELY (GST_BUFFER_SIZE (buf))) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
} else {
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/* FIXME ? actually consider how much concealment is needed */
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "creating concealment data");
data = NULL;
size = 0;
}
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/* FIXME really needed ?; this might lead to skipping samples below
* which kind of messes with subsequent timestamping */
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
#ifdef CELT_GET_LOOKAHEAD_REQUEST
/* what will be 0.11.5, I guess, but no versioning yet in git */
celt_decoder_ctl (dec->state, CELT_GET_LOOKAHEAD_REQUEST, &skip);
#else
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celt_mode_info (dec->mode, CELT_GET_LOOKAHEAD, &skip);
#endif
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}
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res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, dec->frame_size * dec->header.nb_channels * 2,
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GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
GST_LOG_OBJECT (dec, "decoding frame");
#ifdef HAVE_CELT_0_8
error = celt_decode (dec->state, data, size, out_data, dec->frame_size);
#else
error = celt_decode (dec->state, data, size, out_data);
#endif
#ifdef HAVE_CELT_0_11
if (error < 0) {
#else
if (error != CELT_OK) {
#endif
GST_WARNING_OBJECT (dec, "Decoding error: %d", error);
return GST_FLOW_ERROR;
}
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if (skip > 0) {
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GST_ERROR_OBJECT (dec, "skipping %d samples", skip);
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GST_BUFFER_DATA (outbuf) = GST_BUFFER_DATA (outbuf) +
skip * dec->header.nb_channels * 2;
GST_BUFFER_SIZE (outbuf) = GST_BUFFER_SIZE (outbuf) -
skip * dec->header.nb_channels * 2;
}
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res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
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if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
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return res;
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/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
("decoder not initialized"));
return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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static gboolean
gst_celt_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
{
GstCeltDec *dec = GST_CELT_DEC (bdec);
gboolean ret = TRUE;
GstStructure *s;
const GValue *streamheader;
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s = gst_caps_get_structure (caps, 0);
if ((streamheader = gst_structure_get_value (s, "streamheader")) &&
G_VALUE_HOLDS (streamheader, GST_TYPE_ARRAY) &&
gst_value_array_get_size (streamheader) >= 2) {
const GValue *header, *vorbiscomment;
GstBuffer *buf;
GstFlowReturn res = GST_FLOW_OK;
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header = gst_value_array_get_value (streamheader, 0);
if (header && G_VALUE_HOLDS (header, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (header);
res = gst_celt_dec_parse_header (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->streamheader, buf);
}
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vorbiscomment = gst_value_array_get_value (streamheader, 1);
if (vorbiscomment && G_VALUE_HOLDS (vorbiscomment, GST_TYPE_BUFFER)) {
buf = gst_value_get_buffer (vorbiscomment);
res = gst_celt_dec_parse_comments (dec, buf);
if (res != GST_FLOW_OK)
goto done;
gst_buffer_replace (&dec->vorbiscomment, buf);
}
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g_list_foreach (dec->extra_headers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (dec->extra_headers);
dec->extra_headers = NULL;
if (gst_value_array_get_size (streamheader) > 2) {
gint i, n;
n = gst_value_array_get_size (streamheader);
for (i = 2; i < n; i++) {
header = gst_value_array_get_value (streamheader, i);
buf = gst_value_get_buffer (header);
dec->extra_headers =
g_list_prepend (dec->extra_headers, gst_buffer_ref (buf));
}
}
}
done:
return ret;
}
static GstFlowReturn
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gst_celt_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
{
GstFlowReturn res;
GstCeltDec *dec;
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dec = GST_CELT_DEC (bdec);
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/* no fancy draining */
if (G_UNLIKELY (!buf))
return GST_FLOW_OK;
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/* If we have the streamheader and vorbiscomment from the caps already
* ignore them here */
if (dec->streamheader && dec->vorbiscomment) {
if (GST_BUFFER_SIZE (dec->streamheader) == GST_BUFFER_SIZE (buf)
&& memcmp (GST_BUFFER_DATA (dec->streamheader), GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf)) == 0) {
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GST_DEBUG_OBJECT (dec, "found streamheader");
gst_audio_decoder_finish_frame (bdec, NULL, 1);
res = GST_FLOW_OK;
} else if (GST_BUFFER_SIZE (dec->vorbiscomment) == GST_BUFFER_SIZE (buf)
&& memcmp (GST_BUFFER_DATA (dec->vorbiscomment), GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf)) == 0) {
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GST_DEBUG_OBJECT (dec, "found vorbiscomments");
gst_audio_decoder_finish_frame (bdec, NULL, 1);
res = GST_FLOW_OK;
} else {
GList *l;
for (l = dec->extra_headers; l; l = l->next) {
GstBuffer *header = l->data;
if (GST_BUFFER_SIZE (header) == GST_BUFFER_SIZE (buf) &&
memcmp (GST_BUFFER_DATA (header), GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf)) == 0) {
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GST_DEBUG_OBJECT (dec, "found extra header buffer");
gst_audio_decoder_finish_frame (bdec, NULL, 1);
res = GST_FLOW_OK;
goto done;
}
}
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res = gst_celt_dec_parse_data (dec, buf);
}
} else {
/* Otherwise fall back to packet counting and assume that the
* first two packets are the headers. */
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if (dec->packetno == 0) {
GST_DEBUG_OBJECT (dec, "counted streamheader");
res = gst_celt_dec_parse_header (dec, buf);
gst_audio_decoder_finish_frame (bdec, NULL, 1);
} else if (dec->packetno == 1) {
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
res = gst_celt_dec_parse_comments (dec, buf);
gst_audio_decoder_finish_frame (bdec, NULL, 1);
} else if (dec->packetno <= 1 + dec->header.extra_headers) {
GST_DEBUG_OBJECT (dec, "counted extra header");
gst_audio_decoder_finish_frame (bdec, NULL, 1);
res = GST_FLOW_OK;
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} else {
res = gst_celt_dec_parse_data (dec, buf);
}
}
done:
dec->packetno++;
return res;
}