mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
576 lines
16 KiB
C
576 lines
16 KiB
C
/* GStreamer
|
|
* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
|
|
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
|
|
*
|
|
* 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-speexdec
|
|
* @see_also: speexenc, oggdemux
|
|
*
|
|
* This element decodes a Speex stream to raw integer audio.
|
|
* <ulink url="http://www.speex.org/">Speex</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-1.0 -v filesrc location=speex.ogg ! oggdemux ! speexdec ! audioconvert ! audioresample ! alsasink
|
|
* ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
|
|
* documentation of speexenc.
|
|
* </refsect2>
|
|
*
|
|
* Last reviewed on 2006-04-05 (0.10.2)
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include "config.h"
|
|
#endif
|
|
|
|
#include "gstspeexdec.h"
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
#include <gst/tag/tag.h>
|
|
#include <gst/audio/audio.h>
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (speexdec_debug);
|
|
#define GST_CAT_DEFAULT speexdec_debug
|
|
|
|
#define DEFAULT_ENH TRUE
|
|
|
|
enum
|
|
{
|
|
ARG_0,
|
|
ARG_ENH
|
|
};
|
|
|
|
#define FORMAT_STR GST_AUDIO_NE(S16)
|
|
|
|
static GstStaticPadTemplate speex_dec_src_factory =
|
|
GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw, "
|
|
"format = (string) " FORMAT_STR ", "
|
|
"layout = (string) interleaved, "
|
|
"rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")
|
|
);
|
|
|
|
static GstStaticPadTemplate speex_dec_sink_factory =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-speex")
|
|
);
|
|
|
|
#define gst_speex_dec_parent_class parent_class
|
|
G_DEFINE_TYPE (GstSpeexDec, gst_speex_dec, GST_TYPE_AUDIO_DECODER);
|
|
|
|
static gboolean gst_speex_dec_start (GstAudioDecoder * dec);
|
|
static gboolean gst_speex_dec_stop (GstAudioDecoder * dec);
|
|
static gboolean gst_speex_dec_set_format (GstAudioDecoder * bdec,
|
|
GstCaps * caps);
|
|
static GstFlowReturn gst_speex_dec_handle_frame (GstAudioDecoder * dec,
|
|
GstBuffer * buffer);
|
|
|
|
static void gst_speex_dec_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
static void gst_speex_dec_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
|
|
static void
|
|
gst_speex_dec_class_init (GstSpeexDecClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstAudioDecoderClass *base_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
base_class = (GstAudioDecoderClass *) klass;
|
|
|
|
gobject_class->set_property = gst_speex_dec_set_property;
|
|
gobject_class->get_property = gst_speex_dec_get_property;
|
|
|
|
base_class->start = GST_DEBUG_FUNCPTR (gst_speex_dec_start);
|
|
base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_dec_stop);
|
|
base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_dec_set_format);
|
|
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_dec_handle_frame);
|
|
|
|
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_ENH,
|
|
g_param_spec_boolean ("enh", "Enh", "Enable perceptual enhancement",
|
|
DEFAULT_ENH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&speex_dec_src_factory));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&speex_dec_sink_factory));
|
|
gst_element_class_set_static_metadata (gstelement_class,
|
|
"Speex audio decoder", "Codec/Decoder/Audio",
|
|
"decode speex streams to audio", "Wim Taymans <wim@fluendo.com>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (speexdec_debug, "speexdec", 0,
|
|
"speex decoding element");
|
|
}
|
|
|
|
static void
|
|
gst_speex_dec_reset (GstSpeexDec * dec)
|
|
{
|
|
dec->packetno = 0;
|
|
dec->frame_size = 0;
|
|
dec->frame_duration = 0;
|
|
dec->mode = NULL;
|
|
free (dec->header);
|
|
dec->header = NULL;
|
|
speex_bits_destroy (&dec->bits);
|
|
|
|
gst_buffer_replace (&dec->streamheader, NULL);
|
|
gst_buffer_replace (&dec->vorbiscomment, NULL);
|
|
|
|
if (dec->stereo) {
|
|
speex_stereo_state_destroy (dec->stereo);
|
|
dec->stereo = NULL;
|
|
}
|
|
|
|
if (dec->state) {
|
|
speex_decoder_destroy (dec->state);
|
|
dec->state = NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_speex_dec_init (GstSpeexDec * dec)
|
|
{
|
|
dec->enh = DEFAULT_ENH;
|
|
|
|
gst_speex_dec_reset (dec);
|
|
}
|
|
|
|
static gboolean
|
|
gst_speex_dec_start (GstAudioDecoder * dec)
|
|
{
|
|
GstSpeexDec *sd = GST_SPEEX_DEC (dec);
|
|
|
|
GST_DEBUG_OBJECT (dec, "start");
|
|
gst_speex_dec_reset (sd);
|
|
|
|
/* we know about concealment */
|
|
gst_audio_decoder_set_plc_aware (dec, TRUE);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_speex_dec_stop (GstAudioDecoder * dec)
|
|
{
|
|
GstSpeexDec *sd = GST_SPEEX_DEC (dec);
|
|
|
|
GST_DEBUG_OBJECT (dec, "stop");
|
|
gst_speex_dec_reset (sd);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_dec_parse_header (GstSpeexDec * dec, GstBuffer * buf)
|
|
{
|
|
GstMapInfo map;
|
|
GstAudioInfo info;
|
|
static const GstAudioChannelPosition chan_pos[2][2] = {
|
|
{GST_AUDIO_CHANNEL_POSITION_MONO},
|
|
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
|
|
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT}
|
|
};
|
|
|
|
/* get the header */
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
dec->header = speex_packet_to_header ((gchar *) map.data, map.size);
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
if (!dec->header)
|
|
goto no_header;
|
|
|
|
if (dec->header->mode >= SPEEX_NB_MODES || dec->header->mode < 0)
|
|
goto mode_too_old;
|
|
|
|
dec->mode = speex_lib_get_mode (dec->header->mode);
|
|
|
|
/* initialize the decoder */
|
|
dec->state = speex_decoder_init (dec->mode);
|
|
if (!dec->state)
|
|
goto init_failed;
|
|
|
|
speex_decoder_ctl (dec->state, SPEEX_SET_ENH, &dec->enh);
|
|
speex_decoder_ctl (dec->state, SPEEX_GET_FRAME_SIZE, &dec->frame_size);
|
|
|
|
if (dec->header->nb_channels != 1) {
|
|
dec->stereo = speex_stereo_state_init ();
|
|
dec->callback.callback_id = SPEEX_INBAND_STEREO;
|
|
dec->callback.func = speex_std_stereo_request_handler;
|
|
dec->callback.data = dec->stereo;
|
|
speex_decoder_ctl (dec->state, SPEEX_SET_HANDLER, &dec->callback);
|
|
}
|
|
|
|
speex_decoder_ctl (dec->state, SPEEX_SET_SAMPLING_RATE, &dec->header->rate);
|
|
|
|
dec->frame_duration = gst_util_uint64_scale_int (dec->frame_size,
|
|
GST_SECOND, dec->header->rate);
|
|
|
|
speex_bits_init (&dec->bits);
|
|
|
|
/* set caps */
|
|
gst_audio_info_init (&info);
|
|
gst_audio_info_set_format (&info,
|
|
GST_AUDIO_FORMAT_S16,
|
|
dec->header->rate,
|
|
dec->header->nb_channels, chan_pos[dec->header->nb_channels - 1]);
|
|
|
|
if (!gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info))
|
|
goto nego_failed;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_header:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
|
(NULL), ("couldn't read header"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
mode_too_old:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
|
(NULL),
|
|
("Mode number %d does not (yet/any longer) exist in this version",
|
|
dec->header->mode));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
init_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
|
(NULL), ("couldn't initialize decoder"));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
nego_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
|
|
(NULL), ("couldn't negotiate format"));
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_dec_parse_comments (GstSpeexDec * 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_empty ();
|
|
}
|
|
|
|
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, "Speex", NULL);
|
|
|
|
ver = g_strndup (dec->header->speex_version, SPEEX_HEADER_VERSION_LENGTH);
|
|
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->bitrate > 0) {
|
|
gst_tag_list_add (list, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_BITRATE, (guint) dec->header->bitrate, NULL);
|
|
}
|
|
|
|
GST_INFO_OBJECT (dec, "tags: %" GST_PTR_FORMAT, list);
|
|
|
|
gst_audio_decoder_merge_tags (GST_AUDIO_DECODER (dec), list,
|
|
GST_TAG_MERGE_REPLACE);
|
|
gst_tag_list_unref (list);
|
|
|
|
g_free (encoder);
|
|
g_free (ver);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_speex_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
|
|
{
|
|
GstSpeexDec *dec = GST_SPEEX_DEC (bdec);
|
|
gboolean ret = TRUE;
|
|
GstStructure *s;
|
|
const GValue *streamheader;
|
|
|
|
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;
|
|
|
|
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_speex_dec_parse_header (dec, buf);
|
|
if (res != GST_FLOW_OK)
|
|
goto done;
|
|
gst_buffer_replace (&dec->streamheader, buf);
|
|
}
|
|
|
|
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_speex_dec_parse_comments (dec, buf);
|
|
if (res != GST_FLOW_OK)
|
|
goto done;
|
|
gst_buffer_replace (&dec->vorbiscomment, buf);
|
|
}
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_dec_parse_data (GstSpeexDec * dec, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn res = GST_FLOW_OK;
|
|
gint i, fpp;
|
|
SpeexBits *bits;
|
|
GstMapInfo map;
|
|
|
|
if (!dec->frame_duration)
|
|
goto not_negotiated;
|
|
|
|
if (G_LIKELY (gst_buffer_get_size (buf))) {
|
|
/* send data to the bitstream */
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
speex_bits_read_from (&dec->bits, (gchar *) map.data, map.size);
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
fpp = dec->header->frames_per_packet;
|
|
bits = &dec->bits;
|
|
|
|
GST_DEBUG_OBJECT (dec, "received buffer of size %" G_GSIZE_FORMAT
|
|
", fpp %d, %d bits", map.size, fpp, speex_bits_remaining (bits));
|
|
} else {
|
|
/* FIXME ? actually consider how much concealment is needed */
|
|
/* concealment data, pass NULL as the bits parameters */
|
|
GST_DEBUG_OBJECT (dec, "creating concealment data");
|
|
fpp = dec->header->frames_per_packet;
|
|
bits = NULL;
|
|
}
|
|
|
|
/* now decode each frame, catering for unknown number of them (e.g. rtp) */
|
|
for (i = 0; i < fpp; i++) {
|
|
GstBuffer *outbuf;
|
|
gboolean corrupted = FALSE;
|
|
gint ret;
|
|
|
|
GST_LOG_OBJECT (dec, "decoding frame %d/%d, %d bits remaining", i, fpp,
|
|
bits ? speex_bits_remaining (bits) : -1);
|
|
#if 0
|
|
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,
|
|
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;
|
|
}
|
|
#endif
|
|
/* FIXME, we can use a bufferpool because we have fixed size buffers. We
|
|
* could also use an allocator */
|
|
outbuf =
|
|
gst_buffer_new_allocate (NULL,
|
|
dec->frame_size * dec->header->nb_channels * 2, NULL);
|
|
|
|
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
|
|
ret = speex_decode_int (dec->state, bits, (spx_int16_t *) map.data);
|
|
|
|
if (ret == -1) {
|
|
/* uh? end of stream */
|
|
if (fpp == 0 && speex_bits_remaining (bits) < 8) {
|
|
/* if we did not know how many frames to expect, then we get this
|
|
at the end if there are leftover bits to pad to the next byte */
|
|
GST_DEBUG_OBJECT (dec, "Discarding leftover bits");
|
|
} else {
|
|
GST_WARNING_OBJECT (dec, "Unexpected end of stream found");
|
|
}
|
|
corrupted = TRUE;
|
|
} else if (ret == -2) {
|
|
GST_WARNING_OBJECT (dec, "Decoding error: corrupted stream?");
|
|
corrupted = TRUE;
|
|
}
|
|
|
|
if (bits && speex_bits_remaining (bits) < 0) {
|
|
GST_WARNING_OBJECT (dec, "Decoding overflow: corrupted stream?");
|
|
corrupted = TRUE;
|
|
}
|
|
if (dec->header->nb_channels == 2)
|
|
speex_decode_stereo_int ((spx_int16_t *) map.data, dec->frame_size,
|
|
dec->stereo);
|
|
|
|
gst_buffer_unmap (outbuf, &map);
|
|
|
|
if (!corrupted) {
|
|
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
|
|
} else {
|
|
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), NULL, 1);
|
|
gst_buffer_unref (outbuf);
|
|
}
|
|
|
|
if (res != GST_FLOW_OK) {
|
|
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
|
|
break;
|
|
}
|
|
}
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
not_negotiated:
|
|
{
|
|
GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
|
|
("decoder not initialized"));
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
|
|
{
|
|
GstMapInfo map;
|
|
gsize size1, size2;
|
|
gboolean res;
|
|
|
|
size1 = gst_buffer_get_size (buf1);
|
|
size2 = gst_buffer_get_size (buf2);
|
|
|
|
if (size1 != size2)
|
|
return FALSE;
|
|
|
|
gst_buffer_map (buf1, &map, GST_MAP_READ);
|
|
res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;
|
|
gst_buffer_unmap (buf1, &map);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_speex_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn res;
|
|
GstSpeexDec *dec;
|
|
|
|
/* no fancy draining */
|
|
if (G_UNLIKELY (!buf))
|
|
return GST_FLOW_OK;
|
|
|
|
dec = GST_SPEEX_DEC (bdec);
|
|
|
|
/* If we have the streamheader and vorbiscomment from the caps already
|
|
* ignore them here */
|
|
if (dec->streamheader && dec->vorbiscomment) {
|
|
if (memcmp_buffers (dec->streamheader, buf)) {
|
|
GST_DEBUG_OBJECT (dec, "found streamheader");
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
res = GST_FLOW_OK;
|
|
} else if (memcmp_buffers (dec->vorbiscomment, buf)) {
|
|
GST_DEBUG_OBJECT (dec, "found vorbiscomments");
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
res = GST_FLOW_OK;
|
|
} else {
|
|
res = gst_speex_dec_parse_data (dec, buf);
|
|
}
|
|
} else {
|
|
/* Otherwise fall back to packet counting and assume that the
|
|
* first two packets are the headers. */
|
|
switch (dec->packetno) {
|
|
case 0:
|
|
GST_DEBUG_OBJECT (dec, "counted streamheader");
|
|
res = gst_speex_dec_parse_header (dec, buf);
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
break;
|
|
case 1:
|
|
GST_DEBUG_OBJECT (dec, "counted vorbiscomments");
|
|
res = gst_speex_dec_parse_comments (dec, buf);
|
|
gst_audio_decoder_finish_frame (bdec, NULL, 1);
|
|
break;
|
|
default:
|
|
{
|
|
res = gst_speex_dec_parse_data (dec, buf);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
dec->packetno++;
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
gst_speex_dec_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSpeexDec *speexdec;
|
|
|
|
speexdec = GST_SPEEX_DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_ENH:
|
|
g_value_set_boolean (value, speexdec->enh);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_speex_dec_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSpeexDec *speexdec;
|
|
|
|
speexdec = GST_SPEEX_DEC (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_ENH:
|
|
speexdec->enh = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|