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gstreamer/ext/faac/gstfaac.c

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/* GStreamer FAAC (Free AAC Encoder) plugin
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
* Copyright (C) 2009 Mark Nauwelaerts <mnauw@users.sourceforge.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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-faac
* @see_also: faad
*
* faac encodes raw audio to AAC (MPEG-4 part 3) streams.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac ! matroskamux ! filesink location=sine.mkv
* ]| Encode a sine beep as aac and write to matroska container.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/audio/multichannel.h>
#include <gst/pbutils/codec-utils.h>
#include "gstfaac.h"
#define SAMPLE_RATES " 8000, " \
"11025, " \
"12000, " \
"16000, " \
"22050, " \
"24000, " \
"32000, " \
"44100, " \
"48000, " \
"64000, " \
"88200, " \
"96000"
#define SINK_CAPS \
"audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"rate = (int) {" SAMPLE_RATES "}, " \
"channels = (int) [ 1, 6 ] "
/* these don't seem to work? */
#if 0
"audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 32, "
"depth = (int) { 24, 32 }, "
"rate = (int) [ 8000, 96000], "
"channels = (int) [ 1, 6]; "
"audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"rate = (int) [ 8000, 96000], " "channels = (int) [ 1, 6]"
#endif
#define SRC_CAPS \
"audio/mpeg, " \
"mpegversion = (int) 4, " \
"channels = (int) [ 1, 6 ], " \
"rate = (int) {" SAMPLE_RATES "}, " \
"stream-format = (string) { adts, raw }, " \
"base-profile = (string) { main, lc, ssr, ltp }; " \
"audio/mpeg, " \
"mpegversion = (int) 2, " \
"channels = (int) [ 1, 6 ], " \
"rate = (int) {" SAMPLE_RATES "}, " \
"stream-format = (string) { adts, raw }, " \
"profile = (string) { main, lc }"
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SRC_CAPS));
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SINK_CAPS));
enum
{
PROP_0,
PROP_QUALITY,
PROP_BITRATE,
PROP_RATE_CONTROL,
PROP_PROFILE,
PROP_TNS,
PROP_MIDSIDE,
PROP_SHORTCTL
};
enum
{
VBR = 1,
ABR
};
static void gst_faac_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_faac_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_faac_configure_source_pad (GstFaac * faac);
static GstCaps *gst_faac_getcaps (GstAudioEncoder * enc);
static gboolean gst_faac_start (GstAudioEncoder * enc);
static gboolean gst_faac_stop (GstAudioEncoder * enc);
static gboolean gst_faac_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_faac_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
GST_DEBUG_CATEGORY_STATIC (faac_debug);
#define GST_CAT_DEFAULT faac_debug
#define FAAC_DEFAULT_QUALITY 100
#define FAAC_DEFAULT_BITRATE 128 * 1000
#define FAAC_DEFAULT_RATE_CONTROL VBR
#define FAAC_DEFAULT_TNS FALSE
#define FAAC_DEFAULT_MIDSIDE TRUE
#define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL
GST_BOILERPLATE (GstFaac, gst_faac, GstAudioEncoder, GST_TYPE_AUDIO_ENCODER);
static void
gst_faac_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_add_static_pad_template (element_class,
&sink_template);
gst_element_class_set_details_simple (element_class, "AAC audio encoder",
"Codec/Encoder/Audio",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
}
#define GST_TYPE_FAAC_RATE_CONTROL (gst_faac_brtype_get_type ())
static GType
gst_faac_brtype_get_type (void)
{
static GType gst_faac_brtype_type = 0;
if (!gst_faac_brtype_type) {
static GEnumValue gst_faac_brtype[] = {
{VBR, "VBR", "VBR encoding"},
{ABR, "ABR", "ABR encoding"},
{0, NULL, NULL},
};
gst_faac_brtype_type = g_enum_register_static ("GstFaacBrtype",
gst_faac_brtype);
}
return gst_faac_brtype_type;
}
#define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ())
static GType
gst_faac_shortctl_get_type (void)
{
static GType gst_faac_shortctl_type = 0;
if (!gst_faac_shortctl_type) {
static GEnumValue gst_faac_shortctl[] = {
{SHORTCTL_NORMAL, "SHORTCTL_NORMAL", "Normal block type"},
{SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks"},
{SHORTCTL_NOLONG, "SHORTCTL_NOLONG", "No long blocks"},
{0, NULL, NULL},
};
gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl",
gst_faac_shortctl);
}
return gst_faac_shortctl_type;
}
static void
gst_faac_class_init (GstFaacClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_faac_set_property;
gobject_class->get_property = gst_faac_get_property;
base_class->start = GST_DEBUG_FUNCPTR (gst_faac_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_faac_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_faac_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_faac_handle_frame);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_faac_getcaps);
/* properties */
g_object_class_install_property (gobject_class, PROP_QUALITY,
g_param_spec_int ("quality", "Quality (%)",
"Variable bitrate (VBR) quantizer quality in %", 1, 1000,
FAAC_DEFAULT_QUALITY,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (bps)",
"Average Bitrate (ABR) in bits/sec", 8 * 1000, 320 * 1000,
FAAC_DEFAULT_BITRATE,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RATE_CONTROL,
g_param_spec_enum ("rate-control", "Rate Control (ABR/VBR)",
"Encoding bitrate type (VBR/ABR)", GST_TYPE_FAAC_RATE_CONTROL,
FAAC_DEFAULT_RATE_CONTROL,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TNS,
g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
FAAC_DEFAULT_TNS,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MIDSIDE,
g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
FAAC_DEFAULT_MIDSIDE,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SHORTCTL,
g_param_spec_enum ("shortctl", "Block type",
"Block type encorcing",
GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_faac_init (GstFaac * faac, GstFaacClass * klass)
{
}
static void
gst_faac_close_encoder (GstFaac * faac)
{
if (faac->handle)
faacEncClose (faac->handle);
faac->handle = NULL;
}
static gboolean
gst_faac_start (GstAudioEncoder * enc)
{
GstFaac *faac = GST_FAAC (enc);
GST_DEBUG_OBJECT (faac, "start");
return TRUE;
}
static gboolean
gst_faac_stop (GstAudioEncoder * enc)
{
GstFaac *faac = GST_FAAC (enc);
GST_DEBUG_OBJECT (faac, "stop");
gst_faac_close_encoder (faac);
return TRUE;
}
static const GstAudioChannelPosition aac_channel_positions[][8] = {
{GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE}
};
static GstCaps *
gst_faac_getcaps (GstAudioEncoder * enc)
{
static volatile gsize sinkcaps = 0;
if (g_once_init_enter (&sinkcaps)) {
GstCaps *tmp = gst_caps_new_empty ();
GstStructure *s, *t;
gint i, c;
static const int rates[] = {
8000, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
GValue rates_arr = { 0, };
GValue tmp_v = { 0, };
g_value_init (&rates_arr, GST_TYPE_LIST);
g_value_init (&tmp_v, G_TYPE_INT);
for (i = 0; i < G_N_ELEMENTS (rates); i++) {
g_value_set_int (&tmp_v, rates[i]);
gst_value_list_append_value (&rates_arr, &tmp_v);
}
g_value_unset (&tmp_v);
s = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16, "depth", G_TYPE_INT, 16, NULL);
gst_structure_set_value (s, "rate", &rates_arr);
for (i = 1; i <= 6; i++) {
GValue chanpos = { 0 };
GValue pos = { 0 };
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
g_value_init (&chanpos, GST_TYPE_ARRAY);
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (c = 0; c < i; c++) {
g_value_set_enum (&pos, aac_channel_positions[i - 1][c]);
gst_value_array_append_value (&chanpos, &pos);
}
g_value_unset (&pos);
gst_structure_set_value (t, "channel-positions", &chanpos);
g_value_unset (&chanpos);
gst_caps_append_structure (tmp, t);
}
gst_structure_free (s);
g_value_unset (&rates_arr);
GST_DEBUG_OBJECT (enc, "Generated sinkcaps: %" GST_PTR_FORMAT, tmp);
g_once_init_leave (&sinkcaps, (gsize) tmp);
}
return gst_audio_encoder_proxy_getcaps (enc, (GstCaps *) sinkcaps);
}
static gboolean
gst_faac_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstFaac *faac = GST_FAAC (enc);
faacEncHandle *handle;
gint channels, samplerate, width;
gulong samples, bytes, fmt = 0, bps = 0;
gboolean result = FALSE;
/* base class takes care */
channels = GST_AUDIO_INFO_CHANNELS (info);
samplerate = GST_AUDIO_INFO_RATE (info);
width = GST_AUDIO_INFO_WIDTH (info);
if (GST_AUDIO_INFO_IS_INTEGER (info)) {
switch (width) {
case 16:
fmt = FAAC_INPUT_16BIT;
bps = 2;
break;
case 24:
case 32:
fmt = FAAC_INPUT_32BIT;
bps = 4;
break;
default:
g_return_val_if_reached (FALSE);
}
} else {
fmt = FAAC_INPUT_FLOAT;
bps = 4;
}
/* clean up in case of re-configure */
gst_faac_close_encoder (faac);
if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes)))
goto setup_failed;
/* mind channel count */
samples /= channels;
/* ok, record and set up */
faac->format = fmt;
faac->bps = bps;
faac->handle = handle;
faac->bytes = bytes;
faac->samples = samples;
faac->channels = channels;
faac->samplerate = samplerate;
/* finish up */
result = gst_faac_configure_source_pad (faac);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (enc, samples);
gst_audio_encoder_set_frame_samples_max (enc, samples);
gst_audio_encoder_set_frame_max (enc, 1);
done:
return result;
/* ERRORS */
setup_failed:
{
GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL));
goto done;
}
}
/* check downstream caps to configure format */
static void
gst_faac_negotiate (GstFaac * faac)
{
GstCaps *caps;
/* default setup */
faac->profile = LOW;
faac->mpegversion = 4;
faac->outputformat = 0;
caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (faac));
GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, caps);
if (caps && gst_caps_get_size (caps) > 0) {
GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *str = NULL;
gint i = 4;
if ((str = gst_structure_get_string (s, "stream-format"))) {
if (strcmp (str, "adts") == 0) {
GST_DEBUG_OBJECT (faac, "use ADTS format for output");
faac->outputformat = 1;
} else if (strcmp (str, "raw") == 0) {
GST_DEBUG_OBJECT (faac, "use RAW format for output");
faac->outputformat = 0;
} else {
GST_DEBUG_OBJECT (faac, "unknown stream-format: %s", str);
faac->outputformat = 0;
}
}
if ((str = gst_structure_get_string (s, "profile"))) {
if (strcmp (str, "main") == 0) {
faac->profile = MAIN;
} else if (strcmp (str, "lc") == 0) {
faac->profile = LOW;
} else if (strcmp (str, "ssr") == 0) {
faac->profile = SSR;
} else if (strcmp (str, "ltp") == 0) {
faac->profile = LTP;
} else {
faac->profile = LOW;
}
}
if (!gst_structure_get_int (s, "mpegversion", &i) || i == 4) {
faac->mpegversion = 4;
} else {
faac->mpegversion = 2;
}
}
if (caps)
gst_caps_unref (caps);
}
static gboolean
gst_faac_configure_source_pad (GstFaac * faac)
{
GstCaps *srccaps;
gboolean ret = FALSE;
faacEncConfiguration *conf;
guint maxbitrate;
/* negotiate stream format */
gst_faac_negotiate (faac);
/* we negotiated caps update current configuration */
conf = faacEncGetCurrentConfiguration (faac->handle);
conf->mpegVersion = (faac->mpegversion == 4) ? MPEG4 : MPEG2;
conf->aacObjectType = faac->profile;
conf->allowMidside = faac->midside;
conf->useLfe = 0;
conf->useTns = faac->tns;
if (faac->brtype == VBR) {
conf->quantqual = faac->quality;
} else if (faac->brtype == ABR) {
conf->bitRate = faac->bitrate / faac->channels;
}
conf->inputFormat = faac->format;
conf->outputFormat = faac->outputformat;
conf->shortctl = faac->shortctl;
/* check, warn and correct if the max bitrate for the given samplerate is
* exceeded. Maximum of 6144 bit for a channel */
maxbitrate =
(unsigned int) (6144.0 * (double) faac->samplerate / (double) 1024.0 +
.5);
if (conf->bitRate > maxbitrate) {
GST_ELEMENT_WARNING (faac, RESOURCE, SETTINGS, (NULL),
("bitrate %lu exceeds maximum allowed bitrate of %u for samplerate %d. "
"Setting bitrate to %u", conf->bitRate, maxbitrate,
faac->samplerate, maxbitrate));
conf->bitRate = maxbitrate;
}
/* default 0 to start with, libfaac chooses based on bitrate */
conf->bandWidth = 0;
if (!faacEncSetConfiguration (faac->handle, conf))
goto set_failed;
/* let's see what really happened,
* note that this may not really match desired rate */
GST_DEBUG_OBJECT (faac, "average bitrate: %lu kbps",
(conf->bitRate + 500) / 1000 * faac->channels);
GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual);
GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth);
/* now create a caps for it all */
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, faac->mpegversion,
"channels", G_TYPE_INT, faac->channels,
"rate", G_TYPE_INT, faac->samplerate,
"stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"),
NULL);
/* DecoderSpecificInfo is only available for mpegversion=4 */
if (faac->mpegversion == 4) {
guint8 *config = NULL;
gulong config_len = 0;
/* get the config string */
GST_DEBUG_OBJECT (faac, "retrieving decoder info");
faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len);
if (!gst_codec_utils_aac_caps_set_level_and_profile (srccaps, config,
config_len)) {
free (config);
gst_caps_unref (srccaps);
goto invalid_codec_data;
}
if (!faac->outputformat) {
GstBuffer *codec_data;
/* copy it into a buffer */
codec_data = gst_buffer_new_and_alloc (config_len);
memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
/* add to caps */
gst_caps_set_simple (srccaps,
"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
gst_buffer_unref (codec_data);
}
free (config);
} else {
const gchar *profile;
/* Add least add the profile to the caps */
switch (faac->profile) {
case MAIN:
profile = "main";
break;
case LTP:
profile = "ltp";
break;
case SSR:
profile = "ssr";
break;
case LOW:
default:
profile = "lc";
break;
}
gst_caps_set_simple (srccaps, "profile", G_TYPE_STRING, profile, NULL);
/* FIXME: How to get the profile for mpegversion==2? */
}
GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps);
ret = gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (faac), srccaps);
gst_caps_unref (srccaps);
return ret;
/* ERROR */
set_failed:
{
GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration");
return FALSE;
}
invalid_codec_data:
{
GST_ERROR_OBJECT (faac, "Invalid codec data");
return FALSE;
}
}
static GstFlowReturn
gst_faac_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
{
GstFaac *faac = GST_FAAC (enc);
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
gint size, ret_size;
const guint8 *data;
out_buf = gst_buffer_new_and_alloc (faac->bytes);
if (G_LIKELY (in_buf)) {
data = GST_BUFFER_DATA (in_buf);
size = GST_BUFFER_SIZE (in_buf);
} else {
data = NULL;
size = 0;
}
if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
size / faac->bps, GST_BUFFER_DATA (out_buf),
GST_BUFFER_SIZE (out_buf))) < 0))
goto encode_failed;
GST_LOG_OBJECT (faac, "encoder return: %d", ret_size);
if (G_LIKELY (ret_size > 0)) {
GST_BUFFER_SIZE (out_buf) = ret_size;
ret = gst_audio_encoder_finish_frame (enc, out_buf, faac->samples);
} else {
gst_buffer_unref (out_buf);
}
return ret;
/* ERRORS */
encode_failed:
{
gst_buffer_unref (out_buf);
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
return GST_FLOW_ERROR;
}
}
static void
gst_faac_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstFaac *faac = GST_FAAC (object);
GST_OBJECT_LOCK (faac);
switch (prop_id) {
case PROP_QUALITY:
faac->quality = g_value_get_int (value);
break;
case PROP_BITRATE:
faac->bitrate = g_value_get_int (value);
break;
case PROP_RATE_CONTROL:
faac->brtype = g_value_get_enum (value);
break;
case PROP_TNS:
faac->tns = g_value_get_boolean (value);
break;
case PROP_MIDSIDE:
faac->midside = g_value_get_boolean (value);
break;
case PROP_SHORTCTL:
faac->shortctl = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (faac);
}
static void
gst_faac_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstFaac *faac = GST_FAAC (object);
GST_OBJECT_LOCK (faac);
switch (prop_id) {
case PROP_QUALITY:
g_value_set_int (value, faac->quality);
break;
case PROP_BITRATE:
g_value_set_int (value, faac->bitrate);
break;
case PROP_RATE_CONTROL:
g_value_set_enum (value, faac->brtype);
break;
case PROP_TNS:
g_value_set_boolean (value, faac->tns);
break;
case PROP_MIDSIDE:
g_value_set_boolean (value, faac->midside);
break;
case PROP_SHORTCTL:
g_value_set_enum (value, faac->shortctl);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (faac);
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "faac", GST_RANK_SECONDARY,
GST_TYPE_FAAC);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"faac",
"Free AAC Encoder (FAAC)",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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