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gstreamer/subprojects/gst-plugins-bad/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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-faac
* @title: faac
* @see_also: faad
*
* faac encodes raw audio to AAC (MPEG-4 part 3) streams.
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac ! matroskamux ! filesink location=sine.mkv
* ]| Encode a sine beep as aac and write to matroska container.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/audio/audio.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"
/* 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 }, " \
"framed = (boolean) true; " \
"audio/mpeg, " \
"mpegversion = (int) 2, " \
"channels = (int) [ 1, 6 ], " \
"rate = (int) {" SAMPLE_RATES "}, " \
"stream-format = (string) { adts, raw }, " \
"profile = (string) { main, lc }," \
"framed = (boolean) true; "
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SRC_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 GstCaps *gst_faac_enc_generate_sink_caps (void);
static gboolean gst_faac_configure_source_pad (GstFaac * faac,
GstAudioInfo * info);
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
#define gst_faac_parent_class parent_class
G_DEFINE_TYPE (GstFaac, gst_faac, GST_TYPE_AUDIO_ENCODER);
GST_ELEMENT_REGISTER_DEFINE (faac, "faac", GST_RANK_SECONDARY, GST_TYPE_FAAC);
#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 const 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 const 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);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
GstCaps *sink_caps;
GstPadTemplate *sink_templ;
gobject_class->set_property = gst_faac_set_property;
gobject_class->get_property = gst_faac_get_property;
GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
sink_caps = gst_faac_enc_generate_sink_caps ();
sink_templ = gst_pad_template_new ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, sink_caps);
gst_element_class_add_pad_template (gstelement_class, sink_templ);
gst_caps_unref (sink_caps);
gst_element_class_set_static_metadata (gstelement_class, "AAC audio encoder",
"Codec/Encoder/Audio",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
gst_type_mark_as_plugin_api (GST_TYPE_FAAC_RATE_CONTROL, 0);
gst_type_mark_as_plugin_api (GST_TYPE_FAAC_SHORTCTL, 0);
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);
/* 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)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (faac));
}
static void
gst_faac_close_encoder (GstFaac * faac)
{
if (faac->handle)
faacEncClose (faac->handle);
faac->handle = NULL;
}
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_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_LFE1}
};
static GstCaps *
gst_faac_enc_generate_sink_caps (void)
{
GstCaps *caps = 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",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_structure_set_value (s, "rate", &rates_arr);
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, 1, NULL);
gst_caps_append_structure (caps, t);
for (i = 2; i <= 6; i++) {
guint64 channel_mask = 0;
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
for (c = 0; c < i; c++)
channel_mask |= G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask, NULL);
gst_caps_append_structure (caps, t);
}
gst_structure_free (s);
g_value_unset (&rates_arr);
GST_DEBUG ("Generated sinkcaps: %" GST_PTR_FORMAT, caps);
return caps;
}
static void
gst_faac_set_tags (GstFaac * faac)
{
GstTagList *taglist;
/* create a taglist and add a bitrate tag to it */
taglist = gst_tag_list_new_empty ();
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, faac->bitrate, NULL);
gst_audio_encoder_merge_tags (GST_AUDIO_ENCODER (faac), taglist,
GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (taglist);
}
static gboolean
gst_faac_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstFaac *faac = GST_FAAC (enc);
gint width;
gulong fmt = 0;
gboolean result = FALSE;
/* base class takes care */
width = GST_AUDIO_INFO_WIDTH (info);
if (GST_AUDIO_INFO_IS_INTEGER (info)) {
switch (width) {
case 16:
fmt = FAAC_INPUT_16BIT;
break;
case 24:
case 32:
fmt = FAAC_INPUT_32BIT;
break;
default:
g_return_val_if_reached (FALSE);
}
} else {
fmt = FAAC_INPUT_FLOAT;
}
faac->format = fmt;
/* finish up */
result = gst_faac_configure_source_pad (faac, info);
if (!result)
goto done;
gst_faac_set_tags (faac);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (enc, faac->samples);
gst_audio_encoder_set_frame_samples_max (enc, faac->samples);
gst_audio_encoder_set_frame_max (enc, 1);
done:
return result;
}
/* 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_open_encoder (GstFaac * faac, GstAudioInfo * info)
{
faacEncHandle *handle;
faacEncConfiguration *conf;
guint maxbitrate;
gulong samples, bytes;
g_return_val_if_fail (info->rate != 0 && info->channels != 0, FALSE);
/* clean up in case of re-configure */
gst_faac_close_encoder (faac);
if (!(handle = faacEncOpen (info->rate, info->channels, &samples, &bytes)))
goto setup_failed;
/* mind channel count */
samples /= info->channels;
/* record */
faac->handle = handle;
faac->samples = samples;
faac->bytes = bytes;
GST_DEBUG_OBJECT (faac, "faac needs samples %d, output size %d",
faac->samples, faac->bytes);
/* 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 / info->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) info->rate / (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,
info->rate, maxbitrate));
conf->bitRate = maxbitrate;
}
/* default 0 to start with, libfaac chooses based on bitrate */
conf->bandWidth = 0;
if (!faacEncSetConfiguration (faac->handle, conf))
goto setup_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 * info->channels);
GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual);
GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth);
return TRUE;
/* ERRORS */
setup_failed:
{
GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL));
return FALSE;
}
}
static gboolean
gst_faac_configure_source_pad (GstFaac * faac, GstAudioInfo * info)
{
GstCaps *srccaps;
gboolean ret;
/* negotiate stream format */
gst_faac_negotiate (faac);
if (!gst_faac_open_encoder (faac, info))
goto set_failed;
/* now create a caps for it all */
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, faac->mpegversion,
"channels", G_TYPE_INT, info->channels,
"rate", G_TYPE_INT, info->rate,
"stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"),
"framed", G_TYPE_BOOLEAN, TRUE, 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);
gst_buffer_fill (codec_data, 0, 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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (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;
gsize size, ret_size;
int enc_ret;
GstMapInfo map, omap;
guint8 *data;
GstAudioInfo *info =
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (faac));
out_buf = gst_buffer_new_and_alloc (faac->bytes);
gst_buffer_map (out_buf, &omap, GST_MAP_WRITE);
if (G_LIKELY (in_buf)) {
if (memcmp (info->position, aac_channel_positions[info->channels - 1],
sizeof (GstAudioChannelPosition) * info->channels) != 0) {
in_buf = gst_buffer_make_writable (in_buf);
gst_audio_buffer_reorder_channels (in_buf, info->finfo->format,
info->channels, info->position,
aac_channel_positions[info->channels - 1]);
}
gst_buffer_map (in_buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
} else {
data = NULL;
size = 0;
}
if (G_UNLIKELY ((enc_ret = faacEncEncode (faac->handle, (gint32 *) data,
size / (info->finfo->width / 8), omap.data, omap.size)) < 0))
goto encode_failed;
ret_size = enc_ret;
if (in_buf)
gst_buffer_unmap (in_buf, &map);
GST_LOG_OBJECT (faac, "encoder return: %" G_GSIZE_FORMAT, ret_size);
if (ret_size > 0) {
gst_buffer_unmap (out_buf, &omap);
gst_buffer_resize (out_buf, 0, ret_size);
ret = gst_audio_encoder_finish_frame (enc, out_buf, faac->samples);
} else {
gst_buffer_unmap (out_buf, &omap);
gst_buffer_unref (out_buf);
/* re-create encoder after final flush */
if (!in_buf) {
GST_DEBUG_OBJECT (faac, "flushed; recreating encoder");
gst_faac_close_encoder (faac);
if (!gst_faac_open_encoder (faac, gst_audio_encoder_get_audio_info (enc)))
ret = GST_FLOW_ERROR;
}
}
return ret;
/* ERRORS */
encode_failed:
{
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
if (in_buf)
gst_buffer_unmap (in_buf, &map);
gst_buffer_unmap (out_buf, &omap);
gst_buffer_unref (out_buf);
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 (faac, plugin);
}
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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