gstreamer/ext/faac/gstfaac.c
Mark Nauwelaerts c7b374abcb ext/faac/gstfaac.c: Prevent endless loop if buffer alloc error.
Original commit message from CVS:
* ext/faac/gstfaac.c: (gst_faac_sink_event):
Prevent endless loop if buffer alloc error.
2008-10-27 16:32:49 +00:00

808 lines
23 KiB
C

/* GStreamer FAAC (Free AAC Encoder) plugin
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstfaac.h"
#define SINK_CAPS \
"audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"rate = (int) [ 8000, 96000 ], " \
"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, 2 }, " \
"channels = (int) [ 1, 6 ], " \
"rate = (int) [ 8000, 96000 ]"
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));
static const GstElementDetails gst_faac_details =
GST_ELEMENT_DETAILS ("AAC audio encoder",
"Codec/Encoder/Audio",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
enum
{
ARG_0,
ARG_OUTPUTFORMAT,
ARG_BITRATE,
ARG_PROFILE,
ARG_TNS,
ARG_MIDSIDE,
ARG_SHORTCTL
};
static void gst_faac_base_init (GstFaacClass * klass);
static void gst_faac_class_init (GstFaacClass * klass);
static void gst_faac_init (GstFaac * faac);
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_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstFlowReturn gst_faac_chain (GstPad * pad, GstBuffer * data);
static GstStateChangeReturn gst_faac_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
GST_DEBUG_CATEGORY_STATIC (faac_debug);
#define GST_CAT_DEFAULT faac_debug
#define FAAC_DEFAULT_MPEGVERSION 4
GType
gst_faac_get_type (void)
{
static GType gst_faac_type = 0;
if (!gst_faac_type) {
static const GTypeInfo gst_faac_info = {
sizeof (GstFaacClass),
(GBaseInitFunc) gst_faac_base_init,
NULL,
(GClassInitFunc) gst_faac_class_init,
NULL,
NULL,
sizeof (GstFaac),
0,
(GInstanceInitFunc) gst_faac_init,
};
gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstFaac", &gst_faac_info, 0);
}
return gst_faac_type;
}
static void
gst_faac_base_init (GstFaacClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details (element_class, &gst_faac_details);
GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
}
#define GST_TYPE_FAAC_PROFILE (gst_faac_profile_get_type ())
static GType
gst_faac_profile_get_type (void)
{
static GType gst_faac_profile_type = 0;
if (!gst_faac_profile_type) {
static GEnumValue gst_faac_profile[] = {
{MAIN, "MAIN", "Main profile"},
{LOW, "LC", "Low complexity profile"},
{SSR, "SSR", "Scalable sampling rate profile"},
{LTP, "LTP", "Long term prediction profile"},
{0, NULL, NULL},
};
gst_faac_profile_type = g_enum_register_static ("GstFaacProfile",
gst_faac_profile);
}
return gst_faac_profile_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;
}
#define GST_TYPE_FAAC_OUTPUTFORMAT (gst_faac_outputformat_get_type ())
static GType
gst_faac_outputformat_get_type (void)
{
static GType gst_faac_outputformat_type = 0;
if (!gst_faac_outputformat_type) {
static GEnumValue gst_faac_outputformat[] = {
{0, "OUTPUTFORMAT_RAW", "Raw AAC"},
{1, "OUTPUTFORMAT_ADTS", "ADTS headers"},
{0, NULL, NULL},
};
gst_faac_outputformat_type = g_enum_register_static ("GstFaacOutputFormat",
gst_faac_outputformat);
}
return gst_faac_outputformat_type;
}
static void
gst_faac_class_init (GstFaacClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_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;
/* properties */
g_object_class_install_property (gobject_class, ARG_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
8 * 1000, 320 * 1000, 128 * 1000, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_PROFILE,
g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile",
GST_TYPE_FAAC_PROFILE, MAIN, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_TNS,
g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
FALSE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_MIDSIDE,
g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
TRUE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_SHORTCTL,
g_param_spec_enum ("shortctl", "Block type",
"Block type encorcing",
GST_TYPE_FAAC_SHORTCTL, MAIN, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_OUTPUTFORMAT,
g_param_spec_enum ("outputformat", "Output format",
"Format of output frames",
GST_TYPE_FAAC_OUTPUTFORMAT, MAIN, G_PARAM_READWRITE));
/* virtual functions */
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faac_change_state);
}
static void
gst_faac_init (GstFaac * faac)
{
faac->handle = NULL;
faac->samplerate = -1;
faac->channels = -1;
faac->cache = NULL;
faac->cache_time = GST_CLOCK_TIME_NONE;
faac->cache_duration = 0;
faac->next_ts = GST_CLOCK_TIME_NONE;
faac->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_chain));
gst_pad_set_setcaps_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_sink_setcaps));
gst_pad_set_event_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_sink_event));
gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
faac->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (faac->srcpad);
gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
/* default properties */
faac->bitrate = 1000 * 128;
faac->profile = MAIN;
faac->shortctl = SHORTCTL_NORMAL;
faac->outputformat = 0;
faac->tns = FALSE;
faac->midside = TRUE;
}
static gboolean
gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
faacEncHandle *handle;
gint channels, samplerate, width;
gulong samples, bytes, fmt = 0, bps = 0;
gboolean result = FALSE;
if (!gst_caps_is_fixed (caps))
goto done;
GST_OBJECT_LOCK (faac);
if (faac->handle) {
faacEncClose (faac->handle);
faac->handle = NULL;
}
if (faac->cache) {
gst_buffer_unref (faac->cache);
faac->cache = NULL;
}
GST_OBJECT_UNLOCK (faac);
if (!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "rate", &samplerate)) {
goto done;
}
if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes)))
goto done;
if (gst_structure_has_name (structure, "audio/x-raw-int")) {
gst_structure_get_int (structure, "width", &width);
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 if (gst_structure_has_name (structure, "audio/x-raw-float")) {
fmt = FAAC_INPUT_FLOAT;
bps = 4;
}
if (!fmt) {
faacEncClose (handle);
goto done;
}
GST_OBJECT_LOCK (faac);
faac->format = fmt;
faac->bps = bps;
faac->handle = handle;
faac->bytes = bytes;
faac->samples = samples;
faac->channels = channels;
faac->samplerate = samplerate;
GST_OBJECT_UNLOCK (faac);
result = TRUE;
done:
gst_object_unref (faac);
return result;
}
static gboolean
gst_faac_configure_source_pad (GstFaac * faac)
{
GstCaps *allowed_caps;
GstCaps *srccaps;
gboolean ret = FALSE;
gint n, ver, mpegversion;
faacEncConfiguration *conf;
guint maxbitrate;
mpegversion = FAAC_DEFAULT_MPEGVERSION;
allowed_caps = gst_pad_get_allowed_caps (faac->srcpad);
GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
if (allowed_caps == NULL)
return FALSE;
if (gst_caps_is_empty (allowed_caps))
goto empty_caps;
if (!gst_caps_is_any (allowed_caps)) {
for (n = 0; n < gst_caps_get_size (allowed_caps); n++) {
GstStructure *s = gst_caps_get_structure (allowed_caps, n);
if (gst_structure_get_int (s, "mpegversion", &ver) &&
(ver == 4 || ver == 2)) {
mpegversion = ver;
break;
}
}
}
gst_caps_unref (allowed_caps);
/* we negotiated caps update current configuration */
conf = faacEncGetCurrentConfiguration (faac->handle);
conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
conf->aacObjectType = faac->profile;
conf->allowMidside = faac->midside;
conf->useLfe = 0;
conf->useTns = faac->tns;
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 %u exceeds maximum allowed bitrate of %u for samplerate %d. "
"Setting bitrate to %u", conf->bitRate, maxbitrate,
faac->samplerate, maxbitrate));
conf->bitRate = maxbitrate;
}
if (!faacEncSetConfiguration (faac->handle, conf))
goto set_failed;
/* now create a caps for it all */
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, mpegversion,
"channels", G_TYPE_INT, faac->channels,
"rate", G_TYPE_INT, faac->samplerate, NULL);
if (mpegversion == 4) {
GstBuffer *codec_data;
guint8 *config = NULL;
gulong config_len = 0;
/* get the config string */
GST_DEBUG_OBJECT (faac, "retrieving decoder info");
faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len);
/* 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);
}
GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps);
ret = gst_pad_set_caps (faac->srcpad, srccaps);
gst_caps_unref (srccaps);
return ret;
/* ERROR */
empty_caps:
{
gst_caps_unref (allowed_caps);
return FALSE;
}
set_failed:
{
GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration");
return FALSE;
}
}
static gboolean
gst_faac_sink_event (GstPad * pad, GstEvent * event)
{
GstFaac *faac;
gboolean ret;
faac = GST_FAAC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
{
GstBuffer *outbuf;
if (!faac->handle)
ret = FALSE;
else
ret = TRUE;
/* flush first */
GST_DEBUG ("Pushing out remaining buffers because of EOS");
while (ret) {
if (gst_pad_alloc_buffer_and_set_caps (faac->srcpad,
GST_BUFFER_OFFSET_NONE, faac->bytes,
GST_PAD_CAPS (faac->srcpad), &outbuf) == GST_FLOW_OK) {
gint ret_size;
GST_DEBUG ("next_ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (faac->next_ts));
if ((ret_size = faacEncEncode (faac->handle, NULL, 0,
GST_BUFFER_DATA (outbuf), faac->bytes)) > 0) {
GST_BUFFER_SIZE (outbuf) = ret_size;
GST_BUFFER_TIMESTAMP (outbuf) = faac->next_ts;
/* faac seems to always consume a fixed number of input samples,
* therefore extrapolate the duration from that value and the incoming
* bitrate */
GST_BUFFER_DURATION (outbuf) = gst_util_uint64_scale (faac->samples,
GST_SECOND, faac->channels * faac->samplerate);
if (GST_CLOCK_TIME_IS_VALID (faac->next_ts))
faac->next_ts += GST_BUFFER_DURATION (outbuf);
gst_pad_push (faac->srcpad, outbuf);
} else {
gst_buffer_unref (outbuf);
ret = FALSE;
}
} else
ret = FALSE;
}
ret = gst_pad_event_default (pad, event);
break;
}
case GST_EVENT_NEWSEGMENT:
ret = gst_pad_push_event (faac->srcpad, event);
break;
case GST_EVENT_TAG:
ret = gst_pad_event_default (pad, event);
break;
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (faac);
return ret;
}
static GstFlowReturn
gst_faac_chain (GstPad * pad, GstBuffer * inbuf)
{
GstFlowReturn result = GST_FLOW_OK;
GstBuffer *outbuf, *subbuf;
GstFaac *faac;
guint size, ret_size, in_size, frame_size;
faac = GST_FAAC (gst_pad_get_parent (pad));
if (!faac->handle)
goto no_handle;
if (!GST_PAD_CAPS (faac->srcpad)) {
if (!gst_faac_configure_source_pad (faac))
goto nego_failed;
}
GST_DEBUG ("Got buffer time:%" GST_TIME_FORMAT " duration:%" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
size = GST_BUFFER_SIZE (inbuf);
in_size = size;
if (faac->cache)
in_size += GST_BUFFER_SIZE (faac->cache);
frame_size = faac->samples * faac->bps;
while (1) {
/* do we have enough data for one frame? */
if (in_size / faac->bps < faac->samples) {
if (in_size > size) {
GstBuffer *merge;
/* this is panic! we got a buffer, but still don't have enough
* data. Merge them and retry in the next cycle... */
merge = gst_buffer_merge (faac->cache, inbuf);
gst_buffer_unref (faac->cache);
gst_buffer_unref (inbuf);
faac->cache = merge;
} else if (in_size == size) {
/* this shouldn't happen, but still... */
faac->cache = inbuf;
} else if (in_size > 0) {
faac->cache = gst_buffer_create_sub (inbuf, size - in_size, in_size);
GST_BUFFER_DURATION (faac->cache) =
GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (faac->cache) / size;
GST_BUFFER_TIMESTAMP (faac->cache) =
GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
(size - in_size) / size);
gst_buffer_unref (inbuf);
} else {
gst_buffer_unref (inbuf);
}
goto done;
}
/* create the frame */
if (in_size > size) {
GstBuffer *merge;
/* merge */
subbuf = gst_buffer_create_sub (inbuf, 0, frame_size - (in_size - size));
GST_BUFFER_DURATION (subbuf) =
GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
merge = gst_buffer_merge (faac->cache, subbuf);
gst_buffer_unref (faac->cache);
gst_buffer_unref (subbuf);
subbuf = merge;
faac->cache = NULL;
} else {
subbuf = gst_buffer_create_sub (inbuf, size - in_size, frame_size);
GST_BUFFER_DURATION (subbuf) =
GST_BUFFER_DURATION (inbuf) * GST_BUFFER_SIZE (subbuf) / size;
GST_BUFFER_TIMESTAMP (subbuf) =
GST_BUFFER_TIMESTAMP (inbuf) + (GST_BUFFER_DURATION (inbuf) *
(size - in_size) / size);
}
result =
gst_pad_alloc_buffer_and_set_caps (faac->srcpad, GST_BUFFER_OFFSET_NONE,
faac->bytes, GST_PAD_CAPS (faac->srcpad), &outbuf);
if (result != GST_FLOW_OK)
goto done;
if ((ret_size = faacEncEncode (faac->handle,
(gint32 *) GST_BUFFER_DATA (subbuf),
GST_BUFFER_SIZE (subbuf) / faac->bps,
GST_BUFFER_DATA (outbuf), faac->bytes)) < 0) {
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
gst_buffer_unref (inbuf);
gst_buffer_unref (subbuf);
result = GST_FLOW_ERROR;
goto done;
}
if (ret_size > 0) {
GST_BUFFER_SIZE (outbuf) = ret_size;
if (faac->cache_time != GST_CLOCK_TIME_NONE) {
GST_BUFFER_TIMESTAMP (outbuf) = faac->cache_time;
faac->cache_time = GST_CLOCK_TIME_NONE;
} else
GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (subbuf);
GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (subbuf);
if (faac->cache_duration) {
GST_BUFFER_DURATION (outbuf) += faac->cache_duration;
faac->cache_duration = 0;
}
/* Store the value of the next expected timestamp to output
* This is required in order to output the trailing encoded packets
* at EOS with proper timestamps and duration. */
faac->next_ts =
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
GST_DEBUG ("Pushing out buffer time:%" GST_TIME_FORMAT " duration:%"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
result = gst_pad_push (faac->srcpad, outbuf);
} else {
/* FIXME: what I'm doing here isn't fully correct, but there
* really isn't a better way yet.
* Problem is that libfaac caches buffers (for encoding
* purposes), so the timestamp of the outgoing buffer isn't
* the same as the timestamp of the data that I pushed in.
* However, I don't know the delay between those two so I
* cannot really say aything about it. This is a bad guess. */
gst_buffer_unref (outbuf);
if (faac->cache_time != GST_CLOCK_TIME_NONE)
faac->cache_time = GST_BUFFER_TIMESTAMP (subbuf);
faac->cache_duration += GST_BUFFER_DURATION (subbuf);
}
in_size -= frame_size;
gst_buffer_unref (subbuf);
}
done:
gst_object_unref (faac);
return result;
/* ERRORS */
no_handle:
{
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before chain function"));
gst_buffer_unref (inbuf);
result = GST_FLOW_ERROR;
goto done;
}
nego_failed:
{
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
("failed to negotiate MPEG/AAC format with next element"));
gst_buffer_unref (inbuf);
result = GST_FLOW_ERROR;
goto done;
}
}
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 ARG_BITRATE:
faac->bitrate = g_value_get_int (value);
break;
case ARG_PROFILE:
faac->profile = g_value_get_enum (value);
break;
case ARG_TNS:
faac->tns = g_value_get_boolean (value);
break;
case ARG_MIDSIDE:
faac->midside = g_value_get_boolean (value);
break;
case ARG_SHORTCTL:
faac->shortctl = g_value_get_enum (value);
break;
case ARG_OUTPUTFORMAT:
faac->outputformat = 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 ARG_BITRATE:
g_value_set_int (value, faac->bitrate);
break;
case ARG_PROFILE:
g_value_set_enum (value, faac->profile);
break;
case ARG_TNS:
g_value_set_boolean (value, faac->tns);
break;
case ARG_MIDSIDE:
g_value_set_boolean (value, faac->midside);
break;
case ARG_SHORTCTL:
g_value_set_enum (value, faac->shortctl);
break;
case ARG_OUTPUTFORMAT:
g_value_set_enum (value, faac->outputformat);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (faac);
}
static GstStateChangeReturn
gst_faac_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstFaac *faac = GST_FAAC (element);
/* upwards state changes */
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
/* downwards state changes */
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
GST_OBJECT_LOCK (faac);
if (faac->handle) {
faacEncClose (faac->handle);
faac->handle = NULL;
}
if (faac->cache) {
gst_buffer_unref (faac->cache);
faac->cache = NULL;
}
faac->cache_time = GST_CLOCK_TIME_NONE;
faac->cache_duration = 0;
faac->samplerate = -1;
faac->channels = -1;
faac->next_ts = GST_CLOCK_TIME_NONE;
GST_OBJECT_UNLOCK (faac);
break;
}
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "faac", GST_RANK_NONE, 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)