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gstreamer/ext/faac/gstfaac.c
David Schleef d33b0d62aa Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). 
Original commit message from CVS:
Remove all usage of gst_pad_get_caps(), and replace it with
gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
2004-01-12 03:40:18 +00:00

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/* 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 "gstfaac.h"
static GstStaticPadTemplate src_template =
GST_STATIC_PAD_TEMPLATE (
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (
"audio/mpeg, "
"mpegversion = (int) { 4, 2 }, "
"channels = (int) [ 1, 6 ], "
"rate = (int) [ 8000, 96000 ]"
)
);
static GstStaticPadTemplate sink_template =
GST_STATIC_PAD_TEMPLATE (
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_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]; "
"audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) TRUE, "
"width = (int) 32, "
"depth = (int) 24, "
"rate = (int) [ 8000, 96000], "
"channels = (int) [ 1, 6]; "
"audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"depth = (int) 32, " /* sizeof (gfloat) */
"rate = (int) [ 8000, 96000], "
"channels = (int) [ 1, 6]"
)
);
enum {
ARG_0,
ARG_BITRATE,
ARG_PROFILE,
ARG_TNS,
ARG_MIDSIDE,
ARG_SHORTCTL
/* FILL ME */
};
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 GstPadLinkReturn
gst_faac_sinkconnect (GstPad *pad,
const GstCaps *caps);
static GstPadLinkReturn
gst_faac_srcconnect (GstPad *pad,
const GstCaps *caps);
static void gst_faac_chain (GstPad *pad,
GstData *data);
static GstElementStateReturn
gst_faac_change_state (GstElement *element);
static GstElementClass *parent_class = NULL;
/* static guint gst_faac_signals[LAST_SIGNAL] = { 0 }; */
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)
{
GstElementDetails gst_faac_details = {
"Free AAC Encoder (FAAC)",
"Codec/Audio/Encoder",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>",
};
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);
}
#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, "LOW", "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;
}
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_ref (GST_TYPE_ELEMENT);
/* properties */
g_object_class_install_property (gobject_class, ARG_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
8 * 1024, 320 * 1024, 128 * 1024, 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));
/* virtual functions */
gstelement_class->change_state = gst_faac_change_state;
gobject_class->set_property = gst_faac_set_property;
gobject_class->get_property = gst_faac_get_property;
}
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;
GST_FLAG_SET (faac, GST_ELEMENT_EVENT_AWARE);
faac->sinkpad = gst_pad_new_from_template (
gst_static_pad_template_get (&sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
gst_pad_set_chain_function (faac->sinkpad, gst_faac_chain);
gst_pad_set_link_function (faac->sinkpad, gst_faac_sinkconnect);
faac->srcpad = gst_pad_new_from_template (
gst_static_pad_template_get (&src_template), "src");
gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
gst_pad_set_link_function (faac->srcpad, gst_faac_srcconnect);
/* default properties */
faac->bitrate = 1024 * 128;
faac->profile = MAIN;
faac->shortctl = SHORTCTL_NORMAL;
faac->tns = FALSE;
faac->midside = TRUE;
}
static GstPadLinkReturn
gst_faac_sinkconnect (GstPad *pad,
const GstCaps *caps)
{
GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
faacEncHandle *handle;
gint channels, samplerate, depth;
gulong samples, bytes, fmt = 0, bps = 0;
if (!gst_caps_is_fixed (caps))
return GST_PAD_LINK_DELAYED;
if (faac->handle) {
faacEncClose (faac->handle);
faac->handle = NULL;
}
if (faac->cache) {
gst_buffer_unref (faac->cache);
faac->cache = NULL;
}
gst_structure_get_int (structure, "channels", &channels);
gst_structure_get_int (structure, "rate", &samplerate);
gst_structure_get_int (structure, "depth", &depth);
/* open a new handle to the encoder */
if (!(handle = faacEncOpen (samplerate, channels,
&samples, &bytes)))
return GST_PAD_LINK_REFUSED;
switch (depth) {
case 16:
fmt = FAAC_INPUT_16BIT;
bps = 2;
break;
case 24:
fmt = FAAC_INPUT_32BIT; /* 24-in-32, actually */
bps = 4;
break;
case 32:
fmt = FAAC_INPUT_FLOAT; /* see template, this is right */
bps = 4;
break;
}
if (!fmt) {
faacEncClose (handle);
return GST_PAD_LINK_REFUSED;
}
faac->format = fmt;
faac->bps = bps;
faac->handle = handle;
faac->bytes = bytes;
faac->samples = samples;
faac->channels = channels;
faac->samplerate = samplerate;
/* if the other side was already set-up, redo that */
if (GST_PAD_CAPS (faac->srcpad))
return gst_faac_srcconnect (faac->srcpad,
gst_pad_get_allowed_caps (faac->srcpad));
/* else, that'll be done later */
return GST_PAD_LINK_OK;
}
static GstPadLinkReturn
gst_faac_srcconnect (GstPad *pad,
const GstCaps *caps)
{
GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
gint n;
if (!faac->handle ||
(faac->samplerate == -1 || faac->channels == -1)) {
return GST_PAD_LINK_DELAYED;
}
/* we do samplerate/channels ourselves */
for (n = 0; n < gst_caps_get_size (caps); n++) {
GstStructure *structure = gst_caps_get_structure (caps, n);
gst_structure_remove_field (structure, "rate");
gst_structure_remove_field (structure, "channels");
}
/* go through list */
caps = gst_caps_normalize (caps);
for (n = 0; n < gst_caps_get_size (caps); n++) {
GstStructure *structure = gst_caps_get_structure (caps, n);
faacEncConfiguration *conf;
gint mpegversion = 0;
GstCaps *newcaps;
GstPadLinkReturn ret;
gst_structure_get_int (structure, "mpegversion", &mpegversion);
/* new conf */
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;
conf->inputFormat = faac->format;
/* FIXME: this one here means that we do not support direct
* "MPEG audio file" output (like mp3). This means we can
* only mux this into mov/qt (mp4a) or matroska or so. If
* we want to support direct AAC file output, we need ADTS
* headers, and we need to find a way in the caps to detect
* that (that the next element is filesink or any element
* that does want ADTS headers). */
conf->outputFormat = 0; /* raw, no ADTS headers */
conf->shortctl = faac->shortctl;
if (!faacEncSetConfiguration (faac->handle, conf)) {
GST_WARNING ("Faac doesn't support the current conf");
continue;
}
newcaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, mpegversion,
"channels", G_TYPE_INT, faac->channels,
"rate", G_TYPE_INT, faac->samplerate,
NULL);
ret = gst_pad_try_set_caps (faac->srcpad, newcaps);
switch (ret) {
case GST_PAD_LINK_OK:
case GST_PAD_LINK_DONE:
return GST_PAD_LINK_DONE;
case GST_PAD_LINK_DELAYED:
return GST_PAD_LINK_DELAYED;
default:
break;
}
}
return GST_PAD_LINK_REFUSED;
}
static void
gst_faac_chain (GstPad *pad,
GstData *data)
{
GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
GstBuffer *inbuf, *outbuf, *subbuf;
guint size, ret_size, in_size, frame_size;
if (GST_IS_EVENT (data)) {
GstEvent *event = GST_EVENT (data);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* flush first */
while (1) {
outbuf = gst_buffer_new_and_alloc (faac->bytes);
if ((ret_size = faacEncEncode (faac->handle,
NULL, 0,
GST_BUFFER_DATA (outbuf),
faac->bytes)) < 0) {
gst_element_error (GST_ELEMENT (faac), "Error during AAC encoding");
gst_event_unref (event);
gst_buffer_unref (outbuf);
return;
}
if (ret_size > 0) {
GST_BUFFER_SIZE (outbuf) = ret_size;
GST_BUFFER_TIMESTAMP (outbuf) = 0;
GST_BUFFER_DURATION (outbuf) = 0;
gst_pad_push (faac->srcpad, GST_DATA (outbuf));
} else {
break;
}
}
gst_element_set_eos (GST_ELEMENT (faac));
gst_pad_push (faac->srcpad, data);
return;
default:
gst_pad_event_default (pad, event);
return;
}
}
inbuf = GST_BUFFER (data);
if (!faac->handle) {
gst_element_error (GST_ELEMENT (faac),
"No input format negotiated");
gst_buffer_unref (inbuf);
return;
}
if (!GST_PAD_CAPS (faac->srcpad)) {
if (gst_faac_srcconnect (faac->srcpad,
gst_pad_get_allowed_caps (faac->srcpad)) <= 0) {
gst_element_error (GST_ELEMENT (faac),
"Failed to negotiate MPEG/AAC format with next element");
gst_buffer_unref (inbuf);
return;
}
}
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) {
/* this is panic! we got a buffer, but still don't have enough
* data. Merge them and retry in the next cycle... */
faac->cache = gst_buffer_merge (faac->cache, inbuf);
} 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);
}
return;
}
/* create the frame */
if (in_size > size) {
/* 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;
subbuf = gst_buffer_merge (faac->cache, subbuf);
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);
}
outbuf = gst_buffer_new_and_alloc (faac->bytes);
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 (GST_ELEMENT (faac), "Error during AAC encoding");
gst_buffer_unref (inbuf);
gst_buffer_unref (subbuf);
return;
}
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;
}
gst_pad_push (faac->srcpad, GST_DATA (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);
}
}
static void
gst_faac_set_property (GObject *object,
guint prop_id,
const GValue *value,
GParamSpec *pspec)
{
GstFaac *faac = GST_FAAC (object);
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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_faac_get_property (GObject *object,
guint prop_id,
GValue *value,
GParamSpec *pspec)
{
GstFaac *faac = GST_FAAC (object);
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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstElementStateReturn
gst_faac_change_state (GstElement *element)
{
GstFaac *faac = GST_FAAC (element);
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_PAUSED_TO_READY:
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;
break;
default:
break;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
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,
GST_ORIGIN
)
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