gstreamer/ext/a52dec/gsta52dec.c
Andy Wingo 8349527f92 Way, way, way too many files: Remove crack comment from the 2000 era.
Original commit message from CVS:
2005-07-05  Andy Wingo  <wingo@pobox.com>

* Way, way, way too many files:
Remove crack comment from the 2000 era.
2005-07-05 10:51:41 +00:00

612 lines
16 KiB
C

/* GStreamer
* Copyright (C) <2001> David I. Lehn <dlehn@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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include "_stdint.h"
#include <gst/gst.h>
#include <gst/audio/multichannel.h>
#include <a52dec/a52.h>
#include <a52dec/mm_accel.h>
#include "gsta52dec.h"
/* elementfactory information */
static GstElementDetails gst_a52dec_details = {
"ATSC A/52 audio decoder",
"Codec/Decoder/Audio",
"Decodes ATSC A/52 encoded audio streams",
"David I. Lehn <dlehn@users.sourceforge.net>",
};
#ifdef LIBA52_DOUBLE
#define SAMPLE_WIDTH 64
#else
#define SAMPLE_WIDTH 32
#endif
GST_DEBUG_CATEGORY_STATIC (a52dec_debug);
#define GST_CAT_DEFAULT (a52dec_debug)
/* A52Dec signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_DRC
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-ac3")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"width = (int) " G_STRINGIFY (SAMPLE_WIDTH) ", "
"rate = (int) [ 4000, 96000 ], "
"channels = (int) [ 1, 6 ], " "buffer-frames = (int) 0")
);
static void gst_a52dec_base_init (gpointer g_class);
static void gst_a52dec_class_init (GstA52DecClass * klass);
static void gst_a52dec_init (GstA52Dec * a52dec);
static void gst_a52dec_chain (GstPad * pad, GstData * data);
static GstElementStateReturn gst_a52dec_change_state (GstElement * element);
static void gst_a52dec_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_a52dec_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/* static guint gst_a52dec_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_a52dec_get_type (void)
{
static GType a52dec_type = 0;
if (!a52dec_type) {
static const GTypeInfo a52dec_info = {
sizeof (GstA52DecClass),
gst_a52dec_base_init,
NULL, (GClassInitFunc) gst_a52dec_class_init,
NULL,
NULL,
sizeof (GstA52Dec),
0,
(GInstanceInitFunc) gst_a52dec_init,
};
a52dec_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstA52Dec", &a52dec_info, 0);
GST_DEBUG_CATEGORY_INIT (a52dec_debug, "a52dec", 0,
"AC3/A52 software decoder");
}
return a52dec_type;
}
static void
gst_a52dec_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details (element_class, &gst_a52dec_details);
}
static void
gst_a52dec_class_init (GstA52DecClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
g_param_spec_boolean ("drc", "Dynamic Range Compression",
"Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));
gobject_class->set_property = gst_a52dec_set_property;
gobject_class->get_property = gst_a52dec_get_property;
gstelement_class->change_state = gst_a52dec_change_state;
}
static void
gst_a52dec_init (GstA52Dec * a52dec)
{
GST_FLAG_SET (GST_ELEMENT (a52dec), GST_ELEMENT_EVENT_AWARE);
/* create the sink and src pads */
a52dec->sinkpad =
gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
(a52dec), "sink"), "sink");
gst_pad_set_chain_function (a52dec->sinkpad, gst_a52dec_chain);
gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->sinkpad);
a52dec->srcpad =
gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
(a52dec), "src"), "src");
gst_pad_use_explicit_caps (a52dec->srcpad);
gst_element_add_pad (GST_ELEMENT (a52dec), a52dec->srcpad);
a52dec->dynamic_range_compression = FALSE;
a52dec->cache = NULL;
}
static int
gst_a52dec_channels (int flags, GstAudioChannelPosition ** _pos)
{
int chans = 0;
GstAudioChannelPosition *pos = NULL;
/* allocated just for safety. Number makes no sense */
if (_pos) {
pos = g_new (GstAudioChannelPosition, 6);
*_pos = pos;
}
if (flags & A52_LFE) {
chans += 1;
if (pos) {
pos[0] = GST_AUDIO_CHANNEL_POSITION_LFE;
}
}
flags &= A52_CHANNEL_MASK;
switch (flags) {
case A52_3F2R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[4 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
chans += 5;
break;
case A52_2F2R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_LEFT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT;
}
chans += 4;
break;
case A52_3F1R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[3 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
chans += 4;
break;
case A52_2F1R:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_REAR_CENTER;
}
chans += 3;
break;
case A52_3F:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER;
pos[2 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
chans += 3;
break;
/*case A52_CHANNEL: */
case A52_STEREO:
case A52_DOLBY:
if (pos) {
pos[0 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
pos[1 + chans] = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
}
chans += 2;
break;
default:
/* error */
g_warning ("a52dec invalid flags %d", flags);
g_free (pos);
return 0;
}
return chans;
}
static int
gst_a52dec_push (GstA52Dec * a52dec,
GstPad * srcpad, int flags, sample_t * samples, GstClockTime timestamp)
{
GstBuffer *buf;
int chans, n, c;
flags &= (A52_CHANNEL_MASK | A52_LFE);
chans = gst_a52dec_channels (flags, NULL);
if (!chans) {
return 1;
}
buf = gst_buffer_new ();
GST_BUFFER_SIZE (buf) = 256 * chans * (SAMPLE_WIDTH / 8);
GST_BUFFER_DATA (buf) = g_malloc (GST_BUFFER_SIZE (buf));
for (n = 0; n < 256; n++) {
for (c = 0; c < chans; c++) {
((sample_t *) GST_BUFFER_DATA (buf))[n * chans + c] =
samples[c * 256 + n];
}
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = 256 * GST_SECOND / a52dec->sample_rate;
GST_DEBUG_OBJECT (a52dec,
"Pushing buffer with ts %" GST_TIME_FORMAT " duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
gst_pad_push (srcpad, GST_DATA (buf));
return 0;
}
static gboolean
gst_a52dec_reneg (GstPad * pad)
{
GstAudioChannelPosition *pos;
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
gint channels = gst_a52dec_channels (a52dec->using_channels, &pos);
GstCaps *caps;
if (!channels)
return FALSE;
GST_INFO ("a52dec: reneg channels:%d rate:%d\n",
channels, a52dec->sample_rate);
caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, SAMPLE_WIDTH,
"channels", G_TYPE_INT, channels,
"rate", G_TYPE_INT, a52dec->sample_rate,
"buffer-frames", G_TYPE_INT, 0, NULL);
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
g_free (pos);
return gst_pad_set_explicit_caps (pad, caps);
}
static void
gst_a52dec_handle_event (GstA52Dec * a52dec, GstEvent * event)
{
GST_LOG ("Handling event of type %d timestamp %llu", GST_EVENT_TYPE (event),
GST_EVENT_TIMESTAMP (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_DISCONTINUOUS:{
gint64 val;
if (!gst_event_discont_get_value (event, GST_FORMAT_TIME, &val)
|| !GST_CLOCK_TIME_IS_VALID (val)) {
GST_WARNING ("No time discont value in event %p", event);
} else {
a52dec->time = val;
}
}
/* fall-through */
case GST_EVENT_FLUSH:
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
break;
default:
break;
}
gst_pad_event_default (a52dec->sinkpad, event);
}
static void
gst_a52dec_update_streaminfo (GstA52Dec * a52dec)
{
GstTagList *taglist;
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_APPEND,
GST_TAG_BITRATE, (guint) a52dec->bit_rate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (a52dec),
GST_PAD (a52dec->srcpad), a52dec->time, taglist);
}
static gboolean
gst_a52dec_handle_frame (GstA52Dec * a52dec, guint8 * data,
guint length, gint flags, gint sample_rate, gint bit_rate)
{
gint channels, i;
gboolean need_reneg = FALSE;
/* update stream information, renegotiate or re-streaminfo if needed */
need_reneg = FALSE;
if (a52dec->sample_rate != sample_rate) {
need_reneg = TRUE;
a52dec->sample_rate = sample_rate;
}
if (flags) {
a52dec->stream_channels = flags & (A52_CHANNEL_MASK | A52_LFE);
}
if (bit_rate != a52dec->bit_rate) {
a52dec->bit_rate = bit_rate;
gst_a52dec_update_streaminfo (a52dec);
}
/* process */
flags = a52dec->request_channels; /* | A52_ADJUST_LEVEL; */
a52dec->level = 1;
if (a52_frame (a52dec->state, data, &flags, &a52dec->level, a52dec->bias)) {
GST_WARNING ("a52_frame error");
return TRUE;
}
channels = flags & (A52_CHANNEL_MASK | A52_LFE);
if (a52dec->using_channels != channels) {
need_reneg = TRUE;
a52dec->using_channels = channels;
}
/* negotiate if required */
if (need_reneg == TRUE) {
GST_DEBUG ("a52dec reneg: sample_rate:%d stream_chans:%d using_chans:%d\n",
a52dec->sample_rate, a52dec->stream_channels, a52dec->using_channels);
if (!gst_a52dec_reneg (a52dec->srcpad)) {
GST_ELEMENT_ERROR (a52dec, CORE, NEGOTIATION, (NULL), (NULL));
return FALSE;
}
}
if (a52dec->dynamic_range_compression == FALSE) {
a52_dynrng (a52dec->state, NULL, NULL);
}
/* each frame consists of 6 blocks */
for (i = 0; i < 6; i++) {
if (a52_block (a52dec->state)) {
GST_WARNING ("a52_block error %d", i);
} else {
/* push on */
gst_a52dec_push (a52dec, a52dec->srcpad, a52dec->using_channels,
a52dec->samples, a52dec->time);
}
a52dec->time += 256 * GST_SECOND / a52dec->sample_rate;
}
return TRUE;
}
static void
gst_a52dec_chain (GstPad * pad, GstData * _data)
{
GstA52Dec *a52dec = GST_A52DEC (gst_pad_get_parent (pad));
GstBuffer *buf;
guint8 *data;
guint size;
gint length = 0, flags, sample_rate, bit_rate;
/* event handling */
if (GST_IS_EVENT (_data)) {
gst_a52dec_handle_event (a52dec, GST_EVENT (_data));
return;
}
/* merge with cache, if any. Also make sure timestamps match */
buf = GST_BUFFER (_data);
if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
a52dec->time = GST_BUFFER_TIMESTAMP (buf);
GST_DEBUG_OBJECT (a52dec,
"Received buffer with ts %" GST_TIME_FORMAT " duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
}
if (a52dec->cache) {
buf = gst_buffer_join (a52dec->cache, buf);
a52dec->cache = NULL;
}
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* find and read header */
bit_rate = a52dec->bit_rate;
sample_rate = a52dec->sample_rate;
flags = 0;
while (size >= 7) {
length = a52_syncinfo (data, &flags, &sample_rate, &bit_rate);
if (length == 0) {
/* no sync */
data++;
size--;
} else if (length <= size) {
GST_DEBUG ("Sync: %d", length);
if (!gst_a52dec_handle_frame (a52dec, data,
length, flags, sample_rate, bit_rate)) {
size = 0;
break;
}
size -= length;
data += length;
} else {
/* not enough data */
GST_LOG ("Not enough data available");
break;
}
}
/* keep cache */
if (length == 0) {
GST_LOG ("No sync found");
}
if (size > 0) {
a52dec->cache = gst_buffer_create_sub (buf,
GST_BUFFER_SIZE (buf) - size, size);
}
gst_buffer_unref (buf);
}
static GstElementStateReturn
gst_a52dec_change_state (GstElement * element)
{
GstA52Dec *a52dec = GST_A52DEC (element);
GstCPUFlags cpuflags;
uint32_t a52_cpuflags = 0;
switch (GST_STATE_TRANSITION (element)) {
case GST_STATE_NULL_TO_READY:
cpuflags = gst_cpu_get_flags ();
if (cpuflags & GST_CPU_FLAG_MMX)
a52_cpuflags |= MM_ACCEL_X86_MMX;
if (cpuflags & GST_CPU_FLAG_3DNOW)
a52_cpuflags |= MM_ACCEL_X86_3DNOW;
if (cpuflags & GST_CPU_FLAG_MMXEXT)
a52_cpuflags |= MM_ACCEL_X86_MMXEXT;
a52dec->state = a52_init (a52_cpuflags);
break;
case GST_STATE_READY_TO_PAUSED:
a52dec->samples = a52_samples (a52dec->state);
a52dec->bit_rate = -1;
a52dec->sample_rate = -1;
a52dec->stream_channels = A52_CHANNEL;
a52dec->request_channels = A52_3F2R | A52_LFE;
a52dec->using_channels = A52_CHANNEL;
a52dec->level = 1;
a52dec->bias = 0;
a52dec->time = 0;
break;
case GST_STATE_PAUSED_TO_PLAYING:
break;
case GST_STATE_PLAYING_TO_PAUSED:
break;
case GST_STATE_PAUSED_TO_READY:
a52dec->samples = NULL;
if (a52dec->cache) {
gst_buffer_unref (a52dec->cache);
a52dec->cache = NULL;
}
break;
case GST_STATE_READY_TO_NULL:
a52_free (a52dec->state);
a52dec->state = NULL;
break;
default:
break;
}
GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}
static void
gst_a52dec_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstA52Dec *src;
g_return_if_fail (GST_IS_A52DEC (object));
src = GST_A52DEC (object);
switch (prop_id) {
case ARG_DRC:
src->dynamic_range_compression = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_a52dec_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstA52Dec *src;
g_return_if_fail (GST_IS_A52DEC (object));
src = GST_A52DEC (object);
switch (prop_id) {
case ARG_DRC:
g_value_set_boolean (value, src->dynamic_range_compression);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_library_load ("gstaudio"))
return FALSE;
if (!gst_element_register (plugin, "a52dec", GST_RANK_PRIMARY,
GST_TYPE_A52DEC))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"a52dec",
"Decodes ATSC A/52 encoded audio streams",
plugin_init, VERSION, "GPL", GST_PACKAGE, GST_ORIGIN);