gstreamer/ext/sbc/gstsbcdec.c

303 lines
8.5 KiB
C

/* GStreamer SBC audio decoder
* BlueZ - Bluetooth protocol stack for Linux
*
* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
*
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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
*
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include "gstsbcutil.h"
#include "gstsbcdec.h"
#include <gst/audio/audio.h>
/* FIXME: where does this come from? how is it derived? */
#define BUF_SIZE 8192
GST_DEBUG_CATEGORY_STATIC (sbc_dec_debug);
#define GST_CAT_DEFAULT sbc_dec_debug
static void gst_sbc_dec_finalize (GObject * obj);
/* FIXME: port to GstAudioDecoder base class */
#define parent_class gst_sbc_dec_parent_class
G_DEFINE_TYPE (GstSbcDec, gst_sbc_dec, GST_TYPE_ELEMENT);
static GstStaticPadTemplate sbc_dec_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-sbc"));
static GstStaticPadTemplate sbc_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, format=" GST_AUDIO_NE (S16) ", "
"rate = (int) { 16000, 32000, 44100, 48000 }, "
"channels = (int) [ 1, 2 ], layout=interleaved"));
static GstFlowReturn
gst_sbc_dec_flush (GstSbcDec * dec, GstBuffer * outbuf,
gint outoffset, gint channels, gint rate)
{
GstClockTime outtime, duration;
/* we will reuse the same caps object */
if (dec->send_caps) {
GstCaps *caps;
caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, channels,
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_pad_push_event (dec->srcpad, gst_event_new_caps (caps));
gst_caps_unref (caps);
}
/* calculate duration */
outtime = GST_BUFFER_TIMESTAMP (outbuf);
if (dec->next_timestamp != (guint64) - 1 && outtime != (guint64) - 1) {
duration = dec->next_timestamp - outtime;
} else if (outtime != (guint64) - 1) {
/* otherwise calculate duration based on outbuf size */
duration = gst_util_uint64_scale_int (outoffset / (2 * channels),
GST_SECOND, rate) - outtime;
} else {
duration = GST_CLOCK_TIME_NONE;
}
GST_BUFFER_DURATION (outbuf) = duration;
gst_buffer_resize (outbuf, 0, outoffset);
return gst_pad_push (dec->srcpad, outbuf);
}
static GstFlowReturn
sbc_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstSbcDec *dec = GST_SBC_DEC (parent);
GstFlowReturn res = GST_FLOW_OK;
const guint8 *indata;
guint insize;
GstClockTime timestamp;
gboolean discont;
GstMapInfo out_map;
GstBuffer *outbuf;
guint inoffset, outoffset;
gint rate, channels;
discont = GST_BUFFER_IS_DISCONT (buffer);
if (discont) {
/* reset previous buffer */
gst_adapter_clear (dec->adapter);
/* we need a new timestamp to lock onto */
dec->next_sample = -1;
}
gst_adapter_push (dec->adapter, buffer);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
dec->next_timestamp = timestamp;
insize = gst_adapter_available (dec->adapter);
indata = gst_adapter_map (dec->adapter, insize);
inoffset = 0;
outbuf = NULL;
channels = rate = 0;
while (insize > 0) {
gint inconsumed, outlen;
gint outsize;
size_t outconsumed;
if (outbuf == NULL) {
outbuf = gst_buffer_new_and_alloc (BUF_SIZE);
if (discont) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
discont = FALSE;
}
GST_BUFFER_TIMESTAMP (outbuf) = dec->next_timestamp;
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
outsize = out_map.size;
outoffset = 0;
}
GST_INFO_OBJECT (dec, "inoffset %d/%d, outoffset %d/%d", inoffset,
insize, outoffset, outsize);
inconsumed = sbc_decode (&dec->sbc, indata + inoffset, insize,
out_map.data + outoffset, outsize, &outconsumed);
GST_INFO_OBJECT (dec, "consumed %d, produced %d", inconsumed, outconsumed);
if (inconsumed <= 0) {
guint frame_len = sbc_get_frame_length (&dec->sbc);
/* skip a frame */
if (insize > frame_len) {
insize -= frame_len;
inoffset += frame_len;
} else {
insize = 0;
}
continue;
}
inoffset += inconsumed;
if ((gint) insize > inconsumed)
insize -= inconsumed;
else
insize = 0;
outoffset += outconsumed;
outsize -= outconsumed;
rate = gst_sbc_parse_rate_from_sbc (dec->sbc.frequency);
channels = gst_sbc_get_channel_number (dec->sbc.mode);
/* calculate timestamp either from the incomming buffers or
* from our sample counter */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* lock onto timestamp when we have one */
dec->next_sample = gst_util_uint64_scale_int (timestamp,
rate, GST_SECOND);
timestamp = GST_CLOCK_TIME_NONE;
}
if (dec->next_sample != (guint64) - 1) {
/* calculate the next sample */
dec->next_sample += outconsumed / (2 * channels);
dec->next_timestamp = gst_util_uint64_scale_int (dec->next_sample,
GST_SECOND, rate);
}
/* check for space, push outbuf buffer */
outlen = sbc_get_codesize (&dec->sbc);
if (outsize < outlen) {
gst_buffer_unmap (outbuf, &out_map);
res = gst_sbc_dec_flush (dec, outbuf, outoffset, channels, rate);
outbuf = NULL;
if (res != GST_FLOW_OK)
goto done;
}
}
if (outbuf) {
gst_buffer_unmap (outbuf, &out_map);
res = gst_sbc_dec_flush (dec, outbuf, outoffset, channels, rate);
}
done:
gst_adapter_unmap (dec->adapter);
gst_adapter_flush (dec->adapter, inoffset);
return res;
}
static GstStateChangeReturn
gst_sbc_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn result;
GstSbcDec *dec = GST_SBC_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_DEBUG ("Setup subband codec");
sbc_init (&dec->sbc, 0);
dec->send_caps = TRUE;
dec->next_sample = -1;
break;
default:
break;
}
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG ("Finish subband codec");
gst_adapter_clear (dec->adapter);
sbc_finish (&dec->sbc);
dec->send_caps = TRUE;
break;
default:
break;
}
return result;
}
static void
gst_sbc_dec_class_init (GstSbcDecClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
object_class->finalize = gst_sbc_dec_finalize;
element_class->change_state = GST_DEBUG_FUNCPTR (gst_sbc_dec_change_state);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sbc_dec_sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sbc_dec_src_factory));
gst_element_class_set_static_metadata (element_class,
"Bluetooth SBC audio decoder", "Codec/Decoder/Audio",
"Decode an SBC audio stream", "Marcel Holtmann <marcel@holtmann.org>");
GST_DEBUG_CATEGORY_INIT (sbc_dec_debug, "sbcdec", 0, "SBC decoding element");
}
static void
gst_sbc_dec_init (GstSbcDec * self)
{
self->sinkpad =
gst_pad_new_from_static_template (&sbc_dec_sink_factory, "sink");
gst_pad_set_chain_function (self->sinkpad, GST_DEBUG_FUNCPTR (sbc_dec_chain));
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
self->srcpad = gst_pad_new_from_static_template (&sbc_dec_src_factory, "src");
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
self->adapter = gst_adapter_new ();
self->send_caps = TRUE;
}
static void
gst_sbc_dec_finalize (GObject * obj)
{
GstSbcDec *self = GST_SBC_DEC (obj);
g_object_unref (self->adapter);
self->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (obj);
}