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gstreamer/subprojects/gst-plugins-bad/gst/adpcmdec/adpcmdec.c

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/* GStreamer
* Copyright (C) 2009 Pioneers of the Inevitable <songbird@songbirdnest.com>
*
* Authors: Michael Smith <msmith@songbirdnest.com>
*
* 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.
*/
/* Based on ADPCM decoders in libsndfile,
Copyright (C) 1999-2002 Erik de Castro Lopo <erikd@zip.com.au
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/gstaudiodecoder.h>
#define GST_TYPE_ADPCM_DEC \
(adpcmdec_get_type ())
#define GST_ADPCM_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_ADPCM_DEC, ADPCMDec))
#define GST_CAT_DEFAULT adpcmdec_debug
GST_DEBUG_CATEGORY_STATIC (adpcmdec_debug);
static GstStaticPadTemplate adpcmdec_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
" layout=(string){microsoft, dvi}, "
" block_align = (int) [64, 8192], "
" rate = (int)[ 1, MAX ], " "channels = (int)[1,2];")
);
static GstStaticPadTemplate adpcmdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) [1, MAX], channels = (int) [1,2]")
);
enum adpcm_layout
{
LAYOUT_ADPCM_MICROSOFT,
LAYOUT_ADPCM_DVI
};
typedef struct _ADPCMDecClass
{
GstAudioDecoderClass parent_class;
} ADPCMDecClass;
typedef struct _ADPCMDec
{
GstAudioDecoder parent;
enum adpcm_layout layout;
int rate;
int channels;
int blocksize;
} ADPCMDec;
GType adpcmdec_get_type (void);
GST_ELEMENT_REGISTER_DECLARE (adpcmdec);
G_DEFINE_TYPE_WITH_CODE (ADPCMDec, adpcmdec, GST_TYPE_AUDIO_DECODER,
GST_DEBUG_CATEGORY_INIT (adpcmdec_debug, "adpcmdec", 0, "ADPCM Decoders");
);
GST_ELEMENT_REGISTER_DEFINE (adpcmdec, "adpcmdec", GST_RANK_PRIMARY,
GST_TYPE_ADPCM_DEC);
static gboolean
adpcmdec_set_format (GstAudioDecoder * bdec, GstCaps * in_caps)
{
ADPCMDec *dec = (ADPCMDec *) (bdec);
GstStructure *structure = gst_caps_get_structure (in_caps, 0);
const gchar *layout;
GstAudioInfo info;
layout = gst_structure_get_string (structure, "layout");
if (!layout)
return FALSE;
if (g_str_equal (layout, "microsoft"))
dec->layout = LAYOUT_ADPCM_MICROSOFT;
else if (g_str_equal (layout, "dvi"))
dec->layout = LAYOUT_ADPCM_DVI;
else
return FALSE;
if (!gst_structure_get_int (structure, "block_align", &dec->blocksize))
dec->blocksize = -1; /* Not provided */
if (!gst_structure_get_int (structure, "rate", &dec->rate))
return FALSE;
if (!gst_structure_get_int (structure, "channels", &dec->channels))
return FALSE;
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, dec->rate,
dec->channels, NULL);
gst_audio_decoder_set_output_format (bdec, &info);
return TRUE;
}
/*=====================================================================
* From libsndfile:
*
* MS ADPCM Block Layout.
* ======================
* Block is usually 256, 512 or 1024 bytes depending on sample rate.
* For a mono file, the block is laid out as follows:
* byte purpose
* 0 block predictor [0..6]
* 1,2 initial idelta (positive)
* 3,4 sample 1
* 5,6 sample 0
* 7..n packed bytecodes
*
* For a stereo file, the block is laid out as follows:
* byte purpose
* 0 block predictor [0..6] for left channel
* 1 block predictor [0..6] for right channel
* 2,3 initial idelta (positive) for left channel
* 4,5 initial idelta (positive) for right channel
* 6,7 sample 1 for left channel
* 8,9 sample 1 for right channel
* 10,11 sample 0 for left channel
* 12,13 sample 0 for right channel
* 14..n packed bytecodes
*
*=====================================================================
*/
static const int AdaptationTable[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
static const int AdaptCoeff1[] = {
256, 512, 0, 192, 240, 460, 392
};
static const int AdaptCoeff2[] = {
0, -256, 0, 64, 0, -208, -232
};
static gint16
read_sample (const guint8 * data)
{
guint16 val = data[0] | (data[1] << 8);
return *((gint16 *) & val);
}
/* Decode a single block of data from 'data', storing 'n_samples' decoded 16 bit
samples in 'samples'.
All buffer lengths have been verified by the caller
*/
static gboolean
adpcmdec_decode_ms_block (ADPCMDec * dec, int n_samples, const guint8 * data,
gint16 * samples)
{
gint16 pred[2];
gint16 idelta[2];
int idx; /* Current byte offset in 'data' */
int i; /* Current sample index in 'samples' */
/* Read the block header, verify for sanity */
if (dec->channels == 1) {
pred[0] = data[0];
idelta[0] = read_sample (data + 1);
samples[1] = read_sample (data + 3);
samples[0] = read_sample (data + 5);
idx = 7;
i = 2;
if (pred[0] < 0 || pred[0] > 6) {
GST_WARNING_OBJECT (dec, "Invalid block predictor");
return FALSE;
}
}
else {
pred[0] = data[0];
pred[1] = data[1];
idelta[0] = read_sample (data + 2);
idelta[1] = read_sample (data + 4);
samples[2] = read_sample (data + 6);
samples[3] = read_sample (data + 8);
samples[0] = read_sample (data + 10);
samples[1] = read_sample (data + 12);
idx = 14;
i = 4;
if (pred[0] < 0 || pred[0] > 6 || pred[1] < 0 || pred[1] > 6) {
GST_WARNING_OBJECT (dec, "Invalid block predictor");
return FALSE;
}
}
for (; i < n_samples; i++) {
int chan = i % dec->channels;
int bytecode;
int delta;
int current;
int predict;
if (i % 2 == 0) {
bytecode = (data[idx] >> 4) & 0x0F;
} else {
bytecode = data[idx] & 0x0F;
idx++;
}
delta = idelta[chan];
idelta[chan] = (AdaptationTable[bytecode] * delta) >> 8;
if (idelta[chan] < 16)
idelta[chan] = 16;
/* Bytecode is used above as an index into the table. Below, it's used
as a signed 4-bit value; convert appropriately */
if (bytecode & 0x8)
bytecode -= 0x10;
predict = ((samples[i - dec->channels] * AdaptCoeff1[pred[chan]]) +
(samples[i - 2 * dec->channels] * AdaptCoeff2[pred[chan]])
) >> 8;
current = (bytecode * delta) + predict;
/* Clamp to 16 bits, store decoded sample */
samples[i] = CLAMP (current, G_MININT16, G_MAXINT16);
}
return TRUE;
}
static const int ima_indx_adjust[16] = {
-1, -1, -1, -1, 2, 4, 6, 8, -1, -1, -1, -1, 2, 4, 6, 8,
};
static const int ima_step_size[89] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230,
253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327,
3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442,
11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794,
32767
};
/* Decode a single block of data from 'data', storing 'n_samples' decoded 16 bit
samples in 'samples'.
All buffer lengths have been verified by the caller
*/
static gboolean
adpcmdec_decode_ima_block (ADPCMDec * dec, int n_samples, const guint8 * data,
gint16 * samples)
{
gint16 stepindex[2];
int channel;
int idx;
int i, j;
int sample;
if ((n_samples - dec->channels) % 8 != 0) {
GST_WARNING_OBJECT (dec, "Input not correct size");
return FALSE;
}
for (channel = 0; channel < dec->channels; channel++) {
samples[channel] = read_sample (data + channel * 4);
stepindex[channel] = MIN (data[channel * 4 + 2], 88);
if (data[channel * 4 + 3] != 0) {
GST_WARNING_OBJECT (dec, "Synchronisation error");
return FALSE;
}
}
i = dec->channels;
idx = 4 * dec->channels;
while (i < n_samples) {
for (channel = 0; channel < dec->channels; channel++) {
sample = i + channel;
for (j = 0; j < 8; j++) {
int bytecode;
int step;
int diff;
if (j % 2 == 0) {
bytecode = data[idx] & 0x0F;
} else {
bytecode = (data[idx] >> 4) & 0x0F;
idx++;
}
step = ima_step_size[stepindex[channel]];
diff = (2 * (bytecode & 0x7) * step + step) / 8;
if (bytecode & 8)
diff = -diff;
samples[sample] =
CLAMP (samples[sample - dec->channels] + diff, G_MININT16,
G_MAXINT16);
stepindex[channel] =
CLAMP (stepindex[channel] + ima_indx_adjust[bytecode], 0, 88);
sample += dec->channels;
}
}
i += 8 * dec->channels;
}
return TRUE;
}
static GstBuffer *
adpcmdec_decode_block (ADPCMDec * dec, const guint8 * data, int blocksize)
{
gboolean res = FALSE;
GstBuffer *outbuf = NULL;
int outsize;
int samples;
GstMapInfo omap;
if (dec->layout == LAYOUT_ADPCM_MICROSOFT) {
/* Each block has a 3 byte header per channel, plus 4 bytes per channel to
give two initial sample values per channel. Then the remainder gives
two samples per byte */
if (blocksize < 7 * dec->channels)
goto exit;
samples = (blocksize - 7 * dec->channels) * 2 + 2 * dec->channels;
outsize = 2 * samples;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
res = adpcmdec_decode_ms_block (dec, samples, data, (gint16 *) omap.data);
gst_buffer_unmap (outbuf, &omap);
} else if (dec->layout == LAYOUT_ADPCM_DVI) {
/* Each block has a 4 byte header per channel, include an initial sample.
Then the remainder gives two samples per byte */
if (blocksize < 4 * dec->channels)
goto exit;
samples = (blocksize - 4 * dec->channels) * 2 + dec->channels;
outsize = 2 * samples;
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
res = adpcmdec_decode_ima_block (dec, samples, data, (gint16 *) omap.data);
gst_buffer_unmap (outbuf, &omap);
} else {
GST_WARNING_OBJECT (dec, "Unknown layout");
}
if (!res) {
if (outbuf)
gst_buffer_unref (outbuf);
outbuf = NULL;
GST_WARNING_OBJECT (dec, "Decode of block failed");
}
exit:
return outbuf;
}
static GstFlowReturn
adpcmdec_parse (GstAudioDecoder * bdec, GstAdapter * adapter,
gint * offset, gint * length)
{
ADPCMDec *dec = (ADPCMDec *) (bdec);
guint size;
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
if (dec->blocksize < 0) {
/* No explicit blocksize; we just process one input buffer at a time */
*offset = 0;
*length = size;
} else {
if (size >= dec->blocksize) {
*offset = 0;
*length = dec->blocksize;
} else {
return GST_FLOW_EOS;
}
}
return GST_FLOW_OK;
}
static GstFlowReturn
adpcmdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer)
{
ADPCMDec *dec = (ADPCMDec *) (bdec);
GstFlowReturn ret = GST_FLOW_OK;
GstMapInfo map;
GstBuffer *outbuf = NULL;
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
if (!dec->blocksize)
return GST_FLOW_NOT_NEGOTIATED;
gst_buffer_map (buffer, &map, GST_MAP_READ);
outbuf = adpcmdec_decode_block (dec, map.data, dec->blocksize);
gst_buffer_unmap (buffer, &map);
if (outbuf == NULL) {
GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
("frame decode failed"), ret);
}
if (ret == GST_FLOW_OK)
ret = gst_audio_decoder_finish_frame (bdec, outbuf, 1);
return ret;
}
static gboolean
adpcmdec_start (GstAudioDecoder * bdec)
{
ADPCMDec *dec = (ADPCMDec *) bdec;
GST_DEBUG_OBJECT (dec, "start");
dec->blocksize = 0;
dec->rate = 0;
dec->channels = 0;
return TRUE;
}
static gboolean
adpcmdec_stop (GstAudioDecoder * dec)
{
GST_DEBUG_OBJECT (dec, "stop");
return TRUE;
}
static void
adpcmdec_init (ADPCMDec * dec)
{
gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
(dec), TRUE);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
}
static void
adpcmdec_class_init (ADPCMDecClass * klass)
{
GstElementClass *element_class = (GstElementClass *) klass;
GstAudioDecoderClass *base_class = (GstAudioDecoderClass *) klass;
gst_element_class_add_static_pad_template (element_class,
&adpcmdec_sink_template);
gst_element_class_add_static_pad_template (element_class,
&adpcmdec_src_template);
gst_element_class_set_static_metadata (element_class, "ADPCM decoder",
"Codec/Decoder/Audio", "Decode MS and IMA ADPCM audio",
"Pioneers of the Inevitable <songbird@songbirdnest.com>");
base_class->start = GST_DEBUG_FUNCPTR (adpcmdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (adpcmdec_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (adpcmdec_set_format);
base_class->parse = GST_DEBUG_FUNCPTR (adpcmdec_parse);
base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmdec_handle_frame);
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return GST_ELEMENT_REGISTER (adpcmdec, plugin);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, adpcmdec,
"ADPCM decoder", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);
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