gstreamer/gst/adpcmenc/adpcmenc.c

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/* GStreamer
* Copyright (C) 2009 Pioneers of the Inevitable <songbird@songbirdnest.com>
*
* Authors: Peter van Hardenberg <pvh@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 encoders 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>
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#include <gst/audio/gstaudioencoder.h>
#define GST_TYPE_ADPCM_ENC \
(adpcmenc_get_type ())
#define GST_TYPE_ADPCMENC_LAYOUT \
(adpcmenc_layout_get_type ())
#define GST_ADPCM_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_ADPCM_ENC, ADPCMEnc))
#define GST_CAT_DEFAULT adpcmenc_debug
GST_DEBUG_CATEGORY_STATIC (adpcmenc_debug);
static GstStaticPadTemplate adpcmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) [1, MAX], channels = (int) [1,2]")
);
static GstStaticPadTemplate adpcmenc_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-adpcm, "
" layout=(string)dvi, "
" block_align = (int) [64, 8192], "
" rate = (int)[ 1, MAX ], " "channels = (int)[1,2];")
);
#define MIN_ADPCM_BLOCK_SIZE 64
#define MAX_ADPCM_BLOCK_SIZE 8192
#define DEFAULT_ADPCM_BLOCK_SIZE 1024
#define DEFAULT_ADPCM_LAYOUT LAYOUT_ADPCM_DVI
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
};
enum adpcm_properties
{
PROP_0,
PROP_BLOCK_SIZE,
PROP_LAYOUT
};
enum adpcm_layout
{
LAYOUT_ADPCM_DVI
};
static GType
adpcmenc_layout_get_type (void)
{
static GType adpcmenc_layout_type = 0;
if (!adpcmenc_layout_type) {
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static const GEnumValue layout_types[] = {
{LAYOUT_ADPCM_DVI, "DVI/IMA APDCM", "dvi"},
{0, NULL, NULL},
};
adpcmenc_layout_type = g_enum_register_static ("GstADPCMEncLayout",
layout_types);
}
return adpcmenc_layout_type;
}
typedef struct _ADPCMEncClass
{
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GstAudioEncoderClass parent_class;
} ADPCMEncClass;
typedef struct _ADPCMEnc
{
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GstAudioEncoder parent;
enum adpcm_layout layout;
int rate;
int channels;
int blocksize;
int samples_per_block;
guint8 step_index[2];
} ADPCMEnc;
GType adpcmenc_get_type (void);
GST_ELEMENT_REGISTER_DECLARE (adpcmenc);
G_DEFINE_TYPE_WITH_CODE (ADPCMEnc, adpcmenc, GST_TYPE_AUDIO_ENCODER,
GST_DEBUG_CATEGORY_INIT (adpcmenc_debug, "adpcmenc", 0, "ADPCM Encoders");
);
GST_ELEMENT_REGISTER_DEFINE (adpcmenc, "adpcmenc", GST_RANK_PRIMARY,
GST_TYPE_ADPCM_ENC);
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static gboolean
adpcmenc_setup (ADPCMEnc * enc)
{
const int DVI_IMA_HEADER_SIZE = 4;
const int ADPCM_SAMPLES_PER_BYTE = 2;
guint64 sample_bytes;
const char *layout;
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GstCaps *caps;
gboolean ret;
switch (enc->layout) {
case LAYOUT_ADPCM_DVI:
layout = "dvi";
/* IMA ADPCM includes a 4-byte header per channel, */
sample_bytes = enc->blocksize - (DVI_IMA_HEADER_SIZE * enc->channels);
/* two samples per byte, plus a single sample in the header. */
enc->samples_per_block =
((sample_bytes * ADPCM_SAMPLES_PER_BYTE) / enc->channels) + 1;
break;
default:
GST_WARNING_OBJECT (enc, "Invalid layout");
return FALSE;
}
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caps = gst_caps_new_simple ("audio/x-adpcm",
"rate", G_TYPE_INT, enc->rate,
"channels", G_TYPE_INT, enc->channels,
"layout", G_TYPE_STRING, layout,
"block_align", G_TYPE_INT, enc->blocksize, NULL);
ret = gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
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gst_caps_unref (caps);
/* Step index state is carried between blocks. */
enc->step_index[0] = 0;
enc->step_index[1] = 0;
return ret;
}
static gboolean
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adpcmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
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ADPCMEnc *enc = (ADPCMEnc *) (benc);
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enc->rate = GST_AUDIO_INFO_RATE (info);
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
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if (!adpcmenc_setup (enc))
return FALSE;
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/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, enc->samples_per_block);
gst_audio_encoder_set_frame_samples_max (benc, enc->samples_per_block);
gst_audio_encoder_set_frame_max (benc, 1);
return TRUE;
}
static void
adpcmenc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
ADPCMEnc *enc = GST_ADPCM_ENC (object);
switch (prop_id) {
case PROP_BLOCK_SIZE:
enc->blocksize = g_value_get_int (value);
break;
case PROP_LAYOUT:
enc->layout = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
adpcmenc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
ADPCMEnc *enc = GST_ADPCM_ENC (object);
switch (prop_id) {
case PROP_BLOCK_SIZE:
g_value_set_int (value, enc->blocksize);
break;
case PROP_LAYOUT:
g_value_set_enum (value, enc->layout);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static guint8
adpcmenc_encode_ima_sample (gint16 sample, gint16 * prev_sample,
guint8 * stepindex)
{
const int NEGATIVE_SIGN_BIT = 0x8;
int diff, vpdiff, mask, step;
int bytecode = 0x0;
diff = sample - *prev_sample;
step = ima_step_size[*stepindex];
vpdiff = step >> 3;
if (diff < 0) {
diff = -diff;
bytecode = NEGATIVE_SIGN_BIT;
}
mask = 0x4;
while (mask > 0) {
if (diff >= step) {
bytecode |= mask;
diff -= step;
vpdiff += step;
}
step >>= 1;
mask >>= 1;
}
if (bytecode & 8) {
vpdiff = -vpdiff;
}
*prev_sample = CLAMP (*prev_sample + vpdiff, G_MININT16, G_MAXINT16);
*stepindex = CLAMP (*stepindex + ima_indx_adjust[bytecode], 0, 88);
return bytecode;
}
static gboolean
adpcmenc_encode_ima_block (ADPCMEnc * enc, const gint16 * samples,
guint8 * outbuf)
{
const int HEADER_SIZE = 4;
gint16 prev_sample[2] = { 0, 0 };
guint32 write_pos = 0;
guint32 read_pos = 0;
guint8 channel = 0;
/* Write a header for each channel.
* The header consists of a sixteen-bit predicted sound value,
* and an eight bit step_index, carried forward from any previous block.
* These allow seeking within the file.
*/
for (channel = 0; channel < enc->channels; channel++) {
write_pos = channel * HEADER_SIZE;
outbuf[write_pos + 0] = (samples[channel] & 0xFF);
outbuf[write_pos + 1] = (samples[channel] >> 8) & 0xFF;
outbuf[write_pos + 2] = enc->step_index[channel];
outbuf[write_pos + 3] = 0;
prev_sample[channel] = samples[channel];
}
/* raw-audio looks like this for a stereo stream:
* [ L, R, L, R, L, R ... ]
* encoded audio is in eight-sample blocks, two samples to a byte thusly:
* [ LL, LL, LL, LL, RR, RR, RR, RR ... ]
*/
write_pos = HEADER_SIZE * enc->channels;
read_pos = enc->channels; /* the first sample is in the header. */
while (write_pos < enc->blocksize) {
gint8 CHANNEL_CHUNK_SIZE = 8;
for (channel = 0; channel < enc->channels; channel++) {
/* convert eight samples (four bytes) per channel, then swap */
guint32 channel_chunk_base = read_pos + channel;
gint8 chunk;
for (chunk = 0; chunk < CHANNEL_CHUNK_SIZE; chunk++) {
guint8 packed_byte = 0, encoded_sample;
encoded_sample =
adpcmenc_encode_ima_sample (samples[channel_chunk_base +
(chunk * enc->channels)], &prev_sample[channel],
&enc->step_index[channel]);
packed_byte |= encoded_sample & 0x0F;
chunk++;
encoded_sample =
adpcmenc_encode_ima_sample (samples[channel_chunk_base +
(chunk * enc->channels)], &prev_sample[channel],
&enc->step_index[channel]);
packed_byte |= encoded_sample << 4 & 0xF0;
outbuf[write_pos++] = packed_byte;
}
}
/* advance to the next block of 8 samples per channel */
read_pos += CHANNEL_CHUNK_SIZE * enc->channels;
if (read_pos > enc->samples_per_block * enc->channels) {
GST_LOG ("Ran past the end. (Reading %i of %i.)", read_pos,
enc->samples_per_block);
}
}
return TRUE;
}
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static GstBuffer *
adpcmenc_encode_block (ADPCMEnc * enc, const gint16 * samples, int blocksize)
{
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gboolean res = FALSE;
GstBuffer *outbuf = NULL;
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GstMapInfo omap;
if (enc->layout == LAYOUT_ADPCM_DVI) {
outbuf = gst_buffer_new_and_alloc (enc->blocksize);
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gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
res = adpcmenc_encode_ima_block (enc, samples, omap.data);
gst_buffer_unmap (outbuf, &omap);
} else {
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/* should not happen afaics */
g_assert_not_reached ();
GST_WARNING_OBJECT (enc, "Unknown layout");
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res = FALSE;
}
if (!res) {
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if (outbuf)
gst_buffer_unref (outbuf);
outbuf = NULL;
GST_WARNING_OBJECT (enc, "Encode of block failed");
}
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return outbuf;
}
static GstFlowReturn
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adpcmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
{
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ADPCMEnc *enc = (ADPCMEnc *) (benc);
GstFlowReturn ret = GST_FLOW_OK;
gint16 *samples;
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GstBuffer *outbuf;
int input_bytes_per_block;
const int BYTES_PER_SAMPLE = 2;
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GstMapInfo map;
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/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (benc, "no data");
goto done;
}
input_bytes_per_block =
enc->samples_per_block * BYTES_PER_SAMPLE * enc->channels;
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gst_buffer_map (buffer, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < input_bytes_per_block)) {
GST_DEBUG_OBJECT (enc, "discarding trailing data %d", (gint) map.size);
gst_buffer_unmap (buffer, &map);
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done;
}
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samples = (gint16 *) map.data;
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outbuf = adpcmenc_encode_block (enc, samples, enc->blocksize);
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gst_buffer_unmap (buffer, &map);
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ret = gst_audio_encoder_finish_frame (benc, outbuf, enc->samples_per_block);
done:
return ret;
}
static gboolean
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adpcmenc_start (GstAudioEncoder * enc)
{
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GST_DEBUG_OBJECT (enc, "start");
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return TRUE;
}
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static gboolean
adpcmenc_stop (GstAudioEncoder * enc)
{
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GST_DEBUG_OBJECT (enc, "stop");
return TRUE;
}
static void
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adpcmenc_init (ADPCMEnc * enc)
{
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
/* Set defaults. */
enc->blocksize = DEFAULT_ADPCM_BLOCK_SIZE;
enc->layout = DEFAULT_ADPCM_LAYOUT;
}
static void
adpcmenc_class_init (ADPCMEncClass * klass)
{
GObjectClass *gobjectclass = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) klass;
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GstAudioEncoderClass *base_class = (GstAudioEncoderClass *) klass;
gobjectclass->set_property = adpcmenc_set_property;
gobjectclass->get_property = adpcmenc_get_property;
gst_element_class_add_static_pad_template (element_class,
&adpcmenc_sink_template);
gst_element_class_add_static_pad_template (element_class,
&adpcmenc_src_template);
gst_element_class_set_static_metadata (element_class, "ADPCM encoder",
"Codec/Encoder/Audio", "Encode ADPCM audio",
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"Pioneers of the Inevitable <songbird@songbirdnest.com>");
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base_class->start = GST_DEBUG_FUNCPTR (adpcmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (adpcmenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (adpcmenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (adpcmenc_handle_frame);
g_object_class_install_property (gobjectclass, PROP_LAYOUT,
g_param_spec_enum ("layout", "Layout",
"Layout for output stream",
GST_TYPE_ADPCMENC_LAYOUT, DEFAULT_ADPCM_LAYOUT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobjectclass, PROP_BLOCK_SIZE,
g_param_spec_int ("blockalign", "Block Align",
"Block size for output stream",
MIN_ADPCM_BLOCK_SIZE, MAX_ADPCM_BLOCK_SIZE,
DEFAULT_ADPCM_BLOCK_SIZE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_type_mark_as_plugin_api (GST_TYPE_ADPCMENC_LAYOUT, 0);
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return GST_ELEMENT_REGISTER (adpcmenc, plugin);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, adpcmenc,
"ADPCM encoder", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);