/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.org
*
* WavPack support added by Laszlo Ashin <kodest@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../decoder_api.h"
#include "../path.h"
#include <wavpack/wavpack.h>
#include <glib.h>
/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
#define CHUNK_SIZE 1020
#define ERRORLEN 80
static struct {
const char *name;
enum tag_type type;
} tagtypes[] = {
{ "artist", TAG_ITEM_ARTIST },
{ "album", TAG_ITEM_ALBUM },
{ "title", TAG_ITEM_TITLE },
{ "track", TAG_ITEM_TRACK },
{ "name", TAG_ITEM_NAME },
{ "genre", TAG_ITEM_GENRE },
{ "date", TAG_ITEM_DATE },
{ "composer", TAG_ITEM_COMPOSER },
{ "performer", TAG_ITEM_PERFORMER },
{ "comment", TAG_ITEM_COMMENT },
{ "disc", TAG_ITEM_DISC },
};
/** A pointer type for format converter function. */
typedef void (*format_samples_t)(
int bytes_per_sample,
void *buffer, uint32_t count
);
/*
* This function has been borrowed from the tiny player found on
* wavpack.com. Modifications were required because mpd only handles
* max 24-bit samples.
*/
static void
format_samples_int(int bytes_per_sample, void *buffer, uint32_t count)
{
int32_t *src = buffer;
switch (bytes_per_sample) {
case 1: {
uchar *dst = buffer;
/*
* The asserts like the following one are because we do the
* formatting of samples within a single buffer. The size
* of the output samples never can be greater than the size
* of the input ones. Otherwise we would have an overflow.
*/
assert(sizeof(uchar) <= sizeof(uint32_t));
/* pass through and align 8-bit samples */
while (count--) {
*dst++ = *src++;
}
break;
}
case 2: {
uint16_t *dst = buffer;
assert(sizeof(uint16_t) <= sizeof(uint32_t));
/* pass through and align 16-bit samples */
while (count--) {
*dst++ = *src++;
}
break;
}
case 3:
/* do nothing */
break;
case 4: {
uint32_t *dst = buffer;
assert(sizeof(uint32_t) <= sizeof(uint32_t));
/* downsample to 24-bit */
while (count--) {
*dst++ = *src++ >> 8;
}
break;
}
}
}
/*
* This function converts floating point sample data to 24-bit integer.
*/
static void
format_samples_float(mpd_unused int bytes_per_sample, void *buffer,
uint32_t count)
{
int32_t *dst = buffer;
float *src = buffer;
assert(sizeof(int32_t) <= sizeof(float));
while (count--) {
*dst++ = (int32_t)(*src++ + 0.5f);
}
}
/*
* This does the main decoding thing.
* Requires an already opened WavpackContext.
*/
static void
wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek,
struct replay_gain_info *replay_gain_info)
{
struct audio_format audio_format;
format_samples_t format_samples;
char chunk[CHUNK_SIZE];
int samples_requested, samples_got;
float total_time, current_time;
int bytes_per_sample, output_sample_size;
int position;
audio_format.sample_rate = WavpackGetSampleRate(wpc);
audio_format.channels = WavpackGetReducedChannels(wpc);
audio_format.bits = WavpackGetBitsPerSample(wpc);
/* round bitwidth to 8-bit units */
audio_format.bits = (audio_format.bits + 7) & (~7);
/* mpd handles max 24-bit samples */
if (audio_format.bits > 24) {
audio_format.bits = 24;
}
if ((WavpackGetMode(wpc) & MODE_FLOAT) == MODE_FLOAT) {
format_samples = format_samples_float;
} else {
format_samples = format_samples_int;
}
total_time = WavpackGetNumSamples(wpc);
total_time /= audio_format.sample_rate;
bytes_per_sample = WavpackGetBytesPerSample(wpc);
output_sample_size = bytes_per_sample;
if (output_sample_size == 3) {
output_sample_size = 4;
}
output_sample_size *= audio_format.channels;
/* wavpack gives us all kind of samples in a 32-bit space */
samples_requested = sizeof(chunk) / (4 * audio_format.channels);
decoder_initialized(decoder, &audio_format, can_seek, total_time);
position = 0;
do {
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
if (can_seek) {
int where;
where = decoder_seek_where(decoder);
where *= audio_format.sample_rate;
if (WavpackSeekSample(wpc, where)) {
position = where;
decoder_command_finished(decoder);
} else {
decoder_seek_error(decoder);
}
} else {
decoder_seek_error(decoder);
}
}
if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) {
break;
}
samples_got = WavpackUnpackSamples(
wpc, (int32_t *)chunk, samples_requested
);
if (samples_got > 0) {
int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
1000 + 0.5);
position += samples_got;
current_time = position;
current_time /= audio_format.sample_rate;
format_samples(
bytes_per_sample, chunk,
samples_got * audio_format.channels
);
decoder_data(
decoder, NULL, chunk,
samples_got * output_sample_size,
current_time, bitrate,
replay_gain_info
);
}
} while (samples_got > 0);
}
/**
* Locate and parse a floating point tag. Returns true if it was
* found.
*/
static bool
wavpack_tag_float(WavpackContext *wpc, const char *key, float *value_r)
{
char buffer[64];
int ret;
ret = WavpackGetTagItem(wpc, key, buffer, sizeof(buffer));
if (ret <= 0)
return false;
*value_r = atof(buffer);
return true;
}
static struct replay_gain_info *
wavpack_replaygain(WavpackContext *wpc)
{
struct replay_gain_info *replay_gain_info;
bool found = false;
replay_gain_info = replay_gain_info_new();
found |= wavpack_tag_float(
wpc, "replaygain_track_gain",
&replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain
);
found |= wavpack_tag_float(
wpc, "replaygain_track_peak",
&replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak
);
found |= wavpack_tag_float(
wpc, "replaygain_album_gain",
&replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain
);
found |= wavpack_tag_float(
wpc, "replaygain_album_peak",
&replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak
);
if (found) {
return replay_gain_info;
}
replay_gain_info_free(replay_gain_info);
return NULL;
}
/*
* Reads metainfo from the specified file.
*/
static struct tag *
wavpack_tagdup(const char *fname)
{
WavpackContext *wpc;
struct tag *tag;
char error[ERRORLEN];
char *s;
int size, allocated_size;
wpc = WavpackOpenFileInput(fname, error, OPEN_TAGS, 0);
if (wpc == NULL) {
g_warning(
"failed to open WavPack file \"%s\": %s\n",
fname, error
);
return NULL;
}
tag = tag_new();
tag->time = WavpackGetNumSamples(wpc);
tag->time /= WavpackGetSampleRate(wpc);
allocated_size = 0;
s = NULL;
for (unsigned i = 0; i < G_N_ELEMENTS(tagtypes); ++i) {
size = WavpackGetTagItem(wpc, tagtypes[i].name, NULL, 0);
if (size > 0) {
++size; /* EOS */
if (s == NULL) {
s = g_malloc(size);
allocated_size = size;
} else if (size > allocated_size) {
char *t = (char *)g_realloc(s, size);
allocated_size = size;
s = t;
}
WavpackGetTagItem(wpc, tagtypes[i].name, s, size);
tag_add_item(tag, tagtypes[i].type, s);
}
}
g_free(s);
WavpackCloseFile(wpc);
return tag;
}
/*
* mpd input_stream <=> WavpackStreamReader wrapper callbacks
*/
/* This struct is needed for per-stream last_byte storage. */
struct wavpack_input {
struct decoder *decoder;
struct input_stream *is;
/* Needed for push_back_byte() */
int last_byte;
};
/**
* Little wrapper for struct wavpack_input to cast from void *.
*/
static struct wavpack_input *
wpin(void *id)
{
assert(id);
return id;
}
static int32_t
wavpack_input_read_bytes(void *id, void *data, int32_t bcount)
{
uint8_t *buf = (uint8_t *)data;
int32_t i = 0;
if (wpin(id)->last_byte != EOF) {
*buf++ = wpin(id)->last_byte;
wpin(id)->last_byte = EOF;
--bcount;
++i;
}
/* wavpack fails if we return a partial read, so we just wait
until the buffer is full */
while (bcount > 0) {
size_t nbytes = decoder_read(
wpin(id)->decoder, wpin(id)->is, buf, bcount
);
if (nbytes == 0) {
/* EOF, error or a decoder command */
break;
}
i += nbytes;
bcount -= nbytes;
buf += nbytes;
}
return i;
}
static uint32_t
wavpack_input_get_pos(void *id)
{
return wpin(id)->is->offset;
}
static int
wavpack_input_set_pos_abs(void *id, uint32_t pos)
{
return input_stream_seek(wpin(id)->is, pos, SEEK_SET) ? 0 : -1;
}
static int
wavpack_input_set_pos_rel(void *id, int32_t delta, int mode)
{
return input_stream_seek(wpin(id)->is, delta, mode) ? 0 : -1;
}
static int
wavpack_input_push_back_byte(void *id, int c)
{
if (wpin(id)->last_byte == EOF) {
wpin(id)->last_byte = c;
return c;
} else {
return EOF;
}
}
static uint32_t
wavpack_input_get_length(void *id)
{
return wpin(id)->is->size;
}
static int
wavpack_input_can_seek(void *id)
{
return wpin(id)->is->seekable;
}
static WavpackStreamReader mpd_is_reader = {
.read_bytes = wavpack_input_read_bytes,
.get_pos = wavpack_input_get_pos,
.set_pos_abs = wavpack_input_set_pos_abs,
.set_pos_rel = wavpack_input_set_pos_rel,
.push_back_byte = wavpack_input_push_back_byte,
.get_length = wavpack_input_get_length,
.can_seek = wavpack_input_can_seek,
.write_bytes = NULL /* no need to write edited tags */
};
static void
wavpack_input_init(struct wavpack_input *isp, struct decoder *decoder,
struct input_stream *is)
{
isp->decoder = decoder;
isp->is = is;
isp->last_byte = EOF;
}
static bool
wavpack_open_wvc(struct decoder *decoder, struct input_stream *is_wvc,
struct wavpack_input *wpi)
{
char tmp[MPD_PATH_MAX];
const char *utf8url;
char *wvc_url = NULL;
bool ret;
char first_byte;
size_t nbytes;
/*
* As we use dc->utf8url, this function will be bad for
* single files. utf8url is not absolute file path :/
*/
utf8url = decoder_get_url(decoder, tmp);
if (utf8url == NULL) {
return false;
}
wvc_url = g_strconcat(utf8url, "c", NULL);
ret = input_stream_open(is_wvc, wvc_url);
g_free(wvc_url);
if (!ret) {
return false;
}
/*
* And we try to buffer in order to get know
* about a possible 404 error.
*/
nbytes = decoder_read(
decoder, is_wvc, &first_byte, sizeof(first_byte)
);
if (nbytes == 0) {
input_stream_close(is_wvc);
return false;
}
/* push it back */
wavpack_input_init(wpi, decoder, is_wvc);
wpi->last_byte = first_byte;
return true;
}
/*
* Decodes a stream.
*/
static void
wavpack_streamdecode(struct decoder * decoder, struct input_stream *is)
{
char error[ERRORLEN];
WavpackContext *wpc;
struct input_stream is_wvc;
int open_flags = OPEN_2CH_MAX | OPEN_NORMALIZE /*| OPEN_STREAMING*/;
struct wavpack_input isp, isp_wvc;
bool can_seek = is->seekable;
if (wavpack_open_wvc(decoder, &is_wvc, &isp_wvc)) {
open_flags |= OPEN_WVC;
can_seek &= is_wvc.seekable;
}
wavpack_input_init(&isp, decoder, is);
wpc = WavpackOpenFileInputEx(
&mpd_is_reader, &isp, &isp_wvc, error, open_flags, 23
);
if (wpc == NULL) {
g_warning("failed to open WavPack stream: %s\n", error);
return;
}
wavpack_decode(decoder, wpc, can_seek, NULL);
WavpackCloseFile(wpc);
if (open_flags & OPEN_WVC) {
input_stream_close(&is_wvc);
}
}
/*
* Decodes a file.
*/
static void
wavpack_filedecode(struct decoder *decoder, const char *fname)
{
char error[ERRORLEN];
WavpackContext *wpc;
struct replay_gain_info *replay_gain_info;
wpc = WavpackOpenFileInput(
fname, error,
OPEN_TAGS | OPEN_WVC | OPEN_2CH_MAX | OPEN_NORMALIZE, 23
);
if (wpc == NULL) {
g_warning(
"failed to open WavPack file \"%s\": %s\n",
fname, error
);
return;
}
replay_gain_info = wavpack_replaygain(wpc);
wavpack_decode(decoder, wpc, true, replay_gain_info);
if (replay_gain_info) {
replay_gain_info_free(replay_gain_info);
}
WavpackCloseFile(wpc);
}
static char const *const wavpack_suffixes[] = {
"wv",
NULL
};
static char const *const wavpack_mime_types[] = {
"audio/x-wavpack",
NULL
};
const struct decoder_plugin wavpack_plugin = {
.name = "wavpack",
.stream_decode = wavpack_streamdecode,
.file_decode = wavpack_filedecode,
.tag_dup = wavpack_tagdup,
.suffixes = wavpack_suffixes,
.mime_types = wavpack_mime_types
};