quakespasm/Quake/snd_opus.c
Ozkan Sezer dd9f105e42 * snd_codec: store the samplebits value in the snd_info_t struct,
and add a new blocksize field to it which the flac decoder can
use. updated decoder sources for the snd_info_t changes, where
I made minor tidy-ups too, tightening several format checks and
fixing a few gotchas in snd_wave.c and snd_mem.c.
* snd_flac.c: adjusted for snd_info_t changes. no longed storing
metadata->data.stream_info in our private data, but just storing
a pointer to the stream->info structure. No longer checking the
metadata total_samples field (the FLAC__StreamMetadata_StreamInfo
doesn't seem to have any alignment or pack attributes and I don't
safe with its offset across different compilers), but added check
to make sure that we hit and parsed a STREAMINFO metadata instead,
and our new state seems just fine for validating the file.


git-svn-id: svn://svn.code.sf.net/p/quakespasm/code/trunk/quakespasm@859 af15c1b1-3010-417e-b628-4374ebc0bcbd
2013-07-22 11:41:14 +00:00

216 lines
4.9 KiB
C

/*
* Ogg/Opus streaming music support, loosely based on several open source
* Quake engine based projects with many modifications.
*
* Copyright (C) 2012-2013 O.Sezer <sezero@users.sourceforge.net>
*
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
#include "quakedef.h"
#if defined(USE_CODEC_OPUS)
#include "snd_codec.h"
#include "snd_codeci.h"
#include "snd_opus.h"
#include <errno.h>
#include <opusfile.h>
/* CALLBACK FUNCTIONS: */
static int opc_fclose (void *f)
{
return 0; /* we fclose() elsewhere. */
}
static int opc_fread (void *f, unsigned char *buf, int size)
{
int ret;
if (size < 0)
{
errno = EINVAL;
return -1;
}
ret = (int) FS_fread(buf, 1, (size_t)size, (fshandle_t *)f);
if (ret == 0 && errno != 0)
ret = -1;
return ret;
}
static int opc_fseek (void *f, opus_int64 off, int whence)
{
if (f == NULL) return (-1);
return FS_fseek((fshandle_t *)f, (long) off, whence);
}
static opus_int64 opc_ftell (void *f)
{
return (opus_int64) FS_ftell((fshandle_t *)f);
}
static const OpusFileCallbacks opc_qfs =
{
(int (*)(void *, unsigned char *, int)) opc_fread,
(int (*)(void *, opus_int64, int)) opc_fseek,
(opus_int64 (*)(void *)) opc_ftell,
(int (*)(void *)) opc_fclose
};
static qboolean S_OPUS_CodecInitialize (void)
{
return true;
}
static void S_OPUS_CodecShutdown (void)
{
}
static snd_stream_t *S_OPUS_CodecOpenStream (const char *filename)
{
snd_stream_t *stream;
OggOpusFile *opFile;
const OpusHead *op_info;
long numstreams;
int res;
stream = S_CodecUtilOpen(filename, &opus_codec);
if (!stream)
return NULL;
opFile = op_open_callbacks(&stream->fh, &opc_qfs, NULL, 0, &res);
if (!opFile)
{
Con_Printf("%s is not a valid Opus file (error %i).\n",
filename, res);
goto _fail;
}
stream->priv = opFile;
if (!op_seekable(opFile))
{
Con_Printf("Opus stream %s not seekable.\n", filename);
goto _fail;
}
op_info = op_head(opFile, -1);
if (!op_info)
{
Con_Printf("Unable to get stream information for %s.\n", filename);
goto _fail;
}
/* FIXME: handle section changes */
numstreams = op_info->stream_count;
if (numstreams != 1)
{
Con_Printf("More than one (%ld) stream in %s\n",
(long)op_info->stream_count, filename);
goto _fail;
}
if (op_info->channel_count != 1 && op_info->channel_count != 2)
{
Con_Printf("Unsupported number of channels %d in %s\n",
op_info->channel_count, filename);
goto _fail;
}
/* All Opus audio is coded at 48 kHz, and should also be decoded
* at 48 kHz for playback: info->input_sample_rate only tells us
* the sampling rate of the original input before opus encoding.
* S_RawSamples() shall already downsample this, as necessary. */
stream->info.rate = 48000;
stream->info.channels = op_info->channel_count;
/* op_read() yields 16-bit output using native endian ordering: */
stream->info.bits = 16;
stream->info.width = 2;
return stream;
_fail:
if (opFile)
op_free(opFile);
S_CodecUtilClose(&stream);
return NULL;
}
static int S_OPUS_CodecReadStream (snd_stream_t *stream, int bytes, void *buffer)
{
int section; /* FIXME: handle section changes */
int cnt, res, rem;
opus_int16 * ptr;
rem = bytes / stream->info.width;
if (rem / stream->info.channels <= 0)
return 0;
cnt = 0;
ptr = (opus_int16 *) buffer;
while (1)
{
/* op_read() yields 16-bit output using native endian ordering. returns
* the number of samples read per channel on success, or a negative value
* on failure. */
res = op_read((OggOpusFile *)stream->priv, ptr, rem, &section);
if (res <= 0)
break;
cnt += res;
res *= stream->info.channels;
rem -= res;
if (rem <= 0)
break;
ptr += res;
}
if (res < 0)
return res;
cnt *= (stream->info.channels * stream->info.width);
return cnt;
}
static void S_OPUS_CodecCloseStream (snd_stream_t *stream)
{
op_free((OggOpusFile *)stream->priv);
S_CodecUtilClose(&stream);
}
static int S_OPUS_CodecRewindStream (snd_stream_t *stream)
{
return op_pcm_seek ((OggOpusFile *)stream->priv, 0);
}
snd_codec_t opus_codec =
{
CODECTYPE_OPUS,
true, /* always available. */
"opus",
S_OPUS_CodecInitialize,
S_OPUS_CodecShutdown,
S_OPUS_CodecOpenStream,
S_OPUS_CodecReadStream,
S_OPUS_CodecRewindStream,
S_OPUS_CodecCloseStream,
NULL
};
#endif /* USE_CODEC_OPUS */