mirror of
https://github.com/Shpoike/Quakespasm.git
synced 2024-11-10 07:21:58 +00:00
81adf62374
- New console command music_jump: Jump to given order in music, like Unreal's music change. Only for module (tracker) music. - stream layer: Store the loop setting. - umx reader: Replaced questionable byte-swap. - libxmp backend: Handled the loop setting through libxmp apis. - mikmod backend: Enabled in-module loops. Respect global loop setting. More compatible reader callback structure.
211 lines
4.8 KiB
C
211 lines
4.8 KiB
C
/*
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* Ogg/Opus streaming music support, loosely based on several open source
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* Quake engine based projects with many modifications.
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*
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* Copyright (C) 2012-2013 O.Sezer <sezero@users.sourceforge.net>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or (at
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* your option) any later version.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
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*
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* See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*
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*/
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#include "quakedef.h"
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#if defined(USE_CODEC_OPUS)
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#include "snd_codec.h"
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#include "snd_codeci.h"
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#include "snd_opus.h"
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#include <errno.h>
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#include <opusfile.h>
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/* CALLBACK FUNCTIONS: */
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static int opc_fclose (void *f)
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{
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return 0; /* we fclose() elsewhere. */
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}
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static int opc_fread (void *f, unsigned char *buf, int size)
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{
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int ret;
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if (size < 0)
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{
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errno = EINVAL;
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return -1;
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}
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ret = (int) FS_fread(buf, 1, (size_t)size, (fshandle_t *)f);
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if (ret == 0 && errno != 0)
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ret = -1;
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return ret;
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}
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static int opc_fseek (void *f, opus_int64 off, int whence)
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{
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if (f == NULL) return (-1);
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return FS_fseek((fshandle_t *)f, (long) off, whence);
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}
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static opus_int64 opc_ftell (void *f)
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{
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return (opus_int64) FS_ftell((fshandle_t *)f);
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}
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static const OpusFileCallbacks opc_qfs =
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{
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(int (*)(void *, unsigned char *, int)) opc_fread,
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(int (*)(void *, opus_int64, int)) opc_fseek,
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(opus_int64 (*)(void *)) opc_ftell,
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(int (*)(void *)) opc_fclose
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};
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static qboolean S_OPUS_CodecInitialize (void)
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{
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return true;
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}
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static void S_OPUS_CodecShutdown (void)
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{
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}
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static qboolean S_OPUS_CodecOpenStream (snd_stream_t *stream)
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{
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OggOpusFile *opFile;
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const OpusHead *op_info;
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long numstreams;
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int res;
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opFile = op_open_callbacks(&stream->fh, &opc_qfs, NULL, 0, &res);
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if (!opFile)
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{
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Con_Printf("%s is not a valid Opus file (error %i).\n",
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stream->name, res);
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goto _fail;
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}
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stream->priv = opFile;
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if (!op_seekable(opFile))
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{
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Con_Printf("Opus stream %s not seekable.\n", stream->name);
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goto _fail;
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}
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op_info = op_head(opFile, -1);
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if (!op_info)
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{
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Con_Printf("Unable to get stream information for %s.\n", stream->name);
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goto _fail;
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}
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/* FIXME: handle section changes */
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numstreams = op_info->stream_count;
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if (numstreams != 1)
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{
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Con_Printf("More than one (%ld) stream in %s\n",
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(long)op_info->stream_count, stream->name);
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goto _fail;
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}
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if (op_info->channel_count != 1 && op_info->channel_count != 2)
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{
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Con_Printf("Unsupported number of channels %d in %s\n",
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op_info->channel_count, stream->name);
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goto _fail;
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}
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/* All Opus audio is coded at 48 kHz, and should also be decoded
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* at 48 kHz for playback: info->input_sample_rate only tells us
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* the sampling rate of the original input before opus encoding.
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* S_RawSamples() shall already downsample this, as necessary. */
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stream->info.rate = 48000;
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stream->info.channels = op_info->channel_count;
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/* op_read() yields 16-bit output using native endian ordering: */
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stream->info.bits = 16;
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stream->info.width = 2;
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return true;
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_fail:
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if (opFile)
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op_free(opFile);
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return false;
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}
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static int S_OPUS_CodecReadStream (snd_stream_t *stream, int bytes, void *buffer)
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{
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int section; /* FIXME: handle section changes */
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int cnt, res, rem;
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opus_int16 * ptr;
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rem = bytes / stream->info.width;
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if (rem / stream->info.channels <= 0)
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return 0;
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cnt = 0;
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ptr = (opus_int16 *) buffer;
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while (1)
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{
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/* op_read() yields 16-bit output using native endian ordering. returns
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* the number of samples read per channel on success, or a negative value
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* on failure. */
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res = op_read((OggOpusFile *)stream->priv, ptr, rem, §ion);
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if (res <= 0)
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break;
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cnt += res;
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res *= stream->info.channels;
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rem -= res;
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if (rem <= 0)
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break;
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ptr += res;
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}
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if (res < 0)
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return res;
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cnt *= (stream->info.channels * stream->info.width);
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return cnt;
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}
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static void S_OPUS_CodecCloseStream (snd_stream_t *stream)
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{
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op_free((OggOpusFile *)stream->priv);
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S_CodecUtilClose(&stream);
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}
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static int S_OPUS_CodecRewindStream (snd_stream_t *stream)
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{
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return op_pcm_seek ((OggOpusFile *)stream->priv, 0);
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}
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snd_codec_t opus_codec =
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{
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CODECTYPE_OPUS,
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true, /* always available. */
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"opus",
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S_OPUS_CodecInitialize,
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S_OPUS_CodecShutdown,
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S_OPUS_CodecOpenStream,
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S_OPUS_CodecReadStream,
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S_OPUS_CodecRewindStream,
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NULL, /* jump */
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S_OPUS_CodecCloseStream,
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NULL
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};
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#endif /* USE_CODEC_OPUS */
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