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Merge branch 'codec/mp3' into game/eliteforce

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This commit is contained in:
Zack Middleton 2014-10-29 06:39:10 -05:00
commit 4d666212b2
5 changed files with 745 additions and 0 deletions
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15
Makefile
View file

@ -167,6 +167,10 @@ ifndef USE_CURL_DLOPEN
endif
endif
ifndef USE_CODEC_MP3
USE_CODEC_MP3=0
endif
ifndef USE_CODEC_VORBIS
USE_CODEC_VORBIS=0
endif
@ -992,6 +996,16 @@ ifeq ($(USE_CURL),1)
endif
endif
ifeq ($(USE_CODEC_MP3),1)
CLIENT_CFLAGS += -DUSE_CODEC_MP3
MAD_CFLAGS ?= $(shell pkg-config --silence-errors --cflags mad || true)
MAD_LIBS ?= $(shell pkg-config --silence-errors --libs mad || echo -lmad)
CLIENT_CFLAGS += $(MAD_CFLAGS)
CLIENT_LIBS += $(MAD_LIBS)
endif
ifeq ($(USE_CODEC_OPUS),1)
CLIENT_CFLAGS += -DUSE_CODEC_OPUS
ifeq ($(USE_INTERNAL_OPUS),1)
@ -1578,6 +1592,7 @@ Q3OBJ = \
$(B)/client/snd_main.o \
$(B)/client/snd_codec.o \
$(B)/client/snd_codec_wav.o \
$(B)/client/snd_codec_mp3.o \
$(B)/client/snd_codec_ogg.o \
$(B)/client/snd_codec_opus.o \
\

1
README.md
View file

@ -95,6 +95,7 @@ Makefile.local:
USE_OPENAL_DLOPEN - link with OpenAL at runtime
USE_CURL - use libcurl for http/ftp download support
USE_CURL_DLOPEN - link with libcurl at runtime
USE_CODEC_MP3 - enable MP3 support
USE_CODEC_VORBIS - enable Ogg Vorbis support
USE_CODEC_OPUS - enable Ogg Opus support
USE_MUMBLE - enable Mumble support

4
code/client/snd_codec.c
View file

@ -132,6 +132,10 @@ void S_CodecInit()
S_CodecRegister(&ogg_codec);
#endif
#ifdef USE_CODEC_MP3
S_CodecRegister(&mp3_codec);
#endif
// Register wav codec last so that it is always tried first when a file extension was not found
S_CodecRegister(&wav_codec);
}

9
code/client/snd_codec.h
View file

@ -86,6 +86,15 @@ snd_stream_t *S_WAV_CodecOpenStream(const char *filename);
void S_WAV_CodecCloseStream(snd_stream_t *stream);
int S_WAV_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer);
// MP3 codec
#ifdef USE_CODEC_MP3
extern snd_codec_t mp3_codec;
void *S_MP3_CodecLoad(const char *filename, snd_info_t *info);
snd_stream_t *S_MP3_CodecOpenStream(const char *filename);
void S_MP3_CodecCloseStream(snd_stream_t *stream);
int S_MP3_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer);
#endif // USE_CODEC_MP3
// Ogg Vorbis codec
#ifdef USE_CODEC_VORBIS
extern snd_codec_t ogg_codec;

716
code/client/snd_codec_mp3.c Normal file
View file

@ -0,0 +1,716 @@
/*
===========================================================================
Copyright (C) 1999-2005 Id Software, Inc.
Copyright (C) 2005 Stuart Dalton (badcdev@gmail.com)
Copyright (C) 2005-2006 Joerg Dietrich <dietrich_joerg@gmx.de>
Copyright (C) 2006 Thilo Schulz <arny@ats.s.bawue.de>
This file is part of Quake III Arena source code.
Quake III Arena source code 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.
Quake III Arena source code 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 Quake III Arena source code; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
===========================================================================
*/
// MP3 support is enabled by this define
#ifdef USE_CODEC_MP3
// includes for the Q3 sound system
#include "client.h"
#include "snd_codec.h"
// includes for the MP3 codec
#include <mad.h>
#define MP3_SAMPLE_WIDTH 2
#define MP3_PCMSAMPLES_PERSLICE 32
// buffer size used when reading through the mp3
#define MP3_DATA_BUFSIZ 128*1024
// undefine this if you don't want any dithering.
#define MP3_DITHERING
// Q3 MP3 codec
snd_codec_t mp3_codec =
{
"mp3",
S_MP3_CodecLoad,
S_MP3_CodecOpenStream,
S_MP3_CodecReadStream,
S_MP3_CodecCloseStream,
NULL
};
// structure used for info purposes
struct snd_codec_mp3_info
{
byte encbuf[MP3_DATA_BUFSIZ]; // left over bytes not consumed
// by the decoder.
struct mad_stream madstream; // uses encbuf as buffer.
struct mad_frame madframe; // control structures for libmad.
struct mad_synth madsynth;
byte *pcmbuf; // buffer for not-used samples.
int buflen; // length of buffer data.
int pcmbufsize; // amount of allocated memory for
// pcmbuf. This should have at least
// the size of a decoded mp3 frame.
byte *dest; // copy decoded data here.
int destlen; // amount of already copied data.
int destsize; // amount of bytes we must decode.
};
/*************** MP3 utility functions ***************/
/*
=================
S_MP3_ReadData
=================
*/
// feed libmad with data
int S_MP3_ReadData(snd_stream_t *stream, struct mad_stream *madstream, byte *encbuf, int encbufsize)
{
int retval;
int leftover;
if(!stream)
return -1;
leftover = madstream->bufend - madstream->next_frame;
if(leftover > 0)
memmove(encbuf, madstream->this_frame, leftover);
// Fill the buffer right to the end
retval = FS_Read(&encbuf[leftover], encbufsize - leftover, stream->file);
if(retval <= 0)
{
// EOF reached, that's ok.
return 0;
}
mad_stream_buffer(madstream, encbuf, retval + leftover);
return retval;
}
/*
=================
S_MP3_Scanfile
to determine the samplecount, we apparently must get *all* headers :(
I basically used the xmms-mad plugin source to see how this stuff works.
returns a value < 0 on error.
=================
*/
int S_MP3_Scanfile(snd_stream_t *stream)
{
struct mad_stream madstream;
struct mad_header madheader;
int retval;
int samplecount;
byte encbuf[MP3_DATA_BUFSIZ];
// error out on invalid input.
if(!stream)
return -1;
mad_stream_init(&madstream);
mad_header_init(&madheader);
while(1)
{
retval = S_MP3_ReadData(stream, &madstream, encbuf, sizeof(encbuf));
if(retval < 0)
return -1;
else if(retval == 0)
break;
// Start decoding the headers.
while(1)
{
if((retval = mad_header_decode(&madheader, &madstream)) < 0)
{
if(madstream.error == MAD_ERROR_BUFLEN)
{
// We need to read more data
break;
}
if(!MAD_RECOVERABLE (madstream.error))
{
// unrecoverable error... we must bail out.
return retval;
}
mad_stream_skip(&madstream, madstream.skiplen);
continue;
}
// we got a valid header.
if(madheader.layer != MAD_LAYER_III)
{
// we don't support non-mp3s
return -1;
}
if(!stream->info.samples)
{
// This here is the very first frame. Set initial values now,
// that we expect to stay constant throughout the whole mp3.
stream->info.rate = madheader.samplerate;
stream->info.width = MP3_SAMPLE_WIDTH;
stream->info.channels = MAD_NCHANNELS(&madheader);
stream->info.samples = 0;
stream->info.size = 0; // same here.
stream->info.dataofs = 0;
}
else
{
// Check whether something changed that shouldn't.
if(stream->info.rate != madheader.samplerate ||
stream->info.channels != MAD_NCHANNELS(&madheader))
return -1;
}
// Update the counters
samplecount = MAD_NSBSAMPLES(&madheader) * MP3_PCMSAMPLES_PERSLICE;
stream->info.samples += samplecount;
stream->info.size += samplecount * stream->info.channels * stream->info.width;
}
}
// Reset the file pointer so we can do the real decoding.
FS_Seek(stream->file, 0, FS_SEEK_SET);
return 0;
}
/************************ dithering functions ***************************/
#ifdef MP3_DITHERING
// All dithering done here is taken from the GPL'ed xmms-mad plugin.
/* Copyright (C) 1997 Makoto Matsumoto and Takuji Nishimura. */
/* Any feedback is very welcome. For any question, comments, */
/* see http://www.math.keio.ac.jp/matumoto/emt.html or email */
/* matumoto@math.keio.ac.jp */
/* Period parameters */
#define MP3_DITH_N 624
#define MP3_DITH_M 397
#define MATRIX_A 0x9908b0df /* constant vector a */
#define UPPER_MASK 0x80000000 /* most significant w-r bits */
#define LOWER_MASK 0x7fffffff /* least significant r bits */
/* Tempering parameters */
#define TEMPERING_MASK_B 0x9d2c5680
#define TEMPERING_MASK_C 0xefc60000
#define TEMPERING_SHIFT_U(y) (y >> 11)
#define TEMPERING_SHIFT_S(y) (y << 7)
#define TEMPERING_SHIFT_T(y) (y << 15)
#define TEMPERING_SHIFT_L(y) (y >> 18)
static unsigned long mt[MP3_DITH_N]; /* the array for the state vector */
static int mti=MP3_DITH_N+1; /* mti==MP3_DITH_N+1 means mt[MP3_DITH_N] is not initialized */
/* initializing the array with a NONZERO seed */
void sgenrand(unsigned long seed)
{
/* setting initial seeds to mt[MP3_DITH_N] using */
/* the generator Line 25 of Table 1 in */
/* [KNUTH 1981, The Art of Computer Programming */
/* Vol. 2 (2nd Ed.), pp102] */
mt[0]= seed & 0xffffffff;
for (mti=1; mti<MP3_DITH_N; mti++)
mt[mti] = (69069 * mt[mti-1]) & 0xffffffff;
}
unsigned long genrand(void)
{
unsigned long y;
static unsigned long mag01[2]={0x0, MATRIX_A};
/* mag01[x] = x * MATRIX_A for x=0,1 */
if (mti >= MP3_DITH_N) { /* generate MP3_DITH_N words at one time */
int kk;
if (mti == MP3_DITH_N+1) /* if sgenrand() has not been called, */
sgenrand(4357); /* a default initial seed is used */
for (kk=0;kk<MP3_DITH_N-MP3_DITH_M;kk++) {
y = (mt[kk]&UPPER_MASK)|(mt[kk+1]&LOWER_MASK);
mt[kk] = mt[kk+MP3_DITH_M] ^ (y >> 1) ^ mag01[y & 0x1];
}
for (;kk<MP3_DITH_N-1;kk++) {
y = (mt[kk]&UPPER_MASK)|(mt[kk+1]&LOWER_MASK);
mt[kk] = mt[kk+(MP3_DITH_M-MP3_DITH_N)] ^ (y >> 1) ^ mag01[y & 0x1];
}
y = (mt[MP3_DITH_N-1]&UPPER_MASK)|(mt[0]&LOWER_MASK);
mt[MP3_DITH_N-1] = mt[MP3_DITH_M-1] ^ (y >> 1) ^ mag01[y & 0x1];
mti = 0;
}
y = mt[mti++];
y ^= TEMPERING_SHIFT_U(y);
y ^= TEMPERING_SHIFT_S(y) & TEMPERING_MASK_B;
y ^= TEMPERING_SHIFT_T(y) & TEMPERING_MASK_C;
y ^= TEMPERING_SHIFT_L(y);
return y;
}
long triangular_dither_noise(int nbits) {
// parameter nbits : the peak-to-peak amplitude desired (in bits)
// use with nbits set to 2 + nber of bits to be trimmed.
// (because triangular is made from two uniformly distributed processes,
// it starts at 2 bits peak-to-peak amplitude)
// see The Theory of Dithered Quantization by Robert Alexander Wannamaker
// for complete proof of why that's optimal
long v = (genrand()/2 - genrand()/2); // in ]-2^31, 2^31[
//int signe = (v>0) ? 1 : -1;
long P = 1 << (32 - nbits); // the power of 2
v /= P;
// now v in ]-2^(nbits-1), 2^(nbits-1) [
return v;
}
#endif // MP3_DITHERING
/************************ decoder functions ***************************/
/*
=================
S_MP3_Scale
Converts the signal to 16 bit LE-PCM data and does dithering.
- borrowed from xmms-mad plugin source.
=================
*/
/*
* xmms-mad - mp3 plugin for xmms
* Copyright (C) 2001-2002 Sam Clegg
*/
signed int S_MP3_Scale(mad_fixed_t sample)
{
int n_bits_to_loose = MAD_F_FRACBITS + 1 - 16;
#ifdef MP3_DITHERING
int dither;
#endif
// round
sample += (1L << (n_bits_to_loose - 1));
#ifdef MP3_DITHERING
dither = triangular_dither_noise(n_bits_to_loose + 1);
sample += dither;
#endif
/* clip */
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
/* quantize */
return sample >> n_bits_to_loose;
}
/*
=================
S_MP3_PCMCopy
Copy and convert pcm data until bytecount bytes have been written.
return the position in pcm->samples.
indicate the amount of actually written bytes in wrotecnt.
=================
*/
int S_MP3_PCMCopy(byte *buf, struct mad_pcm *pcm, int bufofs,
int sampleofs, int bytecount, int *wrotecnt)
{
int written = 0;
signed int sample;
int framesize = pcm->channels * MP3_SAMPLE_WIDTH;
// add new pcm data.
while(written < bytecount && sampleofs < pcm->length)
{
sample = S_MP3_Scale(pcm->samples[0][sampleofs]);
#ifdef Q3_BIG_ENDIAN
// output to 16 bit big endian PCM
buf[bufofs++] = (sample >> 8) & 0xff;
buf[bufofs++] = sample & 0xff;
#else
// output to 16 bit little endian PCM
buf[bufofs++] = sample & 0xff;
buf[bufofs++] = (sample >> 8) & 0xff;
#endif
if(pcm->channels == 2)
{
sample = S_MP3_Scale(pcm->samples[1][sampleofs]);
#ifdef Q3_BIG_ENDIAN
buf[bufofs++] = (sample >> 8) & 0xff;
buf[bufofs++] = sample & 0xff;
#else
buf[bufofs++] = sample & 0xff;
buf[bufofs++] = (sample >> 8) & 0xff;
#endif
}
sampleofs++;
written += framesize;
}
if(wrotecnt)
*wrotecnt = written;
return sampleofs;
}
/*
=================
S_MP3_Decode
=================
*/
// gets executed for every decoded frame.
int S_MP3_Decode(snd_stream_t *stream)
{
struct snd_codec_mp3_info *mp3info;
struct mad_stream *madstream;
struct mad_frame *madframe;
struct mad_synth *madsynth;
struct mad_pcm *pcm;
int cursize;
int samplecount;
int needcount;
int wrote;
int retval;
if(!stream)
return -1;
mp3info = stream->ptr;
madstream = &mp3info->madstream;
madframe = &mp3info->madframe;
if(mad_frame_decode(madframe, madstream))
{
if(madstream->error == MAD_ERROR_BUFLEN)
{
// we need more data. Read another chunk.
retval = S_MP3_ReadData(stream, madstream, mp3info->encbuf, sizeof(mp3info->encbuf));
// call myself again now that buffer is full.
if(retval > 0)
retval = S_MP3_Decode(stream);
}
else if(MAD_RECOVERABLE(madstream->error))
{
mad_stream_skip(madstream, madstream->skiplen);
return S_MP3_Decode(stream);
}
else
retval = -1;
return retval;
}
// check whether this really is an mp3
if(madframe->header.layer != MAD_LAYER_III)
return -1;
// generate pcm data
madsynth = &mp3info->madsynth;
mad_synth_frame(madsynth, madframe);
pcm = &madsynth->pcm;
// perform a few checks to see whether something changed that shouldn't.
if(stream->info.rate != pcm->samplerate ||
stream->info.channels != pcm->channels)
{
return -1;
}
// see whether we have got enough data now.
cursize = pcm->length * pcm->channels * stream->info.width;
needcount = mp3info->destsize - mp3info->destlen;
// Copy exactly as many samples as required.
samplecount = S_MP3_PCMCopy(mp3info->dest, pcm,
mp3info->destlen, 0, needcount, &wrote);
mp3info->destlen += wrote;
if(samplecount < pcm->length)
{
// Not all samples got copied. Copy the rest into the pcm buffer.
samplecount = S_MP3_PCMCopy(mp3info->pcmbuf, pcm,
mp3info->buflen,
samplecount,
mp3info->pcmbufsize - mp3info->buflen,
&wrote);
mp3info->buflen += wrote;
if(samplecount < pcm->length)
{
// The pcm buffer was not large enough. Make it bigger.
byte *newbuf = Z_Malloc(cursize);
if(mp3info->pcmbuf)
{
memcpy(newbuf, mp3info->pcmbuf, mp3info->buflen);
Z_Free(mp3info->pcmbuf);
}
mp3info->pcmbuf = newbuf;
mp3info->pcmbufsize = cursize;
samplecount = S_MP3_PCMCopy(mp3info->pcmbuf, pcm,
mp3info->buflen,
samplecount,
mp3info->pcmbufsize - mp3info->buflen,
&wrote);
mp3info->buflen += wrote;
}
// we're definitely done.
retval = 0;
}
else if(mp3info->destlen >= mp3info->destsize)
retval = 0;
else
retval = 1;
return retval;
}
/*************** Callback functions for quake3 ***************/
/*
=================
S_MP3_CodecOpenStream
=================
*/
snd_stream_t *S_MP3_CodecOpenStream(const char *filename)
{
snd_stream_t *stream;
struct snd_codec_mp3_info *mp3info;
// Open the stream
stream = S_CodecUtilOpen(filename, &mp3_codec);
if(!stream || stream->length <= 0)
return NULL;
// We have to scan through the MP3 to determine the important mp3 info.
if(S_MP3_Scanfile(stream) < 0)
{
// scanning didn't work out...
S_CodecUtilClose(&stream);
return NULL;
}
// Initialize the mp3 info structure we need for streaming
mp3info = Z_Malloc(sizeof(*mp3info));
if(!mp3info)
{
S_CodecUtilClose(&stream);
return NULL;
}
stream->ptr = mp3info;
// initialize the libmad control structures.
mad_stream_init(&mp3info->madstream);
mad_frame_init(&mp3info->madframe);
mad_synth_init(&mp3info->madsynth);
if(S_MP3_ReadData(stream, &mp3info->madstream, mp3info->encbuf, sizeof(mp3info->encbuf)) <= 0)
{
// we didnt read anything, that's bad.
S_MP3_CodecCloseStream(stream);
return NULL;
}
return stream;
}
/*
=================
S_MP3_CodecCloseStream
=================
*/
// free all memory we allocated.
void S_MP3_CodecCloseStream(snd_stream_t *stream)
{
struct snd_codec_mp3_info *mp3info;
if(!stream)
return;
// free all data in our mp3info tree
if(stream->ptr)
{
mp3info = stream->ptr;
if(mp3info->pcmbuf)
Z_Free(mp3info->pcmbuf);
mad_synth_finish(&mp3info->madsynth);
mad_frame_finish(&mp3info->madframe);
mad_stream_finish(&mp3info->madstream);
Z_Free(stream->ptr);
}
S_CodecUtilClose(&stream);
}
/*
=================
S_MP3_CodecReadStream
=================
*/
int S_MP3_CodecReadStream(snd_stream_t *stream, int bytes, void *buffer)
{
struct snd_codec_mp3_info *mp3info;
int retval;
if(!stream)
return -1;
mp3info = stream->ptr;
// Make sure we get complete frames all the way through.
bytes -= bytes % (stream->info.channels * stream->info.width);
if(mp3info->buflen)
{
if(bytes < mp3info->buflen)
{
// we still have enough bytes in our decoded pcm buffer
memcpy(buffer, mp3info->pcmbuf, bytes);
// remove the portion from our buffer.
mp3info->buflen -= bytes;
memmove(mp3info->pcmbuf, &mp3info->pcmbuf[bytes], mp3info->buflen);
return bytes;
}
else
{
// copy over the samples we already have.
memcpy(buffer, mp3info->pcmbuf, mp3info->buflen);
mp3info->destlen = mp3info->buflen;
mp3info->buflen = 0;
}
}
else
mp3info->destlen = 0;
mp3info->dest = buffer;
mp3info->destsize = bytes;
do
{
retval = S_MP3_Decode(stream);
} while(retval > 0);
// if there was an error return nothing.
if(retval < 0)
return 0;
return mp3info->destlen;
}
/*
=====================================================================
S_MP3_CodecLoad
We handle S_MP3_CodecLoad as a special case of the streaming functions
where we read the whole stream at once.
======================================================================
*/
void *S_MP3_CodecLoad(const char *filename, snd_info_t *info)
{
snd_stream_t *stream;
byte *pcmbuffer;
// check if input is valid
if(!filename)
return NULL;
stream = S_MP3_CodecOpenStream(filename);
if(!stream)
return NULL;
// copy over the info
info->rate = stream->info.rate;
info->width = stream->info.width;
info->channels = stream->info.channels;
info->samples = stream->info.samples;
info->dataofs = stream->info.dataofs;
// allocate enough buffer for all pcm data
pcmbuffer = Hunk_AllocateTempMemory(stream->info.size);
if(!pcmbuffer)
{
S_MP3_CodecCloseStream(stream);
return NULL;
}
info->size = S_MP3_CodecReadStream(stream, stream->info.size, pcmbuffer);
if(info->size <= 0)
{
// we didn't read anything at all. darn.
Hunk_FreeTempMemory(pcmbuffer);
pcmbuffer = NULL;
}
S_MP3_CodecCloseStream(stream);
return pcmbuffer;
}
#endif // USE_CODEC_MP3