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diff --git a/util/audio.cpp b/util/audio.cpp
index 0ddebb8a..69027f12 100644
--- a/util/audio.cpp
+++ b/util/audio.cpp
@@ -1,261 +1,350 @@
#include <string.h>
#include "audio.h"
#include "debug.h"
namespace Util{
#ifdef USE_SDL1
AudioConverter::AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
Encoding outputEncoding, int outputChannels, int outputFrequency){
SDL_BuildAudioCVT(&conversion, inputEncoding, inputChannels, inputFrequency,
outputEncoding, outputChannels, outputFrequency);
}
AudioConverter::~AudioConverter(){
}
int AudioConverter::convertedLength(int length){
return length;
}
int AudioConverter::convert(void * input, int length){
if (conversion.needed){
conversion.buf = (Uint8*) input;
conversion.len = length;
/* then convert to whatever the real output wants */
SDL_ConvertAudio(&conversion);
return conversion.len_cvt;
} else {
return length;
}
}
#else
AudioConverter::AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
Encoding outputEncoding, int outputChannels, int outputFrequency):
buffer(NULL),
bufferSize(0){
input.bytes = inputEncoding;
input.channels = inputChannels;
input.frequency = inputFrequency;
output.bytes = outputEncoding;
output.channels = outputChannels;
output.frequency = outputFrequency;
sizeRatio = (double) byteSize(output) * output.frequency / ((double) byteSize(input) * input.frequency);
}
int AudioConverter::byteSize(const Format & what){
return encodingBytes(what.bytes) * what.channels;
}
/* how many bytes an encoding takes up */
int AudioConverter::encodingBytes(Encoding what){
switch (what){
case Unsigned8: return 1;
case Signed16: return 2;
case Unsigned16: return 2;
case Float32: return 4;
}
return 1;
}
int AudioConverter::convertedLength(int length){
// return length * sizeRatio;
int total = length * sizeRatio;
/* make sure we get an even number of samples */
if (total % byteSize(output) != 0){
total -= total % byteSize(output);
}
return total;
}
double CubicInterpolate(double y0, double y1,
double y2, double y3,
double mu){
double a0,a1,a2,a3,mu2;
mu2 = mu*mu;
a0 = y3 - y2 - y0 + y1;
a1 = y0 - y1 - a0;
a2 = y2 - y0;
a3 = y1;
return (a0*mu*mu2+a1*mu2+a2*mu+a3);
}
template <class Input, class Output>
Output clamp(double input){
Output top = (1 << (sizeof(Output) * 8 - 1)) - 1;
Output bottom = -(1 << (sizeof(Output) * 8 - 1));
if (input > top){
return top;
}
if (input < bottom){
return bottom;
}
return input;
}
+template <>
+signed short clamp<unsigned char>(double input){
+ return ((input - 127) / 255) * (1 << (sizeof(signed short) * 8 - 1));
+}
+
+template <>
+unsigned char clamp<unsigned char>(double input){
+ if (input > 255){
+ input = 255;
+ }
+ if (input < 0){
+ input = 0;
+ }
+ return input;
+}
+
template <>
float clamp<float>(double input){
return input;
}
+
template <>
unsigned short clamp<signed short>(double input){
return (int) clamp<signed short, signed short>(input) + (int) (1 << (sizeof(signed short) * 8 - 1));
}
+template <>
+float clamp<short unsigned int>(double input){
+ double out = input / (1 << (sizeof(unsigned short) * 8));
+ if (out > 1){
+ return 1;
+ }
+ if (out < -1){
+ return -1;
+ }
+ return out;
+}
+
+template <>
+float clamp<unsigned char>(double input){
+ double out = input / (1 << (sizeof(unsigned char) * 8));
+ if (out > 1){
+ return 1;
+ }
+ if (out < -1){
+ return -1;
+ }
+ return out;
+
+}
+
template <>
unsigned char clamp<signed short>(double input){
double out = input / (1 << (sizeof(signed short) * 8 - 1));
if (out > 1){
return 1;
}
if (out < -1){
return -1;
}
/* -1,1 -> 0,255 */
return (unsigned char)((out + 1) * 128);
}
template <>
float clamp<signed short>(double input){
double out = input / (1 << (sizeof(signed short) * 8 - 1));
if (out > 1){
return 1;
}
if (out < -1){
return -1;
}
return out;
}
template <class SizeInput, class SizeOutput>
void doConvertRate(SizeInput * input, int inputSamples, int inputChannels, SizeOutput * buffer, int outputSamples, int outputChannels, double ratio){
for (int sample = 0; sample < outputSamples; sample += 1){
double inputSample = sample / ratio;
for (int channel = 0; channel < outputChannels; channel += 1){
int inputChannel = inputChannels > channel ? channel : inputChannels - 1;
int sample0 = ((int) inputSample - 1) * inputChannels + inputChannel;
int sample1 = ((int) inputSample + 0) * inputChannels + inputChannel;
int sample2 = ((int) inputSample + 1) * inputChannels + inputChannel;
int sample3 = ((int) inputSample + 2) * inputChannels + inputChannel;
if (sample0 < 0){
sample0 = sample1;
}
if (sample2 >= inputSamples * inputChannels){
sample2 = sample1;
}
if (sample3 >= inputSamples * inputChannels){
sample3 = sample2;
}
buffer[sample * outputChannels + channel] = clamp<SizeInput, SizeOutput>(CubicInterpolate(input[sample0], input[sample1], input[sample2], input[sample3], inputSample - (int) inputSample));
// Global::debug(0) << "Input[" << sample << "] " << channel << ": " << input[sample1] << " Output: " << buffer[sample * 2 + channel] << std::endl;
}
}
}
+template <class Input, class Output>
+void doConvert3(void * input, int inputLength, int inputChannels,
+ void * output, int outputLength, int outputChannels,
+ double ratio){
+ doConvertRate<Input, Output>((Input*) input, inputLength / sizeof(Input) / inputChannels, inputChannels,
+ (Output*) output, outputLength / sizeof(Output) / outputChannels, outputChannels,
+ ratio);
+}
+
+template <class Input>
+void doConvert2(void * input, int inputLength, int inputChannels,
+ Encoding outputType, void * output, int outputLength, int outputChannels,
+ double ratio){
+ switch (outputType){
+ case Unsigned8: doConvert3<Input, unsigned char>(input, inputLength, inputChannels, output, outputLength, outputChannels, ratio); break;
+ case Signed16: doConvert3<Input, signed short>(input, inputLength, inputChannels, output, outputLength, outputChannels, ratio); break;
+ case Unsigned16: doConvert3<Input, unsigned short>(input, inputLength, inputChannels, output, outputLength, outputChannels, ratio); break;
+ case Float32: doConvert3<Input, float>(input, inputLength, inputChannels, output, outputLength, outputChannels, ratio); break;
+ }
+}
+
+void doConvert1(Encoding inputType, void * input, int inputLength, int inputChannels,
+ Encoding outputType, void * output, int outputLength, int outputChannels, double ratio){
+ switch (inputType){
+ case Unsigned8: doConvert2<unsigned char>(input, inputLength, inputChannels, outputType, output, outputLength, outputChannels, ratio); break;
+ case Signed16: doConvert2<signed short>(input, inputLength, inputChannels, outputType, output, outputLength, outputChannels, ratio); break;
+ case Unsigned16: doConvert2<unsigned short>(input, inputLength, inputChannels, outputType, output, outputLength, outputChannels, ratio); break;
+ case Float32: doConvert2<float>(input, inputLength, inputChannels, outputType, output, outputLength, outputChannels, ratio); break;
+ }
+}
+
int AudioConverter::convert(void * input, int length){
/* no conversion needed */
if (this->input == this->output){
return length;
}
int total = convertedLength(length);
/*
if (total % byteSize(output) != 0){
total -= total % byteSize(output);
}
*/
/* cache the buffer for future use */
if (total > bufferSize){
delete[] buffer;
bufferSize = total;
buffer = new char[bufferSize];
}
double frequencyRatio = (double) output.frequency / (double) this->input.frequency;
+ doConvert1(this->input.bytes, input, length, this->input.channels, output.bytes, buffer, total, output.channels, frequencyRatio);
+
+ /*
switch (this->input.bytes){
+ case Unsigned8: {
+ switch (this->output.bytes){
+ case Unsigned8: doConvertRate<unsigned char, unsigned char>(
+ (unsigned char*) input, length / sizeof(unsigned char) / this->input.channels, this->input.channels,
+ (unsigned char*) buffer, total / sizeof(unsigned char) / output.channels, output.channels,
+ frequencyRatio); break;
+ case Signed16: doConvertRate<unsigned char, unsigned char>(
+ (unsigned char*) input, length / sizeof(unsigned char) / this->input.channels, this->input.channels,
+ (unsigned char*) buffer, total / sizeof(unsigned char) / output.channels, output.channels,
+ frequencyRatio); break;
+ }
+ }
case Signed16: {
switch (this->output.bytes){
case Signed16: doConvertRate<signed short, signed short>(
(signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
(signed short*) buffer, total / sizeof(signed short) / output.channels, output.channels,
frequencyRatio); break;
case Unsigned8: doConvertRate<signed short, unsigned char>(
(signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
(unsigned char*) buffer, total / sizeof(unsigned char) / output.channels, output.channels,
frequencyRatio); break;
case Unsigned16: doConvertRate<signed short, unsigned short>(
(signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
(unsigned short*) buffer, total / sizeof(unsigned short) / output.channels, output.channels,
frequencyRatio); break;
case Float32: doConvertRate<signed short, float>(
(signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
(float*) buffer, total / sizeof(float) / output.channels, output.channels,
frequencyRatio); break;
}
break;
}
case Float32: {
switch (this->output.bytes){
case Float32: doConvertRate<float, float>(
(float*) input, length / sizeof(float) / this->input.channels, this->input.channels,
(float*) buffer, total / sizeof(float) / output.channels, output.channels,
frequencyRatio); break;
default: break;
}
break;
}
default: break;
}
+ */
/*
if (this->input.bytes == output.bytes){
switch (this->input.bytes){
case Signed16: doConvertRate<signed short, signed short>((signed short*) input, length / sizeof(signed short) / this->input.channels, this->input.channels,
(signed short*) buffer, total / sizeof(signed short) / output.channels, output.channels,
frequencyRatio); break;
case Float32: doConvertRate<float, float>((float*) input, length / sizeof(float) / this->input.channels, this->input.channels,
(float*) buffer, total / sizeof(float) / output.channels, output.channels, frequencyRatio); break;
}
}
*/
memcpy(input, buffer, total);
return total;
}
AudioConverter::~AudioConverter(){
delete[] buffer;
}
bool AudioConverter::Format::operator==(const AudioConverter::Format & him) const {
return this->bytes == him.bytes &&
this->channels == him.channels &&
this->frequency == him.frequency;
}
#endif
}
diff --git a/util/audio.h b/util/audio.h
index 11694a8f..9417e0cb 100644
--- a/util/audio.h
+++ b/util/audio.h
@@ -1,65 +1,58 @@
#ifndef _paintown_audio_h
#define _paintown_audio_h
-#ifdef USE_SDL
-#include <SDL.h>
-#endif
-
/* Deals with audio conversion between any source format and any destination format.
* A format consists of
* byte encoding (8/16/32 bit, signed/unsigned, floating point/integer)
* number of channels (mono/stereo)
* frequency (22050hz, 44100hz, arbitrary hz)
*/
namespace Util{
/* endianness is always native */
enum Encoding{
Unsigned8,
Signed16,
Unsigned16,
Float32
};
class AudioConverter{
public:
AudioConverter(Encoding inputEncoding, int inputChannels, int inputFrequency,
Encoding outputEncoding, int outputChannels, int outputFrequency);
/* given some input length, return how long the converted output will be */
int convertedLength(int length);
/* convert the audio, put the output in the same buffer passed in -- 'input'
* and returns the number of converted samples.
* 'length' is the number of input samples in *bytes*
* 'input' should be large enough to hold convertedLength(length) samples
*/
int convert(void * input, int length);
virtual ~AudioConverter();
protected:
-#ifdef USE_SDL
- SDL_AudioCVT conversion;
-#endif
struct Format{
Encoding bytes;
int channels;
int frequency;
bool operator==(const Format & him) const;
};
int byteSize(const Format & what);
int encodingBytes(Encoding what);
Format input, output;
double sizeRatio;
char * buffer;
int bufferSize;
};
}
#endif
diff --git a/util/sdl/mixer/SConscript b/util/sdl/mixer/SConscript
index 7bf7dada..e208a8f8 100644
--- a/util/sdl/mixer/SConscript
+++ b/util/sdl/mixer/SConscript
@@ -1,22 +1,22 @@
Import('use')
mixer = use.Clone()
try:
if mixer['HAVE_OGG']:
mixer.Append(CPPDEFINES = ['OGG_MUSIC'])
except KeyError:
pass
try:
if mixer['HAVE_MP3_MAD']:
mixer.Append(CPPDEFINES = ['MP3_MAD_MUSIC'])
except KeyError:
pass
source = Split(""" music.c mixer.c music_ogg.c music_flac.c music_mad.c
dynamic_ogg.c effect_position.c effects_internal.c effect_stereoreverse.c
-load_voc.c load_ogg.c load_aiff.c wavestream.c """)
+load_voc.c load_ogg.c load_aiff.c wavestream.c convert.cpp""")
library = mixer.StaticLibrary('sdl-mixer', source)
Return('library')
diff --git a/util/sdl/mixer/convert.cpp b/util/sdl/mixer/convert.cpp
new file mode 100644
index 00000000..83ee54ce
--- /dev/null
+++ b/util/sdl/mixer/convert.cpp
@@ -0,0 +1,57 @@
+#include "convert.h"
+#include "SDL_mixer.h"
+#include "util/audio.h"
+
+Util::Encoding encoding(int format){
+ switch (format){
+ case AUDIO_U8: return Util::Unsigned8;
+ case AUDIO_S16: return Util::Signed16;
+ case AUDIO_U16: return Util::Unsigned16;
+#if SDL_VERSION_ATLEAST(1, 3, 0)
+ case AUDIO_F32: return Util::Float32;
+#endif
+ }
+ return Util::Signed16;
+}
+
+extern "C" void convertAudio(SDL_AudioSpec * wav, SDL_AudioSpec * mixer, Mix_Chunk *chunk){
+ // printf("Convert format %d, channels %d, frequency %d to format %d, channels %d, frequency %d\n", wav->format, wav->channels, wav->freq, mixer->format, mixer->channels, mixer->freq);
+ Util::AudioConverter convert(encoding(wav->format), wav->channels, wav->freq,
+ encoding(mixer->format), mixer->channels, mixer->freq);
+ int size = convert.convertedLength(chunk->alen);
+ unsigned char * data = (unsigned char *) malloc(size > chunk->alen ? size : chunk->alen);
+ memcpy(data, chunk->abuf, chunk->alen);
+ convert.convert(data, chunk->alen);
+ SDL_FreeWAV(chunk->abuf);
+
+ chunk->abuf = data;
+ chunk->alen = size;
+
+ /*
+ SDL_AudioCVT wavecvt;
+ int samplesize;
+ if ( SDL_BuildAudioCVT(&wavecvt,
+ wav->format, wav->channels, wav->freq,
+ mixer->format, mixer->channels, mixer->freq) < 0 ) {
+ SDL_FreeWAV(chunk->abuf);
+ free(chunk);
+ }
+ samplesize = ((wav->format & 0xFF)/8)*wav->channels;
+ wavecvt.len = chunk->alen & ~(samplesize-1);
+ wavecvt.buf = (Uint8 *)malloc(wavecvt.len*wavecvt.len_mult);
+ if ( wavecvt.buf == NULL ) {
+ SDL_SetError("Out of memory");
+ SDL_FreeWAV(chunk->abuf);
+ free(chunk);
+ }
+ memcpy(wavecvt.buf, chunk->abuf, chunk->alen);
+ SDL_FreeWAV(chunk->abuf);
+
+ if ( SDL_ConvertAudio(&wavecvt) < 0 ) {
+ free(wavecvt.buf);
+ free(chunk);
+ }
+ chunk->abuf = wavecvt.buf;
+ chunk->alen = wavecvt.len_cvt;
+ */
+}
diff --git a/util/sdl/mixer/mixer.c b/util/sdl/mixer/mixer.c
index a6e27357..2e0370ce 100644
--- a/util/sdl/mixer/mixer.c
+++ b/util/sdl/mixer/mixer.c
@@ -1,1448 +1,1453 @@
/*
SDL_mixer: An audio mixer library based on the SDL library
Copyright (C) 1997-2009 Sam Lantinga
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
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
slouken@libsdl.org
*/
/* $Id: mixer.c 5243 2009-11-14 19:31:39Z slouken $ */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "SDL_mutex.h"
#include "SDL_endian.h"
#include "SDL_timer.h"
#include "SDL_mixer.h"
#include "load_aiff.h"
#include "load_voc.h"
#include "load_ogg.h"
#include "load_flac.h"
#include "dynamic_flac.h"
#include "dynamic_mod.h"
#include "dynamic_mp3.h"
#include "dynamic_ogg.h"
+#include "convert.h"
+
#define __MIX_INTERNAL_EFFECT__
#include "effects_internal.h"
/* Magic numbers for various audio file formats */
#define RIFF 0x46464952 /* "RIFF" */
#define WAVE 0x45564157 /* "WAVE" */
#define FORM 0x4d524f46 /* "FORM" */
#define OGGS 0x5367674f /* "OggS" */
#define CREA 0x61657243 /* "Crea" */
#define FLAC 0x43614C66 /* "fLaC" */
static int audio_opened = 0;
static SDL_AudioSpec mixer;
typedef struct _Mix_effectinfo
{
Mix_EffectFunc_t callback;
Mix_EffectDone_t done_callback;
void *udata;
struct _Mix_effectinfo *next;
} effect_info;
static struct _Mix_Channel {
Mix_Chunk *chunk;
int playing;
int paused;
Uint8 *samples;
int volume;
int looping;
int tag;
Uint32 expire;
Uint32 start_time;
Mix_Fading fading;
int fade_volume;
int fade_volume_reset;
Uint32 fade_length;
Uint32 ticks_fade;
effect_info *effects;
} *mix_channel = NULL;
static effect_info *posteffects = NULL;
static int num_channels;
static int reserved_channels = 0;
/* Support for hooking into the mixer callback system */
static void (*mix_postmix)(void *udata, Uint8 *stream, int len) = NULL;
static void *mix_postmix_data = NULL;
/* rcg07062001 callback to alert when channels are done playing. */
static void (*channel_done_callback)(int channel) = NULL;
/* Music function declarations */
extern int open_music(SDL_AudioSpec *mixer);
extern void close_music(void);
/* Support for user defined music functions, plus the default one */
extern int volatile music_active;
extern void music_mixer(void *udata, Uint8 *stream, int len);
static void (*mix_music)(void *udata, Uint8 *stream, int len) = music_mixer;
static void *music_data = NULL;
/* rcg06042009 report available decoders at runtime. */
static const char **chunk_decoders = NULL;
static int num_decoders = 0;
int Mix_GetNumChunkDecoders(void)
{
return(num_decoders);
}
const char *Mix_GetChunkDecoder(int index)
{
if ((index < 0) || (index >= num_decoders)) {
return NULL;
}
return(chunk_decoders[index]);
}
static void add_chunk_decoder(const char *decoder)
{
void *ptr = realloc(chunk_decoders, (num_decoders + 1) * sizeof (const char **));
if (ptr == NULL) {
return; /* oh well, go on without it. */
}
chunk_decoders = (const char **) ptr;
chunk_decoders[num_decoders++] = decoder;
}
/* rcg06192001 get linked library's version. */
const SDL_version *Mix_Linked_Version(void)
{
static SDL_version linked_version;
SDL_MIXER_VERSION(&linked_version);
return(&linked_version);
}
static int initialized = 0;
int Mix_Init(int flags)
{
int result = 0;
if (flags & MIX_INIT_FLAC) {
#ifdef FLAC_MUSIC
if ((initialized & MIX_INIT_FLAC) || Mix_InitFLAC() == 0) {
result |= MIX_INIT_FLAC;
}
#else
Mix_SetError("Mixer not built with FLAC support");
#endif
}
if (flags & MIX_INIT_MOD) {
#ifdef MOD_MUSIC
if ((initialized & MIX_INIT_MOD) || Mix_InitMOD() == 0) {
result |= MIX_INIT_MOD;
}
#else
Mix_SetError("Mixer not built with MOD support");
#endif
}
if (flags & MIX_INIT_MP3) {
#ifdef MP3_MUSIC
if ((initialized & MIX_INIT_MP3) || Mix_InitMP3() == 0) {
result |= MIX_INIT_MP3;
}
#else
Mix_SetError("Mixer not built with MP3 support");
#endif
}
if (flags & MIX_INIT_OGG) {
#ifdef OGG_MUSIC
if ((initialized & MIX_INIT_OGG) || Mix_InitOgg() == 0) {
result |= MIX_INIT_OGG;
}
#else
Mix_SetError("Mixer not built with Ogg Vorbis support");
#endif
}
initialized |= result;
return (result);
}
void Mix_Quit()
{
#ifdef FLAC_MUSIC
if (initialized & MIX_INIT_FLAC) {
Mix_QuitFLAC();
}
#endif
#ifdef MOD_MUSIC
if (initialized & MIX_INIT_MOD) {
Mix_QuitMOD();
}
#endif
#ifdef MP3_MUSIC
if (initialized & MIX_INIT_MP3) {
Mix_QuitMP3();
}
#endif
#ifdef OGG_MUSIC
if (initialized & MIX_INIT_OGG) {
Mix_QuitOgg();
}
#endif
initialized = 0;
}
static int _Mix_remove_all_effects(int channel, effect_info **e);
/*
* rcg06122001 Cleanup effect callbacks.
* MAKE SURE SDL_LockAudio() is called before this (or you're in the
* audio callback).
*/
static void _Mix_channel_done_playing(int channel)
{
if (channel_done_callback) {
channel_done_callback(channel);
}
/*
* Call internal function directly, to avoid locking audio from
* inside audio callback.
*/
_Mix_remove_all_effects(channel, &mix_channel[channel].effects);
}
static void *Mix_DoEffects(int chan, void *snd, int len)
{
int posteffect = (chan == MIX_CHANNEL_POST);
effect_info *e = ((posteffect) ? posteffects : mix_channel[chan].effects);
void *buf = snd;
if (e != NULL) { /* are there any registered effects? */
/* if this is the postmix, we can just overwrite the original. */
if (!posteffect) {
buf = malloc(len);
if (buf == NULL) {
return(snd);
}
memcpy(buf, snd, len);
}
for (; e != NULL; e = e->next) {
if (e->callback != NULL) {
e->callback(chan, buf, len, e->udata);
}
}
}
/* be sure to free() the return value if != snd ... */
return(buf);
}
/* Mixing function */
static void mix_channels(void *udata, Uint8 *stream, int len)
{
Uint8 *mix_input;
int i, mixable, volume = SDL_MIX_MAXVOLUME;
Uint32 sdl_ticks;
#if SDL_VERSION_ATLEAST(1, 3, 0)
/* Need to initialize the stream in SDL 1.3+ */
memset(stream, mixer.silence, len);
#endif
/* Mix the music (must be done before the channels are added) */
if ( music_active || (mix_music != music_mixer) ) {
mix_music(music_data, stream, len);
}
/* Mix any playing channels... */
sdl_ticks = SDL_GetTicks();
for ( i=0; i<num_channels; ++i ) {
if( ! mix_channel[i].paused ) {
if ( mix_channel[i].expire > 0 && mix_channel[i].expire < sdl_ticks ) {
/* Expiration delay for that channel is reached */
mix_channel[i].playing = 0;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].expire = 0;
_Mix_channel_done_playing(i);
} else if ( mix_channel[i].fading != MIX_NO_FADING ) {
Uint32 ticks = sdl_ticks - mix_channel[i].ticks_fade;
if( ticks > mix_channel[i].fade_length ) {
Mix_Volume(i, mix_channel[i].fade_volume_reset); /* Restore the volume */
if( mix_channel[i].fading == MIX_FADING_OUT ) {
mix_channel[i].playing = 0;
mix_channel[i].expire = 0;
_Mix_channel_done_playing(i);
}
mix_channel[i].fading = MIX_NO_FADING;
} else {
if( mix_channel[i].fading == MIX_FADING_OUT ) {
Mix_Volume(i, (mix_channel[i].fade_volume * (mix_channel[i].fade_length-ticks))
/ mix_channel[i].fade_length );
} else {
Mix_Volume(i, (mix_channel[i].fade_volume * ticks) / mix_channel[i].fade_length );
}
}
}
if ( mix_channel[i].playing > 0 ) {
int index = 0;
int remaining = len;
while (mix_channel[i].playing > 0 && index < len) {
remaining = len - index;
volume = (mix_channel[i].volume*mix_channel[i].chunk->volume) / MIX_MAX_VOLUME;
mixable = mix_channel[i].playing;
if ( mixable > remaining ) {
mixable = remaining;
}
mix_input = Mix_DoEffects(i, mix_channel[i].samples, mixable);
SDL_MixAudio(stream+index,mix_input,mixable,volume);
if (mix_input != mix_channel[i].samples)
free(mix_input);
mix_channel[i].samples += mixable;
mix_channel[i].playing -= mixable;
index += mixable;
/* rcg06072001 Alert app if channel is done playing. */
if (!mix_channel[i].playing && !mix_channel[i].looping) {
_Mix_channel_done_playing(i);
}
}
/* If looping the sample and we are at its end, make sure
we will still return a full buffer */
while ( mix_channel[i].looping && index < len ) {
int alen = mix_channel[i].chunk->alen;
remaining = len - index;
if (remaining > alen) {
remaining = alen;
}
mix_input = Mix_DoEffects(i, mix_channel[i].chunk->abuf, remaining);
SDL_MixAudio(stream+index, mix_input, remaining, volume);
if (mix_input != mix_channel[i].chunk->abuf)
free(mix_input);
--mix_channel[i].looping;
mix_channel[i].samples = mix_channel[i].chunk->abuf + remaining;
mix_channel[i].playing = mix_channel[i].chunk->alen - remaining;
index += remaining;
}
if ( ! mix_channel[i].playing && mix_channel[i].looping ) {
--mix_channel[i].looping;
mix_channel[i].samples = mix_channel[i].chunk->abuf;
mix_channel[i].playing = mix_channel[i].chunk->alen;
}
}
}
}
/* rcg06122001 run posteffects... */
Mix_DoEffects(MIX_CHANNEL_POST, stream, len);
if ( mix_postmix ) {
mix_postmix(mix_postmix_data, stream, len);
}
}
#if 0
static void PrintFormat(char *title, SDL_AudioSpec *fmt)
{
printf("%s: %d bit %s audio (%s) at %u Hz\n", title, (fmt->format&0xFF),
(fmt->format&0x8000) ? "signed" : "unsigned",
(fmt->channels > 2) ? "surround" :
(fmt->channels > 1) ? "stereo" : "mono", fmt->freq);
}
#endif
/* Open the mixer with a certain desired audio format */
int Mix_OpenAudio(int frequency, Uint16 format, int nchannels, int chunksize)
{
int i;
SDL_AudioSpec desired;
/* If the mixer is already opened, increment open count */
if ( audio_opened ) {
if ( format == mixer.format && nchannels == mixer.channels ) {
++audio_opened;
return(0);
}
while ( audio_opened ) {
Mix_CloseAudio();
}
}
/* Set the desired format and frequency */
desired.freq = frequency;
desired.format = format;
desired.channels = nchannels;
desired.samples = chunksize;
desired.callback = mix_channels;
desired.userdata = NULL;
/* Accept nearly any audio format */
if ( SDL_OpenAudio(&desired, &mixer) < 0 ) {
return(-1);
}
#if 0
PrintFormat("Audio device", &mixer);
#endif
/* Initialize the music players */
if ( open_music(&mixer) < 0 ) {
SDL_CloseAudio();
return(-1);
}
num_channels = MIX_CHANNELS;
mix_channel = (struct _Mix_Channel *) malloc(num_channels * sizeof(struct _Mix_Channel));
/* Clear out the audio channels */
for ( i=0; i<num_channels; ++i ) {
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
mix_channel[i].effects = NULL;
mix_channel[i].paused = 0;
}
Mix_VolumeMusic(SDL_MIX_MAXVOLUME);
_Mix_InitEffects();
/* This list is (currently) decided at build time. */
add_chunk_decoder("WAVE");
add_chunk_decoder("AIFF");
add_chunk_decoder("VOC");
#ifdef OGG_MUSIC
add_chunk_decoder("OGG");
#endif
#ifdef FLAC_MUSIC
add_chunk_decoder("FLAC");
#endif
audio_opened = 1;
SDL_PauseAudio(0);
return(0);
}
/* Dynamically change the number of channels managed by the mixer.
If decreasing the number of channels, the upper channels are
stopped.
*/
int Mix_AllocateChannels(int numchans)
{
if ( numchans<0 || numchans==num_channels )
return(num_channels);
if ( numchans < num_channels ) {
/* Stop the affected channels */
int i;
for(i=numchans; i < num_channels; i++) {
Mix_UnregisterAllEffects(i);
Mix_HaltChannel(i);
}
}
SDL_LockAudio();
mix_channel = (struct _Mix_Channel *) realloc(mix_channel, numchans * sizeof(struct _Mix_Channel));
if ( numchans > num_channels ) {
/* Initialize the new channels */
int i;
for(i=num_channels; i < numchans; i++) {
mix_channel[i].chunk = NULL;
mix_channel[i].playing = 0;
mix_channel[i].looping = 0;
mix_channel[i].volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume = SDL_MIX_MAXVOLUME;
mix_channel[i].fade_volume_reset = SDL_MIX_MAXVOLUME;
mix_channel[i].fading = MIX_NO_FADING;
mix_channel[i].tag = -1;
mix_channel[i].expire = 0;
mix_channel[i].effects = NULL;
mix_channel[i].paused = 0;
}
}
num_channels = numchans;
SDL_UnlockAudio();
return(num_channels);
}
/* Return the actual mixer parameters */
int Mix_QuerySpec(int *frequency, Uint16 *format, int *channels)
{
if ( audio_opened ) {
if ( frequency ) {
*frequency = mixer.freq;
}
if ( format ) {
*format = mixer.format;
}
if ( channels ) {
*channels = mixer.channels;
}
}
return(audio_opened);
}
/*
* !!! FIXME: Ideally, we want a Mix_LoadSample_RW(), which will handle the
* generic setup, then call the correct file format loader.
*/
/* Load a wave file */
Mix_Chunk *Mix_LoadWAV_RW(SDL_RWops *src, int freesrc)
{
Uint32 magic;
Mix_Chunk *chunk;
SDL_AudioSpec wavespec, *loaded;
SDL_AudioCVT wavecvt;
int samplesize;
/* rcg06012001 Make sure src is valid */
if ( ! src ) {
SDL_SetError("Mix_LoadWAV_RW with NULL src");
return(NULL);
}
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
if ( freesrc && src ) {
SDL_RWclose(src);
}
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)malloc(sizeof(Mix_Chunk));
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
if ( freesrc ) {
SDL_RWclose(src);
}
return(NULL);
}
/* Find out what kind of audio file this is */
magic = SDL_ReadLE32(src);
/* Seek backwards for compatibility with older loaders */
SDL_RWseek(src, -(int)sizeof(Uint32), RW_SEEK_CUR);
switch (magic) {
case WAVE:
case RIFF:
loaded = SDL_LoadWAV_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
case FORM:
loaded = Mix_LoadAIFF_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
#ifdef OGG_MUSIC
case OGGS:
loaded = Mix_LoadOGG_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
#endif
#ifdef FLAC_MUSIC
case FLAC:
loaded = Mix_LoadFLAC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
#endif
case CREA:
loaded = Mix_LoadVOC_RW(src, freesrc, &wavespec,
(Uint8 **)&chunk->abuf, &chunk->alen);
break;
default:
SDL_SetError("Unrecognized sound file type");
return(0);
}
if ( !loaded ) {
free(chunk);
return(NULL);
}
#if 0
PrintFormat("Audio device", &mixer);
PrintFormat("-- Wave file", &wavespec);
#endif
/* Build the audio converter and create conversion buffers */
+#if 0
if ( SDL_BuildAudioCVT(&wavecvt,
wavespec.format, wavespec.channels, wavespec.freq,
mixer.format, mixer.channels, mixer.freq) < 0 ) {
SDL_FreeWAV(chunk->abuf);
free(chunk);
return(NULL);
}
samplesize = ((wavespec.format & 0xFF)/8)*wavespec.channels;
wavecvt.len = chunk->alen & ~(samplesize-1);
wavecvt.buf = (Uint8 *)malloc(wavecvt.len*wavecvt.len_mult);
if ( wavecvt.buf == NULL ) {
SDL_SetError("Out of memory");
SDL_FreeWAV(chunk->abuf);
free(chunk);
return(NULL);
}
memcpy(wavecvt.buf, chunk->abuf, chunk->alen);
SDL_FreeWAV(chunk->abuf);
/* Run the audio converter */
if ( SDL_ConvertAudio(&wavecvt) < 0 ) {
free(wavecvt.buf);
free(chunk);
return(NULL);
}
- chunk->allocated = 1;
chunk->abuf = wavecvt.buf;
chunk->alen = wavecvt.len_cvt;
+#endif
+ convertAudio(&wavespec, &mixer, chunk);
+ chunk->allocated = 1;
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
}
/* Load a wave file of the mixer format from a memory buffer */
Mix_Chunk *Mix_QuickLoad_WAV(Uint8 *mem)
{
Mix_Chunk *chunk;
Uint8 magic[4];
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)calloc(1,sizeof(Mix_Chunk));
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
return(NULL);
}
/* Essentially just skip to the audio data (no error checking - fast) */
chunk->allocated = 0;
mem += 12; /* WAV header */
do {
memcpy(magic, mem, 4);
mem += 4;
chunk->alen = ((mem[3]<<24)|(mem[2]<<16)|(mem[1]<<8)|(mem[0]));
mem += 4;
chunk->abuf = mem;
mem += chunk->alen;
} while ( memcmp(magic, "data", 4) != 0 );
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
}
/* Load raw audio data of the mixer format from a memory buffer */
Mix_Chunk *Mix_QuickLoad_RAW(Uint8 *mem, Uint32 len)
{
Mix_Chunk *chunk;
/* Make sure audio has been opened */
if ( ! audio_opened ) {
SDL_SetError("Audio device hasn't been opened");
return(NULL);
}
/* Allocate the chunk memory */
chunk = (Mix_Chunk *)malloc(sizeof(Mix_Chunk));
if ( chunk == NULL ) {
SDL_SetError("Out of memory");
return(NULL);
}
/* Essentially just point at the audio data (no error checking - fast) */
chunk->allocated = 0;
chunk->alen = len;
chunk->abuf = mem;
chunk->volume = MIX_MAX_VOLUME;
return(chunk);
}
/* Free an audio chunk previously loaded */
void Mix_FreeChunk(Mix_Chunk *chunk)
{
int i;
/* Caution -- if the chunk is playing, the mixer will crash */
if ( chunk ) {
/* Guarantee that this chunk isn't playing */
SDL_LockAudio();
if ( mix_channel ) {
for ( i=0; i<num_channels; ++i ) {
if ( chunk == mix_channel[i].chunk ) {
mix_channel[i].playing = 0;
}
}
}
SDL_UnlockAudio();
/* Actually free the chunk */
if ( chunk->allocated ) {
free(chunk->abuf);
}
free(chunk);
}
}
/* Set a function that is called after all mixing is performed.
This can be used to provide real-time visual display of the audio stream
or add a custom mixer filter for the stream data.
*/
void Mix_SetPostMix(void (*mix_func)
(void *udata, Uint8 *stream, int len), void *arg)
{
SDL_LockAudio();
mix_postmix_data = arg;
mix_postmix = mix_func;
SDL_UnlockAudio();
}
/* Add your own music player or mixer function.
If 'mix_func' is NULL, the default music player is re-enabled.
*/
void Mix_HookMusic(void (*mix_func)(void *udata, Uint8 *stream, int len),
void *arg)
{
SDL_LockAudio();
if ( mix_func != NULL ) {
music_data = arg;
mix_music = mix_func;
} else {
music_data = NULL;
mix_music = music_mixer;
}
SDL_UnlockAudio();
}
void *Mix_GetMusicHookData(void)
{
return(music_data);
}
void Mix_ChannelFinished(void (*channel_finished)(int channel))
{
SDL_LockAudio();
channel_done_callback = channel_finished;
SDL_UnlockAudio();
}
/* Reserve the first channels (0 -> n-1) for the application, i.e. don't allocate
them dynamically to the next sample if requested with a -1 value below.
Returns the number of reserved channels.
*/
int Mix_ReserveChannels(int num)
{
if (num > num_channels)
num = num_channels;
reserved_channels = num;
return num;
}
static int checkchunkintegral(Mix_Chunk *chunk)
{
int frame_width = 1;
if ((mixer.format & 0xFF) == 16) frame_width = 2;
frame_width *= mixer.channels;
while (chunk->alen % frame_width) chunk->alen--;
return chunk->alen;
}
/* Play an audio chunk on a specific channel.
If the specified channel is -1, play on the first free channel.
'ticks' is the number of milliseconds at most to play the sample, or -1
if there is no limit.
Returns which channel was used to play the sound.
*/
int Mix_PlayChannelTimed(int which, Mix_Chunk *chunk, int loops, int ticks)
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
Mix_SetError("Tried to play a NULL chunk");
return(-1);
}
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
/* Lock the mixer while modifying the playing channels */
SDL_LockAudio();
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
for ( i=reserved_channels; i<num_channels; ++i ) {
if ( mix_channel[i].playing <= 0 )
break;
}
if ( i == num_channels ) {
Mix_SetError("No free channels available");
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
if ( which >= 0 ) {
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
mix_channel[which].playing = chunk->alen;
mix_channel[which].looping = loops;
mix_channel[which].chunk = chunk;
mix_channel[which].paused = 0;
mix_channel[which].fading = MIX_NO_FADING;
mix_channel[which].start_time = sdl_ticks;
mix_channel[which].expire = (ticks>0) ? (sdl_ticks + ticks) : 0;
}
}
SDL_UnlockAudio();
/* Return the channel on which the sound is being played */
return(which);
}
/* Change the expiration delay for a channel */
int Mix_ExpireChannel(int which, int ticks)
{
int status = 0;
if ( which == -1 ) {
int i;
for ( i=0; i < num_channels; ++ i ) {
status += Mix_ExpireChannel(i, ticks);
}
} else if ( which < num_channels ) {
SDL_LockAudio();
mix_channel[which].expire = (ticks>0) ? (SDL_GetTicks() + ticks) : 0;
SDL_UnlockAudio();
++ status;
}
return(status);
}
/* Fade in a sound on a channel, over ms milliseconds */
int Mix_FadeInChannelTimed(int which, Mix_Chunk *chunk, int loops, int ms, int ticks)
{
int i;
/* Don't play null pointers :-) */
if ( chunk == NULL ) {
return(-1);
}
if ( !checkchunkintegral(chunk)) {
Mix_SetError("Tried to play a chunk with a bad frame");
return(-1);
}
/* Lock the mixer while modifying the playing channels */
SDL_LockAudio();
{
/* If which is -1, play on the first free channel */
if ( which == -1 ) {
for ( i=reserved_channels; i<num_channels; ++i ) {
if ( mix_channel[i].playing <= 0 )
break;
}
if ( i == num_channels ) {
which = -1;
} else {
which = i;
}
}
/* Queue up the audio data for this channel */
if ( which >= 0 ) {
Uint32 sdl_ticks = SDL_GetTicks();
if (Mix_Playing(which))
_Mix_channel_done_playing(which);
mix_channel[which].samples = chunk->abuf;
mix_channel[which].playing = chunk->alen;
mix_channel[which].looping = loops;
mix_channel[which].chunk = chunk;
mix_channel[which].paused = 0;
mix_channel[which].fading = MIX_FADING_IN;
mix_channel[which].fade_volume = mix_channel[which].volume;
mix_channel[which].fade_volume_reset = mix_channel[which].volume;
mix_channel[which].volume = 0;
mix_channel[which].fade_length = (Uint32)ms;
mix_channel[which].start_time = mix_channel[which].ticks_fade = sdl_ticks;
mix_channel[which].expire = (ticks > 0) ? (sdl_ticks+ticks) : 0;
}
}
SDL_UnlockAudio();
/* Return the channel on which the sound is being played */
return(which);
}
/* Set volume of a particular channel */
int Mix_Volume(int which, int volume)
{
int i;
int prev_volume;
if ( which == -1 ) {
prev_volume = 0;
for ( i=0; i<num_channels; ++i ) {
prev_volume += Mix_Volume(i, volume);
}
prev_volume /= num_channels;
} else {
prev_volume = mix_channel[which].volume;
if ( volume >= 0 ) {
if ( volume > SDL_MIX_MAXVOLUME ) {
volume = SDL_MIX_MAXVOLUME;
}
mix_channel[which].volume = volume;
}
}
return(prev_volume);
}
/* Set volume of a particular chunk */
int Mix_VolumeChunk(Mix_Chunk *chunk, int volume)
{
int prev_volume;
prev_volume = chunk->volume;
if ( volume >= 0 ) {
if ( volume > MIX_MAX_VOLUME ) {
volume = MIX_MAX_VOLUME;
}
chunk->volume = volume;
}
return(prev_volume);
}
/* Halt playing of a particular channel */
int Mix_HaltChannel(int which)
{
int i;
if ( which == -1 ) {
for ( i=0; i<num_channels; ++i ) {
Mix_HaltChannel(i);
}
} else {
SDL_LockAudio();
if (mix_channel[which].playing) {
_Mix_channel_done_playing(which);
mix_channel[which].playing = 0;
}
mix_channel[which].expire = 0;
if(mix_channel[which].fading != MIX_NO_FADING) /* Restore volume */
mix_channel[which].volume = mix_channel[which].fade_volume_reset;
mix_channel[which].fading = MIX_NO_FADING;
SDL_UnlockAudio();
}
return(0);
}
/* Halt playing of a particular group of channels */
int Mix_HaltGroup(int tag)
{
int i;
for ( i=0; i<num_channels; ++i ) {
if( mix_channel[i].tag == tag ) {
Mix_HaltChannel(i);
}
}
return(0);
}
/* Fade out a channel and then stop it automatically */
int Mix_FadeOutChannel(int which, int ms)
{
int status;
status = 0;
if ( audio_opened ) {
if ( which == -1 ) {
int i;
for ( i=0; i<num_channels; ++i ) {
status += Mix_FadeOutChannel(i, ms);
}
} else {
SDL_LockAudio();
if ( mix_channel[which].playing &&
(mix_channel[which].volume > 0) &&
(mix_channel[which].fading != MIX_FADING_OUT) ) {
mix_channel[which].fade_volume = mix_channel[which].volume;
mix_channel[which].fading = MIX_FADING_OUT;
mix_channel[which].fade_length = ms;
mix_channel[which].ticks_fade = SDL_GetTicks();
/* only change fade_volume_reset if we're not fading. */
if (mix_channel[which].fading == MIX_NO_FADING) {
mix_channel[which].fade_volume_reset = mix_channel[which].volume;
}
++status;
}
SDL_UnlockAudio();
}
}
return(status);
}
/* Halt playing of a particular group of channels */
int Mix_FadeOutGroup(int tag, int ms)
{
int i;
int status = 0;
for ( i=0; i<num_channels; ++i ) {
if( mix_channel[i].tag == tag ) {
status += Mix_FadeOutChannel(i,ms);
}
}
return(status);
}
Mix_Fading Mix_FadingChannel(int which)
{
return mix_channel[which].fading;
}
/* Check the status of a specific channel.
If the specified mix_channel is -1, check all mix channels.
*/
int Mix_Playing(int which)
{
int status;
status = 0;
if ( which == -1 ) {
int i;
for ( i=0; i<num_channels; ++i ) {
if ((mix_channel[i].playing > 0) ||
(mix_channel[i].looping > 0))
{
++status;
}
}
} else {
if ((mix_channel[which].playing > 0) ||
(mix_channel[which].looping > 0))
{
++status;
}
}
return(status);
}
/* rcg06072001 Get the chunk associated with a channel. */
Mix_Chunk *Mix_GetChunk(int channel)
{
Mix_Chunk *retval = NULL;
if ((channel >= 0) && (channel < num_channels)) {
retval = mix_channel[channel].chunk;
}
return(retval);
}
/* Close the mixer, halting all playing audio */
void Mix_CloseAudio(void)
{
int i;
if ( audio_opened ) {
if ( audio_opened == 1 ) {
for (i = 0; i < num_channels; i++) {
Mix_UnregisterAllEffects(i);
}
Mix_UnregisterAllEffects(MIX_CHANNEL_POST);
close_music();
Mix_HaltChannel(-1);
_Mix_DeinitEffects();
SDL_CloseAudio();
free(mix_channel);
mix_channel = NULL;
/* rcg06042009 report available decoders at runtime. */
free(chunk_decoders);
chunk_decoders = NULL;
num_decoders = 0;
}
--audio_opened;
}
}
/* Pause a particular channel (or all) */
void Mix_Pause(int which)
{
Uint32 sdl_ticks = SDL_GetTicks();
if ( which == -1 ) {
int i;
for ( i=0; i<num_channels; ++i ) {
if ( mix_channel[i].playing > 0 ) {
mix_channel[i].paused = sdl_ticks;
}
}
} else {
if ( mix_channel[which].playing > 0 ) {
mix_channel[which].paused = sdl_ticks;
}
}
}
/* Resume a paused channel */
void Mix_Resume(int which)
{
Uint32 sdl_ticks = SDL_GetTicks();
SDL_LockAudio();
if ( which == -1 ) {
int i;
for ( i=0; i<num_channels; ++i ) {
if ( mix_channel[i].playing > 0 ) {
if(mix_channel[i].expire > 0)
mix_channel[i].expire += sdl_ticks - mix_channel[i].paused;
mix_channel[i].paused = 0;
}
}
} else {
if ( mix_channel[which].playing > 0 ) {
if(mix_channel[which].expire > 0)
mix_channel[which].expire += sdl_ticks - mix_channel[which].paused;
mix_channel[which].paused = 0;
}
}
SDL_UnlockAudio();
}
int Mix_Paused(int which)
{
if ( which > num_channels )
return(0);
if ( which < 0 ) {
int status = 0;
int i;
for( i=0; i < num_channels; ++i ) {
if ( mix_channel[i].paused ) {
++ status;
}
}
return(status);
} else {
return(mix_channel[which].paused != 0);
}
}
/* Change the group of a channel */
int Mix_GroupChannel(int which, int tag)
{
if ( which < 0 || which > num_channels )
return(0);
SDL_LockAudio();
mix_channel[which].tag = tag;
SDL_UnlockAudio();
return(1);
}
/* Assign several consecutive channels to a group */
int Mix_GroupChannels(int from, int to, int tag)
{
int status = 0;
for( ; from <= to; ++ from ) {
status += Mix_GroupChannel(from, tag);
}
return(status);
}
/* Finds the first available channel in a group of channels */
int Mix_GroupAvailable(int tag)
{
int i;
for( i=0; i < num_channels; i ++ ) {
if ( ((tag == -1) || (tag == mix_channel[i].tag)) &&
(mix_channel[i].playing <= 0) )
return i;
}
return(-1);
}
int Mix_GroupCount(int tag)
{
int count = 0;
int i;
for( i=0; i < num_channels; i ++ ) {
if ( mix_channel[i].tag==tag || tag==-1 )
++ count;
}
return(count);
}
/* Finds the "oldest" sample playing in a group of channels */
int Mix_GroupOldest(int tag)
{
int chan = -1;
Uint32 mintime = SDL_GetTicks();
int i;
for( i=0; i < num_channels; i ++ ) {
if ( (mix_channel[i].tag==tag || tag==-1) && mix_channel[i].playing > 0
&& mix_channel[i].start_time <= mintime ) {
mintime = mix_channel[i].start_time;
chan = i;
}
}
return(chan);
}
/* Finds the "most recent" (i.e. last) sample playing in a group of channels */
int Mix_GroupNewer(int tag)
{
int chan = -1;
Uint32 maxtime = 0;
int i;
for( i=0; i < num_channels; i ++ ) {
if ( (mix_channel[i].tag==tag || tag==-1) && mix_channel[i].playing > 0
&& mix_channel[i].start_time >= maxtime ) {
maxtime = mix_channel[i].start_time;
chan = i;
}
}
return(chan);
}
/*
* rcg06122001 The special effects exportable API.
* Please see effect_*.c for internally-implemented effects, such
* as Mix_SetPanning().
*/
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
static int _Mix_register_effect(effect_info **e, Mix_EffectFunc_t f,
Mix_EffectDone_t d, void *arg)
{
effect_info *new_e = malloc(sizeof (effect_info));
if (!e) {
Mix_SetError("Internal error");
return(0);
}
if (f == NULL) {
Mix_SetError("NULL effect callback");
return(0);
}
if (new_e == NULL) {
Mix_SetError("Out of memory");
return(0);
}
new_e->callback = f;
new_e->done_callback = d;
new_e->udata = arg;
new_e->next = NULL;
/* add new effect to end of linked list... */
if (*e == NULL) {
*e = new_e;
} else {
effect_info *cur = *e;
while (1) {
if (cur->next == NULL) {
cur->next = new_e;
break;
}
cur = cur->next;
}
}
return(1);
}
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
static int _Mix_remove_effect(int channel, effect_info **e, Mix_EffectFunc_t f)
{
effect_info *cur;
effect_info *prev = NULL;
effect_info *next = NULL;
if (!e) {
Mix_SetError("Internal error");
return(0);
}
for (cur = *e; cur != NULL; cur = cur->next) {
if (cur->callback == f) {
next = cur->next;
if (cur->done_callback != NULL) {
cur->done_callback(channel, cur->udata);
}
free(cur);
if (prev == NULL) { /* removing first item of list? */
*e = next;
} else {
prev->next = next;
}
return(1);
}
prev = cur;
}
Mix_SetError("No such effect registered");
return(0);
}
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
static int _Mix_remove_all_effects(int channel, effect_info **e)
{
effect_info *cur;
effect_info *next;
if (!e) {
Mix_SetError("Internal error");
return(0);
}
for (cur = *e; cur != NULL; cur = next) {
next = cur->next;
if (cur->done_callback != NULL) {
cur->done_callback(channel, cur->udata);
}
free(cur);
}
*e = NULL;
return(1);
}
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
int _Mix_RegisterEffect_locked(int channel, Mix_EffectFunc_t f,
Mix_EffectDone_t d, void *arg)
{
effect_info **e = NULL;
if (channel == MIX_CHANNEL_POST) {
e = &posteffects;
} else {
if ((channel < 0) || (channel >= num_channels)) {
Mix_SetError("Invalid channel number");
return(0);
}
e = &mix_channel[channel].effects;
}
return _Mix_register_effect(e, f, d, arg);
}
int Mix_RegisterEffect(int channel, Mix_EffectFunc_t f,
Mix_EffectDone_t d, void *arg)
{
int retval;
SDL_LockAudio();
retval = _Mix_RegisterEffect_locked(channel, f, d, arg);
SDL_UnlockAudio();
return retval;
}
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
int _Mix_UnregisterEffect_locked(int channel, Mix_EffectFunc_t f)
{
effect_info **e = NULL;
if (channel == MIX_CHANNEL_POST) {
e = &posteffects;
} else {
if ((channel < 0) || (channel >= num_channels)) {
Mix_SetError("Invalid channel number");
return(0);
}
e = &mix_channel[channel].effects;
}
return _Mix_remove_effect(channel, e, f);
}
int Mix_UnregisterEffect(int channel, Mix_EffectFunc_t f)
{
int retval;
SDL_LockAudio();
retval = _Mix_UnregisterEffect_locked(channel, f);
SDL_UnlockAudio();
return(retval);
}
/* MAKE SURE you hold the audio lock (SDL_LockAudio()) before calling this! */
int _Mix_UnregisterAllEffects_locked(int channel)
{
effect_info **e = NULL;
if (channel == MIX_CHANNEL_POST) {
e = &posteffects;
} else {
if ((channel < 0) || (channel >= num_channels)) {
Mix_SetError("Invalid channel number");
return(0);
}
e = &mix_channel[channel].effects;
}
return _Mix_remove_all_effects(channel, e);
}
int Mix_UnregisterAllEffects(int channel)
{
int retval;
SDL_LockAudio();
retval = _Mix_UnregisterAllEffects_locked(channel);
SDL_UnlockAudio();
return(retval);
}
/* end of mixer.c ... */

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