#include <fcntl.h>
#include <dirent.h>
#include <signal.h>
+#include <math.h>
+
+#include "platform.h"
+
+#if defined(PLATFORM_LINUX)
+#include <sys/ioctl.h>
+#include <linux/soundcard.h>
+#elif defined(PLATFORM_FREEBSD)
+#include <machine/soundcard.h>
+#elif defined(PLATFORM_NETBSD)
+#include <sys/ioctl.h>
+#include <sys/audioio.h>
+#elif defined(PLATFORM_HPUX)
+#include <sys/audio.h>
+#endif
#include "system.h"
#include "sound.h"
#include "misc.h"
#include "setup.h"
+#include "text.h"
+
+
+/* expiration time (in milliseconds) for sound loops */
+#define SOUND_LOOP_EXPIRATION_TIME 200
+
+/* one second fading interval == 1000 ticks (milliseconds) */
+#define SOUND_FADING_INTERVAL 1000
+
+#if defined(AUDIO_STREAMING_DSP)
+#define SOUND_FADING_VOLUME_STEP (SOUND_MAX_VOLUME / 40)
+#define SOUND_FADING_VOLUME_THRESHOLD (SOUND_FADING_VOLUME_STEP * 2)
+#endif
+
+#if !defined(PLATFORM_HPUX)
+#define SND_BLOCKSIZE 4096
+#else
+#define SND_BLOCKSIZE 32768
+#endif
+
+#define SND_TYPE_NONE 0
+#define SND_TYPE_WAV 1
+
+#define MUS_TYPE_NONE 0
+#define MUS_TYPE_WAV 1
+#define MUS_TYPE_MOD 2
+
+#define DEVICENAME_DSP "/dev/dsp"
+#define DEVICENAME_AUDIO "/dev/audio"
+#define DEVICENAME_AUDIOCTL "/dev/audioCtl"
+
+#define SOUND_VOLUME_LEFT(x) (stereo_volume[x])
+#define SOUND_VOLUME_RIGHT(x) (stereo_volume[SOUND_MAX_LEFT2RIGHT-x])
+
+
+#if 0
+struct SoundHeader_SUN
+{
+ unsigned long magic;
+ unsigned long hdr_size;
+ unsigned long data_size;
+ unsigned long encoding;
+ unsigned long sample_rate;
+ unsigned long channels;
+};
+
+struct SoundHeader_8SVX
+{
+ char magic_FORM[4];
+ unsigned long chunk_size;
+ char magic_8SVX[4];
+};
+#endif
+
+#if defined(AUDIO_UNIX_NATIVE)
+struct SoundHeader_WAV
+{
+ unsigned short compression_code;
+ unsigned short num_channels;
+ unsigned long sample_rate;
+ unsigned long bytes_per_second;
+ unsigned short block_align;
+ unsigned short bits_per_sample;
+};
+#endif
+
+struct AudioFormatInfo
+{
+ boolean stereo; /* availability of stereo sound */
+ int format; /* size and endianess of sample data */
+ int sample_rate; /* sample frequency */
+ int fragment_size; /* audio device fragment size in bytes */
+};
+
+struct SampleInfo
+{
+ char *source_filename;
+ int num_references;
+
+ int type;
+ int format;
+ void *data_ptr; /* pointer to first sample (8 or 16 bit) */
+ long data_len; /* number of samples, NOT number of bytes */
+};
+typedef struct SampleInfo SoundInfo;
+typedef struct SampleInfo MusicInfo;
+
+struct SoundControl
+{
+ boolean active;
+
+ int nr;
+ int volume;
+ int stereo_position;
+
+ int state;
+
+ unsigned long playing_starttime;
+ unsigned long playing_pos;
+ int type;
+ int format;
+ void *data_ptr; /* pointer to first sample (8 or 16 bit) */
+ long data_len; /* number of samples, NOT number of bytes */
+
+#if defined(TARGET_ALLEGRO)
+ int voice;
+#endif
+};
+typedef struct SoundControl SoundControl;
struct ListNode
{
static SoundInfo **Sound = NULL;
static MusicInfo **Music = NULL;
static int num_sounds = 0, num_music = 0;
+static int stereo_volume[SOUND_MAX_LEFT2RIGHT + 1];
/* ========================================================================= */
/* THE STUFF BELOW IS ONLY USED BY THE SOUND SERVER CHILD PROCESS */
-static struct AudioFormatInfo afmt;
static struct SoundControl mixer[NUM_MIXER_CHANNELS];
static int mixer_active_channels = 0;
-/* forward declaration of internal functions */
-static void InitAudioDevice(struct AudioFormatInfo *);
-static void Mixer_Main(void);
+#if defined(AUDIO_UNIX_NATIVE)
+static struct AudioFormatInfo afmt;
-#if defined(PLATFORM_UNIX) && !defined(AUDIO_STREAMING_DSP)
+static void Mixer_Main(void);
+#if !defined(AUDIO_STREAMING_DSP)
static unsigned char linear_to_ulaw(int);
static int ulaw_to_linear(unsigned char);
#endif
+#endif
static void ReloadCustomSounds();
static void ReloadCustomMusic();
static void FreeSound(void *);
-#if defined(PLATFORM_UNIX)
+
+/* ------------------------------------------------------------------------- */
+/* functions for native (non-SDL) Unix audio/mixer support */
+/* ------------------------------------------------------------------------- */
+
+#if defined(AUDIO_UNIX_NATIVE)
+
static int OpenAudioDevice(char *audio_device_name)
{
int audio_device_fd;
return TRUE;
}
-#if !defined(TARGET_SDL)
static boolean ForkAudioProcess(void)
{
if (pipe(audio.mixer_pipe) < 0)
return FALSE;
}
- if (audio.mixer_pid == 0) /* we are child process */
- {
- Mixer_Main();
-
- /* never reached */
- exit(0);
- }
- else /* we are parent */
- close(audio.mixer_pipe[0]); /* no reading from pipe needed */
+ if (IS_CHILD_PROCESS(audio.mixer_pid))
+ Mixer_Main(); /* this function never returns */
+ else
+ close(audio.mixer_pipe[0]); /* no reading from pipe needed */
return TRUE;
}
-#endif
void UnixOpenAudio(void)
{
if (audio.device_fd)
close(audio.device_fd);
- if (audio.mixer_pid > 0) /* we are parent process */
+ if (IS_PARENT_PROCESS(audio.mixer_pid))
kill(audio.mixer_pid, SIGTERM);
}
+
+/* ------------------------------------------------------------------------- */
+/* functions for platform specific audio device initialization */
+/* ------------------------------------------------------------------------- */
+
+#if defined(AUDIO_LINUX_IOCTL)
+static void InitAudioDevice_Linux(struct AudioFormatInfo *afmt)
+{
+ /* "ioctl()" expects pointer to 'int' value for stereo flag
+ (boolean is defined as 'char', which will not work here) */
+ unsigned int fragment_spec = 0;
+ int fragment_size_query;
+ int stereo = TRUE;
+ struct
+ {
+ int format_ioctl;
+ int format_result;
+ }
+ formats[] =
+ {
+ /* supported audio format in preferred order */
+ { AFMT_S16_LE, AUDIO_FORMAT_S16 | AUDIO_FORMAT_LE },
+ { AFMT_S16_BE, AUDIO_FORMAT_S16 | AUDIO_FORMAT_BE },
+ { AFMT_U8, AUDIO_FORMAT_U8 },
+ { -1, -1 }
+ };
+ int i;
+
+ /* determine logarithm (log2) of the fragment size */
+ while ((1 << fragment_spec) < afmt->fragment_size)
+ fragment_spec++;
+
+ /* use two fragments (play one fragment, prepare the other);
+ one fragment would result in interrupted audio output, more
+ than two fragments would raise audio output latency to much */
+ fragment_spec |= 0x00020000;
+
+ /* Example for fragment specification:
+ - 2 buffers / 512 bytes (giving 1/16 second resolution for 8 kHz)
+ - (with stereo the effective buffer size will shrink to 256)
+ => fragment_size = 0x00020009 */
+
+ if (ioctl(audio.device_fd, SNDCTL_DSP_SETFRAGMENT, &fragment_spec) < 0)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set fragment size of /dev/dsp -- no sounds");
+
+ i = 0;
+ afmt->format = 0;
+ while (formats[i].format_result != -1)
+ {
+ unsigned int audio_format = formats[i].format_ioctl;
+ if (ioctl(audio.device_fd, SNDCTL_DSP_SETFMT, &audio_format) == 0)
+ {
+ afmt->format = formats[i].format_result;
+ break;
+ }
+ }
+
+ if (afmt->format == 0) /* no supported audio format found */
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set audio format of /dev/dsp -- no sounds");
+
+ /* try if we can use stereo sound */
+ afmt->stereo = TRUE;
+ if (ioctl(audio.device_fd, SNDCTL_DSP_STEREO, &stereo) < 0)
+ afmt->stereo = FALSE;
+
+ if (ioctl(audio.device_fd, SNDCTL_DSP_SPEED, &afmt->sample_rate) < 0)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set sample rate of /dev/dsp -- no sounds");
+
+ /* get the real fragmentation size; this should return 512 */
+ if (ioctl(audio.device_fd, SNDCTL_DSP_GETBLKSIZE, &fragment_size_query) < 0)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot get fragment size of /dev/dsp -- no sounds");
+ if (fragment_size_query != afmt->fragment_size)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set fragment size of /dev/dsp -- no sounds");
+}
+#endif /* AUDIO_LINUX_IOCTL */
+
+#if defined(PLATFORM_NETBSD)
+static void InitAudioDevice_NetBSD(struct AudioFormatInfo *afmt)
+{
+ audio_info_t a_info;
+ boolean stereo = TRUE;
+
+ AUDIO_INITINFO(&a_info);
+ a_info.play.encoding = AUDIO_ENCODING_LINEAR8;
+ a_info.play.precision = 8;
+ a_info.play.channels = 2;
+ a_info.play.sample_rate = sample_rate;
+ a_info.blocksize = fragment_size;
+
+ afmt->format = AUDIO_FORMAT_U8;
+ afmt->stereo = TRUE;
+
+ if (ioctl(audio.device_fd, AUDIO_SETINFO, &a_info) < 0)
+ {
+ /* try to disable stereo */
+ a_info.play.channels = 1;
+
+ afmt->stereo = FALSE;
+
+ if (ioctl(audio.device_fd, AUDIO_SETINFO, &a_info) < 0)
+ Error(ERR_EXIT_SOUND_SERVER,
+ "cannot set sample rate of /dev/audio -- no sounds");
+ }
+}
+#endif /* PLATFORM_NETBSD */
+
+#if defined(PLATFORM_HPUX)
+static void InitAudioDevice_HPUX(struct AudioFormatInfo *afmt)
+{
+ struct audio_describe ainfo;
+ int audio_ctl;
+
+ audio_ctl = open("/dev/audioCtl", O_WRONLY | O_NDELAY);
+ if (audio_ctl == -1)
+ Error(ERR_EXIT_SOUND_SERVER, "cannot open /dev/audioCtl -- no sounds");
+
+ if (ioctl(audio_ctl, AUDIO_DESCRIBE, &ainfo) == -1)
+ Error(ERR_EXIT_SOUND_SERVER, "no audio info -- no sounds");
+
+ if (ioctl(audio_ctl, AUDIO_SET_DATA_FORMAT, AUDIO_FORMAT_ULAW) == -1)
+ Error(ERR_EXIT_SOUND_SERVER, "ulaw audio not available -- no sounds");
+
+ ioctl(audio_ctl, AUDIO_SET_CHANNELS, 1);
+ ioctl(audio_ctl, AUDIO_SET_SAMPLE_RATE, 8000);
+
+ afmt->format = AUDIO_FORMAT_U8;
+ afmt->stereo = FALSE;
+ afmt->sample_rate = 8000;
+
+ close(audio_ctl);
+}
+#endif /* PLATFORM_HPUX */
+
+static void InitAudioDevice(struct AudioFormatInfo *afmt)
+{
+ afmt->stereo = TRUE;
+ afmt->format = AUDIO_FORMAT_UNKNOWN;
+ afmt->sample_rate = DEFAULT_AUDIO_SAMPLE_RATE;
+ afmt->fragment_size = DEFAULT_AUDIO_FRAGMENT_SIZE;
+
+#if defined(AUDIO_LINUX_IOCTL)
+ InitAudioDevice_Linux(afmt);
+#elif defined(PLATFORM_NETBSD)
+ InitAudioDevice_NetBSD(afmt);
+#elif defined(PLATFORM_HPUX)
+ InitAudioDevice_HPUX(afmt);
+#else
+ /* generic /dev/audio stuff might be placed here */
+#endif
+}
+
+
+/* ------------------------------------------------------------------------- */
+/* functions for communication between main process and sound mixer process */
+/* ------------------------------------------------------------------------- */
+
+static void SendSoundControlToMixerProcess(SoundControl *snd_ctrl)
+{
+ if (IS_CHILD_PROCESS(audio.mixer_pid))
+ return;
+
+ if (write(audio.mixer_pipe[1], snd_ctrl, sizeof(SoundControl)) < 0)
+ {
+ Error(ERR_WARN, "cannot pipe to child process -- no sounds");
+ audio.sound_available = audio.sound_enabled = FALSE;
+ return;
+ }
+}
+
+static void ReadSoundControlFromMainProcess(SoundControl *snd_ctrl)
+{
+ if (IS_PARENT_PROCESS(audio.mixer_pid))
+ return;
+
+ if (read(audio.mixer_pipe[0], snd_ctrl, sizeof(SoundControl))
+ != sizeof(SoundControl))
+ Error(ERR_EXIT_SOUND_SERVER, "broken pipe -- no sounds");
+}
+
static void WriteReloadInfoToPipe(char *set_name, int type)
{
- struct SoundControl snd_ctrl;
+ SoundControl snd_ctrl;
TreeInfo *ti = (type == SND_CTRL_RELOAD_SOUNDS ? artwork.snd_current :
artwork.mus_current);
unsigned long str_size1 = strlen(leveldir_current->fullpath) + 1;
unsigned long str_size2 = strlen(ti->basepath) + 1;
unsigned long str_size3 = strlen(ti->fullpath) + 1;
- if (audio.mixer_pid == 0) /* we are child process */
+ if (IS_CHILD_PROCESS(audio.mixer_pid))
return;
if (leveldir_current == NULL) /* should never happen */
}
}
-static void ReadReloadInfoFromPipe(struct SoundControl snd_ctrl)
+static void ReadReloadInfoFromPipe(SoundControl *snd_ctrl)
{
- TreeInfo **ti_ptr = ((snd_ctrl.state & SND_CTRL_RELOAD_SOUNDS) ?
+ TreeInfo **ti_ptr = ((snd_ctrl->state & SND_CTRL_RELOAD_SOUNDS) ?
&artwork.snd_current : &artwork.mus_current);
TreeInfo *ti = *ti_ptr;
unsigned long str_size1, str_size2, str_size3;
if (set_name)
free(set_name);
- set_name = checked_malloc(snd_ctrl.data_len);
+ set_name = checked_malloc(snd_ctrl->data_len);
if (leveldir_current == NULL)
leveldir_current = checked_calloc(sizeof(TreeInfo));
free(ti->fullpath);
if (read(audio.mixer_pipe[0], set_name,
- snd_ctrl.data_len) != snd_ctrl.data_len ||
+ snd_ctrl->data_len) != snd_ctrl->data_len ||
read(audio.mixer_pipe[0], leveldir_current,
sizeof(TreeInfo)) != sizeof(TreeInfo) ||
read(audio.mixer_pipe[0], ti,
str_size3) != str_size3)
Error(ERR_EXIT_SOUND_SERVER, "broken pipe -- no sounds");
- if (snd_ctrl.state & SND_CTRL_RELOAD_SOUNDS)
+ if (snd_ctrl->state & SND_CTRL_RELOAD_SOUNDS)
artwork.sounds_set_current = set_name;
else
artwork.music_set_current = set_name;
}
-#endif /* PLATFORM_UNIX */
+
+#endif /* AUDIO_UNIX_NATIVE */
+
+
+/* ------------------------------------------------------------------------- */
+/* mixer functions */
+/* ------------------------------------------------------------------------- */
void Mixer_InitChannels()
{
mixer_active_channels = 0;
}
-static void Mixer_PlayChannel(int channel)
+static void Mixer_ResetChannelExpiration(int channel)
{
-#if defined(PLATFORM_MSDOS)
- mixer[channel].voice = allocate_voice((SAMPLE *)mixer[channel].data_ptr);
+ mixer[channel].playing_starttime = Counter();
+
+#if defined(TARGET_SDL)
+ if (IS_LOOP(mixer[channel]) && !IS_MUSIC(mixer[channel]))
+ Mix_ExpireChannel(channel, SOUND_LOOP_EXPIRATION_TIME);
+#endif
+}
+
+static boolean Mixer_ChannelExpired(int channel)
+{
+ if (!mixer[channel].active)
+ return TRUE;
+
+ if (IS_LOOP(mixer[channel]) && !IS_MUSIC(mixer[channel]) &&
+ DelayReached(&mixer[channel].playing_starttime,
+ SOUND_LOOP_EXPIRATION_TIME))
+ return TRUE;
+#if defined(TARGET_SDL)
+
+ if (!Mix_Playing(channel))
+ return TRUE;
+
+#elif defined(TARGET_ALLEGRO)
+
+ mixer[channel].playing_pos = voice_get_position(mixer[channel].voice);
+ mixer[channel].volume = voice_get_volume(mixer[channel].voice);
+
+ /* sound sample has completed playing or was completely faded out */
+ if (mixer[channel].playing_pos == -1 || mixer[channel].volume == 0)
+ return TRUE;
+
+#endif /* TARGET_ALLEGRO */
+
+ return FALSE;
+}
+
+static boolean Mixer_AllocateChannel(int channel)
+{
+#if defined(TARGET_ALLEGRO)
+ mixer[channel].voice = allocate_voice((SAMPLE *)mixer[channel].data_ptr);
if (mixer[channel].voice < 0)
- return;
+ return FALSE;
+#endif
+
+ return TRUE;
+}
+static void Mixer_SetChannelProperties(int channel)
+{
+#if defined(TARGET_SDL)
+ Mix_Volume(channel, mixer[channel].volume);
+ Mix_SetPanning(channel,
+ SOUND_VOLUME_LEFT(mixer[channel].stereo_position),
+ SOUND_VOLUME_RIGHT(mixer[channel].stereo_position));
+#elif defined(TARGET_ALLEGRO)
+ voice_set_volume(mixer[channel].voice, mixer[channel].volume);
+ voice_set_pan(mixer[channel].voice, mixer[channel].stereo_position);
+#endif
+}
+
+static void Mixer_StartChannel(int channel)
+{
+#if defined(TARGET_SDL)
+ Mix_PlayChannel(channel, mixer[channel].data_ptr,
+ IS_LOOP(mixer[channel]) ? -1 : 0);
+#elif defined(TARGET_ALLEGRO)
if (IS_LOOP(mixer[channel]))
voice_set_playmode(mixer[channel].voice, PLAYMODE_LOOP);
- voice_set_volume(mixer[channel].voice, snd_ctrl.volume);
- voice_set_pan(mixer[channel].voice, snd_ctrl.stereo);
voice_start(mixer[channel].voice);
#endif
}
+static void Mixer_PlayChannel(int channel)
+{
+ /* start with inactive channel in case something goes wrong */
+ mixer[channel].active = FALSE;
+
+ if (mixer[channel].type != MUS_TYPE_WAV)
+ return;
+
+ if (!Mixer_AllocateChannel(channel))
+ return;
+
+ Mixer_SetChannelProperties(channel);
+ Mixer_StartChannel(channel);
+
+ Mixer_ResetChannelExpiration(channel);
+
+ mixer[channel].playing_pos = 0;
+ mixer[channel].active = TRUE;
+ mixer_active_channels++;
+}
+
+static void Mixer_PlayMusicChannel()
+{
+ Mixer_PlayChannel(audio.music_channel);
+
+#if defined(TARGET_SDL)
+ if (mixer[audio.music_channel].type != MUS_TYPE_WAV)
+ {
+ /* Mix_VolumeMusic() must be called _after_ Mix_PlayMusic() --
+ this looks like a bug in the SDL_mixer library */
+ Mix_PlayMusic(mixer[audio.music_channel].data_ptr, -1);
+ Mix_VolumeMusic(SOUND_MAX_VOLUME);
+ }
+#endif
+}
+
static void Mixer_StopChannel(int channel)
{
-#if defined(PLATFORM_MSDOS)
+ if (!mixer[channel].active)
+ return;
+
+#if defined(TARGET_SDL)
+ Mix_HaltChannel(channel);
+#elif defined(TARGET_ALLEGRO)
voice_set_volume(mixer[channel].voice, 0);
deallocate_voice(mixer[channel].voice);
#endif
+
+ mixer[channel].active = FALSE;
+ mixer_active_channels--;
+}
+
+static void Mixer_StopMusicChannel()
+{
+ Mixer_StopChannel(audio.music_channel);
+
+#if defined(TARGET_SDL)
+ Mix_HaltMusic();
+#endif
}
static void Mixer_FadeChannel(int channel)
{
+ if (!mixer[channel].active)
+ return;
+
mixer[channel].state |= SND_CTRL_FADE;
-#if defined(PLATFORM_MSDOS)
+#if defined(TARGET_SDL)
+ Mix_FadeOutChannel(channel, SOUND_FADING_INTERVAL);
+#elif defined(TARGET_ALLEGRO)
if (voice_check(mixer[channel].voice))
- voice_ramp_volume(mixer[channel].voice, 1000, 0);
- mixer[channel].state &= ~SND_CTRL_IS_LOOP;
+ voice_ramp_volume(mixer[channel].voice, SOUND_FADING_INTERVAL, 0);
#endif
}
-static void Mixer_RemoveSound(int channel)
+static void Mixer_FadeMusicChannel()
{
- if (!mixer_active_channels || !mixer[channel].active)
- return;
+ Mixer_FadeChannel(audio.music_channel);
-#if 0
- printf("REMOVING MIXER SOUND %d\n", channel);
+#if defined(TARGET_SDL)
+ Mix_FadeOutMusic(SOUND_FADING_INTERVAL);
#endif
+}
+
+static void Mixer_UnFadeChannel(int channel)
+{
+ if (!mixer[channel].active || !IS_FADING(mixer[channel]))
+ return;
- Mixer_StopChannel(channel);
+ mixer[channel].state &= ~SND_CTRL_FADE;
+ mixer[channel].volume = SOUND_MAX_VOLUME;
- mixer[channel].active = FALSE;
- mixer_active_channels--;
+#if defined(TARGET_SDL)
+ Mix_ExpireChannel(channel, -1);
+ Mix_Volume(channel, mixer[channel].volume);
+#elif defined(TARGET_ALLEGRO)
+ voice_stop_volumeramp(mixer[channel].voice);
+ voice_ramp_volume(mixer[channel].voice, SOUND_FADING_INTERVAL,
+ mixer[channel].volume);
+#endif
}
-static void Mixer_InsertSound(struct SoundControl snd_ctrl)
+static void Mixer_InsertSound(SoundControl snd_ctrl)
{
SoundInfo *snd_info;
int i, k;
#if 0
- printf("NEW SOUND %d HAS ARRIVED [%d]\n", snd_ctrl.nr, num_sounds);
+ printf("NEW SOUND %d HAS ARRIVED [%d]\n", snd_ctrl.nr, num_sounds);
+#endif
+
+#if 0
+ printf("%d ACTIVE CHANNELS\n", mixer_active_channels);
#endif
if (IS_MUSIC(snd_ctrl))
snd_info = (IS_MUSIC(snd_ctrl) ? Music[snd_ctrl.nr] : Sound[snd_ctrl.nr]);
if (snd_info == NULL)
- {
-#if 0
- printf("sound/music %d undefined\n", snd_ctrl.nr);
-#endif
return;
- }
-
-#if 0
- printf("-> %d\n", mixer_active_channels);
-#endif
- if (mixer_active_channels == audio.num_channels)
- {
- for (i=0; i<audio.num_channels; i++)
- {
- if (mixer[i].data_ptr == NULL)
- {
-#if 1
- printf("THIS SHOULD NEVER HAPPEN! [%d]\n", i);
-#endif
-
- Mixer_RemoveSound(i);
- }
- }
- }
+ /* copy sound sample and format information */
+ snd_ctrl.type = snd_info->type;
+ snd_ctrl.format = snd_info->format;
+ snd_ctrl.data_ptr = snd_info->data_ptr;
+ snd_ctrl.data_len = snd_info->data_len;
- /* if mixer is full, remove oldest sound */
- if (mixer_active_channels == audio.num_channels)
+ /* play music samples on a dedicated music channel */
+ if (IS_MUSIC(snd_ctrl))
{
- int longest = 0, longest_nr = audio.first_sound_channel;
-
- for (i=audio.first_sound_channel; i<audio.num_channels; i++)
- {
-#if !defined(PLATFORM_MSDOS)
- int actual = 100 * mixer[i].playingpos / mixer[i].data_len;
-#else
- int actual = mixer[i].playingpos;
+#if 0
+ printf("PLAY MUSIC WITH VOLUME/STEREO %d/%d\n",
+ snd_ctrl.volume, snd_ctrl.stereo_position);
#endif
- if (!IS_LOOP(mixer[i]) && actual > longest)
- {
- longest = actual;
- longest_nr = i;
- }
- }
+ mixer[audio.music_channel] = snd_ctrl;
+ Mixer_PlayMusicChannel();
- Mixer_RemoveSound(longest_nr);
+ return;
}
/* check if sound is already being played (and how often) */
for (k=0, i=audio.first_sound_channel; i<audio.num_channels; i++)
- if (mixer[i].nr == snd_ctrl.nr)
+ if (mixer[i].active && mixer[i].nr == snd_ctrl.nr)
k++;
- /* restart loop sounds only if they are just fading out */
- if (k >= 1 && IS_LOOP(snd_ctrl))
+#if 0
+ printf("SOUND %d [CURRENTLY PLAYING %d TIMES]\n", snd_ctrl.nr, k);
+#endif
+
+ /* reset expiration delay for already playing loop sounds */
+ if (k > 0 && IS_LOOP(snd_ctrl))
{
for(i=audio.first_sound_channel; i<audio.num_channels; i++)
{
- if (mixer[i].nr == snd_ctrl.nr && IS_FADING(mixer[i]))
+ if (mixer[i].active && mixer[i].nr == snd_ctrl.nr)
{
- mixer[i].state &= ~SND_CTRL_FADE;
- mixer[i].volume = PSND_MAX_VOLUME;
-#if defined(PLATFORM_MSDOS)
- mixer[i].state |= SND_CTRL_LOOP;
- voice_stop_volumeramp(mixer[i].voice);
- voice_ramp_volume(mixer[i].voice, mixer[i].volume, 1000);
+#if 0
+ printf("RESETTING EXPIRATION FOR SOUND %d\n", snd_ctrl.nr);
+#endif
+
+ if (IS_FADING(mixer[i]))
+ Mixer_UnFadeChannel(i);
+
+ /* restore settings like volume and stereo position */
+ mixer[i].volume = snd_ctrl.volume;
+ mixer[i].stereo_position = snd_ctrl.stereo_position;
+
+ Mixer_SetChannelProperties(i);
+ Mixer_ResetChannelExpiration(i);
+
+#if 0
+ printf("RESETTING VOLUME/STEREO FOR SOUND %d TO %d/%d\n",
+ snd_ctrl.nr, snd_ctrl.volume, snd_ctrl.stereo_position);
#endif
}
}
/* don't play sound more than n times simultaneously (with n == 2 for now) */
if (k >= 2)
{
+ unsigned long playing_current = Counter();
int longest = 0, longest_nr = audio.first_sound_channel;
/* look for oldest equal sound */
for(i=audio.first_sound_channel; i<audio.num_channels; i++)
{
+ int playing_time = playing_current - mixer[i].playing_starttime;
int actual;
if (!mixer[i].active || mixer[i].nr != snd_ctrl.nr)
continue;
-#if !defined(PLATFORM_MSDOS)
- actual = 100 * mixer[i].playingpos / mixer[i].data_len;
-#else
- actual = mixer[i].playingpos;
-#endif
+ actual = 1000 * playing_time / mixer[i].data_len;
+
if (actual >= longest)
{
longest = actual;
}
}
- Mixer_RemoveSound(longest_nr);
+ Mixer_StopChannel(longest_nr);
+ }
+
+ /* If all (non-music) channels are active, stop the channel that has
+ played its sound sample most completely (in percent of the sample
+ length). As we cannot currently get the actual playing position
+ of the channel's sound sample when compiling with the SDL mixer
+ library, we use the current playing time (in milliseconds) instead. */
+
+ if (mixer_active_channels ==
+ audio.num_channels - (mixer[audio.music_channel].active ? 0 : 1))
+ {
+ unsigned long playing_current = Counter();
+ int longest = 0, longest_nr = audio.first_sound_channel;
+
+ for (i=audio.first_sound_channel; i<audio.num_channels; i++)
+ {
+ int playing_time = playing_current - mixer[i].playing_starttime;
+ int actual = 1000 * playing_time / mixer[i].data_len;
+
+ if (!IS_LOOP(mixer[i]) && actual > longest)
+ {
+ longest = actual;
+ longest_nr = i;
+ }
+ }
+
+ Mixer_StopChannel(longest_nr);
}
- /* add new sound to mixer */
+ /* add the new sound to the mixer */
for(i=0; i<audio.num_channels; i++)
{
#if 0
if ((i == audio.music_channel && IS_MUSIC(snd_ctrl)) ||
(i != audio.music_channel && !mixer[i].active))
{
- snd_ctrl.data_ptr = snd_info->data_ptr;
- snd_ctrl.data_len = snd_info->data_len;
- snd_ctrl.format = snd_info->format;
-
- snd_ctrl.playingpos = 0;
- snd_ctrl.playingtime = 0;
+#if 0
+ printf("ADDING NEW SOUND %d TO MIXER\n", snd_ctrl.nr);
+#endif
#if 1
+#if defined(AUDIO_UNIX_NATIVE)
if (snd_info->data_len == 0)
{
printf("THIS SHOULD NEVER HAPPEN! [snd_info->data_len == 0]\n");
}
#endif
+#endif
#if 1
if (IS_MUSIC(snd_ctrl) && i == audio.music_channel && mixer[i].active)
printf("THIS SHOULD NEVER HAPPEN! [adding music twice]\n");
#if 1
- Mixer_RemoveSound(i);
+ Mixer_StopChannel(i);
#endif
}
#endif
mixer[i] = snd_ctrl;
- mixer_active_channels++;
-
-#if 0
- printf("NEW SOUND %d ADDED TO MIXER\n", snd_ctrl.nr);
-#endif
-
Mixer_PlayChannel(i);
break;
}
}
-static void HandleSoundRequest(struct SoundControl snd_ctrl)
+static void HandleSoundRequest(SoundControl snd_ctrl)
{
int i;
+#if defined(AUDIO_UNIX_NATIVE)
+ if (IS_PARENT_PROCESS(audio.mixer_pid))
+ {
+ SendSoundControlToMixerProcess(&snd_ctrl);
+ return;
+ }
+#endif
+
+ /* deactivate channels that have expired since the last request */
+ for (i=0; i<audio.num_channels; i++)
+ if (mixer[i].active && Mixer_ChannelExpired(i))
+ Mixer_StopChannel(i);
+
if (IS_RELOADING(snd_ctrl)) /* load new sound or music files */
{
- ReadReloadInfoFromPipe(snd_ctrl);
- Mixer_InitChannels();
+ Mixer_StopMusicChannel();
+ for(i=audio.first_sound_channel; i<audio.num_channels; i++)
+ Mixer_StopChannel(i);
+
+#if defined(AUDIO_UNIX_NATIVE)
CloseAudioDevice(&audio.device_fd);
+ ReadReloadInfoFromPipe(&snd_ctrl);
+#endif
if (snd_ctrl.state & SND_CTRL_RELOAD_SOUNDS)
ReloadCustomSounds();
}
else if (IS_FADING(snd_ctrl)) /* fade out existing sound or music */
{
- if (!mixer_active_channels)
- return;
-
if (IS_MUSIC(snd_ctrl))
{
- Mixer_FadeChannel(audio.music_channel);
+ Mixer_FadeMusicChannel();
return;
}
}
else if (IS_STOPPING(snd_ctrl)) /* stop existing sound or music */
{
- if (!mixer_active_channels)
- return;
-
if (IS_MUSIC(snd_ctrl))
{
- Mixer_RemoveSound(audio.music_channel);
+ Mixer_StopMusicChannel();
return;
}
for(i=audio.first_sound_channel; i<audio.num_channels; i++)
if (mixer[i].nr == snd_ctrl.nr || ALL_SOUNDS(snd_ctrl))
- Mixer_RemoveSound(i);
+ Mixer_StopChannel(i);
+#if defined(AUDIO_UNIX_NATIVE)
if (!mixer_active_channels)
CloseAudioDevice(&audio.device_fd);
+#endif
}
else if (snd_ctrl.active) /* add new sound to mixer */
{
void StartMixer(void)
{
+ int i;
+
+#if 0
+ SDL_version compile_version;
+ const SDL_version *link_version;
+ MIX_VERSION(&compile_version);
+ printf("compiled with SDL_mixer version: %d.%d.%d\n",
+ compile_version.major,
+ compile_version.minor,
+ compile_version.patch);
+ link_version = Mix_Linked_Version();
+ printf("running with SDL_mixer version: %d.%d.%d\n",
+ link_version->major,
+ link_version->minor,
+ link_version->patch);
+#endif
+
if (!audio.sound_available)
return;
-#if defined(PLATFORM_UNIX) && !defined(TARGET_SDL)
+ /* initialize stereo position conversion information */
+ for(i=0; i<=SOUND_MAX_LEFT2RIGHT; i++)
+ stereo_volume[i] =
+ (int)sqrt((float)(SOUND_MAX_LEFT2RIGHT * SOUND_MAX_LEFT2RIGHT - i * i));
+
+#if defined(AUDIO_UNIX_NATIVE)
if (!ForkAudioProcess())
audio.sound_available = FALSE;
#endif
}
-#if defined(PLATFORM_UNIX) && !defined(TARGET_SDL)
+#if defined(AUDIO_UNIX_NATIVE)
-static void CopySampleToMixingBuffer(struct SoundControl *snd_ctrl,
+static void CopySampleToMixingBuffer(SoundControl *snd_ctrl,
int sample_pos, int sample_size,
short *buffer_ptr)
{
#if defined(AUDIO_STREAMING_DSP)
static void Mixer_Main_DSP()
{
- static int stereo_volume[PSND_MAX_LEFT2RIGHT + 1];
- static boolean stereo_volume_calculated = FALSE;
static short premix_first_buffer[SND_BLOCKSIZE];
static short premix_left_buffer[SND_BLOCKSIZE];
static short premix_right_buffer[SND_BLOCKSIZE];
int max_sample_size;
int i, j;
- if (!stereo_volume_calculated)
- {
- for(i=0; i<=PSND_MAX_LEFT2RIGHT; i++)
- stereo_volume[i] =
- (int)sqrt((float)(PSND_MAX_LEFT2RIGHT * PSND_MAX_LEFT2RIGHT - i * i));
-
- stereo_volume_calculated = TRUE;
- }
-
if (!mixer_active_channels)
return;
if (!mixer[i].active)
continue;
+ if (Mixer_ChannelExpired(i))
+ {
+ Mixer_StopChannel(i);
+ continue;
+ }
+
/* pointer, lenght and actual playing position of sound sample */
sample_ptr = mixer[i].data_ptr;
sample_len = mixer[i].data_len;
- sample_pos = mixer[i].playingpos;
+ sample_pos = mixer[i].playing_pos;
sample_size = MIN(max_sample_size, sample_len - sample_pos);
- mixer[i].playingpos += sample_size;
+ mixer[i].playing_pos += sample_size;
/* copy original sample to first mixing buffer */
CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
premix_first_buffer[sample_size + j] =
((short *)sample_ptr)[j];
- mixer[i].playingpos = restarted_sample_size;
+ mixer[i].playing_pos = restarted_sample_size;
sample_size += restarted_sample_size;
}
}
mixer[i].volume -= SOUND_FADING_VOLUME_STEP;
/* adjust volume of actual sound sample */
- if (mixer[i].volume != PSND_MAX_VOLUME)
+ if (mixer[i].volume != SOUND_MAX_VOLUME)
for(j=0; j<sample_size; j++)
premix_first_buffer[j] =
- (mixer[i].volume * (long)premix_first_buffer[j])
- >> PSND_MAX_VOLUME_BITS;
+ mixer[i].volume * (long)premix_first_buffer[j] / SOUND_MAX_VOLUME;
/* fill the last mixing buffer with stereo or mono sound */
if (stereo)
{
- int middle_pos = PSND_MAX_LEFT2RIGHT / 2;
- int left_volume = stereo_volume[middle_pos + mixer[i].stereo];
- int right_volume= stereo_volume[middle_pos - mixer[i].stereo];
+ int left_volume = SOUND_VOLUME_LEFT(mixer[i].stereo_position);
+ int right_volume = SOUND_VOLUME_RIGHT(mixer[i].stereo_position);
for(j=0; j<sample_size; j++)
{
premix_left_buffer[j] =
- (left_volume * premix_first_buffer[j])
- >> PSND_MAX_LEFT2RIGHT_BITS;
+ left_volume * premix_first_buffer[j] / SOUND_MAX_LEFT2RIGHT;
premix_right_buffer[j] =
- (right_volume * premix_first_buffer[j])
- >> PSND_MAX_LEFT2RIGHT_BITS;
+ right_volume * premix_first_buffer[j] / SOUND_MAX_LEFT2RIGHT;
premix_last_buffer[2 * j + 0] += premix_left_buffer[j];
premix_last_buffer[2 * j + 1] += premix_right_buffer[j];
}
/* delete completed sound entries from the mixer */
- if (mixer[i].playingpos >= mixer[i].data_len)
+ if (mixer[i].playing_pos >= mixer[i].data_len)
{
if (IS_LOOP(mixer[i]))
- mixer[i].playingpos = 0;
+ mixer[i].playing_pos = 0;
else
- Mixer_RemoveSound(i);
+ Mixer_StopChannel(i);
}
else if (mixer[i].volume <= SOUND_FADING_VOLUME_THRESHOLD)
- Mixer_RemoveSound(i);
+ Mixer_StopChannel(i);
}
/* prepare final playing buffer according to system audio format */
#else /* !AUDIO_STREAMING_DSP */
-static int Mixer_Main_SimpleAudio(struct SoundControl snd_ctrl)
+static int Mixer_Main_SimpleAudio(SoundControl snd_ctrl)
{
static short premix_first_buffer[SND_BLOCKSIZE];
static byte playing_buffer[SND_BLOCKSIZE];
/* pointer, lenght and actual playing position of sound sample */
sample_ptr = mixer[i].data_ptr;
sample_len = mixer[i].data_len;
- sample_pos = mixer[i].playingpos;
+ sample_pos = mixer[i].playing_pos;
sample_size = MIN(max_sample_size, sample_len - sample_pos);
- mixer[i].playingpos += sample_size;
+ mixer[i].playing_pos += sample_size;
/* copy original sample to first mixing buffer */
CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
premix_first_buffer);
/* adjust volume of actual sound sample */
- if (mixer[i].volume != PSND_MAX_VOLUME)
+ if (mixer[i].volume != SOUND_MAX_VOLUME)
for(j=0; j<sample_size; j++)
premix_first_buffer[j] =
- (mixer[i].volume * (long)premix_first_buffer[j])
- >> PSND_MAX_VOLUME_BITS;
+ mixer[i].volume * (long)premix_first_buffer[j] / SOUND_MAX_VOLUME;
/* might be needed for u-law /dev/audio */
#if 0
linear_to_ulaw(premix_first_buffer[j]);
#endif
- /* delete completed sound entries from the mixer */
- if (mixer[i].playingpos >= mixer[i].data_len)
- Mixer_RemoveSound(i);
-
- for(i=0; i<sample_size; i++)
- playing_buffer[i] = (premix_first_buffer[i] >> 8) ^ 0x80;
-
- /* finally play the sound fragment */
- write(audio.device_fd, playing_buffer, sample_size);
-
- return sample_size;
-}
-#endif /* !AUDIO_STREAMING_DSP */
-
-void Mixer_Main()
-{
- struct SoundControl snd_ctrl;
- fd_set mixer_fdset;
-
- close(audio.mixer_pipe[1]); /* no writing into pipe needed */
-
- Mixer_InitChannels();
-
-#if defined(PLATFORM_HPUX)
- InitAudioDevice(&afmt);
-#endif
-
- FD_ZERO(&mixer_fdset);
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
-
- while(1) /* wait for sound playing commands from client */
- {
- struct timeval delay = { 0, 0 };
-
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
- select(audio.mixer_pipe[0] + 1, &mixer_fdset, NULL, NULL, NULL);
- if (!FD_ISSET(audio.mixer_pipe[0], &mixer_fdset))
- continue;
- if (read(audio.mixer_pipe[0], &snd_ctrl, sizeof(snd_ctrl))
- != sizeof(snd_ctrl))
- Error(ERR_EXIT_SOUND_SERVER, "broken pipe -- no sounds");
-
- HandleSoundRequest(snd_ctrl);
-
-#if defined(AUDIO_STREAMING_DSP)
-
- while (mixer_active_channels &&
- select(audio.mixer_pipe[0] + 1,
- &mixer_fdset, NULL, NULL, &delay) < 1)
- {
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
-
- Mixer_Main_DSP();
- }
-
-#else /* !AUDIO_STREAMING_DSP */
-
- if (!snd_ctrl.active || IS_LOOP(snd_ctrl) ||
- (audio.device_fd = OpenAudioDevice(audio.device_name)) < 0)
- continue;
-
- InitAudioDevice(&afmt);
-
- delay.tv_sec = 0;
- delay.tv_usec = 0;
-
- while (mixer_active_channels &&
- select(audio.mixer_pipe[0] + 1,
- &mixer_fdset, NULL, NULL, &delay) < 1)
- {
- int wait_percent = 90; /* wait 90% of the real playing time */
- int sample_size;
-
- FD_SET(audio.mixer_pipe[0], &mixer_fdset);
-
- sample_size = Mixer_Main_SimpleAudio(snd_ctrl);
-
- delay.tv_sec = 0;
- delay.tv_usec =
- ((sample_size * 10 * wait_percent) / afmt.sample_rate) * 1000;
- }
-
- CloseAudioDevice(&audio.device_fd);
-
- Mixer_InitChannels(); /* remove all sounds from mixer */
-
-#endif /* !AUDIO_STREAMING_DSP */
- }
-}
-#endif /* PLATFORM_UNIX */
-
-#if defined(PLATFORM_MSDOS)
-static void sound_handler(struct SoundControl snd_ctrl)
-{
- for (i=0; i<audio.num_channels; i++)
- {
- if (!mixer[i].active || mixer[i].loop)
- continue;
-
- mixer[i].playingpos = voice_get_position(mixer[i].voice);
- mixer[i].volume = voice_get_volume(mixer[i].voice);
- if (mixer[i].playingpos == -1 || !mixer[i].volume)
- {
- deallocate_voice(mixer[i].voice);
- Mixer_RemoveSound(i);
- }
- }
-
- HandleSoundRequest(snd_ctrl);
-}
-#endif /* PLATFORM_MSDOS */
-
-
-/* ------------------------------------------------------------------------- */
-/* platform dependant audio initialization code */
-/* ------------------------------------------------------------------------- */
-
-#if defined(AUDIO_LINUX_IOCTL)
-static void InitAudioDevice_Linux(struct AudioFormatInfo *afmt)
-{
- /* "ioctl()" expects pointer to 'int' value for stereo flag
- (boolean is defined as 'char', which will not work here) */
- unsigned int fragment_spec = 0;
- int fragment_size_query;
- int stereo = TRUE;
- struct
- {
- int format_ioctl;
- int format_result;
- }
- formats[] =
- {
- /* supported audio format in preferred order */
- { AFMT_S16_LE, AUDIO_FORMAT_S16 | AUDIO_FORMAT_LE },
- { AFMT_S16_BE, AUDIO_FORMAT_S16 | AUDIO_FORMAT_BE },
- { AFMT_U8, AUDIO_FORMAT_U8 },
- { -1, -1 }
- };
- int i;
-
- /* determine logarithm (log2) of the fragment size */
- while ((1 << fragment_spec) < afmt->fragment_size)
- fragment_spec++;
-
- /* use two fragments (play one fragment, prepare the other);
- one fragment would result in interrupted audio output, more
- than two fragments would raise audio output latency to much */
- fragment_spec |= 0x00020000;
-
- /* Example for fragment specification:
- - 2 buffers / 512 bytes (giving 1/16 second resolution for 8 kHz)
- - (with stereo the effective buffer size will shrink to 256)
- => fragment_size = 0x00020009 */
-
- if (ioctl(audio.device_fd, SNDCTL_DSP_SETFRAGMENT, &fragment_spec) < 0)
- Error(ERR_EXIT_SOUND_SERVER,
- "cannot set fragment size of /dev/dsp -- no sounds");
-
- i = 0;
- afmt->format = 0;
- while (formats[i].format_result != -1)
- {
- unsigned int audio_format = formats[i].format_ioctl;
- if (ioctl(audio.device_fd, SNDCTL_DSP_SETFMT, &audio_format) == 0)
- {
- afmt->format = formats[i].format_result;
- break;
- }
- }
-
- if (afmt->format == 0) /* no supported audio format found */
- Error(ERR_EXIT_SOUND_SERVER,
- "cannot set audio format of /dev/dsp -- no sounds");
-
- /* try if we can use stereo sound */
- afmt->stereo = TRUE;
- if (ioctl(audio.device_fd, SNDCTL_DSP_STEREO, &stereo) < 0)
- afmt->stereo = FALSE;
+ /* delete completed sound entries from the mixer */
+ if (mixer[i].playing_pos >= mixer[i].data_len)
+ Mixer_StopChannel(i);
- if (ioctl(audio.device_fd, SNDCTL_DSP_SPEED, &afmt->sample_rate) < 0)
- Error(ERR_EXIT_SOUND_SERVER,
- "cannot set sample rate of /dev/dsp -- no sounds");
+ for(i=0; i<sample_size; i++)
+ playing_buffer[i] = (premix_first_buffer[i] >> 8) ^ 0x80;
- /* get the real fragmentation size; this should return 512 */
- if (ioctl(audio.device_fd, SNDCTL_DSP_GETBLKSIZE, &fragment_size_query) < 0)
- Error(ERR_EXIT_SOUND_SERVER,
- "cannot get fragment size of /dev/dsp -- no sounds");
- if (fragment_size_query != afmt->fragment_size)
- Error(ERR_EXIT_SOUND_SERVER,
- "cannot set fragment size of /dev/dsp -- no sounds");
+ /* finally play the sound fragment */
+ write(audio.device_fd, playing_buffer, sample_size);
+
+ return sample_size;
}
-#endif /* AUDIO_LINUX_IOCTL */
+#endif /* !AUDIO_STREAMING_DSP */
-#if defined(PLATFORM_NETBSD)
-static void InitAudioDevice_NetBSD(struct AudioFormatInfo *afmt)
+void Mixer_Main()
{
- audio_info_t a_info;
- boolean stereo = TRUE;
+ SoundControl snd_ctrl;
+ fd_set mixer_fdset;
- AUDIO_INITINFO(&a_info);
- a_info.play.encoding = AUDIO_ENCODING_LINEAR8;
- a_info.play.precision = 8;
- a_info.play.channels = 2;
- a_info.play.sample_rate = sample_rate;
- a_info.blocksize = fragment_size;
+ close(audio.mixer_pipe[1]); /* no writing into pipe needed */
- afmt->format = AUDIO_FORMAT_U8;
- afmt->stereo = TRUE;
+ Mixer_InitChannels();
- if (ioctl(audio.device_fd, AUDIO_SETINFO, &a_info) < 0)
+#if defined(PLATFORM_HPUX)
+ InitAudioDevice(&afmt);
+#endif
+
+ FD_ZERO(&mixer_fdset);
+ FD_SET(audio.mixer_pipe[0], &mixer_fdset);
+
+ while(1) /* wait for sound playing commands from client */
{
- /* try to disable stereo */
- a_info.play.channels = 1;
+ struct timeval delay = { 0, 0 };
- afmt->stereo = FALSE;
+ FD_SET(audio.mixer_pipe[0], &mixer_fdset);
+ select(audio.mixer_pipe[0] + 1, &mixer_fdset, NULL, NULL, NULL);
+ if (!FD_ISSET(audio.mixer_pipe[0], &mixer_fdset))
+ continue;
- if (ioctl(audio.device_fd, AUDIO_SETINFO, &a_info) < 0)
- Error(ERR_EXIT_SOUND_SERVER,
- "cannot set sample rate of /dev/audio -- no sounds");
- }
-}
-#endif /* PLATFORM_NETBSD */
+ ReadSoundControlFromMainProcess(&snd_ctrl);
-#if defined(PLATFORM_HPUX)
-static void InitAudioDevice_HPUX(struct AudioFormatInfo *afmt)
-{
- struct audio_describe ainfo;
- int audio_ctl;
+ HandleSoundRequest(snd_ctrl);
- audio_ctl = open("/dev/audioCtl", O_WRONLY | O_NDELAY);
- if (audio_ctl == -1)
- Error(ERR_EXIT_SOUND_SERVER, "cannot open /dev/audioCtl -- no sounds");
+#if defined(AUDIO_STREAMING_DSP)
- if (ioctl(audio_ctl, AUDIO_DESCRIBE, &ainfo) == -1)
- Error(ERR_EXIT_SOUND_SERVER, "no audio info -- no sounds");
+ while (mixer_active_channels &&
+ select(audio.mixer_pipe[0] + 1,
+ &mixer_fdset, NULL, NULL, &delay) < 1)
+ {
+ FD_SET(audio.mixer_pipe[0], &mixer_fdset);
- if (ioctl(audio_ctl, AUDIO_SET_DATA_FORMAT, AUDIO_FORMAT_ULAW) == -1)
- Error(ERR_EXIT_SOUND_SERVER, "ulaw audio not available -- no sounds");
+ Mixer_Main_DSP();
+ }
- ioctl(audio_ctl, AUDIO_SET_CHANNELS, 1);
- ioctl(audio_ctl, AUDIO_SET_SAMPLE_RATE, 8000);
+#else /* !AUDIO_STREAMING_DSP */
- afmt->format = AUDIO_FORMAT_U8;
- afmt->stereo = FALSE;
- afmt->sample_rate = 8000;
+ if (!snd_ctrl.active || IS_LOOP(snd_ctrl) ||
+ (audio.device_fd = OpenAudioDevice(audio.device_name)) < 0)
+ continue;
- close(audio_ctl);
-}
-#endif /* PLATFORM_HPUX */
+ InitAudioDevice(&afmt);
-#if defined(PLATFORM_UNIX)
-static void InitAudioDevice(struct AudioFormatInfo *afmt)
-{
- afmt->stereo = TRUE;
- afmt->format = AUDIO_FORMAT_UNKNOWN;
- afmt->sample_rate = DEFAULT_AUDIO_SAMPLE_RATE;
- afmt->fragment_size = DEFAULT_AUDIO_FRAGMENT_SIZE;
+ delay.tv_sec = 0;
+ delay.tv_usec = 0;
-#if defined(AUDIO_LINUX_IOCTL)
- InitAudioDevice_Linux(afmt);
-#elif defined(PLATFORM_NETBSD)
- InitAudioDevice_NetBSD(afmt);
-#elif defined(PLATFORM_HPUX)
- InitAudioDevice_HPUX(afmt);
-#else
- /* generic /dev/audio stuff might be placed here */
-#endif
+ while (mixer_active_channels &&
+ select(audio.mixer_pipe[0] + 1,
+ &mixer_fdset, NULL, NULL, &delay) < 1)
+ {
+ int wait_percent = 90; /* wait 90% of the real playing time */
+ int sample_size;
+
+ FD_SET(audio.mixer_pipe[0], &mixer_fdset);
+
+ sample_size = Mixer_Main_SimpleAudio(snd_ctrl);
+
+ delay.tv_sec = 0;
+ delay.tv_usec =
+ ((sample_size * 10 * wait_percent) / afmt.sample_rate) * 1000;
+ }
+
+ CloseAudioDevice(&audio.device_fd);
+
+ Mixer_InitChannels(); /* remove all sounds from mixer */
+
+#endif /* !AUDIO_STREAMING_DSP */
+ }
}
-#endif /* PLATFORM_UNIX */
+#endif /* AUDIO_UNIX_NATIVE */
+
-#if defined(PLATFORM_UNIX) && !defined(AUDIO_STREAMING_DSP)
+#if defined(AUDIO_UNIX_NATIVE) && !defined(AUDIO_STREAMING_DSP)
/* these two are stolen from "sox"... :) */
return(sample);
}
-#endif /* PLATFORM_UNIX && !AUDIO_STREAMING_DSP */
+#endif /* AUDIO_UNIX_NATIVE && !AUDIO_STREAMING_DSP */
/* THE STUFF ABOVE IS ONLY USED BY THE SOUND SERVER CHILD PROCESS */
static SoundInfo *Load_WAV(char *filename)
{
SoundInfo *snd_info;
-#if !defined(TARGET_SDL) && !defined(PLATFORM_MSDOS)
+#if defined(AUDIO_UNIX_NATIVE)
+ struct SoundHeader_WAV header;
+#if 0
byte sound_header_buffer[WAV_HEADER_SIZE];
+ int i;
+#endif
char chunk_name[CHUNK_ID_LEN + 1];
int chunk_size;
FILE *file;
- int i;
#endif
if (!audio.sound_available)
return NULL;
}
+ snd_info->data_len = ((Mix_Chunk *)snd_info->data_ptr)->alen;
+
#elif defined(TARGET_ALLEGRO)
if ((snd_info->data_ptr = load_sample(filename)) == NULL)
return NULL;
}
-#else /* PLATFORM_UNIX */
+ snd_info->data_len = ((SAMPLE *)snd_info->data_ptr)->len;
+
+#else /* AUDIO_UNIX_NATIVE */
if ((file = fopen(filename, MODE_READ)) == NULL)
{
}
/* read chunk id "RIFF" */
- getFileChunk(file, chunk_name, &chunk_size, BYTE_ORDER_LITTLE_ENDIAN);
+ getFileChunkLE(file, chunk_name, &chunk_size);
if (strcmp(chunk_name, "RIFF") != 0)
{
Error(ERR_WARN, "missing 'RIFF' chunk of sound file '%s'", filename);
}
/* read "RIFF" type id "WAVE" */
- getFileChunk(file, chunk_name, NULL, BYTE_ORDER_LITTLE_ENDIAN);
+ getFileChunkLE(file, chunk_name, NULL);
if (strcmp(chunk_name, "WAVE") != 0)
{
Error(ERR_WARN, "missing 'WAVE' type ID of sound file '%s'", filename);
return NULL;
}
- while (getFileChunk(file, chunk_name, &chunk_size, BYTE_ORDER_LITTLE_ENDIAN))
+ while (getFileChunkLE(file, chunk_name, &chunk_size))
{
if (strcmp(chunk_name, "fmt ") == 0)
{
- /* read header information */
- for (i=0; i < MIN(chunk_size, WAV_HEADER_SIZE); i++)
- sound_header_buffer[i] = fgetc(file);
+ if (chunk_size < WAV_HEADER_SIZE)
+ {
+ Error(ERR_WARN, "sound file '%s': chunk 'fmt ' too short", filename);
+ fclose(file);
+ free(snd_info);
+ return NULL;
+ }
+
+ header.compression_code = getFile16BitLE(file);
+ header.num_channels = getFile16BitLE(file);
+ header.sample_rate = getFile32BitLE(file);
+ header.bytes_per_second = getFile32BitLE(file);
+ header.block_align = getFile16BitLE(file);
+ header.bits_per_sample = getFile16BitLE(file);
if (chunk_size > WAV_HEADER_SIZE)
ReadUnusedBytesFromFile(file, chunk_size - WAV_HEADER_SIZE);
+
+ if (header.compression_code != 1)
+ {
+ Error(ERR_WARN, "sound file '%s': compression code %d not supported",
+ filename, header.compression_code);
+ fclose(file);
+ free(snd_info);
+ return NULL;
+ }
+
+ if (header.num_channels != 1)
+ {
+ Error(ERR_WARN, "sound file '%s': number of %d channels not supported",
+ filename, header.num_channels);
+ fclose(file);
+ free(snd_info);
+ return NULL;
+ }
+
+ if (header.bits_per_sample != 8 && header.bits_per_sample != 16)
+ {
+ Error(ERR_WARN, "sound file '%s': %d bits per sample not supported",
+ filename, header.bits_per_sample);
+ fclose(file);
+ free(snd_info);
+ return NULL;
+ }
+
+ /* warn, but accept wrong sample rate (may be only slightly different) */
+ if (header.sample_rate != DEFAULT_AUDIO_SAMPLE_RATE)
+ Error(ERR_WARN, "sound file '%s': wrong sample rate %d instead of %d",
+ filename, header.sample_rate, DEFAULT_AUDIO_SAMPLE_RATE);
+
+#if 0
+ printf("WAV file: '%s'\n", filename);
+ printf(" Compression code: %d'\n", header.compression_code);
+ printf(" Number of channels: %d'\n", header.num_channels);
+ printf(" Sample rate: %ld'\n", header.sample_rate);
+ printf(" Average bytes per second: %ld'\n", header.bytes_per_second);
+ printf(" Block align: %d'\n", header.block_align);
+ printf(" Significant bits per sample: %d'\n", header.bits_per_sample);
+#endif
}
else if (strcmp(chunk_name, "data") == 0)
{
return NULL;
}
- snd_info->format = AUDIO_FORMAT_U8;
+ if (header.bits_per_sample == 8)
+ snd_info->format = AUDIO_FORMAT_U8;
+ else /* header.bits_per_sample == 16 */
+ {
+ snd_info->format = AUDIO_FORMAT_S16;
+ snd_info->data_len /= 2; /* correct number of samples */
+ }
-#endif /* PLATFORM_UNIX */
+#endif /* AUDIO_UNIX_NATIVE */
snd_info->type = SND_TYPE_WAV;
snd_info->source_filename = getStringCopy(filename);
{
char *filename = getCustomSoundFilename(basename);
+#if 0
+ printf("GOT CUSTOM SOUND FILE '%s'\n", filename);
+#endif
+
if (strcmp(basename, SND_FILE_UNDEFINED) == 0)
{
deleteSoundEntry(snd_info);
void LoadCustomMusic(void)
{
+ static boolean draw_init_text = TRUE; /* only draw at startup */
char *music_directory = getCustomMusicDirectory();
DIR *dir;
struct dirent *dir_entry;
return;
}
+ if (draw_init_text)
+ DrawInitText("Loading music:", 120, FC_GREEN);
+
while ((dir_entry = readdir(dir)) != NULL) /* loop until last dir entry */
{
char *basename = dir_entry->d_name;
char *filename = getPath2(music_directory, basename);
MusicInfo *mus_info = NULL;
+ if (draw_init_text)
+ DrawInitText(basename, 150, FC_YELLOW);
+
if (FileIsSound(basename))
mus_info = Load_WAV(filename);
else if (FileIsMusic(basename))
closedir(dir);
+ draw_init_text = FALSE;
+
if (num_music == 0)
Error(ERR_WARN, "cannot find any valid music files in directory '%s'",
music_directory);
if (!audio.music_available)
return;
-#if defined(TARGET_SDL)
-
- nr = nr % num_music;
-
- if (Music[nr]->type == MUS_TYPE_MOD)
- {
- Mix_PlayMusic(Music[nr]->data_ptr, -1);
- Mix_VolumeMusic(SOUND_MAX_VOLUME); /* must be _after_ Mix_PlayMusic()! */
- }
- else /* play WAV music loop */
- {
- Mix_Volume(audio.music_channel, SOUND_MAX_VOLUME);
- Mix_PlayChannel(audio.music_channel, Music[nr]->data_ptr, -1);
- }
-
-#else
-
PlaySoundMusic(nr);
-
-#endif
}
void PlaySound(int nr)
{
- PlaySoundExt(nr, PSND_MAX_VOLUME, PSND_MIDDLE, SND_CTRL_PLAY_SOUND);
+ PlaySoundExt(nr, SOUND_MAX_VOLUME, SOUND_MIDDLE, SND_CTRL_PLAY_SOUND);
}
-void PlaySoundStereo(int nr, int stereo)
+void PlaySoundStereo(int nr, int stereo_position)
{
- PlaySoundExt(nr, PSND_MAX_VOLUME, stereo, SND_CTRL_PLAY_SOUND);
+ PlaySoundExt(nr, SOUND_MAX_VOLUME, stereo_position, SND_CTRL_PLAY_SOUND);
}
void PlaySoundLoop(int nr)
{
- PlaySoundExt(nr, PSND_MAX_VOLUME, PSND_MIDDLE, SND_CTRL_PLAY_LOOP);
+ PlaySoundExt(nr, SOUND_MAX_VOLUME, SOUND_MIDDLE, SND_CTRL_PLAY_LOOP);
}
void PlaySoundMusic(int nr)
{
- PlaySoundExt(nr, PSND_MAX_VOLUME, PSND_MIDDLE, SND_CTRL_PLAY_MUSIC);
+ PlaySoundExt(nr, SOUND_MAX_VOLUME, SOUND_MIDDLE, SND_CTRL_PLAY_MUSIC);
}
-void PlaySoundExt(int nr, int volume, int stereo, int state)
+void PlaySoundExt(int nr, int volume, int stereo_position, int state)
{
- struct SoundControl snd_ctrl;
+ SoundControl snd_ctrl;
if (!audio.sound_available ||
!audio.sound_enabled ||
audio.sound_deactivated)
return;
- if (volume < PSND_MIN_VOLUME)
- volume = PSND_MIN_VOLUME;
- else if (volume > PSND_MAX_VOLUME)
- volume = PSND_MAX_VOLUME;
+ if (volume < SOUND_MIN_VOLUME)
+ volume = SOUND_MIN_VOLUME;
+ else if (volume > SOUND_MAX_VOLUME)
+ volume = SOUND_MAX_VOLUME;
- if (stereo < PSND_MAX_LEFT)
- stereo = PSND_MAX_LEFT;
- else if (stereo > PSND_MAX_RIGHT)
- stereo = PSND_MAX_RIGHT;
+ if (stereo_position < SOUND_MAX_LEFT)
+ stereo_position = SOUND_MAX_LEFT;
+ else if (stereo_position > SOUND_MAX_RIGHT)
+ stereo_position = SOUND_MAX_RIGHT;
snd_ctrl.active = TRUE;
- snd_ctrl.nr = nr;
+ snd_ctrl.nr = nr;
snd_ctrl.volume = volume;
- snd_ctrl.stereo = stereo;
- snd_ctrl.state = state;
-
-#if defined(TARGET_SDL)
- if (Sound[nr])
- {
- Mix_Volume(-1, SOUND_MAX_VOLUME);
- Mix_PlayChannel(-1, Sound[nr]->data_ptr, (state & SND_CTRL_LOOP ? -1 : 0));
- }
-#elif defined(PLATFORM_UNIX)
- if (audio.mixer_pid == 0) /* we are child process */
- return;
+ snd_ctrl.stereo_position = stereo_position;
+ snd_ctrl.state = state;
- if (write(audio.mixer_pipe[1], &snd_ctrl, sizeof(snd_ctrl)) < 0)
- {
- Error(ERR_WARN, "cannot pipe to child process -- no sounds");
- audio.sound_available = audio.sound_enabled = FALSE;
- return;
- }
-#elif defined(PLATFORM_MSDOS)
- sound_handler(snd_ctrl);
-#endif
+ HandleSoundRequest(snd_ctrl);
}
void FadeMusic(void)
{
- if (!audio.sound_available)
+ if (!audio.music_available)
return;
-#if defined(TARGET_SDL)
- Mix_FadeOutMusic(SOUND_FADING_INTERVAL);
- Mix_FadeOutChannel(audio.music_channel, SOUND_FADING_INTERVAL);
-#else
StopSoundExt(-1, SND_CTRL_FADE_MUSIC);
-#endif
}
void FadeSound(int nr)
void StopMusic(void)
{
-#if defined(TARGET_SDL)
- if (!audio.sound_available)
+ if (!audio.music_available)
return;
- Mix_HaltMusic();
- Mix_HaltChannel(audio.music_channel);
-#else
StopSoundExt(-1, SND_CTRL_STOP_MUSIC);
-#endif
}
void StopSound(int nr)
void StopSoundExt(int nr, int state)
{
- struct SoundControl snd_ctrl;
+ SoundControl snd_ctrl;
if (!audio.sound_available)
return;
snd_ctrl.active = FALSE;
- snd_ctrl.nr = nr;
- snd_ctrl.state = state;
-
-#if defined(TARGET_SDL)
-
- if (state & SND_CTRL_FADE)
- {
- int i;
-
- /*
- for (i=audio.first_sound_channel; i<audio.num_channels; i++)
- */
-
- for (i=0; i<audio.channels; i++)
- if (i != audio.music_channel || snd_ctrl.music)
- Mix_FadeOutChannel(i, SOUND_FADING_INTERVAL);
- if (state & SND_CTRL_MUSIC)
- Mix_FadeOutMusic(SOUND_FADING_INTERVAL);
- }
- else
- {
- int i;
-
- for (i=0; i<audio.channels; i++)
- if (i != audio.music_channel || snd_ctrl.music)
- Mix_HaltChannel(i);
- if (state & SND_CTRL_MUSIC)
- Mix_HaltMusic();
- }
+ snd_ctrl.nr = nr;
+ snd_ctrl.state = state;
-#elif !defined(PLATFORM_MSDOS)
-
- if (audio.mixer_pid == 0) /* we are child process */
- return;
-
- if (write(audio.mixer_pipe[1], &snd_ctrl, sizeof(snd_ctrl)) < 0)
- {
- Error(ERR_WARN, "cannot pipe to child process -- no sounds");
- audio.sound_available = audio.sound_enabled = FALSE;
- return;
- }
-#else /* PLATFORM_MSDOS */
- sound_handler(snd_ctrl);
-#endif
+ HandleSoundRequest(snd_ctrl);
}
ListNode *newListNode()
struct SetupFileList *setup_file_list;
int i;
+#if 0
+ printf("GOT CUSTOM SOUND CONFIG FILE '%s'\n", filename);
+#endif
+
/* always start with reliable default values */
for (i=0; i<num_sounds; i++)
sound_effect[i].filename = NULL;
freeSetupFileList(setup_file_list);
-#if 1
+#if 0
for (i=0; i<num_sounds; i++)
{
printf("'%s' ", sound_effect[i].text);
static void ReloadCustomSounds()
{
+ static boolean draw_init_text = TRUE; /* only draw at startup */
int i;
#if 0
LoadSoundsInfo();
+ if (draw_init_text)
+ DrawInitText("Loading sounds:", 120, FC_GREEN);
+
#if 0
printf("DEBUG: reloading %d sounds ...\n", num_sounds);
#endif
for(i=0; i<num_sounds; i++)
{
+ if (draw_init_text)
+ DrawInitText(sound_effect[i].text, 150, FC_YELLOW);
+
if (sound_effect[i].filename)
LoadSoundToList(sound_effect[i].filename, i);
else
LoadSoundToList(sound_effect[i].default_filename, i);
}
+ draw_init_text = FALSE;
+
/*
printf("list size == %d\n", getNumNodes(SoundFileList));
*/
if (!audio.sound_available)
return;
-#if defined(TARGET_SDL) || defined(TARGET_ALLEGRO)
- ReloadCustomSounds();
-#elif defined(PLATFORM_UNIX)
+#if defined(AUDIO_UNIX_NATIVE)
WriteReloadInfoToPipe(set_name, SND_CTRL_RELOAD_SOUNDS);
+#else
+ ReloadCustomSounds();
#endif
}
if (!audio.music_available)
return;
-#if defined(TARGET_SDL) || defined(TARGET_ALLEGRO)
- ReloadCustomMusic();
-#elif defined(PLATFORM_UNIX)
+#if defined(AUDIO_UNIX_NATIVE)
WriteReloadInfoToPipe(set_name, SND_CTRL_RELOAD_MUSIC);
+#else
+ ReloadCustomMusic();
#endif
}
Mix_FreeChunk(sound->data_ptr);
#elif defined(TARGET_ALLEGRO)
destroy_sample(sound->data_ptr);
-#else /* PLATFORM_UNIX */
+#else /* AUDIO_UNIX_NATIVE */
free(sound->data_ptr);
#endif
}
Mix_FreeChunk(music->data_ptr);
#elif defined(TARGET_ALLEGRO)
destroy_sample(music->data_ptr);
-#else /* PLATFORM_UNIX */
+#else /* AUDIO_UNIX_NATIVE */
free(music->data_ptr);
#endif
}
if (Sound == NULL)
return;
+#if 0
printf("%s: FREEING SOUNDS ...\n",
- audio.mixer_pid == 0 ? "CHILD" : "PARENT");
+ IS_CHILD_PROCESS(audio.mixer_pid) ? "CHILD" : "PARENT");
+#endif
for(i=0; i<num_sounds; i++)
deleteSoundEntry(&Sound[i]);
FreeSound(Sound[i]);
*/
+#if 0
printf("%s: FREEING SOUNDS -- DONE\n",
- audio.mixer_pid == 0 ? "CHILD" : "PARENT");
+ IS_CHILD_PROCESS(audio.mixer_pid) ? "CHILD" : "PARENT");
+#endif
free(Sound);
projects / rocksndiamonds.git / blobdiff