X-Git-Url: https://git.artsoft.org/?a=blobdiff_plain;f=src%2Flibgame%2Fsound.c;h=b29d5dd62616a3dc9b1f245b2cb83707b9cf6397;hb=cb5fe20318d9b2c18cb82bc1f7197150cfba7bc0;hp=ac814e90f30a6fbcae9e2daea299796a9285037f;hpb=71a1616f11627ab005c19fc6fc06ba23f6b4b633;p=rocksndiamonds.git diff --git a/src/libgame/sound.c b/src/libgame/sound.c index ac814e90..b29d5dd6 100644 --- a/src/libgame/sound.c +++ b/src/libgame/sound.c @@ -17,12 +17,136 @@ #include #include #include +#include + +#include "platform.h" + +#if defined(PLATFORM_LINUX) +#include +#include +#elif defined(PLATFORM_FREEBSD) +#include +#elif defined(PLATFORM_NETBSD) +#include +#include +#elif defined(PLATFORM_HPUX) +#include +#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 { @@ -44,29 +168,36 @@ static ListNode *SoundFileList = NULL; 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; @@ -123,7 +254,6 @@ static boolean TestAudioDevices(void) return TRUE; } -#if !defined(TARGET_SDL) static boolean ForkAudioProcess(void) { if (pipe(audio.mixer_pipe) < 0) @@ -138,19 +268,13 @@ static boolean ForkAudioProcess(void) 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) { @@ -175,16 +299,177 @@ void UnixCloseAudio(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); } -static void WriteSoundControlToPipe(struct SoundControl snd_ctrl) + +/* ------------------------------------------------------------------------- */ +/* 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 (audio.mixer_pid == 0) /* we are child process */ + if (IS_CHILD_PROCESS(audio.mixer_pid)) return; - if (write(audio.mixer_pipe[1], &snd_ctrl, sizeof(struct SoundControl)) < 0) + 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; @@ -192,26 +477,26 @@ static void WriteSoundControlToPipe(struct SoundControl snd_ctrl) } } -static void ReadSoundControlFromPipe(struct SoundControl *snd_ctrl) +static void ReadSoundControlFromMainProcess(SoundControl *snd_ctrl) { - if (audio.mixer_pid != 0) /* we are parent process */ + if (IS_PARENT_PROCESS(audio.mixer_pid)) return; - if (read(audio.mixer_pipe[0], snd_ctrl, sizeof(struct SoundControl)) - != sizeof(struct SoundControl)) + 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 */ @@ -248,9 +533,9 @@ static void WriteReloadInfoToPipe(char *set_name, int type) } } -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; @@ -259,7 +544,7 @@ static void ReadReloadInfoFromPipe(struct SoundControl snd_ctrl) 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)); @@ -273,7 +558,7 @@ static void ReadReloadInfoFromPipe(struct SoundControl snd_ctrl) 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, @@ -298,12 +583,18 @@ static void ReadReloadInfoFromPipe(struct SoundControl snd_ctrl) 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() { @@ -314,58 +605,186 @@ 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; @@ -374,6 +793,10 @@ static void Mixer_InsertSound(struct SoundControl snd_ctrl) 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_ctrl.nr = snd_ctrl.nr % num_music; else if (snd_ctrl.nr >= num_sounds) @@ -381,73 +804,61 @@ static void Mixer_InsertSound(struct SoundControl 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; itype; + 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 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= 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= 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= longest) { longest = actual; @@ -480,10 +890,37 @@ static void Mixer_InsertSound(struct SoundControl snd_ctrl) } } - 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 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; idata_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) @@ -517,18 +953,12 @@ static void Mixer_InsertSound(struct SoundControl snd_ctrl) 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; @@ -536,29 +966,33 @@ static void Mixer_InsertSound(struct SoundControl snd_ctrl) } } -static void HandleSoundRequest(struct SoundControl snd_ctrl) +static void HandleSoundRequest(SoundControl snd_ctrl) { int i; -#if defined(PLATFORM_MSDOS) - for (i=0; imajor, + 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) { @@ -637,8 +1089,6 @@ static void CopySampleToMixingBuffer(struct SoundControl *snd_ctrl, #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]; @@ -650,15 +1100,6 @@ static void Mixer_Main_DSP() 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; @@ -689,12 +1130,18 @@ static void Mixer_Main_DSP() 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, @@ -717,7 +1164,7 @@ static void Mixer_Main_DSP() 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; } } @@ -728,27 +1175,23 @@ static void Mixer_Main_DSP() 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> 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> 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]; @@ -761,15 +1204,15 @@ static void Mixer_Main_DSP() } /* 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 */ @@ -809,7 +1252,7 @@ static void Mixer_Main_DSP() #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]; @@ -825,20 +1268,19 @@ static int Mixer_Main_SimpleAudio(struct SoundControl snd_ctrl) /* 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> 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 @@ -848,8 +1290,8 @@ static int Mixer_Main_SimpleAudio(struct SoundControl snd_ctrl) #endif /* delete completed sound entries from the mixer */ - if (mixer[i].playingpos >= mixer[i].data_len) - Mixer_RemoveSound(i); + if (mixer[i].playing_pos >= mixer[i].data_len) + Mixer_StopChannel(i); for(i=0; i> 8) ^ 0x80; @@ -863,239 +1305,82 @@ static int Mixer_Main_SimpleAudio(struct SoundControl snd_ctrl) void Mixer_Main() { - struct SoundControl snd_ctrl; + 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; - - ReadSoundControlFromPipe(&snd_ctrl); - - 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 */ - - -/* ------------------------------------------------------------------------- */ -/* 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++; + close(audio.mixer_pipe[1]); /* no writing into pipe needed */ - /* 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; + Mixer_InitChannels(); - /* 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 defined(PLATFORM_HPUX) + InitAudioDevice(&afmt); +#endif - 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"); + FD_ZERO(&mixer_fdset); + FD_SET(audio.mixer_pipe[0], &mixer_fdset); - i = 0; - afmt->format = 0; - while (formats[i].format_result != -1) + while(1) /* wait for sound playing commands from client */ { - 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; - } - } + struct timeval delay = { 0, 0 }; - if (afmt->format == 0) /* no supported audio format found */ - Error(ERR_EXIT_SOUND_SERVER, - "cannot set audio format of /dev/dsp -- no sounds"); + 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; - /* try if we can use stereo sound */ - afmt->stereo = TRUE; - if (ioctl(audio.device_fd, SNDCTL_DSP_STEREO, &stereo) < 0) - afmt->stereo = FALSE; + ReadSoundControlFromMainProcess(&snd_ctrl); - 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"); + HandleSoundRequest(snd_ctrl); - /* 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(AUDIO_STREAMING_DSP) -#if defined(PLATFORM_NETBSD) -static void InitAudioDevice_NetBSD(struct AudioFormatInfo *afmt) -{ - audio_info_t a_info; - boolean stereo = TRUE; + while (mixer_active_channels && + select(audio.mixer_pipe[0] + 1, + &mixer_fdset, NULL, NULL, &delay) < 1) + { + FD_SET(audio.mixer_pipe[0], &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; + Mixer_Main_DSP(); + } - afmt->format = AUDIO_FORMAT_U8; - afmt->stereo = TRUE; +#else /* !AUDIO_STREAMING_DSP */ - if (ioctl(audio.device_fd, AUDIO_SETINFO, &a_info) < 0) - { - /* try to disable stereo */ - a_info.play.channels = 1; + if (!snd_ctrl.active || IS_LOOP(snd_ctrl) || + (audio.device_fd = OpenAudioDevice(audio.device_name)) < 0) + continue; - afmt->stereo = FALSE; + InitAudioDevice(&afmt); - 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 */ + delay.tv_sec = 0; + delay.tv_usec = 0; -#if defined(PLATFORM_HPUX) -static void InitAudioDevice_HPUX(struct AudioFormatInfo *afmt) -{ - struct audio_describe ainfo; - int audio_ctl; + 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; - audio_ctl = open("/dev/audioCtl", O_WRONLY | O_NDELAY); - if (audio_ctl == -1) - Error(ERR_EXIT_SOUND_SERVER, "cannot open /dev/audioCtl -- no sounds"); + FD_SET(audio.mixer_pipe[0], &mixer_fdset); - if (ioctl(audio_ctl, AUDIO_DESCRIBE, &ainfo) == -1) - Error(ERR_EXIT_SOUND_SERVER, "no audio info -- no sounds"); + sample_size = Mixer_Main_SimpleAudio(snd_ctrl); - if (ioctl(audio_ctl, AUDIO_SET_DATA_FORMAT, AUDIO_FORMAT_ULAW) == -1) - Error(ERR_EXIT_SOUND_SERVER, "ulaw audio not available -- no sounds"); + delay.tv_sec = 0; + delay.tv_usec = + ((sample_size * 10 * wait_percent) / afmt.sample_rate) * 1000; + } - ioctl(audio_ctl, AUDIO_SET_CHANNELS, 1); - ioctl(audio_ctl, AUDIO_SET_SAMPLE_RATE, 8000); + CloseAudioDevice(&audio.device_fd); - afmt->format = AUDIO_FORMAT_U8; - afmt->stereo = FALSE; - afmt->sample_rate = 8000; + Mixer_InitChannels(); /* remove all sounds from mixer */ - close(audio_ctl); +#endif /* !AUDIO_STREAMING_DSP */ + } } -#endif /* PLATFORM_HPUX */ +#endif /* AUDIO_UNIX_NATIVE */ -#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; - -#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 -} -#endif /* PLATFORM_UNIX */ -#if defined(PLATFORM_UNIX) && !defined(AUDIO_STREAMING_DSP) +#if defined(AUDIO_UNIX_NATIVE) && !defined(AUDIO_STREAMING_DSP) /* these two are stolen from "sox"... :) */ @@ -1200,7 +1485,7 @@ static int ulaw_to_linear(unsigned char ulawbyte) 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 */ @@ -1213,12 +1498,15 @@ static int ulaw_to_linear(unsigned char ulawbyte) 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) @@ -1239,6 +1527,8 @@ static SoundInfo *Load_WAV(char *filename) 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) @@ -1248,7 +1538,9 @@ static SoundInfo *Load_WAV(char *filename) 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) { @@ -1258,7 +1550,7 @@ static SoundInfo *Load_WAV(char *filename) } /* 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); @@ -1268,7 +1560,7 @@ static SoundInfo *Load_WAV(char *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); @@ -1277,16 +1569,69 @@ static SoundInfo *Load_WAV(char *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) { @@ -1321,9 +1666,15 @@ static SoundInfo *Load_WAV(char *filename) 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); @@ -1399,6 +1750,10 @@ static void LoadCustomSound(SoundInfo **snd_info, char *basename) { 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); @@ -1469,6 +1824,7 @@ static MusicInfo *Load_MOD(char *filename) void LoadCustomMusic(void) { + static boolean draw_init_text = TRUE; /* only draw at startup */ char *music_directory = getCustomMusicDirectory(); DIR *dir; struct dirent *dir_entry; @@ -1483,12 +1839,18 @@ void LoadCustomMusic(void) 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)) @@ -1506,6 +1868,8 @@ void LoadCustomMusic(void) closedir(dir); + draw_init_text = FALSE; + if (num_music == 0) Error(ERR_WARN, "cannot find any valid music files in directory '%s'", music_directory); @@ -1516,101 +1880,63 @@ void PlayMusic(int nr) 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) - - WriteSoundControlToPipe(snd_ctrl); - -#elif defined(PLATFORM_MSDOS) + snd_ctrl.stereo_position = stereo_position; + snd_ctrl.state = state; HandleSoundRequest(snd_ctrl); - -#endif } 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) @@ -1626,15 +1952,10 @@ void FadeSounds() 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) @@ -1650,51 +1971,16 @@ void StopSounds() 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; idata_ptr); #elif defined(TARGET_ALLEGRO) destroy_sample(sound->data_ptr); -#else /* PLATFORM_UNIX */ +#else /* AUDIO_UNIX_NATIVE */ free(sound->data_ptr); #endif } @@ -1910,7 +2209,7 @@ void FreeMusic(MusicInfo *music) 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 } @@ -1925,8 +2224,10 @@ void FreeAllSounds() 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