+ 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(TARGET_SDL)
+ Mix_FadeOutChannel(channel, SOUND_FADING_INTERVAL);
+#elif defined(TARGET_ALLEGRO)
+ if (voice_check(mixer[channel].voice))
+ voice_ramp_volume(mixer[channel].voice, SOUND_FADING_INTERVAL, 0);
+#endif
+}
+
+static void Mixer_FadeMusicChannel()
+{
+ Mixer_FadeChannel(audio.music_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[channel].state &= ~SND_CTRL_FADE;
+ mixer[channel].volume = SOUND_MAX_VOLUME;
+
+#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(SoundControl snd_ctrl)
+{
+ SoundInfo *snd_info;
+ int i, k;
+ int num_sounds = getSoundListSize();
+ int num_music = getMusicListSize();
+
+#if 0
+ if (IS_MUSIC(snd_ctrl))
+ printf("NEW MUSIC %d ARRIVED [%d/%d] [%d ACTIVE CHANNELS]\n",
+ snd_ctrl.nr, num_music, num_music_noconf, mixer_active_channels);
+ else
+ printf("NEW SOUND %d ARRIVED [%d] [%d ACTIVE CHANNELS]\n",
+ snd_ctrl.nr, num_sounds, mixer_active_channels);
+#endif
+
+#if 0
+ /* !!! TEST ONLY !!! */
+ if (IS_MUSIC(snd_ctrl))
+ snd_ctrl.nr = 0;
+#endif
+
+#if 1
+ if (IS_MUSIC(snd_ctrl))
+ {
+ if (snd_ctrl.nr >= num_music) /* invalid music */
+ return;
+
+ if (snd_ctrl.nr < 0) /* undefined music */
+ {
+ if (num_music_noconf == 0) /* no fallback music available */
+ return;
+
+ snd_ctrl.nr = UNMAP_NOCONF_MUSIC(snd_ctrl.nr) % num_music_noconf;
+ snd_info = Music_NoConf[snd_ctrl.nr];
+ }
+ else
+ snd_info = getMusicInfoEntryFromMusicID(snd_ctrl.nr);
+ }
+ else
+ {
+ if (snd_ctrl.nr < 0 || snd_ctrl.nr >= num_sounds)
+ return;
+
+ snd_info = getSoundInfoEntryFromSoundID(snd_ctrl.nr);
+ }
+
+ /*
+ if (snd_ctrl.nr >= (IS_MUSIC(snd_ctrl) ? num_music : num_sounds))
+ return;
+ */
+#else
+ if (IS_MUSIC(snd_ctrl))
+ {
+ if (num_music_noconf == 0)
+ return;
+
+ snd_ctrl.nr = snd_ctrl.nr % num_music_noconf;
+ }
+ else if (snd_ctrl.nr >= num_sounds)
+ return;
+#endif
+
+#if 0
+#if 1
+ snd_info = (IS_MUSIC(snd_ctrl) ? getMusicInfoEntryFromMusicID(snd_ctrl.nr) :
+ getSoundInfoEntryFromSoundID(snd_ctrl.nr));
+#else
+ snd_info = (IS_MUSIC(snd_ctrl) ? Music_NoConf[snd_ctrl.nr] :
+ getSoundInfoEntryFromSoundID(snd_ctrl.nr));
+#endif
+#endif
+
+ if (snd_info == NULL)
+ return;
+
+ /* 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;
+ snd_ctrl.num_channels = snd_info->num_channels;
+
+ /* play music samples on a dedicated music channel */
+ if (IS_MUSIC(snd_ctrl))
+ {
+#if 0
+ printf("::: slot %d, ptr 0x%08x\n", snd_ctrl.nr, snd_ctrl.data_ptr);
+#endif
+
+ Mixer_StopMusicChannel();
+
+ mixer[audio.music_channel] = snd_ctrl;
+ Mixer_PlayMusicChannel();
+
+ return;
+ }
+
+ /* check if (and how often) this sound sample is already playing */
+ for (k = 0, i = audio.first_sound_channel; i < audio.num_channels; i++)
+ if (mixer[i].active && SAME_SOUND_DATA(mixer[i], snd_ctrl))
+ k++;
+
+#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].active && SAME_SOUND_DATA(mixer[i], snd_ctrl))
+ {
+#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
+ }
+ }
+
+ return;
+ }
+
+#if 0
+ printf("PLAYING NEW SOUND %d\n", snd_ctrl.nr);
+#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 || !SAME_SOUND_NR(mixer[i], snd_ctrl))
+ continue;
+
+ actual = 1000 * playing_time / mixer[i].data_len;
+
+ if (actual >= longest)
+ {
+ longest = actual;
+ longest_nr = i;
+ }
+ }
+
+ 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 DEBUG
+ /* channel allocation sanity check -- should not be needed */
+ if (mixer_active_channels ==
+ audio.num_channels - (mixer[audio.music_channel].active ? 0 : 1))
+ {
+ for (i = audio.first_sound_channel; i < audio.num_channels; i++)
+ {
+ if (!mixer[i].active)
+ {
+ Error(ERR_RETURN, "Mixer_InsertSound: Channel %d inactive", i);
+ Error(ERR_RETURN, "Mixer_InsertSound: This should never happen!");
+
+ mixer_active_channels--;
+ }
+ }
+ }
+#endif
+
+ 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;
+
+#if 0
+#if DEBUG
+ /* print some debugging information about audio channel usage */
+ for (i = audio.first_sound_channel; i < audio.num_channels; i++)
+ {
+ Error(ERR_RETURN, "Mixer_InsertSound: %d [%d]: %ld (%ld)",
+ i, mixer[i].active, mixer[i].data_len, (long)mixer[i].data_ptr);
+ }
+#endif
+#endif
+
+ 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 the new sound to the mixer */
+ for (i = audio.first_sound_channel; i < audio.num_channels; i++)
+ {
+#if 0
+ printf("CHECKING CHANNEL %d FOR SOUND %d ...\n", i, snd_ctrl.nr);
+#endif
+
+ if (!mixer[i].active)
+ {
+#if 0
+ printf("ADDING NEW SOUND %d TO MIXER\n", snd_ctrl.nr);
+#endif
+
+#if defined(AUDIO_UNIX_NATIVE)
+ if (snd_info->data_len == 0)
+ {
+ printf("THIS SHOULD NEVER HAPPEN! [snd_info->data_len == 0]\n");
+ }
+#endif
+
+ mixer[i] = snd_ctrl;
+ Mixer_PlayChannel(i);
+
+ break;
+ }
+ }
+}
+
+static void HandleSoundRequest(SoundControl snd_ctrl)
+{
+ int i;
+
+#if defined(AUDIO_UNIX_NATIVE)
+ if (IS_PARENT_PROCESS())
+ {
+ 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 */
+ {
+ 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
+ ReloadCustomMusic();
+ }
+ else if (IS_FADING(snd_ctrl)) /* fade out existing sound or music */
+ {
+ if (IS_MUSIC(snd_ctrl))
+ {
+ Mixer_FadeMusicChannel();
+ return;
+ }
+
+ for (i = audio.first_sound_channel; i < audio.num_channels; i++)
+ if (SAME_SOUND_NR(mixer[i], snd_ctrl) || ALL_SOUNDS(snd_ctrl))
+ Mixer_FadeChannel(i);
+ }
+ else if (IS_STOPPING(snd_ctrl)) /* stop existing sound or music */
+ {
+ if (IS_MUSIC(snd_ctrl))
+ {
+ Mixer_StopMusicChannel();
+ return;
+ }
+
+ for (i = audio.first_sound_channel; i < audio.num_channels; i++)
+ if (SAME_SOUND_NR(mixer[i], snd_ctrl) || ALL_SOUNDS(snd_ctrl))
+ 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 */
+ {
+ Mixer_InsertSound(snd_ctrl);
+ }
+}
+
+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;
+
+ /* 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(AUDIO_UNIX_NATIVE)
+
+static void CopySampleToMixingBuffer(SoundControl *snd_ctrl,
+ int sample_pos, int sample_size,
+ short *buffer_base_ptr, int buffer_pos,
+ int num_output_channels)
+{
+ short *buffer_ptr = buffer_base_ptr + num_output_channels * buffer_pos;
+ int num_channels = snd_ctrl->num_channels;
+ int stepsize = num_channels;
+ int output_stepsize = num_output_channels;
+ int i, j;
+
+ if (snd_ctrl->format == AUDIO_FORMAT_U8)
+ {
+ byte *sample_ptr = (byte *)snd_ctrl->data_ptr + num_channels * sample_pos;
+
+ for (i = 0; i < num_output_channels; i++)
+ {
+ int offset = (snd_ctrl->num_channels == 1 ? 0 : i);
+
+ for (j = 0; j < sample_size; j++)
+ buffer_ptr[output_stepsize * j + i] =
+ ((short)(sample_ptr[stepsize * j + offset] ^ 0x80)) << 8;
+ }
+ }
+ else /* AUDIO_FORMAT_S16 */
+ {
+ short *sample_ptr= (short *)snd_ctrl->data_ptr + num_channels * sample_pos;
+
+ for (i = 0; i < num_output_channels; i++)
+ {
+ int offset = (snd_ctrl->num_channels == 1 ? 0 : i);
+
+ for (j = 0; j < sample_size; j++)
+ buffer_ptr[output_stepsize * j + i] =
+ sample_ptr[stepsize * j + offset];
+ }
+ }
+}
+
+#if defined(AUDIO_STREAMING_DSP)
+static void Mixer_Main_DSP()
+{
+ static short premix_first_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ static long premix_last_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ static byte playing_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ boolean stereo;
+ int fragment_size;
+ int sample_bytes;
+ int max_sample_size;
+ int num_output_channels;
+ int i, j;
+
+ if (!mixer_active_channels)
+ return;
+
+ if (audio.device_fd < 0)
+ {
+ if ((audio.device_fd = OpenAudioDevice(audio.device_name)) < 0)
+ return;
+
+ InitAudioDevice(&afmt);
+ }
+
+ stereo = afmt.stereo;
+ fragment_size = afmt.fragment_size;
+ sample_bytes = (afmt.format & AUDIO_FORMAT_U8 ? 1 : 2);
+ num_output_channels = (stereo ? 2 : 1);
+ max_sample_size = fragment_size / (num_output_channels * sample_bytes);
+
+ /* first clear the last premixing buffer */
+ memset(premix_last_buffer, 0,
+ max_sample_size * num_output_channels * sizeof(long));
+
+ for (i = 0; i < audio.num_channels; i++)
+ {
+ void *sample_ptr;
+ int sample_len;
+ int sample_pos;
+ int sample_size;
+
+ 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].playing_pos;
+ sample_size = MIN(max_sample_size, sample_len - sample_pos);
+ mixer[i].playing_pos += sample_size;
+
+ /* copy original sample to first mixing buffer */
+ CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
+ premix_first_buffer, 0, num_output_channels);
+
+ /* are we about to restart a looping sound? */
+ if (IS_LOOP(mixer[i]) && sample_size < max_sample_size)
+ {
+ while (sample_size < max_sample_size)
+ {
+ int restarted_sample_size =
+ MIN(max_sample_size - sample_size, sample_len);
+
+ CopySampleToMixingBuffer(&mixer[i], 0, restarted_sample_size,
+ premix_first_buffer, sample_size,
+ num_output_channels);
+
+ mixer[i].playing_pos = restarted_sample_size;
+ sample_size += restarted_sample_size;
+ }
+ }
+
+ /* decrease volume if sound is fading out */
+ if (IS_FADING(mixer[i]) &&
+ mixer[i].volume >= SOUND_FADING_VOLUME_THRESHOLD)
+ mixer[i].volume -= SOUND_FADING_VOLUME_STEP;
+
+ /* adjust volume of actual sound sample */
+ if (mixer[i].volume != SOUND_MAX_VOLUME)
+ for (j = 0; j < sample_size * num_output_channels; j++)
+ premix_first_buffer[j] =
+ mixer[i].volume * (long)premix_first_buffer[j] / SOUND_MAX_VOLUME;
+
+ /* adjust left and right channel volume due to stereo sound position */
+ if (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_first_buffer[2 * j + 0] =
+ left_volume * premix_first_buffer[2 * j + 0] / SOUND_MAX_LEFT2RIGHT;
+ premix_first_buffer[2 * j + 1] =
+ right_volume * premix_first_buffer[2 * j + 1] / SOUND_MAX_LEFT2RIGHT;
+ }
+ }
+
+ /* fill the last mixing buffer with stereo or mono sound */
+ for (j = 0; j < sample_size * num_output_channels; j++)
+ premix_last_buffer[j] += premix_first_buffer[j];
+
+ /* delete completed sound entries from the mixer */
+ if (mixer[i].playing_pos >= mixer[i].data_len)
+ {
+ if (IS_LOOP(mixer[i]))
+ mixer[i].playing_pos = 0;
+ else
+ Mixer_StopChannel(i);
+ }
+ else if (mixer[i].volume <= SOUND_FADING_VOLUME_THRESHOLD)
+ Mixer_StopChannel(i);
+ }
+
+ /* prepare final playing buffer according to system audio format */
+ for (i = 0; i < max_sample_size * num_output_channels; i++)
+ {
+ /* cut off at 17 bit value */
+ if (premix_last_buffer[i] < -65535)
+ premix_last_buffer[i] = -65535;
+ else if (premix_last_buffer[i] > 65535)
+ premix_last_buffer[i] = 65535;
+
+ /* shift to 16 bit value */
+ premix_last_buffer[i] >>= 1;
+
+ if (afmt.format & AUDIO_FORMAT_U8)
+ {
+ playing_buffer[i] = (premix_last_buffer[i] >> 8) ^ 0x80;
+ }
+ else if (afmt.format & AUDIO_FORMAT_LE) /* 16 bit */
+ {
+ playing_buffer[2 * i + 0] = premix_last_buffer[i] & 0xff;
+ playing_buffer[2 * i + 1] = premix_last_buffer[i] >> 8;
+ }
+ else /* big endian */
+ {
+ playing_buffer[2 * i + 0] = premix_last_buffer[i] >> 8;
+ playing_buffer[2 * i + 1] = premix_last_buffer[i] & 0xff;
+ }
+ }
+
+ /* finally play the sound fragment */
+ write(audio.device_fd, playing_buffer, fragment_size);
+
+ if (!mixer_active_channels)
+ CloseAudioDevice(&audio.device_fd);
+}
+
+#else /* !AUDIO_STREAMING_DSP */
+
+static int Mixer_Main_SimpleAudio(SoundControl snd_ctrl)
+{
+ static short premix_first_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ static byte playing_buffer[DEFAULT_AUDIO_FRAGMENT_SIZE];
+ int max_sample_size = DEFAULT_AUDIO_FRAGMENT_SIZE;
+ int num_output_channels = 1;
+ void *sample_ptr;
+ int sample_len;
+ int sample_pos;
+ int sample_size;
+ int i, j;
+
+ i = 1;
+
+ /* 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].playing_pos;
+ sample_size = MIN(max_sample_size, sample_len - sample_pos);
+ mixer[i].playing_pos += sample_size;
+
+ /* copy original sample to first mixing buffer */
+ CopySampleToMixingBuffer(&mixer[i], sample_pos, sample_size,
+ premix_first_buffer, 0, num_output_channels);
+
+ /* adjust volume of actual sound sample */
+ 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] / SOUND_MAX_VOLUME;
+
+ /* might be needed for u-law /dev/audio */
+#if 1
+ for (j = 0; j < sample_size; j++)
+ playing_buffer[j] =
+ linear_to_ulaw(premix_first_buffer[j]);
+#endif
+
+ /* delete completed sound entries from the mixer */
+ if (mixer[i].playing_pos >= mixer[i].data_len)
+ Mixer_StopChannel(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()
+{
+ 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;
+
+ ReadSoundControlFromMainProcess(&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 /* AUDIO_UNIX_NATIVE */
+
+
+#if defined(AUDIO_UNIX_NATIVE) && !defined(AUDIO_STREAMING_DSP)
+
+/* these two are stolen from "sox"... :) */
+
+/*
+** This routine converts from linear to ulaw.
+**
+** Craig Reese: IDA/Supercomputing Research Center
+** Joe Campbell: Department of Defense
+** 29 September 1989
+**
+** References:
+** 1) CCITT Recommendation G.711 (very difficult to follow)
+** 2) "A New Digital Technique for Implementation of Any
+** Continuous PCM Companding Law," Villeret, Michel,
+** et al. 1973 IEEE Int. Conf. on Communications, Vol 1,
+** 1973, pg. 11.12-11.17
+** 3) MIL-STD-188-113,"Interoperability and Performance Standards
+** for Analog-to_Digital Conversion Techniques,"
+** 17 February 1987
+**
+** Input: Signed 16 bit linear sample
+** Output: 8 bit ulaw sample
+*/
+
+#define ZEROTRAP /* turn on the trap as per the MIL-STD */
+#define BIAS 0x84 /* define the add-in bias for 16 bit samples */
+#define CLIP 32635
+
+static unsigned char linear_to_ulaw(int sample)
+{
+ static int exp_lut[256] =
+ {
+ 0,0,1,1,2,2,2,2,3,3,3,3,3,3,3,3,
+ 4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,
+ 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
+ 5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,5,
+ 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
+ 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
+ 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
+ 6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,6,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,
+ 7,7,7,7,7,7,7,7,7,7,7,7,7,7,7,7
+ };
+
+ int sign, exponent, mantissa;
+ unsigned char ulawbyte;
+
+ /* Get the sample into sign-magnitude. */
+ sign = (sample >> 8) & 0x80; /* set aside the sign */
+ if (sign != 0)
+ sample = -sample; /* get magnitude */
+ if (sample > CLIP)
+ sample = CLIP; /* clip the magnitude */
+
+ /* Convert from 16 bit linear to ulaw. */
+ sample = sample + BIAS;
+ exponent = exp_lut[( sample >> 7 ) & 0xFF];
+ mantissa = ( sample >> ( exponent + 3 ) ) & 0x0F;
+ ulawbyte = ~ ( sign | ( exponent << 4 ) | mantissa );
+#ifdef ZEROTRAP
+ if (ulawbyte == 0)
+ ulawbyte = 0x02; /* optional CCITT trap */
+#endif
+
+ return(ulawbyte);
+}
+
+/*
+** This routine converts from ulaw to 16 bit linear.
+**
+** Craig Reese: IDA/Supercomputing Research Center
+** 29 September 1989
+**
+** References:
+** 1) CCITT Recommendation G.711 (very difficult to follow)
+** 2) MIL-STD-188-113,"Interoperability and Performance Standards
+** for Analog-to_Digital Conversion Techniques,"
+** 17 February 1987
+**
+** Input: 8 bit ulaw sample
+** Output: signed 16 bit linear sample
+*/
+
+static int ulaw_to_linear(unsigned char ulawbyte)
+{
+ static int exp_lut[8] = { 0, 132, 396, 924, 1980, 4092, 8316, 16764 };
+ int sign, exponent, mantissa, sample;
+
+ ulawbyte = ~ ulawbyte;
+ sign = ( ulawbyte & 0x80 );
+ exponent = ( ulawbyte >> 4 ) & 0x07;
+ mantissa = ulawbyte & 0x0F;
+ sample = exp_lut[exponent] + ( mantissa << ( exponent + 3 ) );
+ if (sign != 0)
+ sample = -sample;
+
+ return(sample);
+}
+#endif /* AUDIO_UNIX_NATIVE && !AUDIO_STREAMING_DSP */
+
+
+/* THE STUFF ABOVE IS ONLY USED BY THE SOUND SERVER CHILD PROCESS */
+/* ========================================================================= */
+/* THE STUFF BELOW IS ONLY USED BY THE MAIN PROCESS */
+
+#define CHUNK_ID_LEN 4 /* IFF style chunk id length */
+#define WAV_HEADER_SIZE 16 /* size of WAV file header */
+
+static void *Load_WAV(char *filename)
+{
+ SoundInfo *snd_info;
+#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;
+ int data_byte_len;
+ FILE *file;
+#endif
+
+ if (!audio.sound_available)
+ return NULL;
+
+#if 0
+ printf("loading WAV file '%s'\n", filename);
+#endif
+
+ snd_info = checked_calloc(sizeof(SoundInfo));
+
+#if defined(TARGET_SDL)
+
+ if ((snd_info->data_ptr = Mix_LoadWAV(filename)) == NULL)
+ {
+ Error(ERR_WARN, "cannot read sound file '%s'", filename);
+ free(snd_info);
+ 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)
+ {
+ Error(ERR_WARN, "cannot read sound file '%s'", filename);
+ free(snd_info);
+ return NULL;
+ }
+
+ snd_info->data_len = ((SAMPLE *)snd_info->data_ptr)->len;
+
+#else /* AUDIO_UNIX_NATIVE */
+
+ if ((file = fopen(filename, MODE_READ)) == NULL)
+ {
+ Error(ERR_WARN, "cannot open sound file '%s'", filename);
+ free(snd_info);
+ return NULL;
+ }
+
+ /* read chunk id "RIFF" */
+ getFileChunkLE(file, chunk_name, &chunk_size);
+ if (strcmp(chunk_name, "RIFF") != 0)
+ {
+ Error(ERR_WARN, "missing 'RIFF' chunk of sound file '%s'", filename);
+ fclose(file);
+ free(snd_info);
+ return NULL;
+ }
+
+ /* read "RIFF" type id "WAVE" */
+ getFileChunkLE(file, chunk_name, NULL);
+ if (strcmp(chunk_name, "WAVE") != 0)
+ {
+ Error(ERR_WARN, "missing 'WAVE' type ID of sound file '%s'", filename);
+ fclose(file);
+ free(snd_info);
+ return NULL;
+ }
+
+ while (getFileChunkLE(file, chunk_name, &chunk_size))
+ {
+ if (strcmp(chunk_name, "fmt ") == 0)
+ {
+ 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 &&
+ header.num_channels != 2)
+ {
+ 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)
+ {
+ data_byte_len = chunk_size;
+
+ snd_info->data_len = data_byte_len;
+ snd_info->data_ptr = checked_malloc(snd_info->data_len);
+
+ /* read sound data */
+ if (fread(snd_info->data_ptr, 1, snd_info->data_len, file) !=
+ snd_info->data_len)
+ {
+ Error(ERR_WARN,"cannot read 'data' chunk of sound file '%s'",filename);
+ fclose(file);
+ free(snd_info->data_ptr);
+ free(snd_info);
+ return NULL;
+ }
+
+ /* check for odd number of data bytes (data chunk is word aligned) */
+ if ((data_byte_len % 2) == 1)
+ ReadUnusedBytesFromFile(file, 1);
+ }
+ else /* unknown chunk -- ignore */
+ ReadUnusedBytesFromFile(file, chunk_size);
+ }
+
+ fclose(file);
+
+ if (snd_info->data_ptr == NULL)
+ {
+ Error(ERR_WARN, "missing 'data' chunk of sound file '%s'", filename);
+ free(snd_info);
+ return NULL;
+ }
+
+ 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 */
+ }
+
+ snd_info->num_channels = header.num_channels;
+ if (header.num_channels == 2)
+ snd_info->data_len /= 2; /* correct number of samples */
+
+#if 0
+ if (header.num_channels == 1) /* convert mono sound to stereo */
+ {
+ void *buffer_ptr = checked_malloc(data_byte_len * 2);
+ void *sample_ptr = snd_info->data_ptr;
+ int sample_size = snd_info->data_len;
+ int i;
+
+ if (snd_ctrl->format == AUDIO_FORMAT_U8)
+ for (i = 0; i < sample_size; i++)
+ *buffer_ptr++ =
+ ((short)(((byte *)sample_ptr)[i] ^ 0x80)) << 8;
+ else /* AUDIO_FORMAT_S16 */
+ for (i = 0; i < sample_size; i++)
+ *buffer_ptr++ =
+ ((short *)sample_ptr)[i];
+ }
+#endif
+
+#endif /* AUDIO_UNIX_NATIVE */
+
+ snd_info->type = SND_TYPE_WAV;
+ snd_info->source_filename = getStringCopy(filename);
+
+ return snd_info;
+}
+
+static void *Load_MOD(char *filename)
+{
+#if defined(TARGET_SDL)
+ MusicInfo *mod_info;
+
+ if (!audio.sound_available)
+ return NULL;
+
+ mod_info = checked_calloc(sizeof(MusicInfo));
+
+ if ((mod_info->data_ptr = Mix_LoadMUS(filename)) == NULL)
+ {
+ Error(ERR_WARN, "cannot read music file '%s'", filename);
+ free(mod_info);
+ return NULL;
+ }
+
+ mod_info->type = MUS_TYPE_MOD;
+ mod_info->source_filename = getStringCopy(filename);
+
+ return mod_info;
+#else
+ return NULL;
+#endif
+}
+
+static void *Load_WAV_or_MOD(char *filename)
+{
+ if (FileIsSound(filename))
+ return Load_WAV(filename);
+ else if (FileIsMusic(filename))
+ return Load_MOD(filename);
+ else
+ return NULL;
+}
+
+void LoadCustomMusic_NoConf(void)
+{
+ static boolean draw_init_text = TRUE; /* only draw at startup */
+ static char *last_music_directory = NULL;
+ char *music_directory = getCustomMusicDirectory();
+ DIR *dir;
+ struct dirent *dir_entry;
+ int num_music = getMusicListSize();
+
+ if (!audio.sound_available)
+ return;
+
+ if (last_music_directory != NULL &&
+ strcmp(last_music_directory, music_directory) == 0)
+ return; /* old and new music directory are the same */
+
+ if (last_music_directory != NULL)
+ free(last_music_directory);
+ last_music_directory = getStringCopy(music_directory);
+
+ FreeAllMusic_NoConf();
+
+ if ((dir = opendir(music_directory)) == NULL)
+ {
+ Error(ERR_WARN, "cannot read music directory '%s'", music_directory);
+ audio.music_available = FALSE;
+ 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 = NULL;
+ MusicInfo *mus_info = NULL;
+ boolean music_already_used = FALSE;
+ int i;
+
+ /* skip all music files that are configured in music config file */
+ for (i = 0; i < num_music; i++)
+ {
+ struct FileInfo *music = getMusicListEntry(i);
+
+ if (strcmp(basename, music->filename) == 0)
+ {
+ music_already_used = TRUE;
+ break;
+ }
+ }
+
+ if (music_already_used)
+ continue;
+
+#if 0
+ if (FileIsSound(basename) || FileIsMusic(basename))
+ printf("DEBUG: loading music '%s' ...\n", basename);
+#endif
+
+ if (draw_init_text)
+ DrawInitText(basename, 150, FC_YELLOW);
+
+ filename = getPath2(music_directory, basename);
+
+#if 1
+ if (FileIsMusic(basename))
+ mus_info = Load_WAV_or_MOD(filename);
+#else
+ if (FileIsSound(basename))
+ mus_info = Load_WAV(filename);
+ else if (FileIsMusic(basename))
+ mus_info = Load_MOD(filename);
+#endif
+
+ free(filename);
+
+ if (mus_info)
+ {
+ num_music_noconf++;
+ Music_NoConf = checked_realloc(Music_NoConf,
+ num_music_noconf * sizeof(MusicInfo *));
+ Music_NoConf[num_music_noconf - 1] = mus_info;
+ }
+ }
+
+ closedir(dir);
+
+ draw_init_text = FALSE;
+}
+
+int getSoundListSize()
+{
+ return (sound_info->num_file_list_entries +
+ sound_info->num_dynamic_file_list_entries);
+}
+
+int getMusicListSize()
+{
+ return (music_info->num_file_list_entries +
+ music_info->num_dynamic_file_list_entries);
+}
+
+struct FileInfo *getSoundListEntry(int pos)
+{
+ int num_list_entries = sound_info->num_file_list_entries;
+ int list_pos = (pos < num_list_entries ? pos : pos - num_list_entries);
+
+ return (pos < num_list_entries ? &sound_info->file_list[list_pos] :
+ &sound_info->dynamic_file_list[list_pos]);
+}
+
+struct FileInfo *getMusicListEntry(int pos)
+{
+ int num_list_entries = music_info->num_file_list_entries;
+ int list_pos = (pos < num_list_entries ? pos : pos - num_list_entries);
+
+ return (pos < num_list_entries ? &music_info->file_list[list_pos] :
+ &music_info->dynamic_file_list[list_pos]);
+}
+
+static SoundInfo *getSoundInfoEntryFromSoundID(int pos)
+{
+ int num_list_entries = sound_info->num_file_list_entries;
+ int list_pos = (pos < num_list_entries ? pos : pos - num_list_entries);
+ SoundInfo **snd_info =
+ (SoundInfo **)(pos < num_list_entries ? sound_info->artwork_list :
+ sound_info->dynamic_artwork_list);
+
+ return snd_info[list_pos];
+}
+
+static MusicInfo *getMusicInfoEntryFromMusicID(int pos)
+{
+ int num_list_entries = music_info->num_file_list_entries;
+ int list_pos = (pos < num_list_entries ? pos : pos - num_list_entries);
+ MusicInfo **mus_info =
+ (MusicInfo **)(pos < num_list_entries ? music_info->artwork_list :
+ music_info->dynamic_artwork_list);