PcmDevice *
openPcmDevice (int errorLevel, const char *device) {
#ifdef HAVE_HPUX_AUDIO
PcmDevice *pcm;
if ((pcm = malloc(sizeof(*pcm)))) {
long status;
AudioAttrMask mask = 0;
AudioAttributes attributes;
SSPlayParams parameters;
if (!audioServer) {
char *server = "";
audioServer = AOpenAudio(server, &status);
if (status != AENoError) {
logAudioError(errorLevel, status, "AOpenAudio");
audioServer = NULL;
goto noServer;
}
logMessage(LOG_DEBUG, "connected to audio server: %s", AAudioString(audioServer));
ASetCloseDownMode(audioServer, AKeepTransactions, &status);
if (status != AENoError) {
logAudioError(errorLevel, status, "ASetCloseDownMode");
}
}
memset(&attributes, 0, sizeof(attributes));
parameters.gain_matrix = *ASimplePlayer(audioServer);
parameters.play_volume = AUnityGain;
parameters.priority = APriorityUrgent;
parameters.event_mask = 0;
pcm->transaction = APlaySStream(audioServer, mask, &attributes, ¶meters, &pcm->stream, &status);
if (status == AENoError) {
if ((pcm->socket = socket(AF_INET, SOCK_STREAM, 0)) != -1) {
if (connect(pcm->socket, (struct sockaddr *)&pcm->stream.tcp_sockaddr, sizeof(pcm->stream.tcp_sockaddr)) != -1) {
return pcm;
} else {
logSystemError("PCM socket connection");
}
close(pcm->socket);
} else {
logSystemError("PCM socket creation");
}
} else {
logAudioError(errorLevel, status, "APlaySStream");
}
noServer:
free(pcm);
} else {
logSystemError("PCM device allocation");
}
#endif /* HAVE_HPUX_AUDIO */
return NULL;
}
// Initialize Sound Card ------------------------------------------------------
static int Sound_Init_SoundCard (void)
{
int i;
AUDIOINFO Audio_Infos;
ConsolePrintf (Msg_Get (MSG_Sound_Init_Soundcard), Sound.SampleRate);
ConsolePrint ("\n");
Audio_Infos.nDeviceId = Sound.SoundCard;
Audio_Infos.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO; // FIXME: Stereo ?
Audio_Infos.nSampleRate = audio_sample_rate = Sound.SampleRate;
if (AOpenAudio(&Audio_Infos) != AUDIO_ERROR_NONE)
{
Quit_Msg ("%s", Msg_Get (MSG_Sound_Init_Error_Audio));
return (MEKA_ERR_FAIL);
}
// FIXME: original sound engine was trying different sample rate on failure
// Unused
// Maybe it was intended to check out number of channels there ?
// AGetAudioCaps (Audio_Infos.nDeviceId, &Audio_Caps);
// Open voices
if (AOpenVoices(Sound.Voices_Max) != AUDIO_ERROR_NONE)
{
Quit_Msg ("%s", Msg_Get (MSG_Sound_Init_Error_Voices));
return (MEKA_ERR_FAIL);
}
ASetAudioMixerValue (AUDIO_MIXER_MASTER_VOLUME, 256);
// Allocate voices and waveforms
Sound.Voices = Memory_Alloc (sizeof (t_voice) * Sound.Voices_Max);
for (i = 0; i < Sound.Voices_Max; i++)
{
if (ACreateAudioVoice(&Sound.Voices[i].hVoice) != AUDIO_ERROR_NONE)
{
Quit_Msg (Msg_Get (MSG_Sound_Init_Error_Voice_N), i);
return (MEKA_ERR_FAIL);
}
ASetVoicePanning(Sound.Voices[i].hVoice, 128); // Center voice
Sound.Voices[i].lpWave = NULL;
Sound.Voices[i].playing = FALSE;
}
// FIXME: is this needed ?
AUpdateAudio ();
// FIXME: is this needed ?
// Check frame sample rate
audio_sample_rate = nominal_sample_rate = Audio_Infos.nSampleRate;
return (MEKA_ERR_OK);
}
int main(void)
{
AUDIOINFO info;
AUDIOCAPS caps;
UINT rc, nDevId;
/* initialize audio library */
AInitialize();
/* show registered device drivers */
printf("List of registered devices:\n");
for (nDevId = 0; nDevId < AGetAudioNumDevs(); nDevId++) {
AGetAudioDevCaps(nDevId, &caps);
printf(" %2d. %s\n", nDevId, caps.szProductName);
}
printf("\n");
/*
* NOTE: Here we can use any of the above devices, or we can
* use the virtual device AUDIO_DEVICE_MAPPER for detection.
*/
/* open audio device */
info.nDeviceId = AUDIO_DEVICE_MAPPER;
info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO;
info.nSampleRate = 44100;
if ((rc = AOpenAudio(&info)) != AUDIO_ERROR_NONE) {
CHAR szText[80];
AGetErrorText(rc, szText, sizeof(szText) - 1);
printf("ERROR: %s\n", szText);
exit(1);
}
/*
* NOTE: Since the audio device may not support the playback
* format and sampling frequency, the audio system uses the
* closest configuration which is then returned to the user
* in the AUDIOINFO structure.
*
*/
/* print information */
AGetAudioDevCaps(info.nDeviceId, &caps);
printf("%s at %d-bit %s %u Hz detected\n",
caps.szProductName,
info.wFormat & AUDIO_FORMAT_16BITS ? 16 : 8,
info.wFormat & AUDIO_FORMAT_STEREO ? "stereo" : "mono",
info.nSampleRate);
/* close audio device */
ACloseAudio();
return 0;
}
int main(void)
{
AUDIOINFO info;
LPAUDIOMODULE lpModule;
LPAUDIOWAVE lpWave;
HAC hVoice;
BOOL stopped;
/* initialize audio library */
AInitialize();
/* open audio device */
info.nDeviceId = AUDIO_DEVICE_MAPPER;
info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO;
info.nSampleRate = 44100;
AOpenAudio(&info);
/* load module and waveform file */
ALoadModuleFile("test.s3m", &lpModule, 0);
ALoadWaveFile("test.wav", &lpWave, 0);
/* open voices for module and waveform */
AOpenVoices(lpModule->nTracks + 1);
/* play the module file */
APlayModule(lpModule);
ASetModuleVolume(64);
/* play the waveform through a voice */
ACreateAudioVoice(&hVoice);
APlayVoice(hVoice, lpWave);
ASetVoiceVolume(hVoice, 48);
ASetVoicePanning(hVoice, 128);
/* program main execution loop */
printf("Playing module and waveform, press any key to stop.\n");
while (!kbhit()) {
/* update audio system */
AUpdateAudio();
/* restart waveform if stopped */
AGetVoiceStatus(hVoice, &stopped);
if (stopped) APlayVoice(hVoice, lpWave);
/* check if the module is stopped */
AGetModuleStatus(&stopped);
if (stopped) break;
}
/* stop playing the waveform */
AStopVoice(hVoice);
ADestroyAudioVoice(hVoice);
/* stop playing the module */
AStopModule();
ACloseVoices();
/* release the waveform & module */
AFreeWaveFile(lpWave);
AFreeModuleFile(lpModule);
/* close audio device */
ACloseAudio();
return 0;
}
int msdos_init_sound(int *rate, int card)
{
int i;
seal_sample_rate = *rate;
seal_sound_card = card;
if (AInitialize() != AUDIO_ERROR_NONE)
return 1;
/* Ask the user if no sound card was chosen */
if (seal_sound_card == -1)
{
unsigned int k;
printf("\n SELECT YOUR AUDIO DEVICE :\n\n"
" AWE32/64 playback requires onboard DRAM,\n"
" Sound Blaster playback is the most compatible & better for emulation\n\n");
for (k = 0;k < AGetAudioNumDevs();k++)
{
if (AGetAudioDevCaps(k,&caps) == AUDIO_ERROR_NONE)
printf(" %2d. %s\n",k,caps.szProductName);
}
printf("\n");
if (k < 10)
{
i = getch();
seal_sound_card = i - '0';
}
else
scanf("%d",&seal_sound_card);
}
/* initialize SEAL audio library */
if (seal_sound_card == 0) /* silence */
{
/* update the Machine structure to show that sound is disabled */
seal_sample_rate = 0;
exit(0);
return 0;
}
/* open audio device */
/* info.nDeviceId = AUDIO_DEVICE_MAPPER;*/
info.nDeviceId = seal_sound_card;
/* always use 16 bit mixing if possible - better quality and same speed of 8 bit */
info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO | AUDIO_FORMAT_RAW_SAMPLE;
info.nSampleRate = seal_sample_rate;
if (AOpenAudio(&info) != AUDIO_ERROR_NONE)
{
return (1);
}
AGetAudioDevCaps(info.nDeviceId,&caps);
printf("Using `%s' at %d-bit %s %u Hz\n",
caps.szProductName,
info.wFormat & AUDIO_FORMAT_16BITS ? 16 : 8,
info.wFormat & AUDIO_FORMAT_STEREO ? "stereo" : "mono",
info.nSampleRate);
/* open and allocate voices, allocate waveforms */
if (AOpenVoices(NUMVOICES) != AUDIO_ERROR_NONE)
{
printf("voices initialization failed\n");
return 1;
}
for (i = 0; i < NUMVOICES; i++)
{
if (ACreateAudioVoice(&hVoice[i]) != AUDIO_ERROR_NONE)
{
printf("voice #%d creation failed\n",i);
return 1;
}
ASetVoicePanning(hVoice[i],128);
lpWave[i] = 0;
}
/* update the Machine structure to reflect the actual sample rate */
*rate = seal_sample_rate = info.nSampleRate;
{
uclock_t a,b;
LONG start,end;
if ((lpWave[0] = (LPAUDIOWAVE)malloc(sizeof(AUDIOWAVE))) == 0)
return 1;
lpWave[0]->wFormat = AUDIO_FORMAT_8BITS | AUDIO_FORMAT_MONO;
lpWave[0]->nSampleRate = seal_sample_rate;
lpWave[0]->dwLength = 3*seal_sample_rate;
lpWave[0]->dwLoopStart = 0;
lpWave[0]->dwLoopEnd = 3*seal_sample_rate;
if (ACreateAudioData(lpWave[0]) != AUDIO_ERROR_NONE)
{
free(lpWave[0]);
lpWave[0] = 0;
return 1;
}
memset(lpWave[0]->lpData,0,3*seal_sample_rate);
/* upload the data to the audio DRAM local memory */
AWriteAudioData(lpWave[0],0,3*seal_sample_rate);
APrimeVoice(hVoice[0],lpWave[0]);
ASetVoiceFrequency(hVoice[0],seal_sample_rate);
ASetVoiceVolume(hVoice[0],0);
AStartVoice(hVoice[0]);
a = uclock();
/* wait some time to let everything stabilize */
do
{
osd_update_audio();
b = uclock();
} while (b-a < UCLOCKS_PER_SEC/10);
a = uclock();
AGetVoicePosition(hVoice[0],&start);
do
{
osd_update_audio();
b = uclock();
} while (b-a < UCLOCKS_PER_SEC);
AGetVoicePosition(hVoice[0],&end);
nominal_sample_rate = seal_sample_rate;
seal_sample_rate = end - start;
AStopVoice(hVoice[0]);
ADestroyAudioData(lpWave[0]);
free(lpWave[0]);
lpWave[0] = 0;
}
osd_set_mastervolume(0); /* start at maximum volume */
return 0;
}
/* return on error leaves stuff dirty here... */
static int open_alib(audio_output_t *ao)
{
AudioAttributes Attribs;
AudioAttrMask AttribsMask;
AGainEntry gainEntry[4];
SSPlayParams playParams;
SStream audioStream;
AErrorHandler prevHandler;
char server[1];
int i;
long status;
if (audioServer) {
error("openAudio: audio already open");
return -1;
}
prevHandler = ASetErrorHandler(myHandler);
server[0] = '\0';
audioServer = AOpenAudio( server, NULL );
if (audioServer==NULL) {
error("Error: could not open audio\n");
return -1;
}
ao->fn = socket( AF_INET, SOCK_STREAM, 0 );
if(ao->fn<0) {
error("Socket creation failed");
return -1;
}
Attribs.type = ATSampled;
Attribs.attr.sampled_attr.sampling_rate = ao->rate;
Attribs.attr.sampled_attr.channels = ao->channels;
Attribs.attr.sampled_attr.data_format = ADFLin16;
AttribsMask = ASSamplingRateMask | ASChannelsMask | ASDataFormatMask;
gainEntry[0].gain = AUnityGain;
gainEntry[0].u.o.out_ch = AOCTMono;
gainEntry[0].u.o.out_dst = AODTDefaultOutput;
playParams.gain_matrix.type = AGMTOutput; /* gain matrix */
playParams.gain_matrix.num_entries = 1;
playParams.gain_matrix.gain_entries = gainEntry;
playParams.play_volume = AUnityGain; /* play volume */
playParams.priority = APriorityNormal; /* normal priority */
playParams.event_mask = 0; /* don't solicit any events */
xid=APlaySStream(audioServer,AttribsMask,&Attribs,
&playParams,&audioStream,NULL);
status=connect(ao->fn,
(struct sockaddr *) &audioStream.tcp_sockaddr,
sizeof(struct sockaddr_in) );
if (status<0) {
error("Connect failed");
return -1;
}
i=-1;
tcpProtocolEntry=getprotobyname("tcp");
setsockopt(ao->fn,tcpProtocolEntry->p_proto,TCP_NODELAY,&i,sizeof(i));
return ao->fn;
}
int main(void)
{
AUDIOINFO info;
LPAUDIOWAVE lpWave;
HAC hVoice[NUMVOICES];
BOOL stopped;
UINT n, m;
/* initialize audio library */
AInitialize();
/* open audio device */
info.nDeviceId = AUDIO_DEVICE_MAPPER;
info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO;
info.nSampleRate = 44100;
AOpenAudio(&info);
/* load waveform file */
ALoadWaveFile("test.wav", &lpWave, 0);
/* open and allocate voices */
AOpenVoices(NUMVOICES);
for (n = 0; n < NUMVOICES; n++) {
ACreateAudioVoice(&hVoice[n]);
ASetVoiceVolume(hVoice[n], 64);
ASetVoicePanning(hVoice[n], n & 1 ? 0 : 255);
}
/* program main execution loop */
printf("Playing waveform, press any key to stop.\n");
for (n = m = 0; !kbhit() && n < 48 - 7; n++) {
/* play chord C-E-G */
APlayVoice(hVoice[m+0], lpWave);
APlayVoice(hVoice[m+1], lpWave);
APlayVoice(hVoice[m+2], lpWave);
ASetVoiceFrequency(hVoice[m+0], FREQ(aPeriodTable[n+0]));
ASetVoiceFrequency(hVoice[m+1], FREQ(aPeriodTable[n+4]));
ASetVoiceFrequency(hVoice[m+2], FREQ(aPeriodTable[n+7]));
m = (m + 3) % NUMVOICES;
/* wait until note finishes */
do {
/* update audio system */
AUpdateAudio();
AGetVoiceStatus(hVoice[0], &stopped);
} while (!stopped);
}
/* stop and release voices */
for (n = 0; n < NUMVOICES; n++) {
AStopVoice(hVoice[n]);
ADestroyAudioVoice(hVoice[n]);
}
ACloseVoices();
/* release the waveform file */
AFreeWaveFile(lpWave);
/* close audio device */
ACloseAudio();
return 0;
}
int msdos_init_sound(void)
{
#ifdef USE_SEAL
int i;
/* Ask the user if no soundcard was chosen */
if (soundcard == -1)
{
unsigned int k;
printf("\nSelect the audio device:\n");
for (k = 0;k < AGetAudioNumDevs();k++)
{
/* don't show the AWE32, it's too slow, users must choose Sound Blaster */
if (AGetAudioDevCaps(k,&caps) == AUDIO_ERROR_NONE &&
strcmp(caps.szProductName,"Sound Blaster AWE32"))
printf(" %2d. %s\n",k,caps.szProductName);
}
printf("\n");
if (k < 10)
{
i = getch();
soundcard = i - '0';
}
else
scanf("%d",&soundcard);
}
/* initialize SEAL audio library */
if (soundcard == 0) /* silence */
{
/* update the Machine structure to show that sound is disabled */
Machine->sample_rate = 0;
return 0;
}
/* open audio device */
/* info.nDeviceId = AUDIO_DEVICE_MAPPER;*/
info.nDeviceId = soundcard;
/* always use 16 bit mixing if possible - better quality and same speed of 8 bit */
info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_MONO | AUDIO_FORMAT_RAW_SAMPLE;
/* use stereo output if supported */
if (usestereo)
{
if (Machine->drv->sound_attributes & SOUND_SUPPORTS_STEREO)
info.wFormat = AUDIO_FORMAT_16BITS | AUDIO_FORMAT_STEREO | AUDIO_FORMAT_RAW_SAMPLE;
}
info.nSampleRate = Machine->sample_rate;
if (AOpenAudio(&info) != AUDIO_ERROR_NONE)
{
printf("audio initialization failed\n");
return 1;
}
AGetAudioDevCaps(info.nDeviceId,&caps);
logerror("Using %s at %d-bit %s %u Hz\n",
caps.szProductName,
info.wFormat & AUDIO_FORMAT_16BITS ? 16 : 8,
info.wFormat & AUDIO_FORMAT_STEREO ? "stereo" : "mono",
info.nSampleRate);
/* open and allocate voices, allocate waveforms */
if (AOpenVoices(SOUND_CHANNELS) != AUDIO_ERROR_NONE)
{
printf("voices initialization failed\n");
return 1;
}
for (i = 0; i < SOUND_CHANNELS; i++)
{
lpWave[i] = 0;
}
stream_playing = 0;
stream_cache_data = 0;
stream_cache_len = 0;
stream_cache_stereo = 0;
/* update the Machine structure to reflect the actual sample rate */
Machine->sample_rate = info.nSampleRate;
logerror("set sample rate: %d\n",Machine->sample_rate);
if (sampleratedetect)
{
TICKER a,b;
LONG start,end;
if (ACreateAudioVoice(&hVoice[0]) != AUDIO_ERROR_NONE)
return 1;
if ((lpWave[0] = (LPAUDIOWAVE)malloc(sizeof(AUDIOWAVE))) == 0)
{
ADestroyAudioVoice(hVoice[0]);
return 1;
}
lpWave[0]->wFormat = AUDIO_FORMAT_8BITS | AUDIO_FORMAT_MONO;
lpWave[0]->nSampleRate = Machine->sample_rate;
lpWave[0]->dwLength = 3*Machine->sample_rate;
lpWave[0]->dwLoopStart = 0;
lpWave[0]->dwLoopEnd = 3*Machine->sample_rate;
if (ACreateAudioData(lpWave[0]) != AUDIO_ERROR_NONE)
{
free(lpWave[0]);
lpWave[0] = 0;
return 1;
}
memset(lpWave[0]->lpData,0,3*Machine->sample_rate);
/* upload the data to the audio DRAM local memory */
AWriteAudioData(lpWave[0],0,3*Machine->sample_rate);
APrimeVoice(hVoice[0],lpWave[0]);
ASetVoiceFrequency(hVoice[0],Machine->sample_rate);
ASetVoiceVolume(hVoice[0],0);
AStartVoice(hVoice[0]);
a = ticker();
/* wait some time to let everything stabilize */
do
{
AUpdateAudioEx(Machine->sample_rate / Machine->drv->frames_per_second);
b = ticker();
} while (b-a < TICKS_PER_SEC/10);
a = ticker();
AGetVoicePosition(hVoice[0],&start);
do
{
AUpdateAudioEx(Machine->sample_rate / Machine->drv->frames_per_second);
b = ticker();
} while (b-a < TICKS_PER_SEC);
AGetVoicePosition(hVoice[0],&end);
nominal_sample_rate = Machine->sample_rate;
Machine->sample_rate = end - start;
logerror("actual sample rate: %d\n",Machine->sample_rate);
AStopVoice(hVoice[0]);
ADestroyAudioData(lpWave[0]);
free(lpWave[0]);
lpWave[0] = 0;
ADestroyAudioVoice(hVoice[0]);
}
else
nominal_sample_rate = Machine->sample_rate;
#if 0
{
char *blaster_env;
/* Get Soundblaster base address from environment variabler BLASTER */
/* Soundblaster OPL base port, at some compatibles this must be 0x388 */
if(!getenv("BLASTER"))
{
printf("\nBLASTER variable not found, disabling fm sound!\n");
No_OPL = options.no_fm = 1;
}
else
{
blaster_env = getenv("BLASTER");
BaseSb = i = 0;
while ((blaster_env[i] & 0x5f) != 0x41) i++; /* Look for 'A' char */
while (blaster_env[++i] != 0x20) {
BaseSb = (BaseSb << 4) + (blaster_env[i]-0x30);
}
}
}
#endif
#endif
#ifdef USE_ALLEGRO
reserve_voices(1,0);
if (install_sound(DIGI_AUTODETECT,MIDI_NONE,0) != 0)
{
logerror("Allegro install_sound error: %s\n",allegro_error);
return 1;
}
nominal_sample_rate = Machine->sample_rate;
#endif
num_used_opl = 0;
osd_set_mastervolume(attenuation); /* set the startup volume */
return 0;
}
