static int
MINTSTFA_CheckAudio(_THIS)
{
int i;
DEBUG_PRINT((DEBUG_NAME "asked: %d bits, ",
SDL_AUDIO_BITSIZE(this->spec.format)));
DEBUG_PRINT(("float=%d, ", SDL_AUDIO_ISFLOAT(this->spec.format)));
DEBUG_PRINT(("signed=%d, ", SDL_AUDIO_ISSIGNED(this->spec.format)));
DEBUG_PRINT(("big endian=%d, ",
SDL_AUDIO_ISBIGENDIAN(this->spec.format)));
DEBUG_PRINT(("channels=%d, ", this->spec.channels));
DEBUG_PRINT(("freq=%d\n", this->spec.freq));
if (SDL_AUDIO_BITSIZE(this->spec.format) > 16) {
this->spec.format = AUDIO_S16SYS; /* clamp out int32/float32 ... */
}
if (this->spec.channels > 2) {
this->spec.channels = 2; /* no more than stereo! */
}
/* Check formats available */
MINTAUDIO_freqcount = 0;
for (i = 0; i < 16; i++) {
SDL_MintAudio_AddFrequency(this, freqs[i], 0, i, -1);
}
#if 1
for (i = 0; i < MINTAUDIO_freqcount; i++) {
DEBUG_PRINT((DEBUG_NAME "freq %d: %lu Hz, clock %lu, prediv %d\n",
i, MINTAUDIO_frequencies[i].frequency,
MINTAUDIO_frequencies[i].masterclock,
MINTAUDIO_frequencies[i].predivisor));
}
#endif
MINTAUDIO_numfreq = SDL_MintAudio_SearchFrequency(this, this->spec.freq);
this->spec.freq = MINTAUDIO_frequencies[MINTAUDIO_numfreq].frequency;
DEBUG_PRINT((DEBUG_NAME "obtained: %d bits, ",
SDL_AUDIO_BITSIZE(this->spec.format)));
DEBUG_PRINT(("float=%d, ", SDL_AUDIO_ISFLOAT(this->spec.format)));
DEBUG_PRINT(("signed=%d, ", SDL_AUDIO_ISSIGNED(this->spec.format)));
DEBUG_PRINT(("big endian=%d, ",
SDL_AUDIO_ISBIGENDIAN(this->spec.format)));
DEBUG_PRINT(("channels=%d, ", this->spec.channels));
DEBUG_PRINT(("freq=%d\n", this->spec.freq));
return 0;
}
static OSStatus
inputCallback(void *inRefCon,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames,
AudioBufferList * ioData)
{
OSStatus status;
SDL_AudioDevice *this = (SDL_AudioDevice *) inRefCon;
UInt32 remaining, len, frameSize;
AudioBuffer abuf;
AudioBufferList buflist;
void *ptr;
abuf.mNumberChannels = this->spec.channels;
buflist.mNumberBuffers = 1;
frameSize = (SDL_AUDIO_BITSIZE(this->spec.format) / 8) * this->spec.channels;
remaining = inNumberFrames * frameSize;
ptr = this->hidden->buffer;
while (remaining > 0) {
/* render input audio into buffer */
len = this->hidden->bufferSize - this->hidden->bufferOffset;
if (len > remaining) {
len = remaining;
}
ptr = (char*)this->hidden->buffer + this->hidden->bufferOffset;
abuf.mDataByteSize = len;
abuf.mData = ptr;
buflist.mBuffers[0] = abuf;
status = AudioUnitRender(this->hidden->audioUnit, ioActionFlags, inTimeStamp, inBusNumber,
len / frameSize, &buflist);
if (status) {
SDL_SetError("CoreAudio error (%s): %d", "AudioUnitRender", (int) status); \
return status;
}
remaining -= len;
this->hidden->bufferOffset += len;
if (this->hidden->bufferOffset >= this->hidden->bufferSize) {
/* The buffer is full, send it to the user callback */
SDL_LockMutex(this->mixer_lock);
(*this->spec.callback)(this->spec.userdata,
this->hidden->buffer, this->hidden->bufferSize);
SDL_UnlockMutex(this->mixer_lock);
this->hidden->bufferOffset = 0;
}
}
return noErr;
}
void SDL_CalculateAudioSpec(SDL_AudioSpec * spec)
{
switch (spec->format) {
case AUDIO_U8:
spec->silence = 0x80;
break;
default:
spec->silence = 0x00;
break;
}
spec->size = SDL_AUDIO_BITSIZE(spec->format) / 8;
spec->size *= spec->channels;
spec->size *= spec->samples;
}
/*
* Called right before feeding this->hidden->mixbuf to the hardware. Swizzle
* channels from Windows/Mac order to the format alsalib will want.
*/
static SDL_INLINE void
swizzle_alsa_channels(_THIS, void *buffer, Uint32 bufferlen)
{
if (this->spec.channels == 6) {
switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
case 8: swizzle_alsa_channels_6_8bit(buffer, bufferlen); break;
case 16: swizzle_alsa_channels_6_16bit(buffer, bufferlen); break;
case 32: swizzle_alsa_channels_6_32bit(buffer, bufferlen); break;
case 64: swizzle_alsa_channels_6_64bit(buffer, bufferlen); break;
default: SDL_assert(!"unhandled bitsize"); break;
}
}
/* !!! FIXME: update this for 7.1 if needed, later. */
}
static void
ALSA_PlayDevice(_THIS)
{
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
swizzle_alsa_channels(this, this->hidden->mixbuf, frames_left);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
int status;
/* This wait is a work-around for a hang when USB devices are
unplugged. Normally it should not result in any waiting,
but in the case of a USB unplug, it serves as a way to
join the playback thread after the timeout occurs */
status = ALSA_snd_pcm_wait(this->hidden->pcm_handle, 1000);
if (status == 0) {
/*fprintf(stderr, "ALSA timeout waiting for available buffer space\n");*/
SDL_OpenedAudioDeviceDisconnected(this);
return;
}
status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
SDL_OpenedAudioDeviceDisconnected(this);
return;
}
continue;
}
sample_buf += status * frame_size;
frames_left -= status;
}
}
ALenum GetALFormat(const SDL_AudioSpec &spec)
{
int bits = SDL_AUDIO_BITSIZE(spec.format);
int chan = spec.channels;
if (bits == 8)
{
if (chan == 1) return AL_FORMAT_MONO8;
if (chan == 2) return AL_FORMAT_STEREO8;
}
else if (bits == 16)
{
if (chan == 1) return AL_FORMAT_MONO16;
if (chan == 2) return AL_FORMAT_STEREO16;
}
return 0;
}
static void
MINTSTFA_InitAudio(_THIS)
{
void *buffer = SDL_MintAudio_audiobuf[SDL_MintAudio_numbuf];
void *oldpile = (void *) Super(0);
/* Stop replay */
cookie_stfa->sound_enable = STFA_PLAY_DISABLE;
/* Select replay format */
cookie_stfa->sound_control =
MINTAUDIO_frequencies[MINTAUDIO_numfreq].predivisor;
if (SDL_AUDIO_BITSIZE(this->spec.format) == 8) {
cookie_stfa->sound_control |= STFA_FORMAT_8BIT;
} else {
cookie_stfa->sound_control |= STFA_FORMAT_16BIT;
}
if (this->spec.channels == 2) {
cookie_stfa->sound_control |= STFA_FORMAT_STEREO;
} else {
cookie_stfa->sound_control |= STFA_FORMAT_MONO;
}
if (SDL_AUDIO_ISSIGNED(this->spec.format) != 0) {
cookie_stfa->sound_control |= STFA_FORMAT_SIGNED;
} else {
cookie_stfa->sound_control |= STFA_FORMAT_UNSIGNED;
}
if (SDL_AUDIO_ISBIGENDIAN(this->spec.format) != 0) {
cookie_stfa->sound_control |= STFA_FORMAT_BIGENDIAN;
} else {
cookie_stfa->sound_control |= STFA_FORMAT_LITENDIAN;
}
/* Set buffer */
cookie_stfa->sound_start = (unsigned long) buffer;
cookie_stfa->sound_end = (unsigned long) (buffer + this->spec.size);
/* Set interrupt */
cookie_stfa->stfa_it = SDL_MintAudio_StfaInterrupt;
/* Restart replay */
cookie_stfa->sound_enable = STFA_PLAY_ENABLE | STFA_PLAY_REPEAT;
Super(oldpile);
DEBUG_PRINT((DEBUG_NAME "hardware initialized\n"));
}
void *TIMIDITY_CreateFromRW(SDL_RWops *src, int freesrc)
{
TIMIDITY_Music *music;
SDL_AudioSpec spec;
SDL_bool need_stream = SDL_FALSE;
music = (TIMIDITY_Music *)SDL_calloc(1, sizeof(*music));
if (!music) {
SDL_OutOfMemory();
return NULL;
}
SDL_memcpy(&spec, &music_spec, sizeof(spec));
if (spec.channels > 2) {
need_stream = SDL_TRUE;
spec.channels = 2;
}
music->song = Timidity_LoadSong(src, &spec);
if (!music->song) {
TIMIDITY_Delete(music);
return NULL;
}
if (need_stream) {
music->stream = SDL_NewAudioStream(spec.format, spec.channels, spec.freq,
music_spec.format, music_spec.channels, music_spec.freq);
if (!music->stream) {
TIMIDITY_Delete(music);
return NULL;
}
music->buffer_size = spec.samples * (SDL_AUDIO_BITSIZE(spec.format) / 8) * spec.channels;
music->buffer = SDL_malloc(music->buffer_size);
if (!music->buffer) {
SDL_OutOfMemory();
TIMIDITY_Delete(music);
return NULL;
}
}
if (freesrc) {
SDL_RWclose(src);
}
return music;
}
static int
ALSA_CaptureFromDevice(_THIS, void *buffer, int buflen)
{
Uint8 *sample_buf = (Uint8 *) buffer;
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
const int total_frames = buflen / frame_size;
snd_pcm_uframes_t frames_left = total_frames;
SDL_assert((buflen % frame_size) == 0);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
int status = ALSA_snd_pcm_readi(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
/*printf("ALSA: capture error %d\n", status);*/
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA read failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
return -1;
}
continue;
}
/*printf("ALSA: captured %d bytes\n", status * frame_size);*/
sample_buf += status * frame_size;
frames_left -= status;
}
swizzle_alsa_channels(this, buffer, total_frames - frames_left);
return (total_frames - frames_left) * frame_size;
}
static void
NETBSDAUDIO_FlushCapture(_THIS)
{
audio_info_t info;
size_t remain;
Uint8 buf[512];
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
return; /* oh well. */
}
remain = (size_t) (info.record.samples * (SDL_AUDIO_BITSIZE(this->spec.format) / 8));
while (remain > 0) {
const size_t len = SDL_min(sizeof (buf), remain);
const int br = read(this->hidden->audio_fd, buf, len);
if (br <= 0) {
return; /* oh well. */
}
remain -= br;
}
}
/**
** Mix into buffer.
**
** @param buffer Buffer to be filled with samples. Buffer must be big enough.
** @param samples Number of samples.
*/
static void MixIntoBuffer(void *buffer, int samples)
{
// FIXME: can save the memset here, if first channel sets the values
memset(Audio.MixerBuffer, 0, samples * sizeof(*Audio.MixerBuffer));
if (EffectsEnabled) {
// Add channels to mixer buffer
MixChannelsToStereo32(Audio.MixerBuffer, samples);
}
if (MusicEnabled) {
// Add music to mixer buffer
MixMusicToStereo32(Audio.MixerBuffer, samples);
}
ClipMixToStereo16(Audio.MixerBuffer, samples, (short *)buffer);
#ifdef USE_OAML
if (enableOAML && oaml) {
oaml->SetAudioFormat(Audio.Format.freq, Audio.Format.channels, SDL_AUDIO_BITSIZE(Audio.Format.format) / 8);
oaml->MixToBuffer(buffer, samples);
}
#endif
}
static SDL_bool
PrepWaveFormat(_THIS, UINT devId, WAVEFORMATEX *pfmt, const int iscapture)
{
SDL_zerop(pfmt);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
pfmt->wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
pfmt->wFormatTag = WAVE_FORMAT_PCM;
}
pfmt->wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
pfmt->nChannels = this->spec.channels;
pfmt->nSamplesPerSec = this->spec.freq;
pfmt->nBlockAlign = pfmt->nChannels * (pfmt->wBitsPerSample / 8);
pfmt->nAvgBytesPerSec = pfmt->nSamplesPerSec * pfmt->nBlockAlign;
if (iscapture) {
return (waveInOpen(0, devId, pfmt, 0, 0, WAVE_FORMAT_QUERY) == 0);
} else {
return (waveOutOpen(0, devId, pfmt, 0, 0, WAVE_FORMAT_QUERY) == 0);
}
}
static void
ALSA_PlayDevice(_THIS)
{
const Uint8 *sample_buf = (const Uint8 *) this->hidden->mixbuf;
const int frame_size = (((int) SDL_AUDIO_BITSIZE(this->spec.format)) / 8) *
this->spec.channels;
snd_pcm_uframes_t frames_left = ((snd_pcm_uframes_t) this->spec.samples);
swizzle_alsa_channels(this, this->hidden->mixbuf, frames_left);
while ( frames_left > 0 && SDL_AtomicGet(&this->enabled) ) {
/* !!! FIXME: This works, but needs more testing before going live */
/* ALSA_snd_pcm_wait(this->hidden->pcm_handle, -1); */
int status = ALSA_snd_pcm_writei(this->hidden->pcm_handle,
sample_buf, frames_left);
if (status < 0) {
if (status == -EAGAIN) {
/* Apparently snd_pcm_recover() doesn't handle this case -
does it assume snd_pcm_wait() above? */
SDL_Delay(1);
continue;
}
status = ALSA_snd_pcm_recover(this->hidden->pcm_handle, status, 0);
if (status < 0) {
/* Hmm, not much we can do - abort */
fprintf(stderr, "ALSA write failed (unrecoverable): %s\n",
ALSA_snd_strerror(status));
SDL_OpenedAudioDeviceDisconnected(this);
return;
}
continue;
}
sample_buf += status * frame_size;
frames_left -= status;
}
}
static int
SNDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
struct sio_par par;
int status;
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(*this->hidden));
this->hidden->mixlen = this->spec.size;
/* !!! FIXME: SIO_DEVANY can be a specific device... */
if ((this->hidden->dev = SNDIO_sio_open(SIO_DEVANY, SIO_PLAY, 0)) == NULL) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_open() failed");
}
SNDIO_sio_initpar(&par);
par.rate = this->spec.freq;
par.pchan = this->spec.channels;
par.round = this->spec.samples;
par.appbufsz = par.round * 2;
/* Try for a closest match on audio format */
status = -1;
while (test_format && (status < 0)) {
if (!SDL_AUDIO_ISFLOAT(test_format)) {
par.le = SDL_AUDIO_ISLITTLEENDIAN(test_format) ? 1 : 0;
par.sig = SDL_AUDIO_ISSIGNED(test_format) ? 1 : 0;
par.bits = SDL_AUDIO_BITSIZE(test_format);
if (SNDIO_sio_setpar(this->hidden->dev, &par) == 0) {
continue;
}
if (SNDIO_sio_getpar(this->hidden->dev, &par) == 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sio_getpar() failed");
}
if (par.bps != SIO_BPS(par.bits)) {
continue;
}
if ((par.bits == 8 * par.bps) || (par.msb)) {
status = 0;
break;
}
}
test_format = SDL_NextAudioFormat();
}
if (status < 0) {
SNDIO_CloseDevice(this);
return SDL_SetError("sndio: Couldn't find any hardware audio formats");
}
if ((par.bps == 4) && (par.sig) && (par.le))
this->spec.format = AUDIO_S32LSB;
else if ((par.bps == 4) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S32MSB;
else if ((par.bps == 2) && (par.sig) && (par.le))
this->spec.format = AUDIO_S16LSB;
else if ((par.bps == 2) && (par.sig) && (!par.le))
this->spec.format = AUDIO_S16MSB;
else if ((par.bps == 2) && (!par.sig) && (par.le))
this->spec.format = AUDIO_U16LSB;
else if ((par.bps == 2) && (!par.sig) && (!par.le))
this->spec.format = AUDIO_U16MSB;
else if ((par.bps == 1) && (par.sig))
this->spec.format = AUDIO_S8;
else if ((par.bps == 1) && (!par.sig))
this->spec.format = AUDIO_U8;
else {
SNDIO_CloseDevice(this);
return SDL_SetError("sndio: Got unsupported hardware audio format.");
}
this->spec.freq = par.rate;
this->spec.channels = par.pchan;
this->spec.samples = par.round;
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
SNDIO_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->hidden->mixlen);
if (!SNDIO_sio_start(this->hidden->dev)) {
return SDL_SetError("sio_start() failed");
}
/* We're ready to rock and roll. :-) */
return 0;
}
static int
XAUDIO2_OpenDevice(_THIS, const char *devname, int iscapture)
{
HRESULT result = S_OK;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
IXAudio2 *ixa2 = NULL;
IXAudio2SourceVoice *source = NULL;
UINT32 devId = 0; /* 0 == system default device. */
static IXAudio2VoiceCallbackVtbl callbacks_vtable = {
VoiceCBOnVoiceProcessPassStart,
VoiceCBOnVoiceProcessPassEnd,
VoiceCBOnStreamEnd,
VoiceCBOnBufferStart,
VoiceCBOnBufferEnd,
VoiceCBOnLoopEnd,
VoiceCBOnVoiceError
};
static IXAudio2VoiceCallback callbacks = { &callbacks_vtable };
// add WIN_CoInitialize() and WIN_CoUninitialize here;
// to avoid XAudio2Create return hr 0x800401f0 ипн╢╣Всц CoInitialize;
if (iscapture) {
SDL_SetError("XAudio2: capture devices unsupported.");
return 0;
} else if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
SDL_SetError("XAudio2: XAudio2Create() failed.");
return 0;
}
if (devname != NULL) {
UINT32 devcount = 0;
UINT32 i = 0;
if (IXAudio2_GetDeviceCount(ixa2, &devcount) != S_OK) {
IXAudio2_Release(ixa2);
SDL_SetError("XAudio2: IXAudio2_GetDeviceCount() failed.");
return 0;
}
for (i = 0; i < devcount; i++) {
XAUDIO2_DEVICE_DETAILS details;
if (IXAudio2_GetDeviceDetails(ixa2, i, &details) == S_OK) {
char *str = utf16_to_utf8(details.DisplayName);
if (str != NULL) {
const int match = (SDL_strcmp(str, devname) == 0);
SDL_free(str);
if (match) {
devId = i;
break;
}
}
}
}
if (i == devcount) {
IXAudio2_Release(ixa2);
SDL_SetError("XAudio2: Requested device not found.");
return 0;
}
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
IXAudio2_Release(ixa2);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
this->hidden->ixa2 = ixa2;
this->hidden->semaphore = CreateSemaphore(NULL, 1, 2, NULL);
if (this->hidden->semaphore == NULL) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: CreateSemaphore() failed!");
return 0;
}
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Unsupported audio format");
return 0;
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* We feed a Source, it feeds the Mastering, which feeds the device. */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(2 * this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
XAUDIO2_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
this->hidden->nextbuf = this->hidden->mixbuf;
SDL_memset(this->hidden->mixbuf, 0, 2 * this->hidden->mixlen);
/* We use XAUDIO2_DEFAULT_CHANNELS instead of this->spec.channels. On
Xbox360, this means 5.1 output, but on Windows, it means "figure out
what the system has." It might be preferable to let XAudio2 blast
stereo output to appropriate surround sound configurations
instead of clamping to 2 channels, even though we'll configure the
Source Voice for whatever number of channels you supply. */
result = IXAudio2_CreateMasteringVoice(ixa2, &this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS,
this->spec.freq, 0, devId, NULL);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't create mastering voice");
return 0;
}
SDL_zero(waveformat);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
waveformat.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
waveformat.wFormatTag = WAVE_FORMAT_PCM;
}
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
result = IXAudio2_CreateSourceVoice(ixa2, &source, &waveformat,
XAUDIO2_VOICE_NOSRC |
XAUDIO2_VOICE_NOPITCH,
1.0f, &callbacks, NULL, NULL);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't create source voice");
return 0;
}
this->hidden->source = source;
/* Start everything playing! */
result = IXAudio2_StartEngine(ixa2);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't start engine");
return 0;
}
result = IXAudio2SourceVoice_Start(source, 0, XAUDIO2_COMMIT_NOW);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
SDL_SetError("XAudio2: Couldn't start source voice");
return 0;
}
return 1; /* good to go. */
}
static int
COREAUDIO_OpenDevice(_THIS, const char *devname, int iscapture)
{
fprintf(stderr, "OpenDevice\n");
AudioStreamBasicDescription strdesc;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
fprintf(stderr, "OutOfMemory\n");
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Setup a AudioStreamBasicDescription with the requested format */
SDL_memset(&strdesc, '\0', sizeof(AudioStreamBasicDescription));
strdesc.mFormatID = kAudioFormatLinearPCM;
strdesc.mFormatFlags = kLinearPCMFormatFlagIsPacked;
strdesc.mChannelsPerFrame = this->spec.channels;
strdesc.mSampleRate = this->spec.freq;
strdesc.mFramesPerPacket = 1;
while ((!valid_datatype) && (test_format)) {
this->spec.format = test_format;
/* Just a list of valid SDL formats, so people don't pass junk here. */
switch (test_format) {
case AUDIO_U8:
case AUDIO_S8:
case AUDIO_U16LSB:
case AUDIO_S16LSB:
case AUDIO_U16MSB:
case AUDIO_S16MSB:
case AUDIO_S32LSB:
case AUDIO_S32MSB:
case AUDIO_F32LSB:
case AUDIO_F32MSB:
valid_datatype = 1;
strdesc.mBitsPerChannel = SDL_AUDIO_BITSIZE(this->spec.format);
if (SDL_AUDIO_ISBIGENDIAN(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
if (SDL_AUDIO_ISFLOAT(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsFloat;
else if (SDL_AUDIO_ISSIGNED(this->spec.format))
strdesc.mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
break;
}
}
if (!valid_datatype) { /* shouldn't happen, but just in case... */
COREAUDIO_CloseDevice(this);
fprintf(stderr, "Unsupported audio format\n");
return 0;
}
strdesc.mBytesPerFrame =
strdesc.mBitsPerChannel * strdesc.mChannelsPerFrame / 8;
strdesc.mBytesPerPacket =
strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;
if (!prepare_audiounit(this, devname, iscapture, &strdesc)) {
COREAUDIO_CloseDevice(this);
fprintf(stderr, "prepare_audiounit failed\n");
return 0; /* prepare_audiounit() will call SDL_SetError()... */
}
return 1; /* good to go. */
}
/*
===============
SNDDMA_Init
===============
*/
qboolean SNDDMA_Init(void)
{
SDL_AudioSpec desired;
SDL_AudioSpec obtained;
int tmp;
#ifdef USE_SDL_AUDIO_CAPTURE
SDL_version sdlVersion;
#endif
if (snd_inited)
return qtrue;
if (!s_sdlBits) {
s_sdlBits = Cvar_Get("s_sdlBits", "16", CVAR_ARCHIVE);
s_sdlSpeed = Cvar_Get("s_sdlSpeed", "0", CVAR_ARCHIVE);
s_sdlChannels = Cvar_Get("s_sdlChannels", "2", CVAR_ARCHIVE);
s_sdlDevSamps = Cvar_Get("s_sdlDevSamps", "0", CVAR_ARCHIVE);
s_sdlMixSamps = Cvar_Get("s_sdlMixSamps", "0", CVAR_ARCHIVE);
}
Com_Printf( "SDL_Init( SDL_INIT_AUDIO )... " );
if (SDL_Init(SDL_INIT_AUDIO) != 0)
{
Com_Printf( "FAILED (%s)\n", SDL_GetError( ) );
return qfalse;
}
Com_Printf( "OK\n" );
Com_Printf( "SDL audio driver is \"%s\".\n", SDL_GetCurrentAudioDriver( ) );
memset(&desired, '\0', sizeof (desired));
memset(&obtained, '\0', sizeof (obtained));
tmp = ((int) s_sdlBits->value);
if ((tmp != 16) && (tmp != 8))
tmp = 16;
desired.freq = (int) s_sdlSpeed->value;
if(!desired.freq) desired.freq = 22050;
desired.format = ((tmp == 16) ? AUDIO_S16SYS : AUDIO_U8);
// I dunno if this is the best idea, but I'll give it a try...
// should probably check a cvar for this...
if (s_sdlDevSamps->value)
desired.samples = s_sdlDevSamps->value;
else
{
// just pick a sane default.
if (desired.freq <= 11025)
desired.samples = 256;
else if (desired.freq <= 22050)
desired.samples = 512;
else if (desired.freq <= 44100)
desired.samples = 1024;
else
desired.samples = 2048; // (*shrug*)
}
desired.channels = (int) s_sdlChannels->value;
desired.callback = SNDDMA_AudioCallback;
sdlPlaybackDevice = SDL_OpenAudioDevice(NULL, SDL_FALSE, &desired, &obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
if (sdlPlaybackDevice == 0)
{
Com_Printf("SDL_OpenAudioDevice() failed: %s\n", SDL_GetError());
SDL_QuitSubSystem(SDL_INIT_AUDIO);
return qfalse;
}
SNDDMA_PrintAudiospec("SDL_AudioSpec", &obtained);
// dma.samples needs to be big, or id's mixer will just refuse to
// work at all; we need to keep it significantly bigger than the
// amount of SDL callback samples, and just copy a little each time
// the callback runs.
// 32768 is what the OSS driver filled in here on my system. I don't
// know if it's a good value overall, but at least we know it's
// reasonable...this is why I let the user override.
tmp = s_sdlMixSamps->value;
if (!tmp)
tmp = (obtained.samples * obtained.channels) * 10;
if (tmp & (tmp - 1)) // not a power of two? Seems to confuse something.
{
int val = 1;
while (val < tmp)
val <<= 1;
tmp = val;
}
dmapos = 0;
dma.samplebits = SDL_AUDIO_BITSIZE(obtained.format);
dma.isfloat = SDL_AUDIO_ISFLOAT(obtained.format);
dma.channels = obtained.channels;
dma.samples = tmp;
dma.submission_chunk = 1;
dma.speed = obtained.freq;
dmasize = (dma.samples * (dma.samplebits/8));
dma.buffer = calloc(1, dmasize);
#ifdef USE_SDL_AUDIO_CAPTURE
// !!! FIXME: some of these SDL_OpenAudioDevice() values should be cvars.
s_sdlCapture = Cvar_Get( "s_sdlCapture", "1", CVAR_ARCHIVE | CVAR_LATCH );
// !!! FIXME: hopefully pulseaudio capture will be fixed in SDL 2.0.9... https://bugzilla.libsdl.org/show_bug.cgi?id=4087
SDL_GetVersion(&sdlVersion);
if (sdlVersion.major == 2 && sdlVersion.minor == 0 && sdlVersion.patch < 9 && Q_stricmp(SDL_GetCurrentAudioDriver(), "pulseaudio") == 0)
{
Com_Printf("SDL audio capture support disabled (pulseaudio capture does not work correctly with SDL %d.%d.%d)\n", sdlVersion.major, sdlVersion.minor, sdlVersion.patch);
}
else if (!s_sdlCapture->integer)
{
Com_Printf("SDL audio capture support disabled by user ('+set s_sdlCapture 1' to enable)\n");
}
#if USE_MUMBLE
else if (cl_useMumble->integer)
{
Com_Printf("SDL audio capture support disabled for Mumble support\n");
}
#endif
else
{
/* !!! FIXME: list available devices and let cvar specify one, like OpenAL does */
SDL_AudioSpec spec;
SDL_zero(spec);
spec.freq = 48000;
spec.format = AUDIO_S16SYS;
spec.channels = 1;
spec.samples = VOIP_MAX_PACKET_SAMPLES * 4;
sdlCaptureDevice = SDL_OpenAudioDevice(NULL, SDL_TRUE, &spec, NULL, 0);
Com_Printf( "SDL capture device %s.\n",
(sdlCaptureDevice == 0) ? "failed to open" : "opened");
}
sdlMasterGain = 1.0f;
#endif
Com_Printf("Starting SDL audio callback...\n");
SDL_PauseAudioDevice(sdlPlaybackDevice, 0); // start callback.
// don't unpause the capture device; we'll do that in StartCapture.
Com_Printf("SDL audio initialized.\n");
snd_inited = qtrue;
return qtrue;
}
void playfile(void * udata, Uint8 * stream, int len)
{
(void)udata;
int stream_bytespersample = SDL_AUDIO_BITSIZE(got.format)/8;
int stream_datagain = power(2, stream_bytespersample*8)/2;
int channels = got.channels;
int used = 0;
int stream_LE = SDL_AUDIO_ISLITTLEENDIAN(got.format);
while(used < len)
{
for(auto i = 0; i < channels; i++)
{
float transient = 0.0f;
for(auto emitter : emitters)
{
if(!emitter->sample->ready or emitter->playing == false)
continue;
float ratefactor = emitter->sample->samplerate/(float)(got.freq); // stream samples per emitter samples
if(ceil(emitter->position*ratefactor) >= emitter->sample->length)
{
emitter->playing = false;
continue;
}
if(ratefactor == 1)
transient += emitter->sample->sample_from_channel_and_position(i, emitter->position);
else if (ratefactor < 1) // upsample, use triangle filter to artificially create SUPER RETRO SOUNDING highs
{
// convert output position to surrounding input positions
long long point = emitter->position * emitter->sample->samplerate;
long outpoint1 = point/got.freq;
long outpoint2 = point/got.freq + 1;
//float point = ratefactor*emitter->position; // point is position on audio stream
auto a = emitter->sample->sample_from_channel_and_position(i, outpoint1);
auto b = emitter->sample->sample_from_channel_and_position(i, outpoint2);
float fraction = (point%got.freq)/(float)(got.freq);
//float fraction = point-floor(point);
transient += fraction*b + (1-fraction)*a;
}
else // ratefactor > 1
{ // downsample, use triangle filter for laziness's sake
float point = ratefactor*emitter->position; // point is position on emitter stream
int bottom = ceil(point-ratefactor); // window
int top = floor(point+ratefactor);
float windowlen = ratefactor*2;
float calibrate = 0; // convolution normalization
float sample = 0; // output sample
for(float j = ceil(bottom); j < top; j++) // convolution
{
float factor = j>point?j-point:point-j; // distance from output sample
factor = ratefactor - factor; // convolution index of this sample
calibrate += factor;
sample += emitter->sample->sample_from_channel_and_position(i, j) * factor;
}
sample /= calibrate;
transient += sample;
}
}
Sint64 output = transient*stream_datagain;
Uint8 * outbytes = (Uint8 *)&output;
for(auto j = 0; j < stream_bytespersample; j++)
{
*(stream+used
+i*stream_bytespersample
+j)
= stream_LE ? outbytes[j] : outbytes[stream_bytespersample-1-j];
}
}
for(auto emitter : emitters)
{
emitter->position += 1; // n OUTPUT samples into EMITTER (opposed to EMITTER samples)
}
used += stream_bytespersample*channels;
}
}
/* This function tries to create a secondary audio buffer, and returns the
number of audio chunks available in the created buffer.
*/
static int
CreateSecondary(_THIS, HWND focus)
{
LPDIRECTSOUND sndObj = this->hidden->sound;
LPDIRECTSOUNDBUFFER *sndbuf = &this->hidden->mixbuf;
Uint32 chunksize = this->spec.size;
const int numchunks = 8;
HRESULT result = DS_OK;
DSBUFFERDESC format;
LPVOID pvAudioPtr1, pvAudioPtr2;
DWORD dwAudioBytes1, dwAudioBytes2;
WAVEFORMATEX wfmt;
SDL_zero(wfmt);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
wfmt.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
wfmt.wFormatTag = WAVE_FORMAT_PCM;
}
wfmt.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
wfmt.nChannels = this->spec.channels;
wfmt.nSamplesPerSec = this->spec.freq;
wfmt.nBlockAlign = wfmt.nChannels * (wfmt.wBitsPerSample / 8);
wfmt.nAvgBytesPerSec = wfmt.nSamplesPerSec * wfmt.nBlockAlign;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Try to set primary mixing privileges */
if (focus) {
result = IDirectSound_SetCooperativeLevel(sndObj,
focus, DSSCL_PRIORITY);
} else {
result = IDirectSound_SetCooperativeLevel(sndObj,
GetDesktopWindow(),
DSSCL_NORMAL);
}
if (result != DS_OK) {
return SetDSerror("DirectSound SetCooperativeLevel", result);
}
/* Try to create the secondary buffer */
SDL_zero(format);
format.dwSize = sizeof(format);
format.dwFlags = DSBCAPS_GETCURRENTPOSITION2;
if (!focus) {
format.dwFlags |= DSBCAPS_GLOBALFOCUS;
} else {
format.dwFlags |= DSBCAPS_STICKYFOCUS;
}
format.dwBufferBytes = numchunks * chunksize;
if ((format.dwBufferBytes < DSBSIZE_MIN) ||
(format.dwBufferBytes > DSBSIZE_MAX)) {
return SDL_SetError("Sound buffer size must be between %d and %d",
DSBSIZE_MIN / numchunks, DSBSIZE_MAX / numchunks);
}
format.dwReserved = 0;
format.lpwfxFormat = &wfmt;
result = IDirectSound_CreateSoundBuffer(sndObj, &format, sndbuf, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSound CreateSoundBuffer", result);
}
IDirectSoundBuffer_SetFormat(*sndbuf, &wfmt);
/* Silence the initial audio buffer */
result = IDirectSoundBuffer_Lock(*sndbuf, 0, format.dwBufferBytes,
(LPVOID *) & pvAudioPtr1, &dwAudioBytes1,
(LPVOID *) & pvAudioPtr2, &dwAudioBytes2,
DSBLOCK_ENTIREBUFFER);
if (result == DS_OK) {
SDL_memset(pvAudioPtr1, this->spec.silence, dwAudioBytes1);
IDirectSoundBuffer_Unlock(*sndbuf,
(LPVOID) pvAudioPtr1, dwAudioBytes1,
(LPVOID) pvAudioPtr2, dwAudioBytes2);
}
/* We're ready to go */
return (numchunks);
}
int
DSP_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
char audiodev[1024];
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = spec->freq;
/* Initialize our freeable variables, in case we fail */
audio_fd = -1;
mixbuf = NULL;
ulaw_buf = NULL;
/* Determine the audio parameters from the AudioSpec */
switch (SDL_AUDIO_BITSIZE(spec->format)) {
case 8:
{ /* Unsigned 8 bit audio data */
spec->format = AUDIO_U8;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR8;
#endif
}
break;
case 16:
{ /* Signed 16 bit audio data */
spec->format = AUDIO_S16SYS;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR;
#endif
}
break;
default:
{
/* !!! FIXME: fallback to conversion on unsupported types! */
SDL_SetError("Unsupported audio format");
return (-1);
}
}
audio_fmt = spec->format;
/* Open the audio device */
audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1);
if (audio_fd < 0) {
SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
return (-1);
}
ulaw_only = 0; /* modern Suns do support linear audio */
#ifdef AUDIO_SETINFO
for (;;) {
audio_info_t info;
AUDIO_INITINFO(&info); /* init all fields to "no change" */
/* Try to set the requested settings */
info.play.sample_rate = spec->freq;
info.play.channels = spec->channels;
info.play.precision = (enc == AUDIO_ENCODING_ULAW)
? 8 : spec->format & 0xff;
info.play.encoding = enc;
if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0) {
/* Check to be sure we got what we wanted */
if (ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) {
SDL_SetError("Error getting audio parameters: %s",
strerror(errno));
return -1;
}
if (info.play.encoding == enc
&& info.play.precision == (spec->format & 0xff)
&& info.play.channels == spec->channels) {
/* Yow! All seems to be well! */
spec->freq = info.play.sample_rate;
break;
}
}
switch (enc) {
case AUDIO_ENCODING_LINEAR8:
/* unsigned 8bit apparently not supported here */
enc = AUDIO_ENCODING_LINEAR;
spec->format = AUDIO_S16SYS;
break; /* try again */
case AUDIO_ENCODING_LINEAR:
/* linear 16bit didn't work either, resort to µ-law */
enc = AUDIO_ENCODING_ULAW;
spec->channels = 1;
spec->freq = 8000;
spec->format = AUDIO_U8;
ulaw_only = 1;
break;
default:
/* oh well... */
SDL_SetError("Error setting audio parameters: %s",
strerror(errno));
return -1;
}
}
#endif /* AUDIO_SETINFO */
written = 0;
/* We can actually convert on-the-fly to U-Law */
if (ulaw_only) {
spec->freq = desired_freq;
fragsize = (spec->samples * 1000) / (spec->freq / 8);
frequency = 8;
ulaw_buf = (Uint8 *) SDL_malloc(fragsize);
if (ulaw_buf == NULL) {
SDL_OutOfMemory();
return (-1);
}
spec->channels = 1;
} else {
fragsize = spec->samples;
frequency = spec->freq / 1000;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Audio device %s U-Law only\n",
ulaw_only ? "is" : "is not");
fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
spec->format, spec->channels, spec->freq);
#endif
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size);
if (mixbuf == NULL) {
SDL_OutOfMemory();
return (-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return (0);
}
static int
WINMM_OpenDevice(_THIS, const char *devname, int iscapture)
{
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
int valid_datatype = 0;
MMRESULT result;
WAVEFORMATEX waveformat;
UINT_PTR devId = WAVE_MAPPER; /* WAVE_MAPPER == choose system's default */
char *utf8 = NULL;
int i;
if (devname != NULL) { /* specific device requested? */
if (iscapture) {
const int devcount = (int) waveInGetNumDevs();
WAVEINCAPS caps;
for (i = 0; (i < devcount) && (devId == WAVE_MAPPER); i++) {
result = waveInGetDevCaps(i, &caps, sizeof (caps));
if (result != MMSYSERR_NOERROR)
continue;
else if ((utf8 = WIN_StringToUTF8(caps.szPname)) == NULL)
continue;
else if (SDL_strcmp(devname, utf8) == 0)
devId = (UINT_PTR) i;
SDL_free(utf8);
}
} else {
const int devcount = (int) waveOutGetNumDevs();
WAVEOUTCAPS caps;
for (i = 0; (i < devcount) && (devId == WAVE_MAPPER); i++) {
result = waveOutGetDevCaps(i, &caps, sizeof (caps));
if (result != MMSYSERR_NOERROR)
continue;
else if ((utf8 = WIN_StringToUTF8(caps.szPname)) == NULL)
continue;
else if (SDL_strcmp(devname, utf8) == 0)
devId = (UINT_PTR) i;
SDL_free(utf8);
}
}
if (devId == WAVE_MAPPER) {
SDL_SetError("Requested device not found");
return 0;
}
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Initialize the wavebuf structures for closing */
for (i = 0; i < NUM_BUFFERS; ++i)
this->hidden->wavebuf[i].dwUser = 0xFFFF;
while ((!valid_datatype) && (test_format)) {
valid_datatype = 1;
this->spec.format = test_format;
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
break; /* valid. */
default:
valid_datatype = 0;
test_format = SDL_NextAudioFormat();
break;
}
}
if (!valid_datatype) {
WINMM_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
/* Set basic WAVE format parameters */
SDL_memset(&waveformat, '\0', sizeof(waveformat));
waveformat.wFormatTag = WAVE_FORMAT_PCM;
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
if (this->spec.channels > 2)
this->spec.channels = 2; /* !!! FIXME: is this right? */
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
/* Check the buffer size -- minimum of 1/4 second (word aligned) */
if (this->spec.samples < (this->spec.freq / 4))
this->spec.samples = ((this->spec.freq / 4) + 3) & ~3;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
if (iscapture) {
result = waveInOpen(&this->hidden->hin, devId, &waveformat,
(DWORD_PTR) CaptureSound, (DWORD_PTR) this,
CALLBACK_FUNCTION);
} else {
result = waveOutOpen(&this->hidden->hout, devId, &waveformat,
(DWORD_PTR) FillSound, (DWORD_PTR) this,
CALLBACK_FUNCTION);
}
if (result != MMSYSERR_NOERROR) {
WINMM_CloseDevice(this);
SetMMerror("waveOutOpen()", result);
return 0;
}
#ifdef SOUND_DEBUG
/* Check the sound device we retrieved */
{
WAVEOUTCAPS caps;
result = waveOutGetDevCaps((UINT) this->hidden->hout,
&caps, sizeof(caps));
if (result != MMSYSERR_NOERROR) {
WINMM_CloseDevice(this);
SetMMerror("waveOutGetDevCaps()", result);
return 0;
}
printf("Audio device: %s\n", caps.szPname);
}
#endif
/* Create the audio buffer semaphore */
this->hidden->audio_sem =
CreateSemaphore(NULL, NUM_BUFFERS - 1, NUM_BUFFERS, NULL);
if (this->hidden->audio_sem == NULL) {
WINMM_CloseDevice(this);
SDL_SetError("Couldn't create semaphore");
return 0;
}
/* Create the sound buffers */
this->hidden->mixbuf =
(Uint8 *) SDL_malloc(NUM_BUFFERS * this->spec.size);
if (this->hidden->mixbuf == NULL) {
WINMM_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
for (i = 0; i < NUM_BUFFERS; ++i) {
SDL_memset(&this->hidden->wavebuf[i], 0,
sizeof(this->hidden->wavebuf[i]));
this->hidden->wavebuf[i].dwBufferLength = this->spec.size;
this->hidden->wavebuf[i].dwFlags = WHDR_DONE;
this->hidden->wavebuf[i].lpData =
(LPSTR) & this->hidden->mixbuf[i * this->spec.size];
result = waveOutPrepareHeader(this->hidden->hout,
&this->hidden->wavebuf[i],
sizeof(this->hidden->wavebuf[i]));
if (result != MMSYSERR_NOERROR) {
WINMM_CloseDevice(this);
SetMMerror("waveOutPrepareHeader()", result);
return 0;
}
}
return 1; /* Ready to go! */
}
int
main(int argc, char **argv)
{
SDL_AudioSpec spec;
SDL_AudioCVT cvt;
Uint32 len = 0;
Uint8 *data = NULL;
int cvtfreq = 0;
int bitsize = 0;
int blockalign = 0;
int avgbytes = 0;
SDL_RWops *io = NULL;
if (argc != 4) {
fprintf(stderr, "USAGE: %s in.wav out.wav newfreq\n", argv[0]);
return 1;
}
cvtfreq = SDL_atoi(argv[3]);
if (SDL_Init(SDL_INIT_AUDIO) == -1) {
fprintf(stderr, "SDL_Init() failed: %s\n", SDL_GetError());
return 2;
}
if (SDL_LoadWAV(argv[1], &spec, &data, &len) == NULL) {
fprintf(stderr, "failed to load %s: %s\n", argv[1], SDL_GetError());
SDL_Quit();
return 3;
}
if (SDL_BuildAudioCVT(&cvt, spec.format, spec.channels, spec.freq,
spec.format, spec.channels, cvtfreq) == -1) {
fprintf(stderr, "failed to build CVT: %s\n", SDL_GetError());
SDL_FreeWAV(data);
SDL_Quit();
return 4;
}
cvt.len = len;
cvt.buf = (Uint8 *) SDL_malloc(len * cvt.len_mult);
if (cvt.buf == NULL) {
fprintf(stderr, "Out of memory.\n");
SDL_FreeWAV(data);
SDL_Quit();
return 5;
}
SDL_memcpy(cvt.buf, data, len);
if (SDL_ConvertAudio(&cvt) == -1) {
fprintf(stderr, "Conversion failed: %s\n", SDL_GetError());
SDL_free(cvt.buf);
SDL_FreeWAV(data);
SDL_Quit();
return 6;
}
/* write out a WAV header... */
io = SDL_RWFromFile(argv[2], "wb");
if (io == NULL) {
fprintf(stderr, "fopen('%s') failed: %s\n", argv[2], SDL_GetError());
SDL_free(cvt.buf);
SDL_FreeWAV(data);
SDL_Quit();
return 7;
}
bitsize = SDL_AUDIO_BITSIZE(spec.format);
blockalign = (bitsize / 8) * spec.channels;
avgbytes = cvtfreq * blockalign;
SDL_WriteLE32(io, 0x46464952); /* RIFF */
SDL_WriteLE32(io, len * cvt.len_mult + 36);
SDL_WriteLE32(io, 0x45564157); /* WAVE */
SDL_WriteLE32(io, 0x20746D66); /* fmt */
SDL_WriteLE32(io, 16); /* chunk size */
SDL_WriteLE16(io, 1); /* uncompressed */
SDL_WriteLE16(io, spec.channels); /* channels */
SDL_WriteLE32(io, cvtfreq); /* sample rate */
SDL_WriteLE32(io, avgbytes); /* average bytes per second */
SDL_WriteLE16(io, blockalign); /* block align */
SDL_WriteLE16(io, bitsize); /* significant bits per sample */
SDL_WriteLE32(io, 0x61746164); /* data */
SDL_WriteLE32(io, cvt.len_cvt); /* size */
SDL_RWwrite(io, cvt.buf, cvt.len_cvt, 1);
if (SDL_RWclose(io) == -1) {
fprintf(stderr, "fclose('%s') failed: %s\n", argv[2], SDL_GetError());
SDL_free(cvt.buf);
SDL_FreeWAV(data);
SDL_Quit();
return 8;
} // if
SDL_free(cvt.buf);
SDL_FreeWAV(data);
SDL_Quit();
return 0;
} // main
static int
DSOUND_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const DWORD numchunks = 8;
HRESULT result;
SDL_bool valid_format = SDL_FALSE;
SDL_bool tried_format = SDL_FALSE;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
LPGUID guid = (LPGUID) handle;
DWORD bufsize;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_zerop(this->hidden);
/* Open the audio device */
if (iscapture) {
result = pDirectSoundCaptureCreate8(guid, &this->hidden->capture, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSoundCaptureCreate8", result);
}
} else {
result = pDirectSoundCreate8(guid, &this->hidden->sound, NULL);
if (result != DS_OK) {
return SetDSerror("DirectSoundCreate8", result);
}
result = IDirectSound_SetCooperativeLevel(this->hidden->sound,
GetDesktopWindow(),
DSSCL_NORMAL);
if (result != DS_OK) {
return SetDSerror("DirectSound SetCooperativeLevel", result);
}
}
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
tried_format = SDL_TRUE;
this->spec.format = test_format;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
bufsize = numchunks * this->spec.size;
if ((bufsize < DSBSIZE_MIN) || (bufsize > DSBSIZE_MAX)) {
SDL_SetError("Sound buffer size must be between %d and %d",
(DSBSIZE_MIN < numchunks) ? 1 : DSBSIZE_MIN / numchunks,
DSBSIZE_MAX / numchunks);
} else {
int rc;
WAVEFORMATEX wfmt;
SDL_zero(wfmt);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
wfmt.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
wfmt.wFormatTag = WAVE_FORMAT_PCM;
}
wfmt.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
wfmt.nChannels = this->spec.channels;
wfmt.nSamplesPerSec = this->spec.freq;
wfmt.nBlockAlign = wfmt.nChannels * (wfmt.wBitsPerSample / 8);
wfmt.nAvgBytesPerSec = wfmt.nSamplesPerSec * wfmt.nBlockAlign;
rc = iscapture ? CreateCaptureBuffer(this, bufsize, &wfmt) : CreateSecondary(this, bufsize, &wfmt);
if (rc == 0) {
this->hidden->num_buffers = numchunks;
valid_format = SDL_TRUE;
}
}
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
if (tried_format) {
return -1; /* CreateSecondary() should have called SDL_SetError(). */
}
return SDL_SetError("DirectSound: Unsupported audio format");
}
/* Playback buffers will auto-start playing in DSOUND_WaitDevice() */
return 0; /* good to go. */
}
static int
VITAAUD_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
int format, mixlen, i;
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc(sizeof(*this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, sizeof(*this->hidden));
switch (this->spec.format & 0xff) {
case 8:
case 16:
this->spec.format = AUDIO_S16LSB;
break;
default:
return SDL_SetError("Unsupported audio format");
}
/* The sample count must be a multiple of 64. */
this->spec.samples = SCE_AUDIO_SAMPLE_ALIGN(this->spec.samples);
this->spec.freq = 48000;
/* Update the fragment size as size in bytes. */
/* SDL_CalculateAudioSpec(this->spec); MOD */
switch (this->spec.format) {
case AUDIO_U8:
this->spec.silence = 0x80;
break;
default:
this->spec.silence = 0x00;
break;
}
this->spec.size = SDL_AUDIO_BITSIZE(this->spec.format) / 8;
this->spec.size *= this->spec.channels;
this->spec.size *= this->spec.samples;
/* ========================================== */
/* Allocate the mixing buffer. Its size and starting address must
be a multiple of 64 bytes. Our sample count is already a multiple of
64, so spec->size should be a multiple of 64 as well. */
mixlen = this->spec.size * NUM_BUFFERS;
this->hidden->rawbuf = (Uint8 *) memalign(64, mixlen);
if (this->hidden->rawbuf == NULL) {
return SDL_SetError("Couldn't allocate mixing buffer");
}
/* Setup the hardware channel. */
if (this->spec.channels == 1) {
format = SCE_AUDIO_OUT_MODE_MONO;
} else {
format = SCE_AUDIO_OUT_MODE_STEREO;
}
this->hidden->channel = sceAudioOutOpenPort(SCE_AUDIO_OUT_PORT_TYPE_MAIN, this->spec.samples, this->spec.freq, format);
if (this->hidden->channel < 0) {
free(this->hidden->rawbuf);
this->hidden->rawbuf = NULL;
return SDL_SetError("Couldn't reserve hardware channel");
}
memset(this->hidden->rawbuf, 0, mixlen);
for (i = 0; i < NUM_BUFFERS; i++) {
this->hidden->mixbufs[i] = &this->hidden->rawbuf[i * this->spec.size];
}
this->hidden->next_buffer = 0;
return 0;
}
static int
XAUDIO2_OpenDevice(_THIS, const char *devname, int iscapture)
{
HRESULT result = S_OK;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(_this->spec.format);
IXAudio2 *ixa2 = NULL;
IXAudio2SourceVoice *source = NULL;
UINT32 devId = 0; /* 0 == system default device. */
if (iscapture) {
return SDL_SetError("XAudio2: capture devices unsupported.");
} else if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
return SDL_SetError("XAudio2: XAudio2Create() failed at open.");
}
if (devname != NULL) {
UINT32 devcount = 0;
UINT32 i = 0;
if (i == devcount) {
ixa2->Release();
return SDL_SetError("XAudio2: Requested device not found.");
}
}
/* Initialize all variables that we clean on shutdown */
_this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *_this->hidden));
if (_this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(_this->hidden, 0, (sizeof *_this->hidden));
_this->hidden->ixa2 = ixa2;
_this->hidden->semaphore = CreateSemaphoreEx(NULL, 1, 2, NULL, 0, 0);
if (_this->hidden->semaphore == NULL) {
XAUDIO2_CloseDevice(_this);
return SDL_SetError("XAudio2: CreateSemaphore() failed!");
}
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
_this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
XAUDIO2_CloseDevice(_this);
return SDL_SetError("XAudio2: Unsupported audio format");
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&_this->spec);
/* We feed a Source, it feeds the Mastering, which feeds the device. */
_this->hidden->mixlen = _this->spec.size;
_this->hidden->mixbuf = (Uint8 *) SDL_malloc(2 * _this->hidden->mixlen);
if (_this->hidden->mixbuf == NULL) {
XAUDIO2_CloseDevice(_this);
return SDL_OutOfMemory();
}
_this->hidden->nextbuf = _this->hidden->mixbuf;
SDL_memset(_this->hidden->mixbuf, 0, 2 * _this->hidden->mixlen);
/* We use XAUDIO2_DEFAULT_CHANNELS instead of _this->spec.channels. On
Xbox360, _this means 5.1 output, but on Windows, it means "figure out
what the system has." It might be preferable to let XAudio2 blast
stereo output to appropriate surround sound configurations
instead of clamping to 2 channels, even though we'll configure the
Source Voice for whatever number of channels you supply. */
result = ixa2->CreateMasteringVoice(&_this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS, _this->spec.freq, 0,
nullptr, nullptr, AudioCategory_GameMedia);
// XAUDIO2_DEFAULT_CHANNELS,
// _this->spec.freq, 0, devId, NULL);
if (result != S_OK) {
XAUDIO2_CloseDevice(_this);
return SDL_SetError("XAudio2: Couldn't create mastering voice");
}
SDL_zero(waveformat);
if (SDL_AUDIO_ISFLOAT(_this->spec.format)) {
waveformat.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
waveformat.wFormatTag = WAVE_FORMAT_PCM;
}
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(_this->spec.format);
waveformat.nChannels = _this->spec.channels;
waveformat.nSamplesPerSec = _this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
static VoiceCallback callbacks;
XAUDIO2_SEND_DESCRIPTOR descriptors[1];
descriptors[0].pOutputVoice = _this->hidden->mastering;
descriptors[0].Flags = 0;
XAUDIO2_VOICE_SENDS sends = {0};
sends.SendCount = 1;
sends.pSends = descriptors;
result = ixa2->CreateSourceVoice(&source, &waveformat, 0,
1.0f, &callbacks, &sends, nullptr);
if (result != S_OK) {
XAUDIO2_CloseDevice(_this);
return SDL_SetError("XAudio2: Couldn't create source voice");
}
_this->hidden->source = source;
/* Start everything playing! */
result = ixa2->StartEngine();
if (result != S_OK) {
XAUDIO2_CloseDevice(_this);
return SDL_SetError("XAudio2: Couldn't start engine");
}
result = source->Start(0, XAUDIO2_COMMIT_NOW);
if (result != S_OK) {
XAUDIO2_CloseDevice(_this);
return SDL_SetError("XAudio2: Couldn't start source voice");
}
return 0; /* good to go. */
}
static void
HandleAudioProcess(_THIS)
{
Uint8 *buf = NULL;
int byte_len = 0;
int bytes = SDL_AUDIO_BITSIZE(this->spec.format) / 8;
int bytes_in = SDL_AUDIO_BITSIZE(this->convert.src_format) / 8;
/* Only do soemthing if audio is enabled */
if (!this->enabled)
return;
if (this->paused)
return;
if (this->convert.needed) {
if (this->hidden->conv_in_len != 0) {
this->convert.len = this->hidden->conv_in_len * bytes_in * this->spec.channels;
}
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
buf = this->convert.buf;
byte_len = this->convert.len_cvt;
/* size mismatch*/
if (byte_len != this->spec.size) {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size > byte_len ? this->spec.size * 2 : byte_len * 2;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
}
/* copy existing data */
byte_len = copyData(this);
/* read more data*/
while (byte_len < this->spec.size) {
(*this->spec.callback) (this->spec.userdata,
this->convert.buf,
this->convert.len);
SDL_ConvertAudio(&this->convert);
byte_len = copyData(this);
}
byte_len = this->spec.size;
buf = this->hidden->mixbuf + this->hidden->read_off;
this->hidden->read_off += byte_len;
}
} else {
if (!this->hidden->mixbuf) {
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = SDL_malloc(this->hidden->mixlen);
}
(*this->spec.callback) (this->spec.userdata,
this->hidden->mixbuf,
this->hidden->mixlen);
buf = this->hidden->mixbuf;
byte_len = this->hidden->mixlen;
}
if (buf) {
EM_ASM_ARGS({
var numChannels = SDL2.audio.currentOutputBuffer['numberOfChannels'];
for (var c = 0; c < numChannels; ++c) {
var channelData = SDL2.audio.currentOutputBuffer['getChannelData'](c);
if (channelData.length != $1) {
throw 'Web Audio output buffer length mismatch! Destination size: ' + channelData.length + ' samples vs expected ' + $1 + ' samples!';
}
for (var j = 0; j < $1; ++j) {
channelData[j] = getValue($0 + (j*numChannels + c)*4, 'float');
}
}
}, buf, byte_len / bytes / this->spec.channels);
}
static int
XAUDIO2_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
HRESULT result = S_OK;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
IXAudio2 *ixa2 = NULL;
IXAudio2SourceVoice *source = NULL;
#if defined(SDL_XAUDIO2_WIN8)
LPCWSTR devId = NULL;
#else
UINT32 devId = 0; /* 0 == system default device. */
#endif
static IXAudio2VoiceCallbackVtbl callbacks_vtable = {
VoiceCBOnVoiceProcessPassStart,
VoiceCBOnVoiceProcessPassEnd,
VoiceCBOnStreamEnd,
VoiceCBOnBufferStart,
VoiceCBOnBufferEnd,
VoiceCBOnLoopEnd,
VoiceCBOnVoiceError
};
static IXAudio2VoiceCallback callbacks = { &callbacks_vtable };
#if defined(SDL_XAUDIO2_WIN8)
/* !!! FIXME: hook up hotplugging. */
#else
if (handle != NULL) { /* specific device requested? */
/* -1 because we increment the original value to avoid NULL. */
const size_t val = ((size_t) handle) - 1;
devId = (UINT32) val;
}
#endif
if (XAudio2Create(&ixa2, 0, XAUDIO2_DEFAULT_PROCESSOR) != S_OK) {
return SDL_SetError("XAudio2: XAudio2Create() failed at open.");
}
/*
XAUDIO2_DEBUG_CONFIGURATION debugConfig;
debugConfig.TraceMask = XAUDIO2_LOG_ERRORS; //XAUDIO2_LOG_WARNINGS | XAUDIO2_LOG_DETAIL | XAUDIO2_LOG_FUNC_CALLS | XAUDIO2_LOG_TIMING | XAUDIO2_LOG_LOCKS | XAUDIO2_LOG_MEMORY | XAUDIO2_LOG_STREAMING;
debugConfig.BreakMask = XAUDIO2_LOG_ERRORS; //XAUDIO2_LOG_WARNINGS;
debugConfig.LogThreadID = TRUE;
debugConfig.LogFileline = TRUE;
debugConfig.LogFunctionName = TRUE;
debugConfig.LogTiming = TRUE;
ixa2->SetDebugConfiguration(&debugConfig);
*/
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
IXAudio2_Release(ixa2);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
this->hidden->ixa2 = ixa2;
this->hidden->semaphore = SDL_CreateSemaphore(1);
if (this->hidden->semaphore == NULL) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: CreateSemaphore() failed!");
}
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
case AUDIO_F32:
this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Unsupported audio format");
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* We feed a Source, it feeds the Mastering, which feeds the device. */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_malloc(2 * this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
XAUDIO2_CloseDevice(this);
return SDL_OutOfMemory();
}
this->hidden->nextbuf = this->hidden->mixbuf;
SDL_memset(this->hidden->mixbuf, 0, 2 * this->hidden->mixlen);
/* We use XAUDIO2_DEFAULT_CHANNELS instead of this->spec.channels. On
Xbox360, this means 5.1 output, but on Windows, it means "figure out
what the system has." It might be preferable to let XAudio2 blast
stereo output to appropriate surround sound configurations
instead of clamping to 2 channels, even though we'll configure the
Source Voice for whatever number of channels you supply. */
#if SDL_XAUDIO2_WIN8
result = IXAudio2_CreateMasteringVoice(ixa2, &this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS,
this->spec.freq, 0, devId, NULL, AudioCategory_GameEffects);
#else
result = IXAudio2_CreateMasteringVoice(ixa2, &this->hidden->mastering,
XAUDIO2_DEFAULT_CHANNELS,
this->spec.freq, 0, devId, NULL);
#endif
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't create mastering voice");
}
SDL_zero(waveformat);
if (SDL_AUDIO_ISFLOAT(this->spec.format)) {
waveformat.wFormatTag = WAVE_FORMAT_IEEE_FLOAT;
} else {
waveformat.wFormatTag = WAVE_FORMAT_PCM;
}
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
waveformat.cbSize = sizeof(waveformat);
#ifdef __WINRT__
// DLudwig: for now, make XAudio2 do sample rate conversion, just to
// get the loopwave test to work.
//
// TODO, WinRT: consider removing WinRT-specific source-voice creation code from SDL_xaudio2.c
result = IXAudio2_CreateSourceVoice(ixa2, &source, &waveformat,
0,
1.0f, &callbacks, NULL, NULL);
#else
result = IXAudio2_CreateSourceVoice(ixa2, &source, &waveformat,
XAUDIO2_VOICE_NOSRC |
XAUDIO2_VOICE_NOPITCH,
1.0f, &callbacks, NULL, NULL);
#endif
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't create source voice");
}
this->hidden->source = source;
/* Start everything playing! */
result = IXAudio2_StartEngine(ixa2);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't start engine");
}
result = IXAudio2SourceVoice_Start(source, 0, XAUDIO2_COMMIT_NOW);
if (result != S_OK) {
XAUDIO2_CloseDevice(this);
return SDL_SetError("XAudio2: Couldn't start source voice");
}
return 0; /* good to go. */
}
static int
SUNAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture)
{
const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);
SDL_AudioFormat format = 0;
audio_info_t info;
/* We don't care what the devname is...we'll try to open anything. */
/* ...but default to first name in the list... */
if (devname == NULL) {
devname = SDL_GetAudioDeviceName(0, iscapture);
if (devname == NULL) {
return SDL_SetError("No such audio device");
}
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Open the audio device */
this->hidden->audio_fd = open(devname, flags, 0);
if (this->hidden->audio_fd < 0) {
return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));
}
#ifdef AUDIO_SETINFO
int enc;
#endif
int desired_freq = this->spec.freq;
/* Determine the audio parameters from the AudioSpec */
switch (SDL_AUDIO_BITSIZE(this->spec.format)) {
case 8:
{ /* Unsigned 8 bit audio data */
this->spec.format = AUDIO_U8;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR8;
#endif
}
break;
case 16:
{ /* Signed 16 bit audio data */
this->spec.format = AUDIO_S16SYS;
#ifdef AUDIO_SETINFO
enc = AUDIO_ENCODING_LINEAR;
#endif
}
break;
default:
{
/* !!! FIXME: fallback to conversion on unsupported types! */
return SDL_SetError("Unsupported audio format");
}
}
this->hidden->audio_fmt = this->spec.format;
this->hidden->ulaw_only = 0; /* modern Suns do support linear audio */
#ifdef AUDIO_SETINFO
for (;;) {
audio_info_t info;
AUDIO_INITINFO(&info); /* init all fields to "no change" */
/* Try to set the requested settings */
info.play.sample_rate = this->spec.freq;
info.play.channels = this->spec.channels;
info.play.precision = (enc == AUDIO_ENCODING_ULAW)
? 8 : this->spec.format & 0xff;
info.play.encoding = enc;
if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {
/* Check to be sure we got what we wanted */
if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {
return SDL_SetError("Error getting audio parameters: %s",
strerror(errno));
}
if (info.play.encoding == enc
&& info.play.precision == (this->spec.format & 0xff)
&& info.play.channels == this->spec.channels) {
/* Yow! All seems to be well! */
this->spec.freq = info.play.sample_rate;
break;
}
}
switch (enc) {
case AUDIO_ENCODING_LINEAR8:
/* unsigned 8bit apparently not supported here */
enc = AUDIO_ENCODING_LINEAR;
this->spec.format = AUDIO_S16SYS;
break; /* try again */
case AUDIO_ENCODING_LINEAR:
/* linear 16bit didn't work either, resort to µ-law */
enc = AUDIO_ENCODING_ULAW;
this->spec.channels = 1;
this->spec.freq = 8000;
this->spec.format = AUDIO_U8;
this->hidden->ulaw_only = 1;
break;
default:
/* oh well... */
return SDL_SetError("Error setting audio parameters: %s",
strerror(errno));
}
}
#endif /* AUDIO_SETINFO */
this->hidden->written = 0;
/* We can actually convert on-the-fly to U-Law */
if (this->hidden->ulaw_only) {
this->spec.freq = desired_freq;
this->hidden->fragsize = (this->spec.samples * 1000) /
(this->spec.freq / 8);
this->hidden->frequency = 8;
this->hidden->ulaw_buf = (Uint8 *) SDL_malloc(this->hidden->fragsize);
if (this->hidden->ulaw_buf == NULL) {
return SDL_OutOfMemory();
}
this->spec.channels = 1;
} else {
this->hidden->fragsize = this->spec.samples;
this->hidden->frequency = this->spec.freq / 1000;
}
#ifdef DEBUG_AUDIO
fprintf(stderr, "Audio device %s U-Law only\n",
this->hidden->ulaw_only ? "is" : "is not");
fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
this->spec.format, this->spec.channels, this->spec.freq);
#endif
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->spec.size);
if (this->hidden->mixbuf == NULL) {
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 0;
}
static int
MME_OpenDevice(_THIS, const char *devname, int iscapture)
{
int valid_format = 0;
MMRESULT result;
Uint8 *mixbuf = NULL;
int i;
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Set basic WAVE format parameters */
this->hidden->shm = mmeAllocMem(sizeof(*this->hidden->shm));
if (this->hidden->shm == NULL) {
MME_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->shm, '\0', sizeof(*this->hidden->shm));
this->hidden->shm->sound = 0;
this->hidden->shm->wFmt.wf.wFormatTag = WAVE_FORMAT_PCM;
/* Determine the audio parameters from the AudioSpec */
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!valid_format && test_format;) {
valid_format = 1;
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
break;
default:
valid_format = 0;
test_format = SDL_NextAudioFormat();
}
}
if (!valid_format) {
MME_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
this->spec.format = test_format;
this->hidden->shm->wFmt.wBitsPerSample = SDL_AUDIO_BITSIZE(test_format);
/* !!! FIXME: Can this handle more than stereo? */
this->hidden->shm->wFmt.wf.nChannels = this->spec.channels;
this->hidden->shm->wFmt.wf.nSamplesPerSec = this->spec.freq;
this->hidden->shm->wFmt.wf.nBlockAlign =
this->hidden->shm->wFmt.wf.nChannels *
this->hidden->shm->wFmt.wBitsPerSample / 8;
this->hidden->shm->wFmt.wf.nAvgBytesPerSec =
this->hidden->shm->wFmt.wf.nSamplesPerSec *
this->hidden->shm->wFmt.wf.nBlockAlign;
/* Check the buffer size -- minimum of 1/4 second (word aligned) */
if (this->spec.samples < (this->spec.freq / 4))
this->spec.samples = ((this->spec.freq / 4) + 3) & ~3;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
result = waveOutOpen(&(this->hidden->shm->sound),
WAVE_MAPPER,
&(this->hidden->shm->wFmt.wf),
MME_Callback,
NULL, (CALLBACK_FUNCTION | WAVE_OPEN_SHAREABLE));
if (result != MMSYSERR_NOERROR) {
MME_CloseDevice(this);
SetMMerror("waveOutOpen()", result);
return 0;
}
/* Create the sound buffers */
mixbuf = (Uint8 *) mmeAllocBuffer(NUM_BUFFERS * (this->spec.size));
if (mixbuf == NULL) {
MME_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
this->hidden->mixbuf = mixbuf;
for (i = 0; i < NUM_BUFFERS; i++) {
this->hidden->shm->wHdr[i].lpData = &mixbuf[i * (this->spec.size)];
this->hidden->shm->wHdr[i].dwBufferLength = this->spec.size;
this->hidden->shm->wHdr[i].dwFlags = 0;
this->hidden->shm->wHdr[i].dwUser = i;
this->hidden->shm->wHdr[i].dwLoops = 0; /* loop control counter */
this->hidden->shm->wHdr[i].lpNext = NULL; /* reserved for driver */
this->hidden->shm->wHdr[i].reserved = 0;
inUse[i] = FALSE;
}
this->hidden->next_buffer = 0;
return 1;
}
static int
DSOUND_OpenDevice(_THIS, const char *devname, int iscapture)
{
HRESULT result;
WAVEFORMATEX waveformat;
int valid_format = 0;
SDL_AudioFormat test_format = SDL_FirstAudioFormat(this->spec.format);
FindDevGUIDData devguid;
LPGUID guid = NULL;
if (devname != NULL) {
devguid.found = 0;
devguid.devname = devname;
if (iscapture)
pDirectSoundCaptureEnumerateW(FindDevGUID, &devguid);
else
pDirectSoundEnumerateW(FindDevGUID, &devguid);
if (!devguid.found) {
SDL_SetError("DirectSound: Requested device not found");
return 0;
}
guid = &devguid.guid;
}
/* Initialize all variables that we clean on shutdown */
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
if (this->hidden == NULL) {
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
while ((!valid_format) && (test_format)) {
switch (test_format) {
case AUDIO_U8:
case AUDIO_S16:
case AUDIO_S32:
this->spec.format = test_format;
valid_format = 1;
break;
}
test_format = SDL_NextAudioFormat();
}
if (!valid_format) {
DSOUND_CloseDevice(this);
SDL_SetError("DirectSound: Unsupported audio format");
return 0;
}
SDL_memset(&waveformat, 0, sizeof(waveformat));
waveformat.wFormatTag = WAVE_FORMAT_PCM;
waveformat.wBitsPerSample = SDL_AUDIO_BITSIZE(this->spec.format);
waveformat.nChannels = this->spec.channels;
waveformat.nSamplesPerSec = this->spec.freq;
waveformat.nBlockAlign =
waveformat.nChannels * (waveformat.wBitsPerSample / 8);
waveformat.nAvgBytesPerSec =
waveformat.nSamplesPerSec * waveformat.nBlockAlign;
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(&this->spec);
/* Open the audio device */
result = pDirectSoundCreate8(guid, &this->hidden->sound, NULL);
if (result != DS_OK) {
DSOUND_CloseDevice(this);
SetDSerror("DirectSoundCreate", result);
return 0;
}
/* Create the audio buffer to which we write */
this->hidden->num_buffers = CreateSecondary(this, NULL, &waveformat);
if (this->hidden->num_buffers < 0) {
DSOUND_CloseDevice(this);
return 0;
}
/* The buffer will auto-start playing in DSOUND_WaitDevice() */
this->hidden->mixlen = this->spec.size;
return 1; /* good to go. */
}