本文整理汇总了C++中SDL_AllocAudioMem函数的典型用法代码示例。如果您正苦于以下问题:C++ SDL_AllocAudioMem函数的具体用法?C++ SDL_AllocAudioMem怎么用?C++ SDL_AllocAudioMem使用的例子?那么, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了SDL_AllocAudioMem函数的15个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。
示例1: MPFAUD_OpenAudio
static int MPFAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int format;
float bytes_per_sec = 0.0f;
/* Allocate mixing buffer */
this->hidden->mixlen = spec->size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if ( this->hidden->mixbuf == NULL ) {
return(-1);
}
SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
bytes_per_sec = (float) (((spec->format & 0xFF) / 8) *
spec->channels * spec->freq);
switch (spec->format & 0xff) {
case 8:format=8;break;
case 16:format=16;break;
default:
SDL_SetError("Unsupported audio format");
return -1;
}
sys_sound_init(spec->freq,format,spec->channels);
/* We're ready to rock and roll. :-) */
return(0);
}
示例2: DCAUD_OpenAudio
static int DCAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
switch(spec->format&0xff) {
case 8: spec->format = AUDIO_S8; break;
case 16: spec->format = AUDIO_S16LSB; break;
default:
SDL_SetError("Unsupported audio format");
return(-1);
}
/* Update the fragment size as size in bytes */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
this->hidden->mixlen = spec->size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if ( this->hidden->mixbuf == NULL ) {
return(-1);
}
memset(this->hidden->mixbuf, spec->silence, spec->size);
this->hidden->leftpos = 0x11000;
this->hidden->rightpos = 0x11000+spec->size;
this->hidden->playing = 0;
this->hidden->nextbuf = 0;
/* We're ready to rock and roll. :-) */
return(0);
}
示例3: DISKAUD_OpenAudio
static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
const char *fname = DISKAUD_GetOutputFilename();
/* Open the audio device */
this->hidden->audio_fd = open(fname, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR);
if ( this->hidden->audio_fd < 0 ) {
SDL_SetError("Couldn't open %s: %s", fname, strerror(errno));
return(-1);
}
fprintf(stderr, "WARNING: You are using the SDL disk writer"
" audio driver!\n Writing to file [%s].\n", fname);
/* Allocate mixing buffer */
this->hidden->mixlen = spec->size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if ( this->hidden->mixbuf == NULL ) {
return(-1);
}
memset(this->hidden->mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return(0);
}
示例4: DISKAUD_OpenAudio
static int DISKAUD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
const char *fname = DISKAUD_GetOutputFilename();
/* Open the audio device */
this->hidden->output = SDL_RWFromFile(fname, "wb");
if ( this->hidden->output == NULL ) {
return(-1);
}
#if HAVE_STDIO_H
fprintf(stderr, "WARNING: You are using the SDL disk writer"
" audio driver!\n Writing to file [%s].\n", fname);
#endif
/* Allocate mixing buffer */
this->hidden->mixlen = spec->size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if ( this->hidden->mixbuf == NULL ) {
return(-1);
}
SDL_memset(this->hidden->mixbuf, spec->silence, spec->size);
/* We're ready to rock and roll. :-) */
return(0);
}
示例5: DISKAUD_OpenDevice
static int
DISKAUD_OpenDevice(_THIS, const char *devname, int iscapture)
{
const char *envr = SDL_getenv(DISKENVR_WRITEDELAY);
const char *fname = DISKAUD_GetOutputFilename(devname);
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));
/* Open the audio device */
this->hidden->output = SDL_RWFromFile(fname, "wb");
if (this->hidden->output == NULL) {
DISKAUD_CloseDevice(this);
return 0;
}
/* Allocate mixing buffer */
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
DISKAUD_CloseDevice(this);
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
this->hidden->mixlen = this->spec.size;
this->hidden->write_delay =
(envr) ? SDL_atoi(envr) : DISKDEFAULT_WRITEDELAY;
#if HAVE_STDIO_H
fprintf(stderr,
"WARNING: You are using the SDL disk writer audio driver!\n"
" Writing to file [%s].\n", fname);
#endif
/* We're ready to rock and roll. :-) */
return 1;
}
示例6: QSA_OpenDevice
//.........这里部分代码省略.........
/* assumes test_format not 0 on success */
if (test_format == 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Set the audio format */
cparams.format.format = format;
/* Set mono/stereo/4ch/6ch/8ch audio */
cparams.format.voices = this->spec.channels;
/* Set rate */
cparams.format.rate = this->spec.freq;
/* Setup the transfer parameters according to cparams */
status = snd_pcm_plugin_params(this->hidden->audio_handle, &cparams);
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_channel_params", status);
}
/* Make sure channel is setup right one last time */
SDL_memset(&csetup, 0, sizeof(csetup));
if (!this->hidden->iscapture) {
csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
} else {
csetup.channel = SND_PCM_CHANNEL_CAPTURE;
}
/* Setup an audio channel */
if (snd_pcm_plugin_setup(this->hidden->audio_handle, &csetup) < 0) {
QSA_CloseDevice(this);
return SDL_SetError("QSA: Unable to setup channel");
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->pcm_len = this->spec.size;
if (this->hidden->pcm_len == 0) {
this->hidden->pcm_len =
csetup.buf.block.frag_size * this->spec.channels *
(snd_pcm_format_width(format) / 8);
}
/*
* Allocate memory to the audio buffer and initialize with silence
* (Note that buffer size must be a multiple of fragment size, so find
* closest multiple)
*/
this->hidden->pcm_buf =
(Uint8 *) SDL_AllocAudioMem(this->hidden->pcm_len);
if (this->hidden->pcm_buf == NULL) {
QSA_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->pcm_buf, this->spec.silence,
this->hidden->pcm_len);
/* get the file descriptor */
if (!this->hidden->iscapture) {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
this->hidden->audio_fd =
snd_pcm_file_descriptor(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (this->hidden->audio_fd < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_file_descriptor", status);
}
/* Prepare an audio channel */
if (!this->hidden->iscapture) {
/* Prepare audio playback */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_PLAYBACK);
} else {
/* Prepare audio capture */
status =
snd_pcm_plugin_prepare(this->hidden->audio_handle,
SND_PCM_CHANNEL_CAPTURE);
}
if (status < 0) {
QSA_CloseDevice(this);
return QSA_SetError("snd_pcm_plugin_prepare", status);
}
/* We're really ready to rock and roll. :-) */
return 0;
}
示例7: SUNAUDIO_OpenDevice
//.........这里部分代码省略.........
#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;
}
示例8: AL_OpenAudio
static int AL_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
Uint16 test_format = SDL_FirstAudioFormat(spec->format);
long width = 0;
long fmt = 0;
int valid = 0;
#ifdef OLD_IRIX_AUDIO
{
long audio_param[2];
audio_param[0] = AL_OUTPUT_RATE;
audio_param[1] = spec->freq;
valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0);
}
#else
{
ALpv audio_param;
audio_param.param = AL_RATE;
audio_param.value.i = spec->freq;
valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0);
}
#endif
while ((!valid) && (test_format)) {
valid = 1;
spec->format = test_format;
switch (test_format) {
case AUDIO_S8:
width = AL_SAMPLE_8;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_S16SYS:
width = AL_SAMPLE_16;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
default:
valid = 0;
test_format = SDL_NextAudioFormat();
break;
}
if (valid) {
ALconfig audio_config = alNewConfig();
valid = 0;
if (audio_config) {
if (alSetChannels(audio_config, spec->channels) < 0) {
if (spec->channels > 2) { /* can't handle > stereo? */
spec->channels = 2; /* try again below. */
}
}
if ((alSetSampFmt(audio_config, fmt) >= 0) &&
((!width) || (alSetWidth(audio_config, width) >= 0)) &&
(alSetQueueSize(audio_config, spec->samples * 2) >= 0) &&
(alSetChannels(audio_config, spec->channels) >= 0)) {
audio_port = alOpenPort("SDL audio", "w", audio_config);
if (audio_port == NULL) {
/* docs say AL_BAD_CHANNELS happens here, too. */
int err = oserror();
if (err == AL_BAD_CHANNELS) {
spec->channels = 2;
alSetChannels(audio_config, spec->channels);
audio_port = alOpenPort("SDL audio", "w",
audio_config);
}
}
if (audio_port != NULL) {
valid = 1;
}
}
alFreeConfig(audio_config);
}
}
}
if (!valid) {
SDL_SetError("Unsupported audio format");
return (-1);
}
/* 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);
}
示例9: DSP_OpenAudio
//.........这里部分代码省略.........
switch ( test_format ) {
case AUDIO_U8:
if ( value & AFMT_U8 ) {
format = AFMT_U8;
}
break;
case AUDIO_S8:
if ( value & AFMT_S8 ) {
format = AFMT_S8;
}
break;
case AUDIO_S16LSB:
if ( value & AFMT_S16_LE ) {
format = AFMT_S16_LE;
}
break;
case AUDIO_S16MSB:
if ( value & AFMT_S16_BE ) {
format = AFMT_S16_BE;
}
break;
case AUDIO_U16LSB:
if ( value & AFMT_U16_LE ) {
format = AFMT_U16_LE;
}
break;
case AUDIO_U16MSB:
if ( value & AFMT_U16_BE ) {
format = AFMT_U16_BE;
}
break;
default:
format = 0;
break;
}
if ( ! format ) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
/* Set the audio format */
value = format;
if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format) ) {
SDL_SetError("Couldn't set audio format");
return(-1);
}
/* Set the number of channels of output */
value = spec->channels;
#ifdef SNDCTL_DSP_CHANNELS
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
#endif
value = (spec->channels > 1);
ioctl(audio_fd, SNDCTL_DSP_STEREO, &value);
value = (value ? 2 : 1);
#ifdef SNDCTL_DSP_CHANNELS
}
#endif
spec->channels = value;
/* Because some drivers don't allow setting the buffer size
after setting the format, we must re-open the audio device
once we know what format and channels are supported
*/
if ( DSP_ReopenAudio(this, audiodev, format, spec) < 0 ) {
/* Error is set by DSP_ReopenAudio() */
return(-1);
}
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
#ifndef USE_BLOCKING_WRITES
/* Check to see if we need to use select() workaround */
{ char *workaround;
workaround = getenv("SDL_DSP_NOSELECT");
if ( workaround ) {
frame_ticks = (float)(spec->samples*1000)/spec->freq;
next_frame = SDL_GetTicks()+frame_ticks;
}
}
#endif /* !USE_BLOCKING_WRITES */
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
示例10: ESD_OpenDevice
static int
ESD_OpenDevice(_THIS, const char *devname, int iscapture)
{
esd_format_t format = (ESD_STREAM | ESD_PLAY);
SDL_AudioFormat test_format = 0;
int found = 0;
/* 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->audio_fd = -1;
/* Convert audio spec to the ESD audio format */
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!found && test_format; test_format = SDL_NextAudioFormat()) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
found = 1;
switch (test_format) {
case AUDIO_U8:
format |= ESD_BITS8;
break;
case AUDIO_S16SYS:
format |= ESD_BITS16;
break;
default:
found = 0;
break;
}
}
if (!found) {
ESD_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
if (this->spec.channels == 1) {
format |= ESD_MONO;
} else {
format |= ESD_STEREO;
}
#if 0
this->spec.samples = ESD_BUF_SIZE; /* Darn, no way to change this yet */
#endif
/* Open a connection to the ESD audio server */
this->hidden->audio_fd =
SDL_NAME(esd_play_stream) (format, this->spec.freq, NULL,
get_progname());
if (this->hidden->audio_fd < 0) {
ESD_CloseDevice(this);
return SDL_SetError("Couldn't open ESD connection");
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(&this->spec);
this->hidden->frame_ticks =
(float) (this->spec.samples * 1000) / this->spec.freq;
this->hidden->next_frame = SDL_GetTicks() + this->hidden->frame_ticks;
/* Allocate mixing buffer */
this->hidden->mixlen = this->spec.size;
this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen);
if (this->hidden->mixbuf == NULL) {
ESD_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* Get the parent process id (we're the parent of the audio thread) */
this->hidden->parent = getpid();
/* We're ready to rock and roll. :-) */
return 0;
}
示例11: SDL_FS_OpenDevice
//.........这里部分代码省略.........
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));
/* Try for a closest match on audio format */
for (test_format = SDL_FirstAudioFormat(this->spec.format);
!format && test_format;) {
#ifdef DEBUG_AUDIO
fprintf(stderr, "Trying format 0x%4.4x\n", test_format);
#endif
switch (test_format) {
case AUDIO_U8:
fs_format = FSSF_U8;
bytes = 1;
format = 1;
break;
case AUDIO_S16SYS:
fs_format = FSSF_S16;
bytes = 2;
format = 1;
break;
case AUDIO_S32SYS:
fs_format = FSSF_S32;
bytes = 4;
format = 1;
break;
case AUDIO_F32SYS:
fs_format = FSSF_FLOAT;
bytes = 4;
format = 1;
break;
default:
format = 0;
break;
}
if (!format) {
test_format = SDL_NextAudioFormat();
}
}
if (format == 0) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Couldn't find any hardware audio formats");
}
this->spec.format = test_format;
/* Retrieve the main sound interface. */
ret = SDL_NAME(FusionSoundCreate) (&this->hidden->fs);
if (ret) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Unable to initialize FusionSound: %d", ret);
}
this->hidden->mixsamples = this->spec.size / bytes / this->spec.channels;
/* Fill stream description. */
desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT | FSSDF_PREBUFFER;
desc.samplerate = this->spec.freq;
desc.buffersize = this->spec.size * FUSION_BUFFERS;
desc.channels = this->spec.channels;
desc.prebuffer = 10;
desc.sampleformat = fs_format;
ret =
this->hidden->fs->CreateStream(this->hidden->fs, &desc,
&this->hidden->stream);
if (ret) {
SDL_FS_CloseDevice(this);
return SDL_SetError("Unable to create FusionSoundStream: %d", ret);
}
/* See what we got */
desc.flags = FSSDF_SAMPLERATE | FSSDF_BUFFERSIZE |
FSSDF_CHANNELS | FSSDF_SAMPLEFORMAT;
ret = this->hidden->stream->GetDescription(this->hidden->stream, &desc);
this->spec.freq = desc.samplerate;
this->spec.size =
desc.buffersize / FUSION_BUFFERS * bytes * desc.channels;
this->spec.channels = desc.channels;
/* 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) {
SDL_FS_CloseDevice(this);
return SDL_OutOfMemory();
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 0;
}
示例12: DSP_OpenAudio
//.........这里部分代码省略.........
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);
}
示例13: DSP_OpenAudio
//.........这里部分代码省略.........
if ( value & AFMT_S8 ) {
format = AFMT_S8;
}
break;
case AUDIO_U16LSB:
if ( value & AFMT_U16_LE ) {
format = AFMT_U16_LE;
}
break;
case AUDIO_U16MSB:
if ( value & AFMT_U16_BE ) {
format = AFMT_U16_BE;
}
break;
#endif
default:
format = 0;
break;
}
if ( ! format ) {
test_format = SDL_NextAudioFormat();
}
}
if ( format == 0 ) {
SDL_SetError("Couldn't find any hardware audio formats");
return(-1);
}
spec->format = test_format;
/* Set the audio format */
value = format;
if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
(value != format) ) {
perror("SNDCTL_DSP_SETFMT");
SDL_SetError("Couldn't set audio format");
return(-1);
}
/* Set the number of channels of output */
value = spec->channels;
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
perror("SNDCTL_DSP_CHANNELS");
SDL_SetError("Cannot set the number of channels");
return(-1);
}
spec->channels = value;
/* Set the DSP frequency */
value = spec->freq;
if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) {
perror("SNDCTL_DSP_SPEED");
SDL_SetError("Couldn't set audio frequency");
return(-1);
}
spec->freq = value;
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Determine the power of two of the fragment size */
for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
if ( (0x01<<frag_spec) != spec->size ) {
SDL_SetError("Fragment size must be a power of two");
return(-1);
}
frag_spec |= 0x00020000; /* two fragments, for low latency */
/* Set the audio buffering parameters */
#ifdef DEBUG_AUDIO
fprintf(stderr, "Requesting %d fragments of size %d\n",
(frag_spec >> 16), 1<<(frag_spec&0xFFFF));
#endif
if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) {
perror("SNDCTL_DSP_SETFRAGMENT");
fprintf(stderr, "Warning: Couldn't set audio fragment size\n");
}
#ifdef DEBUG_AUDIO
{ audio_buf_info info;
ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info);
fprintf(stderr, "fragments = %d\n", info.fragments);
fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
fprintf(stderr, "fragsize = %d\n", info.fragsize);
fprintf(stderr, "bytes = %d\n", info.bytes);
}
#endif
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
memset(mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* We're ready to rock and roll. :-) */
return(0);
}
示例14: IRIXAUDIO_OpenDevice
//.........这里部分代码省略.........
valid = (ALsetparams(AL_DEFAULT_DEVICE, audio_param, 2) < 0);
}
#else
{
ALpv audio_param;
audio_param.param = AL_RATE;
audio_param.value.i = this->spec.freq;
valid = (alSetParams(AL_DEFAULT_OUTPUT, &audio_param, 1) < 0);
}
#endif
while ((!valid) && (test_format)) {
valid = 1;
this->spec.format = test_format;
switch (test_format) {
case AUDIO_S8:
width = AL_SAMPLE_8;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_S16SYS:
width = AL_SAMPLE_16;
fmt = AL_SAMPFMT_TWOSCOMP;
break;
case AUDIO_F32SYS:
width = 0; /* not used here... */
fmt = AL_SAMPFMT_FLOAT;
break;
/* Docs say there is int24, but not int32.... */
default:
valid = 0;
test_format = SDL_NextAudioFormat();
break;
}
if (valid) {
ALconfig audio_config = alNewConfig();
valid = 0;
if (audio_config) {
if (alSetChannels(audio_config, this->spec.channels) < 0) {
if (this->spec.channels > 2) { /* can't handle > stereo? */
this->spec.channels = 2; /* try again below. */
}
}
if ((alSetSampFmt(audio_config, fmt) >= 0) &&
((!width) || (alSetWidth(audio_config, width) >= 0)) &&
(alSetQueueSize(audio_config, this->spec.samples * 2) >=
0)
&& (alSetChannels(audio_config, this->spec.channels) >=
0)) {
this->hidden->audio_port = alOpenPort("SDL audio", "w",
audio_config);
if (this->hidden->audio_port == NULL) {
/* docs say AL_BAD_CHANNELS happens here, too. */
int err = oserror();
if (err == AL_BAD_CHANNELS) {
this->spec.channels = 2;
alSetChannels(audio_config, this->spec.channels);
this->hidden->audio_port =
alOpenPort("SDL audio", "w", audio_config);
}
}
if (this->hidden->audio_port != NULL) {
valid = 1;
}
}
alFreeConfig(audio_config);
}
}
}
if (!valid) {
IRIXAUDIO_CloseDevice(this);
SDL_SetError("Unsupported audio format");
return 0;
}
/* 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) {
IRIXAUDIO_CloseDevice(this);
SDL_OutOfMemory();
return 0;
}
SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size);
/* We're ready to rock and roll. :-) */
return 1;
}
示例15: PULSE_OpenAudio
static int PULSE_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
int state;
Uint16 test_format;
pa_sample_spec paspec;
pa_buffer_attr paattr;
pa_channel_map pacmap;
pa_stream_flags_t flags = 0;
paspec.format = PA_SAMPLE_INVALID;
for ( test_format = SDL_FirstAudioFormat(spec->format); test_format; ) {
switch ( test_format ) {
case AUDIO_U8:
paspec.format = PA_SAMPLE_U8;
break;
case AUDIO_S16LSB:
paspec.format = PA_SAMPLE_S16LE;
break;
case AUDIO_S16MSB:
paspec.format = PA_SAMPLE_S16BE;
break;
}
if ( paspec.format != PA_SAMPLE_INVALID )
break;
}
if (paspec.format == PA_SAMPLE_INVALID ) {
SDL_SetError("Couldn't find any suitable audio formats");
return(-1);
}
spec->format = test_format;
paspec.channels = spec->channels;
paspec.rate = spec->freq;
/* Calculate the final parameters for this audio specification */
#ifdef PA_STREAM_ADJUST_LATENCY
spec->samples /= 2; /* Mix in smaller chunck to avoid underruns */
#endif
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
if ( mixbuf == NULL ) {
return(-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* Reduced prebuffering compared to the defaults. */
#ifdef PA_STREAM_ADJUST_LATENCY
paattr.tlength = mixlen * 4; /* 2x original requested bufsize */
paattr.prebuf = -1;
paattr.maxlength = -1;
paattr.minreq = mixlen; /* -1 can lead to pa_stream_writable_size()
>= mixlen never becoming true */
flags = PA_STREAM_ADJUST_LATENCY;
#else
paattr.tlength = mixlen*2;
paattr.prebuf = mixlen*2;
paattr.maxlength = mixlen*2;
paattr.minreq = mixlen;
#endif
/* The SDL ALSA output hints us that we use Windows' channel mapping */
/* http://bugzilla.libsdl.org/show_bug.cgi?id=110 */
SDL_NAME(pa_channel_map_init_auto)(
&pacmap, spec->channels, PA_CHANNEL_MAP_WAVEEX);
/* Set up a new main loop */
if (!(mainloop = SDL_NAME(pa_mainloop_new)())) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_new() failed");
return(-1);
}
mainloop_api = SDL_NAME(pa_mainloop_get_api)(mainloop);
if (!(context = SDL_NAME(pa_context_new)(mainloop_api, get_progname()))) {
PULSE_CloseAudio(this);
SDL_SetError("pa_context_new() failed");
return(-1);
}
/* Connect to the PulseAudio server */
if (SDL_NAME(pa_context_connect)(context, NULL, 0, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("Could not setup connection to PulseAudio");
return(-1);
}
do {
if (SDL_NAME(pa_mainloop_iterate)(mainloop, 1, NULL) < 0) {
PULSE_CloseAudio(this);
SDL_SetError("pa_mainloop_iterate() failed");
return(-1);
}
state = SDL_NAME(pa_context_get_state)(context);
if (!PA_CONTEXT_IS_GOOD(state)) {
PULSE_CloseAudio(this);
SDL_SetError("Could not connect to PulseAudio");
return(-1);
//.........这里部分代码省略.........