int audriv_write(char *buff, int n)
/* audio に buff を n バイト分流し込みます．
* audriv_set_noblock_write() で非ブロック・モードが設定された
* 場合は，この関数の呼び出しは即座に処理が返ります．
* 返り値は実際に流し込まれたバイト数であり，非ブロック・モードが設定
* されている場合は，引数 n より少ない場合があります．
* 失敗すると -1 を返し，成功すると，実際に流し込まれたバイト数を返します．
*/
{
n /= play_frame_width;
if(audio_write_noblocking)
{
int size;
size = alGetFillable(out);
if(size < n)
n = size;
}
add_sample_counter(n);
if(play_encoding != AENC_G711_ULAW)
{
alWriteFrames(out, buff, n);
return n * play_frame_width;
}
else
{
/* AENC_G711_ULAW */
int i, m, ret;
short samps[BUFSIZ];
n *= play_frame_width;
ret = n;
while(n > 0)
{
m = n;
if(m > BUFSIZ)
m = BUFSIZ;
for(i = 0; i < m; i++)
samps[i] = AUDIO_U2S(buff[i]);
alWriteFrames(out, samps, m / play_frame_width);
buff += m;
n -= m;
}
return ret;
}
}
void WritePlay(
struct FFTSound *fftsound,
void *port,double **buffer,int size
){
struct Sgiplay *sgiplay=(struct Sgiplay *)port;
alWriteFrames(sgiplay->out_port, buffer, size);
}
void
ALAudioDevice::update() {
ADR_GUARD("ALAudioDevice::update");
// how much data can we write?
const int filled = alGetFilled(m_port);
int can_write = 5000 - filled; // empty portion of the buffer
// write 1024 frames at a time
static const int BUFFER_SIZE = 1024;
u8 buffer[BUFFER_SIZE * 4];
while (can_write > 0) {
int transfer_count = std::min(can_write, BUFFER_SIZE);
ADR_LOG("reading");
read(transfer_count, buffer);
ADR_LOG("writing");
alWriteFrames(m_port, buffer, transfer_count);
can_write -= transfer_count;
}
usleep(50000); // 50 milliseconds
}
static void AL_PlayAudio(_THIS)
{
/* Write the audio data out */
if ( alWriteFrames(audio_port, mixbuf, this->spec.samples) < 0 ) {
/* Assume fatal error, for now */
this->enabled = 0;
}
}
static void
IRIXAUDIO_PlayDevice(_THIS)
{
/* Write the audio data out */
ALport port = this->hidden->audio_port;
Uint8 *mixbuf = this->hidden->mixbuf;
if (alWriteFrames(port, mixbuf, this->spec.samples) < 0) {
/* Assume fatal error, for now */
this->enabled = 0;
}
}
static int
irix_dsp_write(struct sysdep_dsp_struct *dsp, unsigned char *data, int count)
{
struct irix_dsp_priv_data *priv = dsp->_priv;
int playcnt;
int maxsize;
/*
* We write as many samples as possible (up to count) without blocking
*/
maxsize = alGetFillable(priv->devAudio);
playcnt = (count <= maxsize) ? count : maxsize;
alWriteFrames(priv->devAudio, data, playcnt);
return playcnt;
}
void SNDDMA_Submit(void)
{
int nFillable, nFilled, nPos;
int nFrames, nFramesLeft;
unsigned endtime;
if (!sgisnd_aport) return;
nFillable = alGetFillable( sgisnd_aport );
nFilled = QSND_BUFFER_FRAMES - nFillable;
nFrames = dma.samples >> (dma.channels - 1);
if (paintedtime - soundtime < nFrames)
nFrames = paintedtime - soundtime;
if (nFrames <= QSND_SKID) return;
nPos = UST_TO_BUFFPOS( sgisnd_startframe );
// dump re-written contents of the buffer
if (sgisnd_lastframewritten > sgisnd_startframe)
{
alDiscardFrames( sgisnd_aport, sgisnd_lastframewritten - sgisnd_startframe );
}
else if ((int)(sgisnd_startframe - sgisnd_lastframewritten) >= QSND_BUFFER_FRAMES)
{
// blow away everything if we've underflowed
alDiscardFrames( sgisnd_aport, QSND_BUFFER_FRAMES );
}
// don't block
if (nFrames > nFillable) nFrames = nFillable;
// account for stereo
nFramesLeft = nFrames;
if (nPos + nFrames * dma.channels > QSND_BUFFER_SIZE)
{
int nFramesAtEnd = (QSND_BUFFER_SIZE - nPos) >> (dma.channels - 1);
alWriteFrames( sgisnd_aport, &dma_buffer[nPos], nFramesAtEnd );
nPos = 0;
nFramesLeft -= nFramesAtEnd;
}
void
sgi_play(void)
{
struct audio_packet *packet;
ssize_t len;
unsigned int i;
STREAM out;
int gf;
while (1)
{
if (rdpsnd_queue_empty())
return;
packet = rdpsnd_queue_current_packet();
out = &packet->s;
len = out->end - out->p;
alWriteFrames(output_port, out->p, len / combinedFrameSize);
out->p += len;
if (out->p == out->end)
{
gf = alGetFilled(output_port);
if (gf < (4 * maxFillable / 10))
{
rdpsnd_queue_next(0);
}
else
{
#if (defined(IRIX_DEBUG))
/* fprintf(stderr,"Busy playing...\n"); */
#endif
usleep(10);
return;
}
}
}
}
main (int argc, char **argv)
{
AFfilehandle file;
AFframecount count, frameCount;
int channelCount, sampleFormat, sampleWidth;
float frameSize;
void *buffer;
double sampleRate;
ALport outport;
ALconfig outportconfig;
if (argc < 2)
usage();
file = afOpenFile(argv[1], "r", NULL);
if (file == AF_NULL_FILEHANDLE)
{
fprintf(stderr, "Could not open file %s.\n", argv[1]);
exit(EXIT_FAILURE);
}
frameCount = afGetFrameCount(file, AF_DEFAULT_TRACK);
frameSize = afGetVirtualFrameSize(file, AF_DEFAULT_TRACK, 1);
channelCount = afGetVirtualChannels(file, AF_DEFAULT_TRACK);
sampleRate = afGetRate(file, AF_DEFAULT_TRACK);
afGetVirtualSampleFormat(file, AF_DEFAULT_TRACK, &sampleFormat,
&sampleWidth);
if (sampleFormat == AF_SAMPFMT_UNSIGNED)
{
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, sampleWidth);
}
printf("frame count: %lld\n", frameCount);
printf("frame size: %d bytes\n", (int) frameSize);
printf("channel count: %d\n", channelCount);
printf("sample rate: %.2f Hz\n", sampleRate);
buffer = malloc(BUFFERED_FRAME_COUNT * frameSize);
outportconfig = alNewConfig();
setwidth(outportconfig, sampleWidth);
setsampleformat(outportconfig, sampleFormat);
alSetChannels(outportconfig, channelCount);
count = afReadFrames(file, AF_DEFAULT_TRACK, buffer, BUFFERED_FRAME_COUNT);
outport = alOpenPort("irixread", "w", outportconfig);
setrate(outport, sampleRate);
do
{
printf("count = %lld\n", count);
alWriteFrames(outport, buffer, count);
count = afReadFrames(file, AF_DEFAULT_TRACK, buffer,
BUFFERED_FRAME_COUNT);
} while (count > 0);
waitport(outport);
alClosePort(outport);
alFreeConfig(outportconfig);
afCloseFile(file);
}
void
wave_out_play(void)
{
struct audio_packet *packet;
ssize_t len;
unsigned int i;
uint8 swap;
STREAM out;
static BOOL swapped = False;
int gf;
while (1)
{
if (queue_lo == queue_hi)
{
This->dsp_bu = False;
return;
}
packet = &packet_queue[queue_lo];
out = &packet->s;
/* Swap the current packet, but only once */
if (g_swapaudio && !swapped)
{
for (i = 0; i < out->end - out->p; i += 2)
{
swap = *(out->p + i);
*(out->p + i) = *(out->p + i + 1);
*(out->p + i + 1) = swap;
}
swapped = True;
}
len = out->end - out->p;
alWriteFrames(output_port, out->p, len / combinedFrameSize);
out->p += len;
if (out->p == out->end)
{
gf = alGetFilled(output_port);
if (gf < (4 * maxFillable / 10))
{
rdpsnd_send_completion(packet->tick, packet->index);
free(out->data);
queue_lo = (queue_lo + 1) % MAX_QUEUE;
swapped = False;
}
else
{
#if (defined(IRIX_DEBUG))
/* fprintf(stderr,"Busy playing...\n"); */
#endif
This->dsp_bu = True;
usleep(10);
return;
}
}
}
}
static int al_play (ao_instance_t * _instance, int flags, sample_t * _samples)
{
al_instance_t * instance = (al_instance_t *) _instance;
int16_t int16_samples[256*6];
int chans = -1;
#ifdef LIBDTS_DOUBLE
convert_t samples[256 * 6];
int i;
for (i = 0; i < 256 * 6; i++)
samples[i] = _samples[i];
#else
convert_t * samples = _samples;
#endif
chans = channels_multi (flags);
flags &= DTS_CHANNEL_MASK | DTS_LFE;
if (instance->set_params) {
ALconfig config;
ALpv params[2];
config = alNewConfig ();
if (!config) {
fprintf (stderr, "alNewConfig failed\n");
return 1;
}
if (alSetChannels (config, chans)) {
fprintf (stderr, "alSetChannels failed\n");
return 1;
}
if (alSetConfig (instance->port, config)) {
fprintf (stderr, "alSetConfig failed\n");
return 1;
}
alFreeConfig (config);
params[0].param = AL_MASTER_CLOCK;
params[0].value.i = AL_CRYSTAL_MCLK_TYPE;
params[1].param = AL_RATE;
params[1].value.ll = alIntToFixed (instance->sample_rate);
if (alSetParams (alGetResource (instance->port), params, 2) < 0) {
fprintf (stderr, "alSetParams failed\n");
return 1;
}
instance->flags = flags;
instance->set_params = 0;
} else if ((flags == DTS_DOLBY) && (instance->flags == DTS_STEREO)) {
fprintf (stderr, "Switching from stereo to dolby surround\n");
instance->flags = DTS_DOLBY;
} else if ((flags == DTS_STEREO) && (instance->flags == DTS_DOLBY)) {
fprintf (stderr, "Switching from dolby surround to stereo\n");
instance->flags = DTS_STEREO;
} else if (flags != instance->flags)
return 1;
convert2s16_multi (samples, int16_samples, flags);
alWriteFrames (instance->port, int16_samples, 256);
return 0;
}
main()
{
LS_DATA ls_data;
FILE *fp;
double gains[MAX_CHANNELS];
int azimuth=-10;
int elevation=14;
double *gainptr, gain;
short *inptr,*outptr;
short out[BUFFER_LENGTH][MAX_CHANNELS];
ALconfig c;
AFfilehandle fh;
ALport p;
short *in;
long frames;
int i,j,k;
int numchannels = 8; /* change this according your output device*/
int ls_set_dim = 3;
int ls_num = 8;
int ls_dirs[MAX_FIELD_AM]={-30,0, 30,0, -45,45, 45,45, -90,0,
90,0, 180,0, 180,45};
/* change these according to your loudspeaker positioning*/
/* defining loudspeaker data */
define_loudspeakers(&ls_data, ls_set_dim, ls_num, ls_dirs);
/* ls_data is a struct containing matrices etc
ls_set_dim is 2 if loudspeakers are on a (horizontal) plane
ls_set_dim is 3 if also elevated or descended loudpeakers exist
ls_num is the number of loudspeakers
ls_dirs is an array containing the angular directions of loudsp*/
/* gain factors for virtual source in direction
(int azimuth, int elevation) */
vbap(gains, &ls_data, azimuth, elevation);
/* panning monophonic stream float *in
to multiple outputs float *out[]
with gain factors float *gains */
/* input audio*/
if((fh=afOpenFile("myaiff.aiff","r",0))==NULL){
fprintf(stderr, "Could not open file myaiff.aiff\n");
exit(-1);
}
frames=AFgetframecnt(fh,AF_DEFAULT_TRACK);
if(afGetChannels(fh, AF_DEFAULT_TRACK) != 1){
fprintf(stderr, "Supports only mono aiff-files\n");
exit(-1);
}
in= malloc(frames*sizeof(short)*2);
afReadFrames(fh,AF_DEFAULT_TRACK,in,frames);
/*opening the audio port*/
c = alNewConfig();
if (!c) {
printf("Couldn't create ALconfig:%s\n", alGetErrorString(oserror()));
exit(-1);
}
alSetChannels(c,numchannels);
ALsetqueuesize(c,BUFFER_LENGTH);
p = alOpenPort("alVBAP example","w",c);
if (!p) {
printf("port open failed:%s\n", alGetErrorString(oserror()));
exit(-1);
}
fprintf(stderr,"\nPanning audio");
for(j=0;j<(frames/BUFFER_LENGTH);j++){
inptr = &(in[j*BUFFER_LENGTH]); /* audio to be panned */
outptr= out[0];
for (i=0; i<BUFFER_LENGTH; i++){ /* panning */
gainptr=gains;
for (k=0; k<numchannels; k++){
*outptr++ = (short) ((double) *inptr * *gainptr++);
}
inptr++;
}
alWriteFrames(p, out, BUFFER_LENGTH); /* write frames */
fprintf(stderr,".");
}
/*write rest samples out*/
inptr = &(in[j*BUFFER_LENGTH]); /* partial buffer */
outptr= out[0];
for (i=0; i<(frames-BUFFER_LENGTH*j); i++){ /* panning */
gainptr=gains;
for (k=0; k<numchannels; k++){
*outptr++ = (short) ((double) *inptr * *gainptr++);
}
inptr++;
}
for (;i<BUFFER_LENGTH; i++){ /* zeros */
for (k=0; k<numchannels; k++){
*outptr++ = 0;
}
}
alWriteFrames(p, out, BUFFER_LENGTH); /* write frames */
fprintf(stderr,".");
printf("\n\nDone!\n\n");
}