int32 ad_close (ad_rec_t *r)
{
if (r->audio)
alClosePort(r->audio);
free(r);
return 0;
}
void ClosePlay(void *port){
struct Sgiplay *sgiplay=(struct Sgiplay *)port;
type j;
j = alClosePort(sgiplay->out_port);
j = alFreeConfig(sgiplay->out_c);
free(sgiplay);
}
static void AL_CloseAudio(_THIS)
{
if ( mixbuf != NULL ) {
SDL_FreeAudioMem(mixbuf);
mixbuf = NULL;
}
if ( audio_port != NULL ) {
alClosePort(audio_port);
audio_port = NULL;
}
}
void audriv_play_close(void)
/* 演奏用にオープンされた audio を閉じます．すでに閉じている
* 場合はなにも行いません．
*/
{
if(!out)
return;
while(audriv_play_active() == 1)
audriv_wait_play();
alClosePort(out);
out = NULL;
}
static void
IRIXAUDIO_CloseDevice(_THIS)
{
if (this->hidden != NULL) {
if (this->hidden->mixbuf != NULL) {
SDL_FreeAudioMem(this->hidden->mixbuf);
this->hidden->mixbuf = NULL;
}
if (this->hidden->audio_port != NULL) {
alClosePort(this->hidden->audio_port);
this->hidden->audio_port = NULL;
}
SDL_free(this->hidden);
this->hidden = NULL;
}
}
long audriv_play_stop(void)
/* 演奏を即座に停止し，停止直前のサンプル数を返します．
* audriv_play_stop() の呼び出しによって，audio は閉じます．
* audio が既に閉じている場合に audriv_play_stop() を呼び出した場合は 0 を
* 返します．
* エラーの場合は -1 を返します．
*/
{
long samp;
if(!out)
return 0;
samp = audriv_play_samples();
alClosePort(out);
reset_samples = play_counter = 0;
out = NULL;
return samp;
}
static void
irix_dsp_destroy(struct sysdep_dsp_struct *dsp)
{
struct irix_dsp_priv_data *priv = dsp->_priv;
#ifdef IRIX_DEBUG
fprintf(stderr, "Destroying sound channel.\n");
#endif
if (priv)
{
if (priv->port_status >= 0)
alClosePort(priv->devAudio);
free(priv);
}
free(dsp);
}
void audriv_free_audio(void)
/* audio の後処理を行います．
*/
{
if(!out_config)
return;
if(out != NULL)
{
alClosePort(out);
out = NULL;
}
alFreeConfig(out_config);
out_config = NULL;
if(out_config != NULL)
alFreeConfig(out_config);
out_config = NULL;
}
// close audio device
static void uninit(int immed) {
/* TODO: samplerate should be set back to the value before mplayer was started! */
mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit);
if (ao_config) {
alFreeConfig(ao_config);
ao_config = NULL;
}
if (ao_port) {
if (!immed)
while(alGetFilled(ao_port) > 0) sginap(1);
alClosePort(ao_port);
ao_port = NULL;
}
}
void
sgi_close(void)
{
/* Ack all remaining packets */
#if (defined(IRIX_DEBUG))
fprintf(stderr, "sgi_close: begin\n");
#endif
while (!rdpsnd_queue_empty())
rdpsnd_queue_next(0);
alDiscardFrames(output_port, 0);
alClosePort(output_port);
output_port = (ALport) 0;
alFreeConfig(audioconfig);
#if (defined(IRIX_DEBUG))
fprintf(stderr, "sgi_close: returning\n");
#endif
}
void
wave_out_close(void)
{
/* Ack all remaining packets */
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_close: begin\n");
#endif
while (queue_lo != queue_hi)
{
rdpsnd_send_completion(packet_queue[queue_lo].tick, packet_queue[queue_lo].index);
free(packet_queue[queue_lo].s.data);
queue_lo = (queue_lo + 1) % MAX_QUEUE;
}
alDiscardFrames(output_port, 0);
alClosePort(output_port);
alFreeConfig(audioconfig);
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_close: returning\n");
#endif
}
Bool audriv_play_open(void)
/* audio を演奏用に開き，いつでも audriv_write() により演奏可能な
* 状態にします．既に開いている場合はなにも行いません．
* 成功した場合は True を，失敗した場合は False を返します．
*/
{
if(out)
return True;
if(audriv_al_set_width(out_config, play_encoding) < 0)
{
audriv_err(ALERROR);
return False;
}
alSetChannels(out_config, play_nchannels);
out = alOpenPort("audriv", "w", out_config);
if(!out)
{
audriv_err(ALERROR);
return False;
}
if(audriv_al_set_rate(out, play_sample_rate) == False)
{
audriv_err(ALERROR);
alClosePort(out);
out = NULL;
return False;
}
alSetConfig(out, out_config);
reset_samples = 0;
play_counter = 0;
return True;
}
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);
}
BOOL
wave_out_set_format(WAVEFORMATEX * pwfx)
{
int channels;
int frameSize, channelCount;
ALpv params;
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: init...\n");
#endif
g_swapaudio = False;
if (pwfx->wBitsPerSample == 8)
width = AL_SAMPLE_8;
else if (pwfx->wBitsPerSample == 16)
{
width = AL_SAMPLE_16;
/* Do we need to swap the 16bit values? (Are we BigEndian) */
#if (defined(B_ENDIAN))
g_swapaudio = 1;
#else
g_swapaudio = 0;
#endif
}
/* Limited support to configure an opened audio port in IRIX. The
number of channels is a static setting and can not be changed after
a port is opened. So if the number of channels remains the same, we
can configure other settings; otherwise we have to reopen the audio
port, using same config. */
channels = pwfx->nChannels;
g_snd_rate = pwfx->nSamplesPerSec;
alSetSampFmt(audioconfig, AL_SAMPFMT_TWOSCOMP);
alSetWidth(audioconfig, width);
if (channels != alGetChannels(audioconfig))
{
alClosePort(output_port);
alSetChannels(audioconfig, channels);
output_port = alOpenPort("rdpsnd", "w", audioconfig);
if (output_port == (ALport) 0)
{
fprintf(stderr, "wave_out_set_format: alOpenPort failed: %s\n",
alGetErrorString(oserror()));
return False;
}
}
resource = alGetResource(output_port);
maxFillable = alGetFillable(output_port);
channelCount = alGetChannels(audioconfig);
frameSize = alGetWidth(audioconfig);
if (frameSize == 0 || channelCount == 0)
{
fprintf(stderr, "wave_out_set_format: bad frameSize or channelCount\n");
return False;
}
combinedFrameSize = frameSize * channelCount;
params.param = AL_RATE;
params.value.ll = (long long) g_snd_rate << 32;
if (alSetParams(resource, ¶ms, 1) < 0)
{
fprintf(stderr, "wave_set_format: alSetParams failed: %s\n",
alGetErrorString(oserror()));
return False;
}
if (params.sizeOut < 0)
{
fprintf(stderr, "wave_set_format: invalid rate %d\n", g_snd_rate);
return False;
}
#if (defined(IRIX_DEBUG))
fprintf(stderr, "wave_out_set_format: returning...\n");
#endif
return True;
}
/*
==============
SNDDMA_Shutdown
Reset the sound device for exiting
===============
*/
void SNDDMA_Shutdown(void)
{
if (sgisnd_aport) alClosePort( sgisnd_aport ), sgisnd_aport = NULL;
return;
}
static void al_close (ao_instance_t * _instance)
{
al_instance_t * instance = (al_instance_t *) _instance;
alClosePort (instance->port);
}
ALAudioDevice::~ALAudioDevice() {
ADR_GUARD("ALAudioDevice::~ALAudioDevice");
alClosePort(m_port);
}