AuroraOpenALSoft/Alc/backends/qsa.c
2013-11-25 17:29:39 -08:00

1170 lines
31 KiB
C

/**
* OpenAL cross platform audio library
* Copyright (C) 2011-2013 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <sched.h>
#include <errno.h>
#include <memory.h>
#include <sys/select.h>
#include <sys/asoundlib.h>
#include <sys/neutrino.h>
#include "alMain.h"
#include "alu.h"
#include "threads.h"
typedef struct
{
snd_pcm_t* pcmHandle;
int audio_fd;
snd_pcm_channel_setup_t csetup;
snd_pcm_channel_params_t cparams;
ALvoid* buffer;
ALsizei size;
volatile int killNow;
althread_t thread;
} qsa_data;
typedef struct
{
ALCchar* name;
int card;
int dev;
} DevMap;
static const ALCchar qsaDevice[]="QSA Default";
static DevMap* allDevNameMap;
static ALuint numDevNames;
static DevMap* allCaptureDevNameMap;
static ALuint numCaptureDevNames;
static const struct
{
int32_t format;
} formatlist[]=
{
{SND_PCM_SFMT_FLOAT_LE},
{SND_PCM_SFMT_S32_LE},
{SND_PCM_SFMT_U32_LE},
{SND_PCM_SFMT_S16_LE},
{SND_PCM_SFMT_U16_LE},
{SND_PCM_SFMT_S8},
{SND_PCM_SFMT_U8},
{0},
};
static const struct
{
int32_t rate;
} ratelist[]=
{
{192000},
{176400},
{96000},
{88200},
{48000},
{44100},
{32000},
{24000},
{22050},
{16000},
{12000},
{11025},
{8000},
{0},
};
static const struct
{
int32_t channels;
} channellist[]=
{
{8},
{7},
{6},
{4},
{2},
{1},
{0},
};
static DevMap* deviceList(int type, ALuint* count)
{
snd_ctl_t* handle;
snd_pcm_info_t pcminfo;
int max_cards, card, err, dev, num_devices, idx;
DevMap* dev_list;
char name[1024];
struct snd_ctl_hw_info info;
void* temp;
idx=0;
num_devices=0;
max_cards=snd_cards();
if (max_cards<=0)
{
return 0;
}
dev_list=malloc(sizeof(DevMap)*1);
dev_list[0].name=strdup(qsaDevice);
num_devices=1;
for (card=0; card<max_cards; card++)
{
if ((err=snd_ctl_open(&handle, card))<0)
{
continue;
}
if ((err=snd_ctl_hw_info(handle, &info))<0)
{
snd_ctl_close(handle);
continue;
}
for (dev=0; dev<(int)info.pcmdevs; dev++)
{
if ((err=snd_ctl_pcm_info(handle, dev, &pcminfo)) < 0)
{
continue;
}
if ((type==SND_PCM_CHANNEL_PLAYBACK && (pcminfo.flags&SND_PCM_INFO_PLAYBACK)) ||
(type==SND_PCM_CHANNEL_CAPTURE && (pcminfo.flags&SND_PCM_INFO_CAPTURE)))
{
temp=realloc(dev_list, sizeof(DevMap)*(num_devices+1));
if (temp)
{
dev_list=temp;
snprintf(name, sizeof(name), "%s [%s] (hw:%d,%d)", info.name, pcminfo.name, card, dev);
dev_list[num_devices].name=strdup(name);
dev_list[num_devices].card=card;
dev_list[num_devices].dev=dev;
num_devices++;
}
}
}
snd_ctl_close (handle);
}
*count=num_devices;
return dev_list;
}
FORCE_ALIGN static ALuint qsa_proc_playback(ALvoid* ptr)
{
ALCdevice* device=(ALCdevice*)ptr;
qsa_data* data=(qsa_data*)device->ExtraData;
char* write_ptr;
int avail;
snd_pcm_channel_status_t status;
struct sched_param param;
fd_set wfds;
int selectret;
struct timeval timeout;
SetRTPriority();
SetThreadName(MIXER_THREAD_NAME);
/* Increase default 10 priority to 11 to avoid jerky sound */
SchedGet(0, 0, &param);
param.sched_priority=param.sched_curpriority+1;
SchedSet(0, 0, SCHED_NOCHANGE, &param);
ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
while (!data->killNow)
{
ALint len=data->size;
write_ptr=data->buffer;
avail=len/frame_size;
aluMixData(device, write_ptr, avail);
while (len>0 && !data->killNow)
{
FD_ZERO(&wfds);
FD_SET(data->audio_fd, &wfds);
timeout.tv_sec=2;
timeout.tv_usec=0;
/* Select also works like time slice to OS */
selectret=select(data->audio_fd+1, NULL, &wfds, NULL, &timeout);
switch (selectret)
{
case -1:
aluHandleDisconnect(device);
return 1;
case 0:
break;
default:
if (FD_ISSET(data->audio_fd, &wfds))
{
break;
}
break;
}
int wrote=snd_pcm_plugin_write(data->pcmHandle, write_ptr, len);
if (wrote<=0)
{
if ((errno==EAGAIN) || (errno==EWOULDBLOCK))
{
continue;
}
memset(&status, 0, sizeof (status));
status.channel=SND_PCM_CHANNEL_PLAYBACK;
snd_pcm_plugin_status(data->pcmHandle, &status);
/* we need to reinitialize the sound channel if we've underrun the buffer */
if ((status.status==SND_PCM_STATUS_UNDERRUN) ||
(status.status==SND_PCM_STATUS_READY))
{
if ((snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK))<0)
{
aluHandleDisconnect(device);
break;
}
}
}
else
{
write_ptr+=wrote;
len-=wrote;
}
}
}
return 0;
}
/************/
/* Playback */
/************/
static ALCenum qsa_open_playback(ALCdevice* device, const ALCchar* deviceName)
{
qsa_data* data;
char driver[64];
int status;
int card, dev;
strncpy(driver, GetConfigValue("qsa", "device", qsaDevice), sizeof(driver)-1);
driver[sizeof(driver)-1]=0;
data=(qsa_data*)calloc(1, sizeof(qsa_data));
if (data==NULL)
{
return ALC_OUT_OF_MEMORY;
}
if (!deviceName)
{
deviceName=driver;
}
if (strcmp(deviceName, qsaDevice)==0)
{
if (!deviceName)
{
deviceName=qsaDevice;
}
status=snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK);
}
else
{
size_t idx;
if (!allDevNameMap)
{
allDevNameMap=deviceList(SND_PCM_CHANNEL_PLAYBACK, &numDevNames);
}
for (idx=0; idx<numDevNames; idx++)
{
if (allDevNameMap[idx].name && strcmp(deviceName, allDevNameMap[idx].name)==0)
{
if (idx>0)
{
break;
}
}
}
if (idx==numDevNames)
{
free(data);
return ALC_INVALID_DEVICE;
}
status=snd_pcm_open(&data->pcmHandle, allDevNameMap[idx].card, allDevNameMap[idx].dev, SND_PCM_OPEN_PLAYBACK);
}
if (status<0)
{
free(data);
return ALC_INVALID_DEVICE;
}
data->audio_fd=snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK);
if (data->audio_fd<0)
{
free(data);
return ALC_INVALID_DEVICE;
}
device->DeviceName=strdup(deviceName);
device->ExtraData=data;
return ALC_NO_ERROR;
}
static void qsa_close_playback(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
if (data->buffer!=NULL)
{
free(data->buffer);
data->buffer=NULL;
}
snd_pcm_close(data->pcmHandle);
free(data);
device->ExtraData=NULL;
}
static ALCboolean qsa_reset_playback(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
int32_t format=-1;
switch(device->FmtType)
{
case DevFmtByte:
format=SND_PCM_SFMT_S8;
break;
case DevFmtUByte:
format=SND_PCM_SFMT_U8;
break;
case DevFmtShort:
format=SND_PCM_SFMT_S16_LE;
break;
case DevFmtUShort:
format=SND_PCM_SFMT_U16_LE;
break;
case DevFmtInt:
format=SND_PCM_SFMT_S32_LE;
break;
case DevFmtUInt:
format=SND_PCM_SFMT_U32_LE;
break;
case DevFmtFloat:
format=SND_PCM_SFMT_FLOAT_LE;
break;
}
/* we actually don't want to block on writes */
snd_pcm_nonblock_mode(data->pcmHandle, 1);
/* Disable mmap to control data transfer to the audio device */
snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP);
snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_BUFFER_PARTIAL_BLOCKS);
// configure a sound channel
memset(&data->cparams, 0, sizeof(data->cparams));
data->cparams.channel=SND_PCM_CHANNEL_PLAYBACK;
data->cparams.mode=SND_PCM_MODE_BLOCK;
data->cparams.start_mode=SND_PCM_START_FULL;
data->cparams.stop_mode=SND_PCM_STOP_STOP;
data->cparams.buf.block.frag_size=device->UpdateSize*
ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType);
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
data->cparams.format.interleave=1;
data->cparams.format.rate=device->Frequency;
data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans);
data->cparams.format.format=format;
if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
{
int original_rate=data->cparams.format.rate;
int original_voices=data->cparams.format.voices;
int original_format=data->cparams.format.format;
int it;
int jt;
for (it=0; it<1; it++)
{
/* Check for second pass */
if (it==1)
{
original_rate=ratelist[0].rate;
original_voices=channellist[0].channels;
original_format=formatlist[0].format;
}
do {
/* At first downgrade sample format */
jt=0;
do {
if (formatlist[jt].format==data->cparams.format.format)
{
data->cparams.format.format=formatlist[jt+1].format;
break;
}
if (formatlist[jt].format==0)
{
data->cparams.format.format=0;
break;
}
jt++;
} while(1);
if (data->cparams.format.format==0)
{
data->cparams.format.format=original_format;
/* At secod downgrade sample rate */
jt=0;
do {
if (ratelist[jt].rate==data->cparams.format.rate)
{
data->cparams.format.rate=ratelist[jt+1].rate;
break;
}
if (ratelist[jt].rate==0)
{
data->cparams.format.rate=0;
break;
}
jt++;
} while(1);
if (data->cparams.format.rate==0)
{
data->cparams.format.rate=original_rate;
data->cparams.format.format=original_format;
/* At third downgrade channels number */
jt=0;
do {
if(channellist[jt].channels==data->cparams.format.voices)
{
data->cparams.format.voices=channellist[jt+1].channels;
break;
}
if (channellist[jt].channels==0)
{
data->cparams.format.voices=0;
break;
}
jt++;
} while(1);
}
if (data->cparams.format.voices==0)
{
break;
}
}
data->cparams.buf.block.frag_size=device->UpdateSize*
data->cparams.format.voices*
snd_pcm_format_width(data->cparams.format.format)/8;
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
{
continue;
}
else
{
break;
}
} while(1);
if (data->cparams.format.voices!=0)
{
break;
}
}
if (data->cparams.format.voices==0)
{
return ALC_FALSE;
}
}
if ((snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK))<0)
{
return ALC_FALSE;
}
memset(&data->csetup, 0, sizeof(data->csetup));
data->csetup.channel=SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_plugin_setup(data->pcmHandle, &data->csetup)<0)
{
return ALC_FALSE;
}
/* now fill back to the our AL device */
device->Frequency=data->cparams.format.rate;
switch (data->cparams.format.voices)
{
case 1:
device->FmtChans=DevFmtMono;
break;
case 2:
device->FmtChans=DevFmtStereo;
break;
case 4:
device->FmtChans=DevFmtQuad;
break;
case 6:
device->FmtChans=DevFmtX51;
break;
case 7:
device->FmtChans=DevFmtX61;
break;
case 8:
device->FmtChans=DevFmtX71;
break;
default:
device->FmtChans=DevFmtMono;
break;
}
switch (data->cparams.format.format)
{
case SND_PCM_SFMT_S8:
device->FmtType=DevFmtByte;
break;
case SND_PCM_SFMT_U8:
device->FmtType=DevFmtUByte;
break;
case SND_PCM_SFMT_S16_LE:
device->FmtType=DevFmtShort;
break;
case SND_PCM_SFMT_U16_LE:
device->FmtType=DevFmtUShort;
break;
case SND_PCM_SFMT_S32_LE:
device->FmtType=DevFmtInt;
break;
case SND_PCM_SFMT_U32_LE:
device->FmtType=DevFmtUInt;
break;
case SND_PCM_SFMT_FLOAT_LE:
device->FmtType=DevFmtFloat;
break;
default:
device->FmtType=DevFmtShort;
break;
}
SetDefaultChannelOrder(device);
device->UpdateSize=data->csetup.buf.block.frag_size/
(ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType));
device->NumUpdates=data->csetup.buf.block.frags;
data->size=data->csetup.buf.block.frag_size;
data->buffer=malloc(data->size);
if (!data->buffer)
{
return ALC_FALSE;
}
return ALC_TRUE;
}
static ALCboolean qsa_start_playback(ALCdevice* device)
{
qsa_data *data = (qsa_data*)device->ExtraData;
if(!StartThread(&data->thread, qsa_proc_playback, device))
return ALC_FALSE;
return ALC_TRUE;
}
static void qsa_stop_playback(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
if (data->thread)
{
data->killNow=1;
StopThread(data->thread);
data->thread=NULL;
}
data->killNow=0;
}
/***********/
/* Capture */
/***********/
static ALCenum qsa_open_capture(ALCdevice* device, const ALCchar* deviceName)
{
qsa_data* data;
int format=-1;
char driver[64];
int card, dev;
int status;
strncpy(driver, GetConfigValue("qsa", "capture", qsaDevice), sizeof(driver)-1);
driver[sizeof(driver)-1]=0;
data=(qsa_data*)calloc(1, sizeof(qsa_data));
if (data==NULL)
{
return ALC_OUT_OF_MEMORY;
}
if (!deviceName)
{
deviceName=driver;
}
if (strcmp(deviceName, qsaDevice)==0)
{
if (!deviceName)
{
deviceName=qsaDevice;
}
status=snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE);
}
else
{
size_t idx;
if (!allCaptureDevNameMap)
{
allCaptureDevNameMap=deviceList(SND_PCM_CHANNEL_CAPTURE, &numDevNames);
}
for (idx=0; idx<numDevNames; idx++)
{
if (allCaptureDevNameMap[idx].name && strcmp(deviceName, allCaptureDevNameMap[idx].name)==0)
{
if (idx>0)
{
break;
}
}
}
if (idx==numDevNames)
{
free(data);
return ALC_INVALID_DEVICE;
}
status=snd_pcm_open(&data->pcmHandle, allCaptureDevNameMap[idx].card, allCaptureDevNameMap[idx].dev, SND_PCM_OPEN_CAPTURE);
}
if (status<0)
{
free(data);
return ALC_INVALID_DEVICE;
}
data->audio_fd=snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE);
if (data->audio_fd<0)
{
free(data);
return ALC_INVALID_DEVICE;
}
device->DeviceName=strdup(deviceName);
device->ExtraData=data;
switch (device->FmtType)
{
case DevFmtByte:
format=SND_PCM_SFMT_S8;
break;
case DevFmtUByte:
format=SND_PCM_SFMT_U8;
break;
case DevFmtShort:
format=SND_PCM_SFMT_S16_LE;
break;
case DevFmtUShort:
format=SND_PCM_SFMT_U16_LE;
break;
case DevFmtInt:
format=SND_PCM_SFMT_S32_LE;
break;
case DevFmtUInt:
format=SND_PCM_SFMT_U32_LE;
break;
case DevFmtFloat:
format=SND_PCM_SFMT_FLOAT_LE;
break;
}
/* we actually don't want to block on reads */
snd_pcm_nonblock_mode(data->pcmHandle, 1);
/* Disable mmap to control data transfer to the audio device */
snd_pcm_plugin_set_disable(data->pcmHandle, PLUGIN_DISABLE_MMAP);
/* configure a sound channel */
memset(&data->cparams, 0, sizeof(data->cparams));
data->cparams.mode=SND_PCM_MODE_BLOCK;
data->cparams.channel=SND_PCM_CHANNEL_CAPTURE;
data->cparams.start_mode=SND_PCM_START_GO;
data->cparams.stop_mode=SND_PCM_STOP_STOP;
data->cparams.buf.block.frag_size=device->UpdateSize*
ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType);
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
data->cparams.format.interleave=1;
data->cparams.format.rate=device->Frequency;
data->cparams.format.voices=ChannelsFromDevFmt(device->FmtChans);
data->cparams.format.format=format;
if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
{
int original_rate=data->cparams.format.rate;
int original_voices=data->cparams.format.voices;
int original_format=data->cparams.format.format;
int it;
int jt;
for (it=0; it<1; it++)
{
/* Check for second pass */
if (it==1)
{
original_rate=ratelist[0].rate;
original_voices=channellist[0].channels;
original_format=formatlist[0].format;
}
do {
/* At first downgrade sample format */
jt=0;
do {
if (formatlist[jt].format==data->cparams.format.format)
{
data->cparams.format.format=formatlist[jt+1].format;
break;
}
if (formatlist[jt].format==0)
{
data->cparams.format.format=0;
break;
}
jt++;
} while(1);
if (data->cparams.format.format==0)
{
data->cparams.format.format=original_format;
/* At secod downgrade sample rate */
jt=0;
do {
if (ratelist[jt].rate==data->cparams.format.rate)
{
data->cparams.format.rate=ratelist[jt+1].rate;
break;
}
if (ratelist[jt].rate==0)
{
data->cparams.format.rate=0;
break;
}
jt++;
} while(1);
if (data->cparams.format.rate==0)
{
data->cparams.format.rate=original_rate;
data->cparams.format.format=original_format;
/* At third downgrade channels number */
jt=0;
do {
if(channellist[jt].channels==data->cparams.format.voices)
{
data->cparams.format.voices=channellist[jt+1].channels;
break;
}
if (channellist[jt].channels==0)
{
data->cparams.format.voices=0;
break;
}
jt++;
} while(1);
}
if (data->cparams.format.voices==0)
{
break;
}
}
data->cparams.buf.block.frag_size=device->UpdateSize*
data->cparams.format.voices*
snd_pcm_format_width(data->cparams.format.format)/8;
data->cparams.buf.block.frags_max=device->NumUpdates;
data->cparams.buf.block.frags_min=device->NumUpdates;
if ((snd_pcm_plugin_params(data->pcmHandle, &data->cparams))<0)
{
continue;
}
else
{
break;
}
} while(1);
if (data->cparams.format.voices!=0)
{
break;
}
}
if (data->cparams.format.voices==0)
{
return ALC_INVALID_VALUE;
}
}
/* now fill back to the our AL device */
device->Frequency=data->cparams.format.rate;
switch (data->cparams.format.voices)
{
case 1:
device->FmtChans=DevFmtMono;
break;
case 2:
device->FmtChans=DevFmtStereo;
break;
case 4:
device->FmtChans=DevFmtQuad;
break;
case 6:
device->FmtChans=DevFmtX51;
break;
case 7:
device->FmtChans=DevFmtX61;
break;
case 8:
device->FmtChans=DevFmtX71;
break;
default:
device->FmtChans=DevFmtMono;
break;
}
switch (data->cparams.format.format)
{
case SND_PCM_SFMT_S8:
device->FmtType=DevFmtByte;
break;
case SND_PCM_SFMT_U8:
device->FmtType=DevFmtUByte;
break;
case SND_PCM_SFMT_S16_LE:
device->FmtType=DevFmtShort;
break;
case SND_PCM_SFMT_U16_LE:
device->FmtType=DevFmtUShort;
break;
case SND_PCM_SFMT_S32_LE:
device->FmtType=DevFmtInt;
break;
case SND_PCM_SFMT_U32_LE:
device->FmtType=DevFmtUInt;
break;
case SND_PCM_SFMT_FLOAT_LE:
device->FmtType=DevFmtFloat;
break;
default:
device->FmtType=DevFmtShort;
break;
}
return ALC_NO_ERROR;
}
static void qsa_close_capture(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
if (data->pcmHandle!=NULL)
{
snd_pcm_close(data->pcmHandle);
}
free(data);
device->ExtraData=NULL;
}
static void qsa_start_capture(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
int rstatus;
if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0)
{
ERR("capture prepare failed: %s\n", snd_strerror(rstatus));
return;
}
memset(&data->csetup, 0, sizeof(data->csetup));
data->csetup.channel=SND_PCM_CHANNEL_CAPTURE;
if ((rstatus=snd_pcm_plugin_setup(data->pcmHandle, &data->csetup))<0)
{
ERR("capture setup failed: %s\n", snd_strerror(rstatus));
return;
}
snd_pcm_capture_go(data->pcmHandle);
device->UpdateSize=data->csetup.buf.block.frag_size/
(ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType));
device->NumUpdates=data->csetup.buf.block.frags;
}
static void qsa_stop_capture(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
snd_pcm_capture_flush(data->pcmHandle);
}
static ALCuint qsa_available_samples(ALCdevice* device)
{
qsa_data* data=(qsa_data*)device->ExtraData;
snd_pcm_channel_status_t status;
ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
ALint free_size;
int rstatus;
memset(&status, 0, sizeof (status));
status.channel=SND_PCM_CHANNEL_CAPTURE;
snd_pcm_plugin_status(data->pcmHandle, &status);
if ((status.status==SND_PCM_STATUS_OVERRUN) ||
(status.status==SND_PCM_STATUS_READY))
{
if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0)
{
ERR("capture prepare failed: %s\n", snd_strerror(rstatus));
aluHandleDisconnect(device);
return 0;
}
snd_pcm_capture_go(data->pcmHandle);
return 0;
}
free_size=data->csetup.buf.block.frag_size*data->csetup.buf.block.frags;
free_size-=status.free;
return free_size/frame_size;
}
static ALCenum qsa_capture_samples(ALCdevice *device, ALCvoid *buffer, ALCuint samples)
{
qsa_data* data=(qsa_data*)device->ExtraData;
char* read_ptr;
snd_pcm_channel_status_t status;
fd_set rfds;
int selectret;
struct timeval timeout;
int bytes_read;
ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
ALint len=samples*frame_size;
int rstatus;
read_ptr=buffer;
while (len>0)
{
FD_ZERO(&rfds);
FD_SET(data->audio_fd, &rfds);
timeout.tv_sec=2;
timeout.tv_usec=0;
/* Select also works like time slice to OS */
bytes_read=0;
selectret=select(data->audio_fd+1, &rfds, NULL, NULL, &timeout);
switch (selectret)
{
case -1:
aluHandleDisconnect(device);
return ALC_INVALID_DEVICE;
case 0:
break;
default:
if (FD_ISSET(data->audio_fd, &rfds))
{
bytes_read=snd_pcm_plugin_read(data->pcmHandle, read_ptr, len);
break;
}
break;
}
if (bytes_read<=0)
{
if ((errno==EAGAIN) || (errno==EWOULDBLOCK))
{
continue;
}
memset(&status, 0, sizeof (status));
status.channel=SND_PCM_CHANNEL_CAPTURE;
snd_pcm_plugin_status(data->pcmHandle, &status);
/* we need to reinitialize the sound channel if we've overrun the buffer */
if ((status.status==SND_PCM_STATUS_OVERRUN) ||
(status.status==SND_PCM_STATUS_READY))
{
if ((rstatus=snd_pcm_plugin_prepare(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE))<0)
{
ERR("capture prepare failed: %s\n", snd_strerror(rstatus));
aluHandleDisconnect(device);
return ALC_INVALID_DEVICE;
}
snd_pcm_capture_go(data->pcmHandle);
}
}
else
{
read_ptr+=bytes_read;
len-=bytes_read;
}
}
return ALC_NO_ERROR;
}
static ALint64 qsa_get_latency(ALCdevice* device)
{
ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType);
return (ALint64)(device->UpdateSize*device->NumUpdates/frame_size)*
1000000000/device->Frequency;
}
BackendFuncs qsa_funcs=
{
qsa_open_playback,
qsa_close_playback,
qsa_reset_playback,
qsa_start_playback,
qsa_stop_playback,
qsa_open_capture,
qsa_close_capture,
qsa_start_capture,
qsa_stop_capture,
qsa_capture_samples,
qsa_available_samples,
qsa_get_latency,
};
ALCboolean alc_qsa_init(BackendFuncs* func_list)
{
*func_list=qsa_funcs;
return ALC_TRUE;
}
void alc_qsa_deinit(void)
{
ALuint i;
for (i=0; i<numDevNames; ++i)
{
free(allDevNameMap[i].name);
}
free(allDevNameMap);
allDevNameMap=NULL;
numDevNames=0;
for (i=0; i<numCaptureDevNames; ++i)
{
free(allCaptureDevNameMap[i].name);
}
free(allCaptureDevNameMap);
allCaptureDevNameMap=NULL;
numCaptureDevNames=0;
}
void alc_qsa_probe(enum DevProbe type)
{
ALuint i;
switch (type)
{
case ALL_DEVICE_PROBE:
for (i=0; i<numDevNames; ++i)
{
free(allDevNameMap[i].name);
}
free(allDevNameMap);
allDevNameMap=deviceList(SND_PCM_CHANNEL_PLAYBACK, &numDevNames);
for (i=0; i<numDevNames; ++i)
{
AppendAllDevicesList(allDevNameMap[i].name);
}
break;
case CAPTURE_DEVICE_PROBE:
for (i=0; i<numCaptureDevNames; ++i)
{
free(allCaptureDevNameMap[i].name);
}
free(allCaptureDevNameMap);
allCaptureDevNameMap=deviceList(SND_PCM_CHANNEL_CAPTURE, &numCaptureDevNames);
for (i=0; i<numCaptureDevNames; ++i)
{
AppendCaptureDeviceList(allCaptureDevNameMap[i].name);
}
break;
}
}