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/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "backends/qsa.h"
#include <stdlib.h>
#include <stdio.h>
#include <sched.h>
#include <errno.h>
#include <memory.h>
#include <poll.h>
#include <thread>
#include <memory>
#include <algorithm>
#include "alMain.h"
#include "alu.h"
#include "threads.h"
#include <sys/asoundlib.h>
#include <sys/neutrino.h>
namespace {
struct qsa_data {
snd_pcm_t* pcmHandle{nullptr};
int audio_fd{-1};
snd_pcm_channel_setup_t csetup{};
snd_pcm_channel_params_t cparams{};
ALvoid* buffer{nullptr};
ALsizei size{0};
std::atomic<ALenum> mKillNow{AL_TRUE};
std::thread mThread;
};
struct DevMap {
ALCchar* name;
int card;
int dev;
};
al::vector<DevMap> DeviceNameMap;
al::vector<DevMap> CaptureNameMap;
constexpr ALCchar qsaDevice[] = "QSA Default";
constexpr 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},
};
constexpr struct {
int32_t rate;
} ratelist[] = {
{192000},
{176400},
{96000},
{88200},
{48000},
{44100},
{32000},
{24000},
{22050},
{16000},
{12000},
{11025},
{8000},
{0},
};
constexpr struct {
int32_t channels;
} channellist[] = {
{8},
{7},
{6},
{4},
{2},
{1},
{0},
};
void deviceList(int type, al::vector<DevMap> *devmap)
{
snd_ctl_t* handle;
snd_pcm_info_t pcminfo;
int max_cards, card, err, dev;
DevMap entry;
char name[1024];
snd_ctl_hw_info info;
max_cards = snd_cards();
if(max_cards < 0)
return;
std::for_each(devmap->begin(), devmap->end(),
[](const DevMap &entry) -> void
{ free(entry.name); }
);
devmap->clear();
entry.name = strdup(qsaDevice);
entry.card = 0;
entry.dev = 0;
devmap->push_back(entry);
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)))
{
snprintf(name, sizeof(name), "%s [%s] (hw:%d,%d)", info.name, pcminfo.name, card, dev);
entry.name = strdup(name);
entry.card = card;
entry.dev = dev;
devmap->push_back(entry);
TRACE("Got device \"%s\", card %d, dev %d\n", name, card, dev);
}
}
snd_ctl_close(handle);
}
}
/* Wrappers to use an old-style backend with the new interface. */
struct PlaybackWrapper final : public BackendBase {
PlaybackWrapper(ALCdevice *device) noexcept : BackendBase{device} { }
~PlaybackWrapper() override;
ALCenum open(const ALCchar *name) override;
ALCboolean reset() override;
ALCboolean start() override;
void stop() override;
std::unique_ptr<qsa_data> mExtraData;
static constexpr inline const char *CurrentPrefix() noexcept { return "PlaybackWrapper::"; }
DEF_NEWDEL(PlaybackWrapper)
};
FORCE_ALIGN static int qsa_proc_playback(void *ptr)
{
PlaybackWrapper *self = static_cast<PlaybackWrapper*>(ptr);
ALCdevice *device = self->mDevice;
qsa_data *data = self->mExtraData.get();
snd_pcm_channel_status_t status;
sched_param param;
char* write_ptr;
ALint len;
int sret;
SetRTPriority();
althrd_setname(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);
const ALint frame_size = device->frameSizeFromFmt();
self->lock();
while(!data->mKillNow.load(std::memory_order_acquire))
{
pollfd pollitem{};
pollitem.fd = data->audio_fd;
pollitem.events = POLLOUT;
/* Select also works like time slice to OS */
self->unlock();
sret = poll(&pollitem, 1, 2000);
self->lock();
if(sret == -1)
{
if(errno == EINTR || errno == EAGAIN)
continue;
ERR("poll error: %s\n", strerror(errno));
aluHandleDisconnect(device, "Failed waiting for playback buffer: %s", strerror(errno));
break;
}
if(sret == 0)
{
ERR("poll timeout\n");
continue;
}
len = data->size;
write_ptr = static_cast<char*>(data->buffer);
aluMixData(device, write_ptr, len/frame_size);
while(len>0 && !data->mKillNow.load(std::memory_order_acquire))
{
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, "Playback recovery failed");
break;
}
}
}
else
{
write_ptr += wrote;
len -= wrote;
}
}
}
self->unlock();
return 0;
}
/************/
/* Playback */
/************/
static ALCenum qsa_open_playback(PlaybackWrapper *self, const ALCchar* deviceName)
{
ALCdevice *device = self->mDevice;
int card, dev;
int status;
std::unique_ptr<qsa_data> data{new qsa_data{}};
data->mKillNow.store(AL_TRUE, std::memory_order_relaxed);
if(!deviceName)
deviceName = qsaDevice;
if(strcmp(deviceName, qsaDevice) == 0)
status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK);
else
{
if(DeviceNameMap.empty())
deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap);
auto iter = std::find_if(DeviceNameMap.begin(), DeviceNameMap.end(),
[deviceName](const DevMap &entry) -> bool
{ return entry.name && strcmp(deviceName, entry.name) == 0; }
);
if(iter == DeviceNameMap.cend())
return ALC_INVALID_DEVICE;
status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_PLAYBACK);
}
if(status < 0)
return ALC_INVALID_DEVICE;
data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_PLAYBACK);
if(data->audio_fd < 0)
{
snd_pcm_close(data->pcmHandle);
return ALC_INVALID_DEVICE;
}
device->DeviceName = deviceName;
self->mExtraData = std::move(data);
return ALC_NO_ERROR;
}
static void qsa_close_playback(PlaybackWrapper *self)
{
qsa_data *data = self->mExtraData.get();
if (data->buffer!=NULL)
{
free(data->buffer);
data->buffer=NULL;
}
snd_pcm_close(data->pcmHandle);
self->mExtraData = nullptr;
}
static ALCboolean qsa_reset_playback(PlaybackWrapper *self)
{
ALCdevice *device = self->mDevice;
qsa_data *data = self->mExtraData.get();
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 * device->frameSizeFromFmt();
data->cparams.buf.block.frags_max=device->BufferSize / device->UpdateSize;
data->cparams.buf.block.frags_min=data->cparams.buf.block.frags_max;
data->cparams.format.interleave=1;
data->cparams.format.rate=device->Frequency;
data->cparams.format.voices=device->channelsFromFmt();
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 / device->frameSizeFromFmt();
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(PlaybackWrapper *self)
{
qsa_data *data = self->mExtraData.get();
try {
data->mKillNow.store(AL_FALSE, std::memory_order_release);
data->mThread = std::thread(qsa_proc_playback, self);
return ALC_TRUE;
}
catch(std::exception& e) {
ERR("Could not create playback thread: %s\n", e.what());
}
catch(...) {
}
return ALC_FALSE;
}
static void qsa_stop_playback(PlaybackWrapper *self)
{
qsa_data *data = self->mExtraData.get();
if(data->mKillNow.exchange(AL_TRUE, std::memory_order_acq_rel) || !data->mThread.joinable())
return;
data->mThread.join();
}
PlaybackWrapper::~PlaybackWrapper()
{
if(mExtraData)
qsa_close_playback(this);
}
ALCenum PlaybackWrapper::open(const ALCchar *name)
{ return qsa_open_playback(this, name); }
ALCboolean PlaybackWrapper::reset()
{ return qsa_reset_playback(this); }
ALCboolean PlaybackWrapper::start()
{ return qsa_start_playback(this); }
void PlaybackWrapper::stop()
{ qsa_stop_playback(this); }
/***********/
/* Capture */
/***********/
struct CaptureWrapper final : public BackendBase {
CaptureWrapper(ALCdevice *device) noexcept : BackendBase{device} { }
~CaptureWrapper() override;
ALCenum open(const ALCchar *name) override;
ALCboolean start() override;
void stop() override;
ALCenum captureSamples(void *buffer, ALCuint samples) override;
ALCuint availableSamples() override;
std::unique_ptr<qsa_data> mExtraData;
static constexpr inline const char *CurrentPrefix() noexcept { return "CaptureWrapper::"; }
DEF_NEWDEL(CaptureWrapper)
};
static ALCenum qsa_open_capture(CaptureWrapper *self, const ALCchar *deviceName)
{
ALCdevice *device = self->mDevice;
int card, dev;
int format=-1;
int status;
std::unique_ptr<qsa_data> data{new qsa_data{}};
if(!deviceName)
deviceName = qsaDevice;
if(strcmp(deviceName, qsaDevice) == 0)
status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE);
else
{
if(CaptureNameMap.empty())
deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap);
auto iter = std::find_if(CaptureNameMap.cbegin(), CaptureNameMap.cend(),
[deviceName](const DevMap &entry) -> bool
{ return entry.name && strcmp(deviceName, entry.name) == 0; }
);
if(iter == CaptureNameMap.cend())
return ALC_INVALID_DEVICE;
status = snd_pcm_open(&data->pcmHandle, iter->card, iter->dev, SND_PCM_OPEN_CAPTURE);
}
if(status < 0)
return ALC_INVALID_DEVICE;
data->audio_fd = snd_pcm_file_descriptor(data->pcmHandle, SND_PCM_CHANNEL_CAPTURE);
if(data->audio_fd < 0)
{
snd_pcm_close(data->pcmHandle);
return ALC_INVALID_DEVICE;
}
device->DeviceName = deviceName;
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 * device->frameSizeFromFmt();
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=device->channelsFromFmt();
data->cparams.format.format=format;
if(snd_pcm_plugin_params(data->pcmHandle, &data->cparams) < 0)
{
snd_pcm_close(data->pcmHandle);
return ALC_INVALID_VALUE;
}
self->mExtraData = std::move(data);
return ALC_NO_ERROR;
}
static void qsa_close_capture(CaptureWrapper *self)
{
qsa_data *data = self->mExtraData.get();
if (data->pcmHandle!=nullptr)
snd_pcm_close(data->pcmHandle);
data->pcmHandle = nullptr;
self->mExtraData = nullptr;
}
static void qsa_start_capture(CaptureWrapper *self)
{
qsa_data *data = self->mExtraData.get();
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);
}
static void qsa_stop_capture(CaptureWrapper *self)
{
qsa_data *data = self->mExtraData.get();
snd_pcm_capture_flush(data->pcmHandle);
}
static ALCuint qsa_available_samples(CaptureWrapper *self)
{
ALCdevice *device = self->mDevice;
qsa_data *data = self->mExtraData.get();
snd_pcm_channel_status_t status;
ALint frame_size = device->frameSizeFromFmt();
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, "Failed capture recovery: %s", snd_strerror(rstatus));
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(CaptureWrapper *self, ALCvoid *buffer, ALCuint samples)
{
ALCdevice *device = self->mDevice;
qsa_data *data = self->mExtraData.get();
char* read_ptr;
snd_pcm_channel_status_t status;
int selectret;
int bytes_read;
ALint frame_size=device->frameSizeFromFmt();
ALint len=samples*frame_size;
int rstatus;
read_ptr = static_cast<char*>(buffer);
while (len>0)
{
pollfd pollitem{};
pollitem.fd = data->audio_fd;
pollitem.events = POLLOUT;
/* Select also works like time slice to OS */
bytes_read=0;
selectret = poll(&pollitem, 1, 2000);
switch (selectret)
{
case -1:
aluHandleDisconnect(device, "Failed to check capture samples");
return ALC_INVALID_DEVICE;
case 0:
break;
default:
bytes_read=snd_pcm_plugin_read(data->pcmHandle, read_ptr, len);
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, "Failed capture recovery: %s",
snd_strerror(rstatus));
return ALC_INVALID_DEVICE;
}
snd_pcm_capture_go(data->pcmHandle);
}
}
else
{
read_ptr+=bytes_read;
len-=bytes_read;
}
}
return ALC_NO_ERROR;
}
CaptureWrapper::~CaptureWrapper()
{
if(mExtraData)
qsa_close_capture(this);
}
ALCenum CaptureWrapper::open(const ALCchar *name)
{ return qsa_open_capture(this, name); }
ALCboolean CaptureWrapper::start()
{ qsa_start_capture(this); return ALC_TRUE; }
void CaptureWrapper::stop()
{ qsa_stop_capture(this); }
ALCenum CaptureWrapper::captureSamples(void *buffer, ALCuint samples)
{ return qsa_capture_samples(this, buffer, samples); }
ALCuint CaptureWrapper::availableSamples()
{ return qsa_available_samples(this); }
} // namespace
bool QSABackendFactory::init()
{ return true; }
bool QSABackendFactory::querySupport(BackendType type)
{ return (type == BackendType::Playback || type == BackendType::Capture); }
void QSABackendFactory::probe(DevProbe type, std::string *outnames)
{
auto add_device = [outnames](const DevMap &entry) -> void
{
const char *n = entry.name;
if(n && n[0])
outnames->append(n, strlen(n)+1);
};
switch (type)
{
case DevProbe::Playback:
deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap);
std::for_each(DeviceNameMap.cbegin(), DeviceNameMap.cend(), add_device);
break;
case DevProbe::Capture:
deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap);
std::for_each(CaptureNameMap.cbegin(), CaptureNameMap.cend(), add_device);
break;
}
}
BackendPtr QSABackendFactory::createBackend(ALCdevice *device, BackendType type)
{
if(type == BackendType::Playback)
return BackendPtr{new PlaybackWrapper{device}};
if(type == BackendType::Capture)
return BackendPtr{new CaptureWrapper{device}};
return nullptr;
}
BackendFactory &QSABackendFactory::getFactory()
{
static QSABackendFactory factory{};
return factory;
}