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authorChris Robinson <[email protected]>2020-03-29 20:37:58 -0700
committerChris Robinson <[email protected]>2020-03-29 20:37:58 -0700
commitd70912c0345e402e8aed9835ce450330cd6a7d36 (patch)
treefa0dd7f701dfc6fb017c11bc42f93ddec63c4374 /alc
parentfed80e0c101c5e54f7912e6c61d8553e33c11162 (diff)
Remove the QSA backend
It's been broken for who knows how long, and could really do with a rewrite for the new interface anyway.
Diffstat (limited to 'alc')
-rw-r--r--alc/alc.cpp6
-rw-r--r--alc/backends/qsa.cpp960
-rw-r--r--alc/backends/qsa.h19
3 files changed, 0 insertions, 985 deletions
diff --git a/alc/alc.cpp b/alc/alc.cpp
index efc822ff..9cd99f9d 100644
--- a/alc/alc.cpp
+++ b/alc/alc.cpp
@@ -124,9 +124,6 @@
#ifdef HAVE_OSS
#include "backends/oss.h"
#endif
-#ifdef HAVE_QSA
-#include "backends/qsa.h"
-#endif
#ifdef HAVE_DSOUND
#include "backends/dsound.h"
#endif
@@ -187,9 +184,6 @@ BackendInfo BackendList[] = {
#ifdef HAVE_OSS
{ "oss", OSSBackendFactory::getFactory },
#endif
-#ifdef HAVE_QSA
- { "qsa", QSABackendFactory::getFactory },
-#endif
#ifdef HAVE_DSOUND
{ "dsound", DSoundBackendFactory::getFactory },
#endif
diff --git a/alc/backends/qsa.cpp b/alc/backends/qsa.cpp
deleted file mode 100644
index ef43e080..00000000
--- a/alc/backends/qsa.cpp
+++ /dev/null
@@ -1,960 +0,0 @@
-/**
- * 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 "alcmain.h"
-#include "alexcpt.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;
-
- void open(const ALCchar *name) override;
- bool reset() override;
- bool start() override;
- void stop() override;
-
- std::unique_ptr<qsa_data> mExtraData;
-
- 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();
-
- 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 */
- sret = poll(&pollitem, 1, 2000);
- 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;
- }
- }
- }
-
- 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);
-}
-
-void PlaybackWrapper::open(const ALCchar *name)
-{
- if(auto err = qsa_open_playback(this, name))
- throw al::backend_exception{ALC_INVALID_VALUE, "%d", err};
-}
-
-bool PlaybackWrapper::reset()
-{
- if(!qsa_reset_playback(this))
- throw al::backend_exception{ALC_INVALID_VALUE, ""};
- return true;
-}
-
-bool 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;
-
- void open(const ALCchar *name) override;
- bool start() override;
- void stop() override;
- ALCenum captureSamples(al::byte *buffer, ALCuint samples) override;
- ALCuint availableSamples() override;
-
- std::unique_ptr<qsa_data> mExtraData;
-
- 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);
-}
-
-void CaptureWrapper::open(const ALCchar *name)
-{
- if(auto err = qsa_open_capture(this, name))
- throw al::backend_exception{ALC_INVALID_VALUE, "%d", err};
-}
-
-bool CaptureWrapper::start()
-{ qsa_start_capture(this); return true; }
-
-void CaptureWrapper::stop()
-{ qsa_stop_capture(this); }
-
-ALCenum CaptureWrapper::captureSamples(al::byte *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;
-}
diff --git a/alc/backends/qsa.h b/alc/backends/qsa.h
deleted file mode 100644
index da548bba..00000000
--- a/alc/backends/qsa.h
+++ /dev/null
@@ -1,19 +0,0 @@
-#ifndef BACKENDS_QSA_H
-#define BACKENDS_QSA_H
-
-#include "backends/base.h"
-
-struct QSABackendFactory final : public BackendFactory {
-public:
- bool init() override;
-
- bool querySupport(BackendType type) override;
-
- void probe(DevProbe type, std::string *outnames) override;
-
- BackendPtr createBackend(ALCdevice *device, BackendType type) override;
-
- static BackendFactory &getFactory();
-};
-
-#endif /* BACKENDS_QSA_H */