diff options
author | Chris Robinson <[email protected]> | 2020-03-29 20:37:58 -0700 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2020-03-29 20:37:58 -0700 |
commit | d70912c0345e402e8aed9835ce450330cd6a7d36 (patch) | |
tree | fa0dd7f701dfc6fb017c11bc42f93ddec63c4374 /alc | |
parent | fed80e0c101c5e54f7912e6c61d8553e33c11162 (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.cpp | 6 | ||||
-rw-r--r-- | alc/backends/qsa.cpp | 960 | ||||
-rw-r--r-- | alc/backends/qsa.h | 19 |
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, ¶m); - param.sched_priority=param.sched_curpriority+1; - SchedSet(0, 0, SCHED_NOCHANGE, ¶m); - - 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 */ |