/** * 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 #include #include #include #include #include #include #include #include #include "alMain.h" #include "alu.h" #include "threads.h" #include #include 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 mKillNow{AL_TRUE}; std::thread mThread; }; struct DevMap { ALCchar* name; int card; int dev; }; al::vector DeviceNameMap; al::vector 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) { 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 mExtraData; DEF_NEWDEL(PlaybackWrapper) }; FORCE_ALIGN static int qsa_proc_playback(void *ptr) { PlaybackWrapper *self = static_cast(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(); 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(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 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 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 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(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; }