/** * 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 #include #include #include #include #include #include #include #include "alMain.h" #include "alu.h" #include "threads.h" #include "backends/base.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; althrd_t thread; } qsa_data; typedef struct { ALCchar* name; int card; int dev; } DevMap; TYPEDEF_VECTOR(DevMap, vector_DevMap) static vector_DevMap DeviceNameMap; static vector_DevMap CaptureNameMap; static const ALCchar qsaDevice[] = "QSA Default"; 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 void deviceList(int type, vector_DevMap *devmap) { snd_ctl_t* handle; snd_pcm_info_t pcminfo; int max_cards, card, err, dev; DevMap entry; char name[1024]; struct snd_ctl_hw_info info; max_cards = snd_cards(); if(max_cards < 0) return; VECTOR_RESIZE(*devmap, 0, max_cards+1); entry.name = strdup(qsaDevice); entry.card = 0; entry.dev = 0; VECTOR_PUSH_BACK(*devmap, 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; VECTOR_PUSH_BACK(*devmap, 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. */ typedef struct PlaybackWrapper { DERIVE_FROM_TYPE(ALCbackend); qsa_data *ExtraData; } PlaybackWrapper; static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device); static void PlaybackWrapper_Destruct(PlaybackWrapper *self); static ALCenum PlaybackWrapper_open(PlaybackWrapper *self, const ALCchar *name); static ALCboolean PlaybackWrapper_reset(PlaybackWrapper *self); static ALCboolean PlaybackWrapper_start(PlaybackWrapper *self); static void PlaybackWrapper_stop(PlaybackWrapper *self); static DECLARE_FORWARD2(PlaybackWrapper, ALCbackend, ALCenum, captureSamples, void*, ALCuint) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, ALCuint, availableSamples) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, lock) static DECLARE_FORWARD(PlaybackWrapper, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(PlaybackWrapper) DEFINE_ALCBACKEND_VTABLE(PlaybackWrapper); FORCE_ALIGN static int qsa_proc_playback(void *ptr) { PlaybackWrapper *self = ptr; ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->ExtraData; snd_pcm_channel_status_t status; struct sched_param param; struct timeval timeout; char* write_ptr; fd_set wfds; ALint len; int sret; SetRTPriority(); althrd_setname(althrd_current(), 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 = FrameSizeFromDevFmt( device->FmtChans, device->FmtType, device->AmbiOrder ); V0(device->Backend,lock)(); while(!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 */ V0(device->Backend,unlock)(); sret = select(data->audio_fd+1, NULL, &wfds, NULL, &timeout); V0(device->Backend,lock)(); if(sret == -1) { ERR("select error: %s\n", strerror(errno)); aluHandleDisconnect(device); break; } if(sret == 0) { ERR("select timeout\n"); continue; } len = data->size; write_ptr = data->buffer; aluMixData(device, write_ptr, len/frame_size); while(len>0 && !data->killNow) { 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; } } } V0(device->Backend,unlock)(); return 0; } /************/ /* Playback */ /************/ static ALCenum qsa_open_playback(PlaybackWrapper *self, const ALCchar* deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data; int card, dev; int status; data = (qsa_data*)calloc(1, sizeof(qsa_data)); if(data == NULL) return ALC_OUT_OF_MEMORY; if(!deviceName) deviceName = qsaDevice; if(strcmp(deviceName, qsaDevice) == 0) status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK); else { const DevMap *iter; if(VECTOR_SIZE(DeviceNameMap) == 0) deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap); #define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0) VECTOR_FIND_IF(iter, const DevMap, DeviceNameMap, MATCH_DEVNAME); #undef MATCH_DEVNAME if(iter == VECTOR_END(DeviceNameMap)) { free(data); return ALC_INVALID_DEVICE; } status = snd_pcm_open(&data->pcmHandle, iter->card, iter->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) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_DEVICE; } alstr_copy_cstr(&device->DeviceName, deviceName); self->ExtraData = data; return ALC_NO_ERROR; } static void qsa_close_playback(PlaybackWrapper *self) { qsa_data *data = self->ExtraData; if (data->buffer!=NULL) { free(data->buffer); data->buffer=NULL; } snd_pcm_close(data->pcmHandle); free(data); self->ExtraData = NULL; } static ALCboolean qsa_reset_playback(PlaybackWrapper *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->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 * FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); 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, device->AmbiOrder); 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/ FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); 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->ExtraData; data->killNow = 0; if(althrd_create(&data->thread, qsa_proc_playback, self) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void qsa_stop_playback(PlaybackWrapper *self) { qsa_data *data = self->ExtraData; int res; if(data->killNow) return; data->killNow = 1; althrd_join(data->thread, &res); } static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(PlaybackWrapper, ALCbackend, self); self->ExtraData = NULL; } static void PlaybackWrapper_Destruct(PlaybackWrapper *self) { if(self->ExtraData) qsa_close_playback(self); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum PlaybackWrapper_open(PlaybackWrapper *self, const ALCchar *name) { return qsa_open_playback(self, name); } static ALCboolean PlaybackWrapper_reset(PlaybackWrapper *self) { return qsa_reset_playback(self); } static ALCboolean PlaybackWrapper_start(PlaybackWrapper *self) { return qsa_start_playback(self); } static void PlaybackWrapper_stop(PlaybackWrapper *self) { qsa_stop_playback(self); } /***********/ /* Capture */ /***********/ typedef struct CaptureWrapper { DERIVE_FROM_TYPE(ALCbackend); qsa_data *ExtraData; } CaptureWrapper; static void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device); static void CaptureWrapper_Destruct(CaptureWrapper *self); static ALCenum CaptureWrapper_open(CaptureWrapper *self, const ALCchar *name); static DECLARE_FORWARD(CaptureWrapper, ALCbackend, ALCboolean, reset) static ALCboolean CaptureWrapper_start(CaptureWrapper *self); static void CaptureWrapper_stop(CaptureWrapper *self); static ALCenum CaptureWrapper_captureSamples(CaptureWrapper *self, void *buffer, ALCuint samples); static ALCuint CaptureWrapper_availableSamples(CaptureWrapper *self); static DECLARE_FORWARD(CaptureWrapper, ALCbackend, ClockLatency, getClockLatency) static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, lock) static DECLARE_FORWARD(CaptureWrapper, ALCbackend, void, unlock) DECLARE_DEFAULT_ALLOCATORS(CaptureWrapper) DEFINE_ALCBACKEND_VTABLE(CaptureWrapper); static ALCenum qsa_open_capture(CaptureWrapper *self, const ALCchar *deviceName) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data; int card, dev; int format=-1; int status; data=(qsa_data*)calloc(1, sizeof(qsa_data)); if (data==NULL) { return ALC_OUT_OF_MEMORY; } if(!deviceName) deviceName = qsaDevice; if(strcmp(deviceName, qsaDevice) == 0) status = snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE); else { const DevMap *iter; if(VECTOR_SIZE(CaptureNameMap) == 0) deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap); #define MATCH_DEVNAME(iter) ((iter)->name && strcmp(deviceName, (iter)->name)==0) VECTOR_FIND_IF(iter, const DevMap, CaptureNameMap, MATCH_DEVNAME); #undef MATCH_DEVNAME if(iter == VECTOR_END(CaptureNameMap)) { free(data); return ALC_INVALID_DEVICE; } status = snd_pcm_open(&data->pcmHandle, iter->card, iter->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) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_DEVICE; } alstr_copy_cstr(&device->DeviceName, deviceName); self->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* FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); 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, device->AmbiOrder); data->cparams.format.format=format; if(snd_pcm_plugin_params(data->pcmHandle, &data->cparams) < 0) { snd_pcm_close(data->pcmHandle); free(data); return ALC_INVALID_VALUE; } return ALC_NO_ERROR; } static void qsa_close_capture(CaptureWrapper *self) { qsa_data *data = self->ExtraData; if (data->pcmHandle!=NULL) snd_pcm_close(data->pcmHandle); free(data); self->ExtraData = NULL; } static void qsa_start_capture(CaptureWrapper *self) { qsa_data *data = self->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); } static void qsa_stop_capture(CaptureWrapper *self) { qsa_data *data = self->ExtraData; snd_pcm_capture_flush(data->pcmHandle); } static ALCuint qsa_available_samples(CaptureWrapper *self) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->ExtraData; snd_pcm_channel_status_t status; ALint frame_size = FrameSizeFromDevFmt(device->FmtChans, device->FmtType, device->AmbiOrder); 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(CaptureWrapper *self, ALCvoid *buffer, ALCuint samples) { ALCdevice *device = STATIC_CAST(ALCbackend,self)->mDevice; qsa_data *data = self->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, device->AmbiOrder); 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 void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device) { ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(CaptureWrapper, ALCbackend, self); self->ExtraData = NULL; } static void CaptureWrapper_Destruct(CaptureWrapper *self) { if(self->ExtraData) qsa_close_capture(self); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); } static ALCenum CaptureWrapper_open(CaptureWrapper *self, const ALCchar *name) { return qsa_open_capture(self, name); } static ALCboolean CaptureWrapper_start(CaptureWrapper *self) { qsa_start_capture(self); return ALC_TRUE; } static void CaptureWrapper_stop(CaptureWrapper *self) { qsa_stop_capture(self); } static ALCenum CaptureWrapper_captureSamples(CaptureWrapper *self, void *buffer, ALCuint samples) { return qsa_capture_samples(self, buffer, samples); } static ALCuint CaptureWrapper_availableSamples(CaptureWrapper *self) { return qsa_available_samples(self); } typedef struct ALCqsaBackendFactory { DERIVE_FROM_TYPE(ALCbackendFactory); } ALCqsaBackendFactory; #define ALCQSABACKENDFACTORY_INITIALIZER { { GET_VTABLE2(ALCqsaBackendFactory, ALCbackendFactory) } } static ALCboolean ALCqsaBackendFactory_init(ALCqsaBackendFactory* UNUSED(self)); static void ALCqsaBackendFactory_deinit(ALCqsaBackendFactory* UNUSED(self)); static ALCboolean ALCqsaBackendFactory_querySupport(ALCqsaBackendFactory* UNUSED(self), ALCbackend_Type type); static void ALCqsaBackendFactory_probe(ALCqsaBackendFactory* UNUSED(self), enum DevProbe type); static ALCbackend* ALCqsaBackendFactory_createBackend(ALCqsaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type); DEFINE_ALCBACKENDFACTORY_VTABLE(ALCqsaBackendFactory); static ALCboolean ALCqsaBackendFactory_init(ALCqsaBackendFactory* UNUSED(self)) { return ALC_TRUE; } static void ALCqsaBackendFactory_deinit(ALCqsaBackendFactory* UNUSED(self)) { #define FREE_NAME(iter) free((iter)->name) VECTOR_FOR_EACH(DevMap, DeviceNameMap, FREE_NAME); VECTOR_DEINIT(DeviceNameMap); VECTOR_FOR_EACH(DevMap, CaptureNameMap, FREE_NAME); VECTOR_DEINIT(CaptureNameMap); #undef FREE_NAME } static ALCboolean ALCqsaBackendFactory_querySupport(ALCqsaBackendFactory* UNUSED(self), ALCbackend_Type type) { if(type == ALCbackend_Playback || type == ALCbackend_Capture) return ALC_TRUE; return ALC_FALSE; } static void ALCqsaBackendFactory_probe(ALCqsaBackendFactory* UNUSED(self), enum DevProbe type) { switch (type) { case ALL_DEVICE_PROBE: #define FREE_NAME(iter) free((iter)->name) VECTOR_FOR_EACH(DevMap, DeviceNameMap, FREE_NAME); VECTOR_RESIZE(DeviceNameMap, 0, 0); #undef FREE_NAME deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap); #define APPEND_DEVICE(iter) AppendAllDevicesList((iter)->name) VECTOR_FOR_EACH(const DevMap, DeviceNameMap, APPEND_DEVICE); #undef APPEND_DEVICE break; case CAPTURE_DEVICE_PROBE: #define FREE_NAME(iter) free((iter)->name) VECTOR_FOR_EACH(DevMap, CaptureNameMap, FREE_NAME); VECTOR_RESIZE(CaptureNameMap, 0, 0); #undef FREE_NAME deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap); #define APPEND_DEVICE(iter) AppendCaptureDeviceList((iter)->name) VECTOR_FOR_EACH(const DevMap, CaptureNameMap, APPEND_DEVICE); #undef APPEND_DEVICE break; } } static ALCbackend* ALCqsaBackendFactory_createBackend(ALCqsaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type) { if(type == ALCbackend_Playback) { PlaybackWrapper *backend; NEW_OBJ(backend, PlaybackWrapper)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } if(type == ALCbackend_Capture) { CaptureWrapper *backend; NEW_OBJ(backend, CaptureWrapper)(device); if(!backend) return NULL; return STATIC_CAST(ALCbackend, backend); } return NULL; } ALCbackendFactory *ALCqsaBackendFactory_getFactory(void) { static ALCqsaBackendFactory factory = ALCQSABACKENDFACTORY_INITIALIZER; return STATIC_CAST(ALCbackendFactory, &factory); }