/** * 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" 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; static const ALCchar qsaDevice[] = "QSA Default"; static DevMap* allDevNameMap; static ALuint numDevNames; static DevMap* allCaptureDevNameMap; static ALuint numCaptureDevNames; 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 DevMap *deviceList(int type, ALuint *count) { snd_ctl_t* handle; snd_pcm_info_t pcminfo; int max_cards, card, err, dev, num_devices, idx; DevMap* dev_list; char name[1024]; struct snd_ctl_hw_info info; void* temp; idx=0; num_devices=0; max_cards=snd_cards(); if (max_cards<=0) { return 0; } dev_list=malloc(sizeof(DevMap)*1); dev_list[0].name=strdup(qsaDevice); num_devices=1; for (card=0; cardExtraData; char* write_ptr; int avail; snd_pcm_channel_status_t status; struct sched_param param; fd_set wfds; int selectret; struct timeval timeout; 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); ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType); while (!data->killNow) { ALint len=data->size; write_ptr=data->buffer; avail=len/frame_size; aluMixData(device, write_ptr, avail); while (len>0 && !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 */ selectret=select(data->audio_fd+1, NULL, &wfds, NULL, &timeout); switch (selectret) { case -1: aluHandleDisconnect(device); return 1; case 0: break; default: if (FD_ISSET(data->audio_fd, &wfds)) { break; } break; } 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; } } } return 0; } /************/ /* Playback */ /************/ static ALCenum qsa_open_playback(ALCdevice* device, const ALCchar* deviceName) { qsa_data* data; char driver[64]; int status; int card, dev; strncpy(driver, GetConfigValue("qsa", "device", qsaDevice), sizeof(driver)-1); driver[sizeof(driver)-1]=0; data=(qsa_data*)calloc(1, sizeof(qsa_data)); if (data==NULL) { return ALC_OUT_OF_MEMORY; } if (!deviceName) { deviceName=driver; } if (strcmp(deviceName, qsaDevice)==0) { if (!deviceName) { deviceName=qsaDevice; } status=snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_PLAYBACK); } else { size_t idx; if (!allDevNameMap) { allDevNameMap=deviceList(SND_PCM_CHANNEL_PLAYBACK, &numDevNames); } for (idx=0; idx0) { break; } } } if (idx==numDevNames) { free(data); return ALC_INVALID_DEVICE; } status=snd_pcm_open(&data->pcmHandle, allDevNameMap[idx].card, allDevNameMap[idx].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) { free(data); return ALC_INVALID_DEVICE; } al_string_copy_cstr(&device->DeviceName, deviceName); device->ExtraData = data; return ALC_NO_ERROR; } static void qsa_close_playback(ALCdevice* device) { qsa_data* data=(qsa_data*)device->ExtraData; if (data->buffer!=NULL) { free(data->buffer); data->buffer=NULL; } snd_pcm_close(data->pcmHandle); free(data); device->ExtraData=NULL; } static ALCboolean qsa_reset_playback(ALCdevice* device) { qsa_data* data=(qsa_data*)device->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* ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType); 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); 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/ (ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType)); 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(ALCdevice* device) { qsa_data *data = (qsa_data*)device->ExtraData; data->killNow = 0; if(althrd_create(&data->thread, qsa_proc_playback, device) != althrd_success) return ALC_FALSE; return ALC_TRUE; } static void qsa_stop_playback(ALCdevice* device) { qsa_data *data = (qsa_data*)device->ExtraData; int res; if(data->killNow) return; data->killNow = 1; althrd_join(data->thread, &res); } /***********/ /* Capture */ /***********/ static ALCenum qsa_open_capture(ALCdevice* device, const ALCchar* deviceName) { qsa_data* data; int format=-1; char driver[64]; int card, dev; int status; strncpy(driver, GetConfigValue("qsa", "capture", qsaDevice), sizeof(driver)-1); driver[sizeof(driver)-1]=0; data=(qsa_data*)calloc(1, sizeof(qsa_data)); if (data==NULL) { return ALC_OUT_OF_MEMORY; } if (!deviceName) { deviceName=driver; } if (strcmp(deviceName, qsaDevice)==0) { if (!deviceName) { deviceName=qsaDevice; } status=snd_pcm_open_preferred(&data->pcmHandle, &card, &dev, SND_PCM_OPEN_CAPTURE); } else { size_t idx; if (!allCaptureDevNameMap) { allCaptureDevNameMap=deviceList(SND_PCM_CHANNEL_CAPTURE, &numDevNames); } for (idx=0; idx0) { break; } } } if (idx==numDevNames) { free(data); return ALC_INVALID_DEVICE; } status=snd_pcm_open(&data->pcmHandle, allCaptureDevNameMap[idx].card, allCaptureDevNameMap[idx].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) { free(data); return ALC_INVALID_DEVICE; } al_string_copy_cstr(&device->DeviceName, deviceName); device->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* ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType); 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); 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_INVALID_VALUE; } } /* 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; } return ALC_NO_ERROR; } static void qsa_close_capture(ALCdevice* device) { qsa_data* data=(qsa_data*)device->ExtraData; if (data->pcmHandle!=NULL) { snd_pcm_close(data->pcmHandle); } free(data); device->ExtraData=NULL; } static void qsa_start_capture(ALCdevice* device) { qsa_data* data=(qsa_data*)device->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); device->UpdateSize=data->csetup.buf.block.frag_size/ (ChannelsFromDevFmt(device->FmtChans)*BytesFromDevFmt(device->FmtType)); device->NumUpdates=data->csetup.buf.block.frags; } static void qsa_stop_capture(ALCdevice* device) { qsa_data* data=(qsa_data*)device->ExtraData; snd_pcm_capture_flush(data->pcmHandle); } static ALCuint qsa_available_samples(ALCdevice* device) { qsa_data* data=(qsa_data*)device->ExtraData; snd_pcm_channel_status_t status; ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType); 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(ALCdevice *device, ALCvoid *buffer, ALCuint samples) { qsa_data* data=(qsa_data*)device->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); 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 ALint64 qsa_get_latency(ALCdevice* device) { ALint frame_size=FrameSizeFromDevFmt(device->FmtChans, device->FmtType); return (ALint64)(device->UpdateSize*device->NumUpdates/frame_size)* 1000000000/device->Frequency; } BackendFuncs qsa_funcs= { qsa_open_playback, qsa_close_playback, qsa_reset_playback, qsa_start_playback, qsa_stop_playback, qsa_open_capture, qsa_close_capture, qsa_start_capture, qsa_stop_capture, qsa_capture_samples, qsa_available_samples, qsa_get_latency, }; ALCboolean alc_qsa_init(BackendFuncs* func_list) { *func_list=qsa_funcs; return ALC_TRUE; } void alc_qsa_deinit(void) { ALuint i; for (i=0; i