diff options
author | Chris Robinson <[email protected]> | 2018-11-13 01:56:34 -0800 |
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committer | Chris Robinson <[email protected]> | 2018-11-13 01:56:34 -0800 |
commit | 0ff349a714eac9bc0cf6cbdaafbddcec79f2442e (patch) | |
tree | 36a6804f92b45c9ac053eec7020716de58c436b9 /Alc/backends/qsa.cpp | |
parent | 09ea1d58f63ad3fe248b1e59c0a3634447ce672d (diff) |
Convert the QSA backend to C++
This may very well not work, since there's no testing and my IDE is not able to
show real problems over the incompatibilities with ALSA headers.
Diffstat (limited to 'Alc/backends/qsa.cpp')
-rw-r--r-- | Alc/backends/qsa.cpp | 1078 |
1 files changed, 1078 insertions, 0 deletions
diff --git a/Alc/backends/qsa.cpp b/Alc/backends/qsa.cpp new file mode 100644 index 00000000..778e9e44 --- /dev/null +++ b/Alc/backends/qsa.cpp @@ -0,0 +1,1078 @@ +/** + * 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 <stdlib.h> +#include <stdio.h> +#include <sched.h> +#include <errno.h> +#include <memory.h> +#include <sys/select.h> +#include <sys/asoundlib.h> +#include <sys/neutrino.h> + +#include "alMain.h" +#include "alu.h" +#include "threads.h" + +#include "backends/base.h" + + +namespace { + +struct qsa_data { + snd_pcm_t* pcmHandle; + int audio_fd; + + snd_pcm_channel_setup_t csetup; + snd_pcm_channel_params_t cparams; + + ALvoid* buffer; + ALsizei size; + + ATOMIC(ALenum) killNow; + althrd_t thread; +}; + +struct DevMap { + ALCchar* name; + int card; + int dev; +}; +TYPEDEF_VECTOR(DevMap, vector_DevMap) + +vector_DevMap DeviceNameMap; +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, 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; + +#define FREE_NAME(iter) free((iter)->name) + VECTOR_FOR_EACH(DevMap, *devmap, FREE_NAME); +#undef FREE_NAME + 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); + } +} + +} // namespace + + +/* Wrappers to use an old-style backend with the new interface. */ +struct PlaybackWrapper final : public ALCbackend { + qsa_data *ExtraData; +}; + +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(!ATOMIC_LOAD(&data->killNow, almemory_order_acquire)) + { + 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, "Failed waiting for playback buffer: %s", strerror(errno)); + break; + } + if(sret == 0) + { + ERR("select timeout\n"); + continue; + } + + len = data->size; + write_ptr = static_cast<char*>(data->buffer); + aluMixData(device, write_ptr, len/frame_size); + while(len>0 && !ATOMIC_LOAD(&data->killNow, almemory_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; + } + } + } + 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; + ATOMIC_INIT(&data->killNow, AL_TRUE); + + 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; + + ATOMIC_STORE(&data->killNow, AL_FALSE, almemory_order_release); + 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(ATOMIC_EXCHANGE(&data->killNow, AL_TRUE, almemory_order_acq_rel)) + return; + althrd_join(data->thread, &res); +} + + +static void PlaybackWrapper_Construct(PlaybackWrapper *self, ALCdevice *device) +{ + new (self) PlaybackWrapper{}; + 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)); + self->~PlaybackWrapper(); +} + +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 */ +/***********/ + +struct CaptureWrapper final : public ALCbackend { + qsa_data *ExtraData; +}; + +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, "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 = 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 = static_cast<char*>(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, "Failed to check capture samples"); + 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, "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; +} + + +static void CaptureWrapper_Construct(CaptureWrapper *self, ALCdevice *device) +{ + new (self) CaptureWrapper{}; + 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)); + self->~CaptureWrapper(); +} + +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); +} + + +struct ALCqsaBackendFactory final : public ALCbackendFactory { + ALCqsaBackendFactory() noexcept; +}; + +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, al_string *outnames); +static ALCbackend* ALCqsaBackendFactory_createBackend(ALCqsaBackendFactory* UNUSED(self), ALCdevice *device, ALCbackend_Type type); +DEFINE_ALCBACKENDFACTORY_VTABLE(ALCqsaBackendFactory); + +ALCqsaBackendFactory::ALCqsaBackendFactory() noexcept + : ALCbackendFactory{GET_VTABLE2(ALCqsaBackendFactory, ALCbackendFactory)} +{ } + + +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, al_string *outnames) +{ + switch (type) + { +#define APPEND_OUTNAME(e) do { \ + const char *n_ = (e)->name; \ + if(n_ && n_[0]) \ + alstr_append_range(outnames, n_, n_+strlen(n_)+1); \ +} while(0) + case ALL_DEVICE_PROBE: + deviceList(SND_PCM_CHANNEL_PLAYBACK, &DeviceNameMap); + VECTOR_FOR_EACH(const DevMap, DeviceNameMap, APPEND_OUTNAME); + break; + + case CAPTURE_DEVICE_PROBE: + deviceList(SND_PCM_CHANNEL_CAPTURE, &CaptureNameMap); + VECTOR_FOR_EACH(const DevMap, CaptureNameMap, APPEND_OUTNAME); + break; +#undef APPEND_OUTNAME + } +} + +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{}; + return STATIC_CAST(ALCbackendFactory, &factory); +} |