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authorChris Robinson <[email protected]>2013-03-14 01:29:20 -0700
committerChris Robinson <[email protected]>2013-03-14 01:29:20 -0700
commitd237410d1d40d65664381d5dbbe3a66e3291d1fa (patch)
treebec7ad7bfb185722c374bb1b1edeeac3d324ed9f
parent2486f13dae69fda15dec9e4aa1c61e7b235d62ff (diff)
Add a QSA backend for QNX
-rw-r--r--Alc/ALc.c3
-rw-r--r--Alc/backends/qsa.c1170
-rw-r--r--CMakeLists.txt30
-rw-r--r--OpenAL32/Include/alMain.h5
-rw-r--r--alsoftrc.sample2
-rw-r--r--config.h.in3
6 files changed, 1211 insertions, 2 deletions
diff --git a/Alc/ALc.c b/Alc/ALc.c
index 094c974c..e859a6c2 100644
--- a/Alc/ALc.c
+++ b/Alc/ALc.c
@@ -62,6 +62,9 @@ static struct BackendInfo BackendList[] = {
#ifdef HAVE_SNDIO
{ "sndio", alc_sndio_init, alc_sndio_deinit, alc_sndio_probe, EmptyFuncs },
#endif
+#ifdef HAVE_QSA
+ { "qsa", alc_qsa_init, alc_qsa_deinit, alc_qsa_probe, EmptyFuncs },
+#endif
#ifdef HAVE_MMDEVAPI
{ "mmdevapi", alcMMDevApiInit, alcMMDevApiDeinit, alcMMDevApiProbe, EmptyFuncs },
#endif
diff --git a/Alc/backends/qsa.c b/Alc/backends/qsa.c
new file mode 100644
index 00000000..c659ce2f
--- /dev/null
+++ b/Alc/backends/qsa.c
@@ -0,0 +1,1170 @@
+/**
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, 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"
+
+typedef struct
+{
+ snd_pcm_t* pcmHandle;
+ int audio_fd;
+
+ ALvoid* buffer;
+ ALsizei size;
+
+ volatile int killNow;
+ ALvoid* thread;
+
+ snd_pcm_channel_setup_t csetup;
+ snd_pcm_channel_params_t cparams;
+} 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; 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)))
+ {
+ temp=realloc(dev_list, sizeof(DevMap)*(num_devices+1));
+ if (temp)
+ {
+ dev_list=temp;
+ snprintf(name, sizeof(name), "%s [%s] (hw:%d,%d)", info.name, pcminfo.name, card, dev);
+ dev_list[num_devices].name=strdup(name);
+ dev_list[num_devices].card=card;
+ dev_list[num_devices].dev=dev;
+ num_devices++;
+ }
+ }
+ }
+ snd_ctl_close (handle);
+ }
+
+ *count=num_devices;
+
+ return dev_list;
+}
+
+static ALuint qsa_proc_playback(ALvoid* ptr)
+{
+ ALCdevice* device=(ALCdevice*)ptr;
+ qsa_data* data=(qsa_data*)device->ExtraData;
+ char* write_ptr;
+ int avail;
+ snd_pcm_channel_status_t status;
+ struct sched_param param;
+ fd_set wfds;
+ int selectret;
+ struct timeval timeout;
+
+ SetRTPriority();
+
+ /* Increase default 10 priority to 11 to avoid jerky sound */
+ SchedGet(0, 0, &param);
+ param.sched_priority=param.sched_curpriority+1;
+ SchedSet(0, 0, SCHED_NOCHANGE, &param);
+
+ 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; idx<numDevNames; idx++)
+ {
+ if (allDevNameMap[idx].name && strcmp(deviceName, allDevNameMap[idx].name)==0)
+ {
+ if (idx>0)
+ {
+ 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;
+ }
+
+ device->DeviceName=strdup(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->thread=StartThread(qsa_proc_playback, device);
+ if (data->thread==NULL)
+ {
+ return ALC_FALSE;
+ }
+
+ return ALC_TRUE;
+}
+
+static void qsa_stop_playback(ALCdevice* device)
+{
+ qsa_data* data=(qsa_data*)device->ExtraData;
+
+ if (data->thread)
+ {
+ data->killNow=1;
+ StopThread(data->thread);
+ data->thread=NULL;
+ }
+ data->killNow=0;
+}
+
+/***********/
+/* 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; idx<numDevNames; idx++)
+ {
+ if (allCaptureDevNameMap[idx].name && strcmp(deviceName, allCaptureDevNameMap[idx].name)==0)
+ {
+ if (idx>0)
+ {
+ 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;
+ }
+
+ device->DeviceName=strdup(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,
+ ALCdevice_LockDefault,
+ ALCdevice_UnlockDefault,
+ 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<numDevNames; ++i)
+ {
+ free(allDevNameMap[i].name);
+ }
+ free(allDevNameMap);
+ allDevNameMap=NULL;
+ numDevNames=0;
+
+ for (i=0; i<numCaptureDevNames; ++i)
+ {
+ free(allCaptureDevNameMap[i].name);
+ }
+ free(allCaptureDevNameMap);
+ allCaptureDevNameMap=NULL;
+ numCaptureDevNames=0;
+}
+
+void alc_qsa_probe(enum DevProbe type)
+{
+ ALuint i;
+
+ switch (type)
+ {
+ case ALL_DEVICE_PROBE:
+ for (i=0; i<numDevNames; ++i)
+ {
+ free(allDevNameMap[i].name);
+ }
+ free(allDevNameMap);
+
+ allDevNameMap=deviceList(SND_PCM_CHANNEL_PLAYBACK, &numDevNames);
+ for (i=0; i<numDevNames; ++i)
+ {
+ AppendAllDevicesList(allDevNameMap[i].name);
+ }
+ break;
+ case CAPTURE_DEVICE_PROBE:
+ for (i=0; i<numCaptureDevNames; ++i)
+ {
+ free(allCaptureDevNameMap[i].name);
+ }
+ free(allCaptureDevNameMap);
+
+ allCaptureDevNameMap=deviceList(SND_PCM_CHANNEL_CAPTURE, &numCaptureDevNames);
+ for (i=0; i<numCaptureDevNames; ++i)
+ {
+ AppendCaptureDeviceList(allCaptureDevNameMap[i].name);
+ }
+ break;
+ }
+}
diff --git a/CMakeLists.txt b/CMakeLists.txt
index 4d7c5c97..ed7eb96b 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -37,6 +37,7 @@ OPTION(ALSOFT_BACKEND_ALSA "Check for ALSA backend" ON)
OPTION(ALSOFT_BACKEND_OSS "Check for OSS backend" ON)
OPTION(ALSOFT_BACKEND_SOLARIS "Check for Solaris backend" ON)
OPTION(ALSOFT_BACKEND_SNDIO "Check for SndIO backend" ON)
+OPTION(ALSOFT_BACKEND_QSA "Check for QSA backend" ON)
OPTION(ALSOFT_BACKEND_MMDEVAPI "Check for MMDevApi" ON)
OPTION(ALSOFT_BACKEND_DSOUND "Check for DirectSound backend" ON)
OPTION(ALSOFT_BACKEND_WINMM "Check for Windows Multimedia backend" ON)
@@ -50,6 +51,7 @@ OPTION(ALSOFT_REQUIRE_ALSA "Require ALSA backend" OFF)
OPTION(ALSOFT_REQUIRE_OSS "Require OSS backend" OFF)
OPTION(ALSOFT_REQUIRE_SOLARIS "Require Solaris backend" OFF)
OPTION(ALSOFT_REQUIRE_SNDIO "Require SndIO backend" OFF)
+OPTION(ALSOFT_REQUIRE_QSA "Require QSA backend" OFF)
OPTION(ALSOFT_REQUIRE_MMDEVAPI "Require MMDevApi" OFF)
OPTION(ALSOFT_REQUIRE_DSOUND "Require DirectSound backend" OFF)
OPTION(ALSOFT_REQUIRE_WINMM "Require Windows Multimedia backend" OFF)
@@ -76,6 +78,12 @@ ELSE()
SET(LIBNAME openal)
ENDIF()
+# QNX's gcc do not uses /usr/include and /usr/lib pathes by default
+IF ("${CMAKE_C_PLATFORM_ID}" STREQUAL "QNX")
+ ADD_DEFINITIONS("-I/usr/include")
+ SET(EXTRA_LIBS ${EXTRA_LIBS} -L/usr/lib)
+ENDIF()
+
IF(NOT LIBTYPE)
SET(LIBTYPE SHARED)
ENDIF()
@@ -509,6 +517,7 @@ SET(HAVE_ALSA 0)
SET(HAVE_OSS 0)
SET(HAVE_SOLARIS 0)
SET(HAVE_SNDIO 0)
+SET(HAVE_QSA 0)
SET(HAVE_DSOUND 0)
SET(HAVE_MMDEVAPI 0)
SET(HAVE_WINMM 0)
@@ -582,6 +591,27 @@ IF(ALSOFT_REQUIRE_SNDIO AND NOT HAVE_SNDIO)
MESSAGE(FATAL_ERROR "Failed to enabled required SndIO backend")
ENDIF()
+# Check QSA backend
+IF (ALSOFT_BACKEND_QSA AND "${CMAKE_C_PLATFORM_ID}" STREQUAL "QNX")
+ CHECK_INCLUDE_FILE(sys/asoundlib.h HAVE_SYS_ASOUNDLIB_H)
+ IF(HAVE_SYS_ASOUNDLIB_H)
+ CHECK_SHARED_FUNCTION_EXISTS(snd_pcm_open "sys/asoundlib.h" asound "" HAVE_LIBASOUND)
+ IF(HAVE_LIBASOUND OR HAVE_DLFCN_H)
+ SET(HAVE_QSA 1)
+ SET(ALC_OBJS ${ALC_OBJS} Alc/backends/qsa.c)
+ SET(EXTRA_LIBS asound ${EXTRA_LIBS})
+ IF(HAVE_DLFCN_H)
+ SET(BACKENDS "${BACKENDS} QSA,")
+ ELSE()
+ SET(BACKENDS "${BACKENDS} QSA \(linked\),")
+ ENDIF()
+ ENDIF()
+ ENDIF()
+ENDIF()
+IF(ALSOFT_REQUIRE_QSA AND NOT HAVE_QSA)
+ MESSAGE(FATAL_ERROR "Failed to enabled required QSA backend")
+ENDIF()
+
# Check for MMDevApi backend
IF(HAVE_WINDOWS_H)
IF(ALSOFT_BACKEND_MMDEVAPI)
diff --git a/OpenAL32/Include/alMain.h b/OpenAL32/Include/alMain.h
index 1110e89e..198ab436 100644
--- a/OpenAL32/Include/alMain.h
+++ b/OpenAL32/Include/alMain.h
@@ -118,7 +118,7 @@ void *GetSymbol(void *handle, const char *name);
typedef void *volatile XchgPtr;
-#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1))
+#if defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)) && !defined(__QNXNTO__)
typedef ALuint RefCount;
static __inline RefCount IncrementRef(volatile RefCount *ptr)
{ return __sync_add_and_fetch(ptr, 1); }
@@ -449,6 +449,9 @@ void alc_ca_probe(enum DevProbe type);
ALCboolean alc_opensl_init(BackendFuncs *func_list);
void alc_opensl_deinit(void);
void alc_opensl_probe(enum DevProbe type);
+ALCboolean alc_qsa_init(BackendFuncs *func_list);
+void alc_qsa_deinit(void);
+void alc_qsa_probe(enum DevProbe type);
ALCboolean alc_null_init(BackendFuncs *func_list);
void alc_null_deinit(void);
void alc_null_probe(enum DevProbe type);
diff --git a/alsoftrc.sample b/alsoftrc.sample
index 27aee718..837e01cf 100644
--- a/alsoftrc.sample
+++ b/alsoftrc.sample
@@ -121,7 +121,7 @@
# followed by all other available backends, while 'oss' will list OSS only).
# Backends prepended with - won't be available for use (eg. '-oss,' will allow
# all available backends except OSS). An empty list means the default.
-#drivers = pulse,alsa,core,oss,solaris,sndio,mmdevapi,dsound,winmm,port,opensl,null,wave
+#drivers = pulse,alsa,core,oss,solaris,sndio,qsa,mmdevapi,dsound,winmm,port,opensl,null,wave
## excludefx:
# Sets which effects to exclude, preventing apps from using them. This can
diff --git a/config.h.in b/config.h.in
index 06c34c81..6f45fa84 100644
--- a/config.h.in
+++ b/config.h.in
@@ -41,6 +41,9 @@
/* Define if we have the SndIO backend */
#cmakedefine HAVE_SNDIO
+/* Define if we have the QSA backend */
+#cmakedefine HAVE_QSA
+
/* Define if we have the MMDevApi backend */
#cmakedefine HAVE_MMDEVAPI