🛠️🐜 Antkeeper superbuild with dependencies included https://antkeeper.com
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
 
 
 
 
 
 

751 lines
20 KiB

/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
#include "config.h"
#include "backends/oss.h"
#include <sys/ioctl.h>
#include <sys/types.h>
#include <sys/time.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <cstdlib>
#include <cstdio>
#include <cstring>
#include <memory.h>
#include <unistd.h>
#include <cerrno>
#include <poll.h>
#include <cmath>
#include <atomic>
#include <thread>
#include <vector>
#include <string>
#include <algorithm>
#include <functional>
#include "alMain.h"
#include "alu.h"
#include "alconfig.h"
#include "ringbuffer.h"
#include "compat.h"
#include <sys/soundcard.h>
/*
* The OSS documentation talks about SOUND_MIXER_READ, but the header
* only contains MIXER_READ. Play safe. Same for WRITE.
*/
#ifndef SOUND_MIXER_READ
#define SOUND_MIXER_READ MIXER_READ
#endif
#ifndef SOUND_MIXER_WRITE
#define SOUND_MIXER_WRITE MIXER_WRITE
#endif
#if defined(SOUND_VERSION) && (SOUND_VERSION < 0x040000)
#define ALC_OSS_COMPAT
#endif
#ifndef SNDCTL_AUDIOINFO
#define ALC_OSS_COMPAT
#endif
/*
* FreeBSD strongly discourages the use of specific devices,
* such as those returned in oss_audioinfo.devnode
*/
#ifdef __FreeBSD__
#define ALC_OSS_DEVNODE_TRUC
#endif
namespace {
constexpr char DefaultName[] = "OSS Default";
const char *DefaultPlayback{"/dev/dsp"};
const char *DefaultCapture{"/dev/dsp"};
struct DevMap {
std::string name;
std::string device_name;
template<typename StrT0, typename StrT1>
DevMap(StrT0&& name_, StrT1&& devname_)
: name{std::forward<StrT0>(name_)}, device_name{std::forward<StrT1>(devname_)}
{ }
};
bool checkName(const al::vector<DevMap> &list, const std::string &name)
{
return std::find_if(list.cbegin(), list.cend(),
[&name](const DevMap &entry) -> bool
{ return entry.name == name; }
) != list.cend();
}
al::vector<DevMap> PlaybackDevices;
al::vector<DevMap> CaptureDevices;
#ifdef ALC_OSS_COMPAT
#define DSP_CAP_OUTPUT 0x00020000
#define DSP_CAP_INPUT 0x00010000
void ALCossListPopulate(al::vector<DevMap> *devlist, int type)
{
devlist->emplace_back(DefaultName, (type==DSP_CAP_INPUT) ? DefaultCapture : DefaultPlayback);
}
#else
void ALCossListAppend(al::vector<DevMap> *list, const char *handle, size_t hlen, const char *path, size_t plen)
{
#ifdef ALC_OSS_DEVNODE_TRUC
for(size_t i{0};i < plen;i++)
{
if(path[i] == '.')
{
if(strncmp(path + i, handle + hlen + i - plen, plen - i) == 0)
hlen = hlen + i - plen;
plen = i;
}
}
#endif
if(handle[0] == '\0')
{
handle = path;
hlen = plen;
}
std::string basename{handle, hlen};
basename.erase(std::find(basename.begin(), basename.end(), '\0'), basename.end());
std::string devname{path, plen};
devname.erase(std::find(devname.begin(), devname.end(), '\0'), devname.end());
auto iter = std::find_if(list->cbegin(), list->cend(),
[&devname](const DevMap &entry) -> bool
{ return entry.device_name == devname; }
);
if(iter != list->cend())
return;
int count{1};
std::string newname{basename};
while(checkName(PlaybackDevices, newname))
{
newname = basename;
newname += " #";
newname += std::to_string(++count);
}
list->emplace_back(std::move(newname), std::move(devname));
const DevMap &entry = list->back();
TRACE("Got device \"%s\", \"%s\"\n", entry.name.c_str(), entry.device_name.c_str());
}
void ALCossListPopulate(al::vector<DevMap> *devlist, int type_flag)
{
int fd{open("/dev/mixer", O_RDONLY)};
if(fd < 0)
{
TRACE("Could not open /dev/mixer: %s\n", strerror(errno));
goto done;
}
oss_sysinfo si;
if(ioctl(fd, SNDCTL_SYSINFO, &si) == -1)
{
TRACE("SNDCTL_SYSINFO failed: %s\n", strerror(errno));
goto done;
}
for(int i{0};i < si.numaudios;i++)
{
oss_audioinfo ai;
ai.dev = i;
if(ioctl(fd, SNDCTL_AUDIOINFO, &ai) == -1)
{
ERR("SNDCTL_AUDIOINFO (%d) failed: %s\n", i, strerror(errno));
continue;
}
if(!(ai.caps&type_flag) || ai.devnode[0] == '\0')
continue;
const char *handle;
size_t len;
if(ai.handle[0] != '\0')
{
len = strnlen(ai.handle, sizeof(ai.handle));
handle = ai.handle;
}
else
{
len = strnlen(ai.name, sizeof(ai.name));
handle = ai.name;
}
ALCossListAppend(devlist, handle, len, ai.devnode,
strnlen(ai.devnode, sizeof(ai.devnode)));
}
done:
if(fd >= 0)
close(fd);
fd = -1;
const char *defdev{(type_flag==DSP_CAP_INPUT) ? DefaultCapture : DefaultPlayback};
auto iter = std::find_if(devlist->cbegin(), devlist->cend(),
[defdev](const DevMap &entry) -> bool
{ return entry.device_name == defdev; }
);
if(iter == devlist->cend())
devlist->insert(devlist->begin(), DevMap{DefaultName, defdev});
else
{
DevMap entry{std::move(*iter)};
devlist->erase(iter);
devlist->insert(devlist->begin(), std::move(entry));
}
devlist->shrink_to_fit();
}
#endif
int log2i(ALCuint x)
{
int y = 0;
while (x > 1)
{
x >>= 1;
y++;
}
return y;
}
struct OSSPlayback final : public BackendBase {
OSSPlayback(ALCdevice *device) noexcept : BackendBase{device} { }
~OSSPlayback() override;
int mixerProc();
ALCenum open(const ALCchar *name) override;
ALCboolean reset() override;
ALCboolean start() override;
void stop() override;
int mFd{-1};
al::vector<ALubyte> mMixData;
std::atomic<bool> mKillNow{true};
std::thread mThread;
static constexpr inline const char *CurrentPrefix() noexcept { return "OSSPlayback::"; }
DEF_NEWDEL(OSSPlayback)
};
OSSPlayback::~OSSPlayback()
{
if(mFd != -1)
close(mFd);
mFd = -1;
}
int OSSPlayback::mixerProc()
{
SetRTPriority();
althrd_setname(MIXER_THREAD_NAME);
const int frame_size{mDevice->frameSizeFromFmt()};
lock();
while(!mKillNow.load(std::memory_order_acquire) &&
mDevice->Connected.load(std::memory_order_acquire))
{
pollfd pollitem{};
pollitem.fd = mFd;
pollitem.events = POLLOUT;
unlock();
int pret{poll(&pollitem, 1, 1000)};
lock();
if(pret < 0)
{
if(errno == EINTR || errno == EAGAIN)
continue;
ERR("poll failed: %s\n", strerror(errno));
aluHandleDisconnect(mDevice, "Failed waiting for playback buffer: %s", strerror(errno));
break;
}
else if(pret == 0)
{
WARN("poll timeout\n");
continue;
}
ALubyte *write_ptr{mMixData.data()};
size_t to_write{mMixData.size()};
aluMixData(mDevice, write_ptr, to_write/frame_size);
while(to_write > 0 && !mKillNow.load(std::memory_order_acquire))
{
ssize_t wrote{write(mFd, write_ptr, to_write)};
if(wrote < 0)
{
if(errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR)
continue;
ERR("write failed: %s\n", strerror(errno));
aluHandleDisconnect(mDevice, "Failed writing playback samples: %s",
strerror(errno));
break;
}
to_write -= wrote;
write_ptr += wrote;
}
}
unlock();
return 0;
}
ALCenum OSSPlayback::open(const ALCchar *name)
{
const char *devname{DefaultPlayback};
if(!name)
name = DefaultName;
else
{
if(PlaybackDevices.empty())
ALCossListPopulate(&PlaybackDevices, DSP_CAP_OUTPUT);
auto iter = std::find_if(PlaybackDevices.cbegin(), PlaybackDevices.cend(),
[&name](const DevMap &entry) -> bool
{ return entry.name == name; }
);
if(iter == PlaybackDevices.cend())
return ALC_INVALID_VALUE;
devname = iter->device_name.c_str();
}
mFd = ::open(devname, O_WRONLY);
if(mFd == -1)
{
ERR("Could not open %s: %s\n", devname, strerror(errno));
return ALC_INVALID_VALUE;
}
mDevice->DeviceName = name;
return ALC_NO_ERROR;
}
ALCboolean OSSPlayback::reset()
{
int numFragmentsLogSize;
int log2FragmentSize;
unsigned int periods;
audio_buf_info info;
ALuint frameSize;
int numChannels;
int ossFormat;
int ossSpeed;
const char *err;
switch(mDevice->FmtType)
{
case DevFmtByte:
ossFormat = AFMT_S8;
break;
case DevFmtUByte:
ossFormat = AFMT_U8;
break;
case DevFmtUShort:
case DevFmtInt:
case DevFmtUInt:
case DevFmtFloat:
mDevice->FmtType = DevFmtShort;
/* fall-through */
case DevFmtShort:
ossFormat = AFMT_S16_NE;
break;
}
periods = mDevice->BufferSize / mDevice->UpdateSize;
numChannels = mDevice->channelsFromFmt();
ossSpeed = mDevice->Frequency;
frameSize = numChannels * mDevice->bytesFromFmt();
/* According to the OSS spec, 16 bytes (log2(16)) is the minimum. */
log2FragmentSize = maxi(log2i(mDevice->UpdateSize*frameSize), 4);
numFragmentsLogSize = (periods << 16) | log2FragmentSize;
#define CHECKERR(func) if((func) < 0) { \
err = #func; \
goto err; \
}
/* Don't fail if SETFRAGMENT fails. We can handle just about anything
* that's reported back via GETOSPACE */
ioctl(mFd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize);
CHECKERR(ioctl(mFd, SNDCTL_DSP_SETFMT, &ossFormat));
CHECKERR(ioctl(mFd, SNDCTL_DSP_CHANNELS, &numChannels));
CHECKERR(ioctl(mFd, SNDCTL_DSP_SPEED, &ossSpeed));
CHECKERR(ioctl(mFd, SNDCTL_DSP_GETOSPACE, &info));
if(0)
{
err:
ERR("%s failed: %s\n", err, strerror(errno));
return ALC_FALSE;
}
#undef CHECKERR
if(mDevice->channelsFromFmt() != numChannels)
{
ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(mDevice->FmtChans),
numChannels);
return ALC_FALSE;
}
if(!((ossFormat == AFMT_S8 && mDevice->FmtType == DevFmtByte) ||
(ossFormat == AFMT_U8 && mDevice->FmtType == DevFmtUByte) ||
(ossFormat == AFMT_S16_NE && mDevice->FmtType == DevFmtShort)))
{
ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(mDevice->FmtType),
ossFormat);
return ALC_FALSE;
}
mDevice->Frequency = ossSpeed;
mDevice->UpdateSize = info.fragsize / frameSize;
mDevice->BufferSize = info.fragments * mDevice->UpdateSize;
SetDefaultChannelOrder(mDevice);
mMixData.resize(mDevice->UpdateSize * mDevice->frameSizeFromFmt());
return ALC_TRUE;
}
ALCboolean OSSPlayback::start()
{
try {
mKillNow.store(false, std::memory_order_release);
mThread = std::thread{std::mem_fn(&OSSPlayback::mixerProc), this};
return ALC_TRUE;
}
catch(std::exception& e) {
ERR("Could not create playback thread: %s\n", e.what());
}
catch(...) {
}
return ALC_FALSE;
}
void OSSPlayback::stop()
{
if(mKillNow.exchange(true, std::memory_order_acq_rel) || !mThread.joinable())
return;
mThread.join();
if(ioctl(mFd, SNDCTL_DSP_RESET) != 0)
ERR("Error resetting device: %s\n", strerror(errno));
}
struct OSScapture final : public BackendBase {
OSScapture(ALCdevice *device) noexcept : BackendBase{device} { }
~OSScapture() override;
int recordProc();
ALCenum open(const ALCchar *name) override;
ALCboolean start() override;
void stop() override;
ALCenum captureSamples(ALCvoid *buffer, ALCuint samples) override;
ALCuint availableSamples() override;
int mFd{-1};
RingBufferPtr mRing{nullptr};
std::atomic<bool> mKillNow{true};
std::thread mThread;
static constexpr inline const char *CurrentPrefix() noexcept { return "OSScapture::"; }
DEF_NEWDEL(OSScapture)
};
OSScapture::~OSScapture()
{
if(mFd != -1)
close(mFd);
mFd = -1;
}
int OSScapture::recordProc()
{
SetRTPriority();
althrd_setname(RECORD_THREAD_NAME);
const int frame_size{mDevice->frameSizeFromFmt()};
while(!mKillNow.load(std::memory_order_acquire))
{
pollfd pollitem{};
pollitem.fd = mFd;
pollitem.events = POLLIN;
int sret{poll(&pollitem, 1, 1000)};
if(sret < 0)
{
if(errno == EINTR || errno == EAGAIN)
continue;
ERR("poll failed: %s\n", strerror(errno));
aluHandleDisconnect(mDevice, "Failed to check capture samples: %s", strerror(errno));
break;
}
else if(sret == 0)
{
WARN("poll timeout\n");
continue;
}
auto vec = mRing->getWriteVector();
if(vec.first.len > 0)
{
ssize_t amt{read(mFd, vec.first.buf, vec.first.len*frame_size)};
if(amt < 0)
{
ERR("read failed: %s\n", strerror(errno));
aluHandleDisconnect(mDevice, "Failed reading capture samples: %s",
strerror(errno));
break;
}
mRing->writeAdvance(amt/frame_size);
}
}
return 0;
}
ALCenum OSScapture::open(const ALCchar *name)
{
const char *devname{DefaultCapture};
if(!name)
name = DefaultName;
else
{
if(CaptureDevices.empty())
ALCossListPopulate(&CaptureDevices, DSP_CAP_INPUT);
auto iter = std::find_if(CaptureDevices.cbegin(), CaptureDevices.cend(),
[&name](const DevMap &entry) -> bool
{ return entry.name == name; }
);
if(iter == CaptureDevices.cend())
return ALC_INVALID_VALUE;
devname = iter->device_name.c_str();
}
mFd = ::open(devname, O_RDONLY);
if(mFd == -1)
{
ERR("Could not open %s: %s\n", devname, strerror(errno));
return ALC_INVALID_VALUE;
}
int ossFormat{};
switch(mDevice->FmtType)
{
case DevFmtByte:
ossFormat = AFMT_S8;
break;
case DevFmtUByte:
ossFormat = AFMT_U8;
break;
case DevFmtShort:
ossFormat = AFMT_S16_NE;
break;
case DevFmtUShort:
case DevFmtInt:
case DevFmtUInt:
case DevFmtFloat:
ERR("%s capture samples not supported\n", DevFmtTypeString(mDevice->FmtType));
return ALC_INVALID_VALUE;
}
int periods{4};
int numChannels{mDevice->channelsFromFmt()};
int frameSize{numChannels * mDevice->bytesFromFmt()};
int ossSpeed{static_cast<int>(mDevice->Frequency)};
int log2FragmentSize{log2i(mDevice->BufferSize * frameSize / periods)};
/* according to the OSS spec, 16 bytes are the minimum */
log2FragmentSize = std::max(log2FragmentSize, 4);
int numFragmentsLogSize{(periods << 16) | log2FragmentSize};
audio_buf_info info;
const char *err;
#define CHECKERR(func) if((func) < 0) { \
err = #func; \
goto err; \
}
CHECKERR(ioctl(mFd, SNDCTL_DSP_SETFRAGMENT, &numFragmentsLogSize));
CHECKERR(ioctl(mFd, SNDCTL_DSP_SETFMT, &ossFormat));
CHECKERR(ioctl(mFd, SNDCTL_DSP_CHANNELS, &numChannels));
CHECKERR(ioctl(mFd, SNDCTL_DSP_SPEED, &ossSpeed));
CHECKERR(ioctl(mFd, SNDCTL_DSP_GETISPACE, &info));
if(0)
{
err:
ERR("%s failed: %s\n", err, strerror(errno));
close(mFd);
mFd = -1;
return ALC_INVALID_VALUE;
}
#undef CHECKERR
if(mDevice->channelsFromFmt() != numChannels)
{
ERR("Failed to set %s, got %d channels instead\n", DevFmtChannelsString(mDevice->FmtChans),
numChannels);
close(mFd);
mFd = -1;
return ALC_INVALID_VALUE;
}
if(!((ossFormat == AFMT_S8 && mDevice->FmtType == DevFmtByte) ||
(ossFormat == AFMT_U8 && mDevice->FmtType == DevFmtUByte) ||
(ossFormat == AFMT_S16_NE && mDevice->FmtType == DevFmtShort)))
{
ERR("Failed to set %s samples, got OSS format %#x\n", DevFmtTypeString(mDevice->FmtType), ossFormat);
close(mFd);
mFd = -1;
return ALC_INVALID_VALUE;
}
mRing = CreateRingBuffer(mDevice->BufferSize, frameSize, false);
if(!mRing)
{
ERR("Ring buffer create failed\n");
close(mFd);
mFd = -1;
return ALC_OUT_OF_MEMORY;
}
mDevice->DeviceName = name;
return ALC_NO_ERROR;
}
ALCboolean OSScapture::start()
{
try {
mKillNow.store(false, std::memory_order_release);
mThread = std::thread{std::mem_fn(&OSScapture::recordProc), this};
return ALC_TRUE;
}
catch(std::exception& e) {
ERR("Could not create record thread: %s\n", e.what());
}
catch(...) {
}
return ALC_FALSE;
}
void OSScapture::stop()
{
if(mKillNow.exchange(true, std::memory_order_acq_rel) || !mThread.joinable())
return;
mThread.join();
if(ioctl(mFd, SNDCTL_DSP_RESET) != 0)
ERR("Error resetting device: %s\n", strerror(errno));
}
ALCenum OSScapture::captureSamples(ALCvoid *buffer, ALCuint samples)
{
mRing->read(buffer, samples);
return ALC_NO_ERROR;
}
ALCuint OSScapture::availableSamples()
{ return mRing->readSpace(); }
} // namespace
BackendFactory &OSSBackendFactory::getFactory()
{
static OSSBackendFactory factory{};
return factory;
}
bool OSSBackendFactory::init()
{
ConfigValueStr(nullptr, "oss", "device", &DefaultPlayback);
ConfigValueStr(nullptr, "oss", "capture", &DefaultCapture);
return true;
}
bool OSSBackendFactory::querySupport(BackendType type)
{ return (type == BackendType::Playback || type == BackendType::Capture); }
void OSSBackendFactory::probe(DevProbe type, std::string *outnames)
{
auto add_device = [outnames](const DevMap &entry) -> void
{
#ifdef HAVE_STAT
struct stat buf;
if(stat(entry.device_name.c_str(), &buf) == 0)
#endif
{
/* Includes null char. */
outnames->append(entry.name.c_str(), entry.name.length()+1);
}
};
switch(type)
{
case DevProbe::Playback:
PlaybackDevices.clear();
ALCossListPopulate(&PlaybackDevices, DSP_CAP_OUTPUT);
std::for_each(PlaybackDevices.cbegin(), PlaybackDevices.cend(), add_device);
break;
case DevProbe::Capture:
CaptureDevices.clear();
ALCossListPopulate(&CaptureDevices, DSP_CAP_INPUT);
std::for_each(CaptureDevices.cbegin(), CaptureDevices.cend(), add_device);
break;
}
}
BackendPtr OSSBackendFactory::createBackend(ALCdevice *device, BackendType type)
{
if(type == BackendType::Playback)
return BackendPtr{new OSSPlayback{device}};
if(type == BackendType::Capture)
return BackendPtr{new OSScapture{device}};
return nullptr;
}