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audio-orchestra/audio/orchestra/api/Alsa.cpp

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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
* @fork from RTAudio
*/
#if defined(ORCHESTRA_BUILD_ALSA)
#include <alsa/asoundlib.h>
#include <unistd.h>
#include <audio/orchestra/Interface.h>
#include <audio/orchestra/debug.h>
#include <etk/stdTools.h>
#include <etk/thread/tools.h>
#include <limits.h>
#include <audio/orchestra/api/Alsa.h>
extern "C" {
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sched.h>
#include <errno.h>
#include <getopt.h>
#include <sys/time.h>
#include <math.h>
#include <poll.h>
}
#undef __class__
#define __class__ "api::Alsa"
audio::orchestra::Api* audio::orchestra::api::Alsa::create() {
return new audio::orchestra::api::Alsa();
}
namespace audio {
namespace orchestra {
namespace api {
class AlsaPrivate {
public:
snd_pcm_t *handle;
bool xrun[2];
std11::condition_variable runnable_cv;
bool runnable;
std11::thread* thread;
bool threadRunning;
bool mmapInterface; //!< enable or disable mmap mode...
enum timestampMode timeMode; //!< the timestamp of the flow came from the harware.
std::vector<snd_pcm_channel_area_t> areas;
AlsaPrivate() :
handle(nullptr),
runnable(false),
thread(nullptr),
threadRunning(false),
mmapInterface(false),
timeMode(timestampMode_soft) {
xrun[0] = false;
xrun[1] = false;
// TODO : Wait thread ...
}
};
}
}
}
audio::orchestra::api::Alsa::Alsa() :
m_private(std::make_shared<audio::orchestra::api::AlsaPrivate>()) {
// Nothing to do here.
}
audio::orchestra::api::Alsa::~Alsa() {
if (m_state != audio::orchestra::state_closed) {
closeStream();
}
}
uint32_t audio::orchestra::api::Alsa::getDeviceCount() {
unsigned nDevices = 0;
int32_t result= -1;
int32_t subdevice = -1;
int32_t card = -1;
char name[64];
snd_ctl_t *handle;
// Count cards and devices
card = -1;
snd_card_next(&card);
while (card >= 0) {
sprintf(name, "hw:%d", card);
result = snd_ctl_open(&handle, name, 0);
if (result < 0) {
ATA_ERROR("control open, card = " << card << ", " << snd_strerror(result) << ".");
// TODO : Return error audio::orchestra::error_warning;
goto nextcard;
}
subdevice = -1;
while(1) {
result = snd_ctl_pcm_next_device(handle, &subdevice);
if (result < 0) {
ATA_ERROR("control next device, card = " << card << ", " << snd_strerror(result) << ".");
// TODO : Return error audio::orchestra::error_warning;
break;
}
if (subdevice < 0) {
break;
}
// input/output
nDevices+=2;
}
nextcard:
snd_ctl_close(handle);
snd_card_next(&card);
}
return nDevices;
}
bool audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(const std::string& _deviceName, audio::orchestra::DeviceInfo& _info, int32_t _cardId, int32_t _subdevice, int32_t _localDeviceId, bool _input) {
int32_t result;
snd_ctl_t *chandle;
int32_t openMode = SND_PCM_ASYNC;
snd_pcm_stream_t stream;
snd_pcm_info_t *pcminfo;
snd_pcm_info_alloca(&pcminfo);
snd_pcm_t *phandle;
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(&params);
// First try for playback unless default _device (which has subdev -1)
_info.input = _input;
if (_input == true) {
stream = SND_PCM_STREAM_CAPTURE;
} else {
stream = SND_PCM_STREAM_PLAYBACK;
}
snd_pcm_info_set_stream(pcminfo, stream);
std::vector<std::string> listElement = etk::split(_deviceName, ',');
if (listElement.size() == 0) {
ATA_ERROR("can not get control interface = '" << _deviceName << "' Can not plit at ',' ...");
return false;
}
if (listElement.size() == 1) {
// need to check if it is an input or output:
listElement = etk::split(_deviceName, '_');
}
ATA_INFO("Open control : " << listElement[0]);
result = snd_ctl_open(&chandle, listElement[0].c_str(), SND_CTL_NONBLOCK);
if (result < 0) {
ATA_ERROR("control open, card = " << listElement[0] << ", " << snd_strerror(result) << ".");
return false;
}
if (_subdevice != -1) {
snd_pcm_info_set_device(pcminfo, _subdevice);
snd_pcm_info_set_subdevice(pcminfo, 0);
result = snd_ctl_pcm_info(chandle, pcminfo);
if (result < 0) {
// Device probably doesn't support IO.
return false;
}
}
// ALSA doesn't provide default devices so we'll use the first available one.
if (_localDeviceId == 0) {
_info.isDefault = true;
}
snd_pcm_info_set_stream(pcminfo, stream);
// Now try for capture unless default device (with subdev = -1)
if (_subdevice != -1) {
result = snd_ctl_pcm_info(chandle, pcminfo);
snd_ctl_close(chandle);
if (result < 0) {
// Device probably doesn't support capture.
return false;
}
}
// open device:
result = snd_pcm_open(&phandle, _deviceName.c_str(), stream, openMode | SND_PCM_NONBLOCK);
if (result < 0) {
ATA_ERROR("snd_pcm_open error for device (" << _deviceName << "), " << snd_strerror(result) << ". (seems to be already open)");
return true;
}
// The device is open ... fill the parameter structure.
result = snd_pcm_hw_params_any(phandle, params);
if (result < 0) {
snd_pcm_close(phandle);
ATA_ERROR("snd_pcm_hw_params error for device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
unsigned int value;
result = snd_pcm_hw_params_get_channels_max(params, &value);
if (result < 0) {
snd_pcm_close(phandle);
ATA_ERROR("error getting device (" << _deviceName << ") input channels, " << snd_strerror(result) << ".");
return false;
}
ATA_ERROR("Input channel = " << value);
for (int32_t iii=0; iii<value; ++iii) {
_info.channels.push_back(audio::channel_unknow);
}
// Test our discrete set of sample rate values.
_info.sampleRates.clear();
for (std::vector<uint32_t>::const_iterator it(audio::orchestra::genericSampleRate().begin());
it != audio::orchestra::genericSampleRate().end();
++it ) {
if (snd_pcm_hw_params_test_rate(phandle, params, *it, 0) == 0) {
_info.sampleRates.push_back(*it);
}
}
if (_info.sampleRates.size() == 0) {
snd_pcm_close(phandle);
ATA_ERROR("no supported sample rates found for device (" << _deviceName << ").");
return false;
}
// Probe the supported data formats ... we don't care about endian-ness just yet
snd_pcm_format_t format;
_info.nativeFormats.clear();
format = SND_PCM_FORMAT_S8;
if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
_info.nativeFormats.push_back(audio::format_int8);
}
format = SND_PCM_FORMAT_S16;
if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
_info.nativeFormats.push_back(audio::format_int16);
}
format = SND_PCM_FORMAT_S24;
if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
_info.nativeFormats.push_back(audio::format_int24);
}
format = SND_PCM_FORMAT_S32;
if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
_info.nativeFormats.push_back(audio::format_int32);
}
format = SND_PCM_FORMAT_FLOAT;
if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
_info.nativeFormats.push_back(audio::format_float);
}
format = SND_PCM_FORMAT_FLOAT64;
if (snd_pcm_hw_params_test_format(phandle, params, format) == 0) {
_info.nativeFormats.push_back(audio::format_double);
}
// Check that we have at least one supported format
if (_info.nativeFormats.size() == 0) {
ATA_ERROR("pcm device (" << _deviceName << ") data format not supported by RtAudio.");
// TODO : Return audio::orchestra::error_warning;
return false;
}
// Get the device name
if (_cardId != -1) {
char *cardname;
char name[1024];
result = snd_card_get_name(_cardId, &cardname);
if (result >= 0) {
sprintf(name, "hw:%s,%d", cardname, _subdevice);
}
_info.name = name;
} else {
_info.name = _deviceName;
}
// That's all ... close the device and return
snd_pcm_close(phandle);
return true;
}
audio::orchestra::DeviceInfo audio::orchestra::api::Alsa::getDeviceInfo(uint32_t _device) {
audio::orchestra::DeviceInfo info;
unsigned nDevices = 0;
int32_t result = -1;
int32_t subdevice = -1;
int32_t card = -1;
bool isInput = false;
char name[64];
snd_ctl_t *chandle;
// Count cards and devices
card = -1;
snd_card_next(&card);
while (card >= 0) {
sprintf(name, "hw:%d", card);
result = snd_ctl_open(&chandle, name, SND_CTL_NONBLOCK);
if (result < 0) {
ATA_WARNING("control open, card = " << card << ", " << snd_strerror(result) << ".");
goto nextcard;
}
subdevice = -1;
while(1) {
result = snd_ctl_pcm_next_device(chandle, &subdevice);
if (result < 0) {
ATA_WARNING("control next device, card = " << card << ", " << snd_strerror(result) << ".");
break;
}
if (subdevice < 0) {
break;
}
if (nDevices == _device) {
// for input
sprintf(name, "hw:%d,%d", card, subdevice);
isInput = true;
goto foundDevice;
}
if (nDevices+1 == _device) {
// for output
sprintf(name, "hw:%d,%d", card, subdevice);
isInput = false;
goto foundDevice;
}
nDevices+=2;
}
nextcard:
snd_ctl_close(chandle);
snd_card_next(&card);
}
if (nDevices == 0) {
ATA_ERROR("no devices found!");
// TODO : audio::orchestra::error_invalidUse;
return info;
}
if (_device >= nDevices) {
ATA_ERROR("device ID is invalid!");
// TODO : audio::orchestra::error_invalidUse;
return info;
}
foundDevice:
snd_ctl_close(chandle);
// If a stream is already open, we cannot probe the stream devices.
// Thus, use the saved results.
if ( m_state != audio::orchestra::state_closed
&& ( m_device[0] == _device
|| m_device[1] == _device)) {
if (_device >= m_devices.size()) {
ATA_ERROR("device ID was not present before stream was opened.");
// TODO : return audio::orchestra::error_warning;
return info;
}
return m_devices[_device];
}
bool ret = audio::orchestra::api::Alsa::getNamedDeviceInfoLocal(name, info, card, subdevice, _device, isInput);
if (ret == false) {
// TODO : ...
return info;
}
return info;
}
void audio::orchestra::api::Alsa::saveDeviceInfo() {
m_devices.clear();
uint32_t nDevices = getDeviceCount();
m_devices.resize(nDevices);
for (uint32_t iii=0; iii<nDevices; ++iii) {
m_devices[iii] = getDeviceInfo(iii);
}
}
bool audio::orchestra::api::Alsa::probeDeviceOpen(uint32_t _device,
audio::orchestra::mode _mode,
uint32_t _channels,
uint32_t _firstChannel,
uint32_t _sampleRate,
audio::format _format,
uint32_t *_bufferSize,
const audio::orchestra::StreamOptions& _options) {
// I'm not using the "plug" interface ... too much inconsistent behavior.
unsigned nDevices = 0;
int32_t result, subdevice, card;
char name[64];
snd_ctl_t *chandle;
// Count cards and devices
card = -1;
// NOTE : Find the device name : [BEGIN]
snd_card_next(&card);
while (card >= 0) {
sprintf(name, "hw:%d", card);
result = snd_ctl_open(&chandle, name, SND_CTL_NONBLOCK);
if (result < 0) {
ATA_ERROR("control open, card = " << card << ", " << snd_strerror(result) << ".");
return false;
}
subdevice = -1;
while(1) {
result = snd_ctl_pcm_next_device(chandle, &subdevice);
if (result < 0) break;
if (subdevice < 0) break;
if (nDevices == _device) {
sprintf(name, "hw:%d,%d", card, subdevice);
snd_ctl_close(chandle);
goto foundDevice;
}
nDevices++;
}
snd_ctl_close(chandle);
snd_card_next(&card);
}
if (nDevices == 0) {
// This should not happen because a check is made before this function is called.
ATA_ERROR("no devices found!");
return false;
}
if (_device >= nDevices) {
// This should not happen because a check is made before this function is called.
ATA_ERROR("device ID is invalid!");
return false;
}
// NOTE : Find the device name : [ END ]
foundDevice:
return probeDeviceOpenName(name, _mode, _channels, _firstChannel, _sampleRate, _format, _bufferSize, _options);
}
bool audio::orchestra::api::Alsa::probeDeviceOpenName(const std::string& _deviceName,
audio::orchestra::mode _mode,
uint32_t _channels,
uint32_t _firstChannel,
uint32_t _sampleRate,
audio::format _format,
uint32_t *_bufferSize,
const audio::orchestra::StreamOptions& _options) {
ATA_DEBUG("Probe ALSA device : ");
ATA_DEBUG(" _deviceName=" << _deviceName);
ATA_DEBUG(" _mode=" << _mode);
ATA_DEBUG(" _channels=" << _channels);
ATA_DEBUG(" _firstChannel=" << _firstChannel);
ATA_DEBUG(" _sampleRate=" << _sampleRate);
ATA_DEBUG(" _format=" << _format);
// I'm not using the "plug" interface ... too much inconsistent behavior.
unsigned nDevices = 0;
int32_t result, subdevice, card;
char name[64];
snd_ctl_t *chandle;
// The getDeviceInfo() function will not work for a device that is
// already open. Thus, we'll probe the system before opening a
// stream and save the results for use by getDeviceInfo().
// TODO : ...
{
// only do once
this->saveDeviceInfo();
}
snd_pcm_stream_t stream;
if (_mode == audio::orchestra::mode_output) {
stream = SND_PCM_STREAM_PLAYBACK;
} else {
stream = SND_PCM_STREAM_CAPTURE;
}
//int32_t openMode = SND_PCM_NONBLOCK;
int32_t openMode = SND_PCM_ASYNC;
//int32_t openMode = SND_PCM_ASYNC | SND_PCM_NONBLOCK;
result = snd_pcm_open(&m_private->handle, _deviceName.c_str(), stream, openMode);
ATA_DEBUG("Configure Mode : SND_PCM_ASYNC");
if (result < 0) {
if (_mode == audio::orchestra::mode_output) {
ATA_ERROR("pcm device (" << _deviceName << ") won't open for output.");
} else {
ATA_ERROR("pcm device (" << _deviceName << ") won't open for input.");
}
return false;
}
// Fill the parameter structure.
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_alloca(&hw_params);
result = snd_pcm_hw_params_any(m_private->handle, hw_params);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error getting pcm device (" << _deviceName << ") parameters, " << snd_strerror(result) << ".");
return false;
}
#if 1
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_MMAP_INTERLEAVED");
result = snd_pcm_hw_params_set_access(m_private->handle, hw_params, SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (result >= 0) {
m_deviceInterleaved[modeToIdTable(_mode)] = true;
m_private->mmapInterface = true;
} else {
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_MMAP_NONINTERLEAVED");
result = snd_pcm_hw_params_set_access(m_private->handle, hw_params, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
if (result >= 0) {
m_deviceInterleaved[modeToIdTable(_mode)] = false;
m_private->mmapInterface = true;
} else {
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_RW_INTERLEAVED");
result = snd_pcm_hw_params_set_access(m_private->handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (result >= 0) {
m_deviceInterleaved[modeToIdTable(_mode)] = true;
m_private->mmapInterface = false;
} else {
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_RW_NONINTERLEAVED");
result = snd_pcm_hw_params_set_access(m_private->handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
if (result >= 0) {
m_deviceInterleaved[modeToIdTable(_mode)] = false;
m_private->mmapInterface = false;
} else {
ATA_ERROR("Can not open the interface ...");
return false;
}
}
}
}
#else
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_RW_INTERLEAVED");
result = snd_pcm_hw_params_set_access(m_private->handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (result >= 0) {
m_deviceInterleaved[modeToIdTable(_mode)] = true;
m_private->mmapInterface = false;
} else {
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_RW_NONINTERLEAVED");
result = snd_pcm_hw_params_set_access(m_private->handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
if (result >= 0) {
m_deviceInterleaved[modeToIdTable(_mode)] = false;
m_private->mmapInterface = false;
} else {
ATA_ERROR("Can not open the interface ...");
return false;
}
}
#endif
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error setting pcm device (" << _deviceName << ") access, " << snd_strerror(result) << ".");
return false;
}
// Determine how to set the device format.
m_userFormat = _format;
snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
if (_format == audio::format_int8) {
deviceFormat = SND_PCM_FORMAT_S8;
} else if (_format == audio::format_int16) {
deviceFormat = SND_PCM_FORMAT_S16;
} else if (_format == audio::format_int24) {
deviceFormat = SND_PCM_FORMAT_S24;
} else if (_format == audio::format_int32) {
deviceFormat = SND_PCM_FORMAT_S32;
} else if (_format == audio::format_float) {
deviceFormat = SND_PCM_FORMAT_FLOAT;
} else if (_format == audio::format_double) {
deviceFormat = SND_PCM_FORMAT_FLOAT64;
}
if (snd_pcm_hw_params_test_format(m_private->handle, hw_params, deviceFormat) == 0) {
m_deviceFormat[modeToIdTable(_mode)] = _format;
} else {
// If we get here, no supported format was found.
snd_pcm_close(m_private->handle);
ATA_ERROR("pcm device " << _deviceName << " data format not supported: " << _format);
// TODO : display list of all supported format ..
return false;
}
ATA_DEBUG("configure format: " << _format);
result = snd_pcm_hw_params_set_format(m_private->handle, hw_params, deviceFormat);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error setting pcm device (" << _deviceName << ") data format, " << snd_strerror(result) << ".");
return false;
}
// Determine whether byte-swaping is necessary.
m_doByteSwap[modeToIdTable(_mode)] = false;
if (deviceFormat != SND_PCM_FORMAT_S8) {
result = snd_pcm_format_cpu_endian(deviceFormat);
if (result == 0) {
ATA_DEBUG("configure swap Byte");
m_doByteSwap[modeToIdTable(_mode)] = true;
} else if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error getting pcm device (" << _deviceName << ") endian-ness, " << snd_strerror(result) << ".");
return false;
}
}
ATA_DEBUG("Set frequency " << _sampleRate);
// Set the sample rate.
result = snd_pcm_hw_params_set_rate_near(m_private->handle, hw_params, (uint32_t*) &_sampleRate, 0);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error setting sample rate on device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
// Determine the number of channels for this device. We support a possible
// minimum device channel number > than the value requested by the user.
m_nUserChannels[modeToIdTable(_mode)] = _channels;
uint32_t value;
result = snd_pcm_hw_params_get_channels_max(hw_params, &value);
uint32_t deviceChannels = value;
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("requested channel parameters not supported by device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
if (deviceChannels < _channels + _firstChannel) {
snd_pcm_close(m_private->handle);
ATA_ERROR("requested channel " << _channels << " have : " << deviceChannels );
return false;
}
result = snd_pcm_hw_params_get_channels_min(hw_params, &value);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error getting minimum channels for device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
deviceChannels = value;
ATA_DEBUG("Device Channel : " << deviceChannels);
if (deviceChannels < _channels + _firstChannel) {
deviceChannels = _channels + _firstChannel;
}
ATA_DEBUG("snd_pcm_hw_params_set_channels: " << deviceChannels);
m_nDeviceChannels[modeToIdTable(_mode)] = deviceChannels;
// Set the device channels.
result = snd_pcm_hw_params_set_channels(m_private->handle, hw_params, deviceChannels);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error setting channels for device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
ATA_DEBUG("configure channels : " << deviceChannels);
// Set the buffer (or period) size.
int32_t dir = 0;
snd_pcm_uframes_t periodSize = *_bufferSize;
result = snd_pcm_hw_params_set_period_size_near(m_private->handle, hw_params, &periodSize, &dir);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error setting period size for device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
*_bufferSize = periodSize;
ATA_DEBUG("configure periode size :" << periodSize);
// Set the buffer number, which in ALSA is referred to as the "period".
uint32_t periods = 0;
if (_options.flags.m_minimizeLatency == true) {
periods = 2;
}
/* TODO : Chouse the number of low level buffer ...
if (_options.numberOfBuffers > 0) {
periods = _options.numberOfBuffers;
}
*/
if (periods < 2) {
periods = 4; // a fairly safe default value
}
result = snd_pcm_hw_params_set_periods_near(m_private->handle, hw_params, &periods, &dir);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error setting periods for device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
ATA_DEBUG("configure Buffer number: " << periods);
m_sampleRate = _sampleRate;
// If attempting to setup a duplex stream, the bufferSize parameter
// MUST be the same in both directions!
if ( m_mode == audio::orchestra::mode_output
&& _mode == audio::orchestra::mode_input
&& *_bufferSize != m_bufferSize) {
snd_pcm_close(m_private->handle);
ATA_ERROR("system error setting buffer size for duplex stream on device (" << _deviceName << ").");
return false;
}
m_bufferSize = *_bufferSize;
ATA_INFO("configure buffer size = " << m_bufferSize*periods << " periode time in ms=" << int64_t(double(m_bufferSize)/double(m_sampleRate)*1000000000.0) << "ns");
// check if the hardware provide hardware clock :
if (snd_pcm_hw_params_is_monotonic(hw_params) == 0) {
ATA_INFO("ALSA Audio timestamp is NOT monotonic (Generate with the start timestamp)");
if (_options.mode == timestampMode_Hardware) {
ATA_WARNING("Can not select Harware timeStamp ==> the IO is not monotonic ==> select ");
m_private->timeMode = timestampMode_trigered;
} else {
m_private->timeMode = _options.mode;
}
} else {
ATA_DEBUG("ALSA Audio timestamp is monotonic (can came from harware)");
m_private->timeMode = _options.mode;
}
// Install the hardware configuration
result = snd_pcm_hw_params(m_private->handle, hw_params);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error installing hardware configuration on device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
snd_pcm_uframes_t val;
// Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
snd_pcm_sw_params_t *swParams = nullptr;
snd_pcm_sw_params_alloca(&swParams);
snd_pcm_sw_params_current(m_private->handle, swParams);
#if 0
ATA_DEBUG("configure start_threshold: " << int64_t(*_bufferSize));
snd_pcm_sw_params_set_start_threshold(m_private->handle, swParams, *_bufferSize);
#else
//ATA_DEBUG("configure start_threshold: " << int64_t(1));
//snd_pcm_sw_params_set_start_threshold(m_private->handle, swParams, 1);
ATA_DEBUG("configure start_threshold: " << int64_t(0));
snd_pcm_sw_params_set_start_threshold(m_private->handle, swParams, 0);
#endif
#if 1
ATA_DEBUG("configure stop_threshold: " << ULONG_MAX);
snd_pcm_sw_params_set_stop_threshold(m_private->handle, swParams, ULONG_MAX);
#else
ATA_DEBUG("configure stop_threshold: " << m_bufferSize*periods);
snd_pcm_sw_params_set_stop_threshold(m_private->handle, swParams, m_bufferSize*periods);
#endif
//ATA_DEBUG("configure silence_threshold: " << 0);
//snd_pcm_sw_params_set_silence_threshold(m_private->handle, swParams, 0);
// The following two settings were suggested by Theo Veenker
#if 1
snd_pcm_sw_params_set_avail_min(m_private->handle, swParams, *_bufferSize*periods/2);
snd_pcm_sw_params_get_avail_min(swParams, &val);
ATA_DEBUG("configure set availlable min: " << *_bufferSize*periods/2 << " really set: " << val);
#endif
#if 0
int valInt;
snd_pcm_sw_params_get_period_event(swParams, &valInt);
ATA_DEBUG("configure get period_event: " << valInt);
snd_pcm_sw_params_set_xfer_align(m_private->handle, swParams, 1);
// here are two options for a fix
//snd_pcm_sw_params_set_silence_size(m_private->handle, swParams, ULONG_MAX);
#endif
//snd_pcm_sw_params_set_tstamp_mode(m_private->handle, swParams, SND_PCM_TSTAMP_ENABLE);
ATA_DEBUG("configuration: ");
//ATA_DEBUG(" start_mode: " << snd_pcm_start_mode_name(snd_pcm_sw_params_get_start_mode(swParams)));
//ATA_DEBUG(" xrun_mode: " << snd_pcm_xrun_mode_name(snd_pcm_sw_params_get_xrun_mode(swParams)));
snd_pcm_tstamp_t valTsMode;
snd_pcm_sw_params_get_tstamp_mode(swParams, &valTsMode);
ATA_DEBUG(" tstamp_mode: " << snd_pcm_tstamp_mode_name(valTsMode));
//ATA_DEBUG(" period_step: " << swParams->period_step);
//ATA_DEBUG(" sleep_min: " << swParams->sleep_min);
snd_pcm_sw_params_get_avail_min(swParams, &val);
ATA_DEBUG(" avail_min: " << val);
//snd_pcm_sw_params_get_xfer_align(swParams, &val);
//ATA_DEBUG(" xfer_align: " << val);
snd_pcm_sw_params_get_silence_threshold(swParams, &val);
ATA_DEBUG(" silence_threshold: " << val);
snd_pcm_sw_params_get_silence_size(swParams, &val);
ATA_DEBUG(" silence_size: " << val);
snd_pcm_sw_params_get_boundary(swParams, &val);
ATA_DEBUG(" boundary: " << val);
result = snd_pcm_sw_params(m_private->handle, swParams);
if (result < 0) {
snd_pcm_close(m_private->handle);
ATA_ERROR("error installing software configuration on device (" << _deviceName << "), " << snd_strerror(result) << ".");
return false;
}
{
snd_pcm_uframes_t _period_size = 0;
snd_pcm_uframes_t _buffer_size = 0;
snd_pcm_hw_params_get_period_size(hw_params, &_period_size, &dir);
snd_pcm_hw_params_get_buffer_size(hw_params, &_buffer_size);
ATA_DEBUG("ploooooo _period_size=" << _period_size << " _buffer_size=" << _buffer_size);
}
// Set flags for buffer conversion
m_doConvertBuffer[modeToIdTable(_mode)] = false;
if (m_userFormat != m_deviceFormat[modeToIdTable(_mode)]) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
if (m_nUserChannels[modeToIdTable(_mode)] < m_nDeviceChannels[modeToIdTable(_mode)]) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
if ( m_deviceInterleaved[modeToIdTable(_mode)] == false
&& m_nUserChannels[modeToIdTable(_mode)] > 1) {
m_doConvertBuffer[modeToIdTable(_mode)] = true;
}
// Allocate necessary internal buffers.
uint64_t bufferBytes;
bufferBytes = m_nUserChannels[modeToIdTable(_mode)] * *_bufferSize * audio::getFormatBytes(m_userFormat);
m_userBuffer[modeToIdTable(_mode)].resize(bufferBytes, 0);
if (m_userBuffer[modeToIdTable(_mode)].size() == 0) {
ATA_ERROR("error allocating user buffer memory.");
goto error;
}
// allocate areas interface:
m_private->areas.resize(m_nUserChannels[modeToIdTable(_mode)]);
for (size_t iii=0; iii<m_private->areas.size(); ++iii) {
m_private->areas[iii].addr = &m_userBuffer[modeToIdTable(_mode)][0];
m_private->areas[iii].first = iii * audio::getFormatBytes(m_userFormat);
m_private->areas[iii].step = m_private->areas.size() * audio::getFormatBytes(m_userFormat);
}
// Generate conbverters:
if (m_doConvertBuffer[modeToIdTable(_mode)]) {
bufferBytes = m_nDeviceChannels[modeToIdTable(_mode)] * audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]);
bufferBytes *= *_bufferSize;
if (m_deviceBuffer) {
free(m_deviceBuffer);
m_deviceBuffer = nullptr;
}
m_deviceBuffer = (char *) calloc(bufferBytes, 1);
if (m_deviceBuffer == nullptr) {
ATA_ERROR("error allocating device buffer memory.");
goto error;
}
}
m_nBuffers = periods;
ATA_INFO("ALSA NB buffer = " << m_nBuffers);
// TODO : m_device[modeToIdTable(_mode)] = _device;
m_state = audio::orchestra::state_stopped;
// Setup the buffer conversion information structure.
if (m_doConvertBuffer[modeToIdTable(_mode)]) {
setConvertInfo(_mode, _firstChannel);
}
m_mode = _mode;
// Setup callback thread.
m_private->threadRunning = true;
ATA_INFO("create thread ...");
m_private->thread = new std11::thread(&audio::orchestra::api::Alsa::alsaCallbackEvent, this);
if (m_private->thread == nullptr) {
m_private->threadRunning = false;
ATA_ERROR("creating callback thread!");
goto error;
}
etk::thread::setPriority(*m_private->thread, -6);
return true;
error:
if (m_private->handle) {
snd_pcm_close(m_private->handle);
m_private->handle = nullptr;
}
for (int32_t iii=0; iii<2; ++iii) {
m_userBuffer[iii].clear();
}
if (m_deviceBuffer) {
free(m_deviceBuffer);
m_deviceBuffer = 0;
}
m_state = audio::orchestra::state_closed;
return false;
}
enum audio::orchestra::error audio::orchestra::api::Alsa::closeStream() {
if (m_state == audio::orchestra::state_closed) {
ATA_ERROR("no open stream to close!");
return audio::orchestra::error_warning;
}
m_private->threadRunning = false;
m_mutex.lock();
if (m_state == audio::orchestra::state_stopped) {
m_private->runnable = true;
m_private->runnable_cv.notify_one();
}
m_mutex.unlock();
if (m_private->thread != nullptr) {
m_private->thread->join();
m_private->thread = nullptr;
}
if (m_state == audio::orchestra::state_running) {
m_state = audio::orchestra::state_stopped;
snd_pcm_drop(m_private->handle);
}
// close all stream :
if (m_private->handle) {
snd_pcm_close(m_private->handle);
m_private->handle = nullptr;
}
for (int32_t iii=0; iii<2; ++iii) {
m_userBuffer[iii].clear();
}
if (m_deviceBuffer) {
free(m_deviceBuffer);
m_deviceBuffer = 0;
}
m_mode = audio::orchestra::mode_unknow;
m_state = audio::orchestra::state_closed;
return audio::orchestra::error_none;
}
enum audio::orchestra::error audio::orchestra::api::Alsa::startStream() {
// TODO : Check return ...
//audio::orchestra::Api::startStream();
// This method calls snd_pcm_prepare if the device isn't already in that state.
if (verifyStream() != audio::orchestra::error_none) {
return audio::orchestra::error_fail;
}
if (m_state == audio::orchestra::state_running) {
ATA_ERROR("the stream is already running!");
return audio::orchestra::error_warning;
}
std11::unique_lock<std11::mutex> lck(m_mutex);
int32_t result = 0;
snd_pcm_state_t state;
if (m_private->handle == nullptr) {
ATA_ERROR("send nullptr to alsa ...");
}
state = snd_pcm_state(m_private->handle);
if (state != SND_PCM_STATE_PREPARED) {
ATA_ERROR("prepare stream");
result = snd_pcm_prepare(m_private->handle);
if (result < 0) {
ATA_ERROR("error preparing pcm device: ERR=" << snd_strerror(result) << ".");
goto unlock;
}
}
m_state = audio::orchestra::state_running;
unlock:
m_private->runnable = true;
m_private->runnable_cv.notify_one();
if (result >= 0) {
return audio::orchestra::error_none;
}
return audio::orchestra::error_systemError;
}
enum audio::orchestra::error audio::orchestra::api::Alsa::stopStream() {
if (verifyStream() != audio::orchestra::error_none) {
return audio::orchestra::error_fail;
}
if (m_state == audio::orchestra::state_stopped) {
ATA_ERROR("the stream is already stopped!");
return audio::orchestra::error_warning;
}
m_state = audio::orchestra::state_stopped;
std11::unique_lock<std11::mutex> lck(m_mutex);
int32_t result = 0;
if (m_mode == audio::orchestra::mode_output) {
result = snd_pcm_drain(m_private->handle);
} else {
result = snd_pcm_drop(m_private->handle);
}
if (result < 0) {
ATA_ERROR("error draining output pcm device, " << snd_strerror(result) << ".");
goto unlock;
}
unlock:
if (result >= 0) {
return audio::orchestra::error_none;
}
return audio::orchestra::error_systemError;
}
enum audio::orchestra::error audio::orchestra::api::Alsa::abortStream() {
if (verifyStream() != audio::orchestra::error_none) {
return audio::orchestra::error_fail;
}
if (m_state == audio::orchestra::state_stopped) {
ATA_ERROR("the stream is already stopped!");
return audio::orchestra::error_warning;
}
m_state = audio::orchestra::state_stopped;
std11::unique_lock<std11::mutex> lck(m_mutex);
int32_t result = 0;
result = snd_pcm_drop(m_private->handle);
if (result < 0) {
ATA_ERROR("error aborting output pcm device, " << snd_strerror(result) << ".");
goto unlock;
}
unlock:
if (result >= 0) {
return audio::orchestra::error_none;
}
return audio::orchestra::error_systemError;
}
void audio::orchestra::api::Alsa::alsaCallbackEvent(void *_userData) {
audio::orchestra::api::Alsa* myClass = reinterpret_cast<audio::orchestra::api::Alsa*>(_userData);
myClass->callbackEvent();
}
/**
* @briefTransfer method - write and wait for room in buffer using poll
*/
static int32_t wait_for_poll(snd_pcm_t* _handle, struct pollfd* _ufds, unsigned int _count) {
uint16_t revents;
while (true) {
poll(_ufds, _count, -1);
snd_pcm_poll_descriptors_revents(_handle, _ufds, _count, &revents);
if (revents & POLLERR) {
return -EIO;
}
if (revents & POLLOUT) {
return 0;
}
if (revents & POLLIN) {
return 0;
}
}
}
void audio::orchestra::api::Alsa::callbackEvent() {
// Lock while the system is not started ...
if (m_state == audio::orchestra::state_stopped) {
std11::unique_lock<std11::mutex> lck(m_mutex);
while (!m_private->runnable) {
m_private->runnable_cv.wait(lck);
}
if (m_state != audio::orchestra::state_running) {
return;
}
}
etk::thread::setName("Alsa IO-" + m_name);
//Wait data with poll
std::vector<struct pollfd> ufds;
signed short *ptr;
int32_t err, count, cptr, init;
count = snd_pcm_poll_descriptors_count(m_private->handle);
if (count <= 0) {
ATA_CRITICAL("Invalid poll descriptors count");
}
ufds.resize(count);
if (ufds.size() == 0) {
ATA_CRITICAL("No enough memory\n");
}
if ((err = snd_pcm_poll_descriptors(m_private->handle, &(ufds[0]), count)) < 0) {
ATA_CRITICAL("Unable to obtain poll descriptors for playback: "<< snd_strerror(err));
}
while (m_private->threadRunning == true) {
// have data or need data ...
if (m_private->mmapInterface == false) {
if (m_mode == audio::orchestra::mode_input) {
callbackEventOneCycleRead();
} else {
callbackEventOneCycleWrite();
}
} else {
if (m_mode == audio::orchestra::mode_input) {
callbackEventOneCycleMMAPRead();
} else {
callbackEventOneCycleMMAPWrite();
}
}
ATA_VERBOSE("Poll [Start] " << count);
err = wait_for_poll(m_private->handle, &(ufds[0]), count);
ATA_VERBOSE("Poll [STOP] " << err);
if (err < 0) {
ATA_ERROR(" POLL timeout ...");
return;
}
}
ATA_DEBUG("End of thread");
}
audio::Time audio::orchestra::api::Alsa::getStreamTime() {
//ATA_DEBUG("mode : " << m_private->timeMode);
if (m_private->timeMode == timestampMode_Hardware) {
snd_pcm_status_t *status = nullptr;
snd_pcm_status_alloca(&status);
// get harware timestamp all the time:
snd_pcm_status(m_private->handle, status);
#if 1
snd_timestamp_t timestamp;
snd_pcm_status_get_tstamp(status, &timestamp);
m_startTime = audio::Time(timestamp.tv_sec, timestamp.tv_usec * 1000);
#else
#if 1
snd_htimestamp_t timestamp;
snd_pcm_status_get_htstamp(status, &timestamp);
m_startTime = audio::Time(timestamp.tv_sec, timestamp.tv_nsec);
#else
snd_htimestamp_t timestamp;
snd_pcm_status_get_audio_htstamp(status, &timestamp);
m_startTime = audio::Time(timestamp.tv_sec, timestamp.tv_nsec);
return m_startTime;
#endif
#endif
ATA_VERBOSE("snd_pcm_status_get_htstamp : " << m_startTime);
snd_pcm_sframes_t delay = snd_pcm_status_get_delay(status);
audio::Duration timeDelay = audio::Duration(0, delay*1000000000LL/int64_t(m_sampleRate));
ATA_VERBOSE("delay : " << timeDelay.count() << " ns");
//return m_startTime + m_duration;
if (m_mode == audio::orchestra::mode_output) {
// output
m_startTime += timeDelay;
} else {
// input
m_startTime -= timeDelay;
}
return m_startTime;
} else if (m_private->timeMode == timestampMode_trigered) {
if (m_startTime == audio::Time()) {
snd_pcm_status_t *status = nullptr;
snd_pcm_status_alloca(&status);
// get harware timestamp all the time:
snd_pcm_status(m_private->handle, status);
// get start time:
snd_timestamp_t timestamp;
snd_pcm_status_get_trigger_tstamp(status, &timestamp);
m_startTime = audio::Time(timestamp.tv_sec, timestamp.tv_usec);
ATA_VERBOSE("snd_pcm_status_get_trigger_tstamp : " << m_startTime);
}
return m_startTime + m_duration;
} else {
// softaware mode ...
if (m_startTime == audio::Time()) {
m_startTime = audio::Time::now();
ATA_ERROR("START TIOMESTAMP : " << m_startTime);
audio::Duration timeDelay = audio::Duration(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
if (m_mode == audio::orchestra::mode_output) {
// output
m_startTime += timeDelay;
} else {
// input
m_startTime -= timeDelay;
}
m_duration = audio::Duration(0);
}
return m_startTime + m_duration;
}
return m_startTime + m_duration;
}
void audio::orchestra::api::Alsa::callbackEventOneCycleRead() {
if (m_state == audio::orchestra::state_closed) {
ATA_CRITICAL("the stream is closed ... this shouldn't happen!");
return; // TODO : notify appl: audio::orchestra::error_warning;
}
int32_t doStopStream = 0;
audio::Time streamTime;
std::vector<enum audio::orchestra::status> status;
if (m_private->xrun[0] == true) {
status.push_back(audio::orchestra::status_underflow);
m_private->xrun[0] = false;
}
int32_t result;
char *buffer;
int32_t channels;
snd_pcm_sframes_t frames;
audio::format format;
if (m_state == audio::orchestra::state_stopped) {
// !!! goto unlock;
}
std11::unique_lock<std11::mutex> lck(m_mutex);
// Setup parameters.
if (m_doConvertBuffer[1]) {
buffer = m_deviceBuffer;
channels = m_nDeviceChannels[1];
format = m_deviceFormat[1];
} else {
buffer = &m_userBuffer[1][0];
channels = m_nUserChannels[1];
format = m_userFormat;
}
// Read samples from device in interleaved/non-interleaved format.
if (m_deviceInterleaved[1]) {
result = snd_pcm_readi(m_private->handle, buffer, m_bufferSize);
} else {
void *bufs[channels];
size_t offset = m_bufferSize * audio::getFormatBytes(format);
for (int32_t i=0; i<channels; i++)
bufs[i] = (void *) (buffer + (i * offset));
result = snd_pcm_readn(m_private->handle, bufs, m_bufferSize);
}
{
snd_pcm_state_t state = snd_pcm_state(m_private->handle);
ATA_VERBOSE("plop : " << state);
if (state == SND_PCM_STATE_XRUN) {
ATA_ERROR("Xrun...");
}
}
// get timestamp : (to init here ...
streamTime = getStreamTime();
if (result < (int) m_bufferSize) {
// Either an error or overrun occured.
if (result == -EPIPE) {
snd_pcm_state_t state = snd_pcm_state(m_private->handle);
if (state == SND_PCM_STATE_XRUN) {
m_private->xrun[1] = true;
result = snd_pcm_prepare(m_private->handle);
if (result < 0) {
ATA_ERROR("error preparing device after overrun, " << snd_strerror(result) << ".");
}
} else {
ATA_ERROR("error, current state is " << snd_pcm_state_name(state) << ", " << snd_strerror(result) << ".");
}
} else {
ATA_ERROR("audio read error, " << snd_strerror(result) << ".");
usleep(10000);
}
// TODO : Notify application ... audio::orchestra::error_warning;
goto noInput;
}
// Do byte swapping if necessary.
if (m_doByteSwap[1]) {
byteSwapBuffer(buffer, m_bufferSize * channels, format);
}
// Do buffer conversion if necessary.
if (m_doConvertBuffer[1]) {
convertBuffer(&m_userBuffer[1][0], m_deviceBuffer, m_convertInfo[1]);
}
// Check stream latency
result = snd_pcm_delay(m_private->handle, &frames);
if (result == 0 && frames > 0) {
ATA_VERBOSE("Delay in the Input " << frames << " chunk");
m_latency[1] = frames;
}
noInput:
streamTime = getStreamTime();
{
audio::Time startCall = audio::Time::now();
doStopStream = m_callback(&m_userBuffer[1][0],
streamTime,// - audio::Duration(m_latency[1]*1000000000LL/int64_t(m_sampleRate)),
nullptr,
audio::Time(),
m_bufferSize,
status);
audio::Time stopCall = audio::Time::now();
audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
audio::Duration timeProcess = stopCall - startCall;
if (timeDelay <= timeProcess) {
ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
}
}
if (doStopStream == 2) {
abortStream();
return;
}
unlock:
audio::orchestra::Api::tickStreamTime();
if (doStopStream == 1) {
this->stopStream();
}
}
void audio::orchestra::api::Alsa::callbackEventOneCycleWrite() {
if (m_state == audio::orchestra::state_closed) {
ATA_CRITICAL("the stream is closed ... this shouldn't happen!");
return; // TODO : notify appl: audio::orchestra::error_warning;
}
int32_t doStopStream = 0;
audio::Time streamTime;
std::vector<enum audio::orchestra::status> status;
if (m_private->xrun[1] == true) {
status.push_back(audio::orchestra::status_overflow);
m_private->xrun[1] = false;
}
int32_t result;
char *buffer;
int32_t channels;
snd_pcm_sframes_t frames;
audio::format format;
if (m_state == audio::orchestra::state_stopped) {
// !!! goto unlock;
}
streamTime = getStreamTime();
{
audio::Time startCall = audio::Time::now();
doStopStream = m_callback(nullptr,
audio::Time(),
&m_userBuffer[0][0],
streamTime,// + audio::Duration(m_latency[0]*1000000000LL/int64_t(m_sampleRate)),
m_bufferSize,
status);
audio::Time stopCall = audio::Time::now();
audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
audio::Duration timeProcess = stopCall - startCall;
if (timeDelay <= timeProcess) {
ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
}
}
if (doStopStream == 2) {
abortStream();
return;
}
std11::unique_lock<std11::mutex> lck(m_mutex);
// Setup parameters and do buffer conversion if necessary.
if (m_doConvertBuffer[0]) {
buffer = m_deviceBuffer;
convertBuffer(buffer, &m_userBuffer[0][0], m_convertInfo[0]);
channels = m_nDeviceChannels[0];
format = m_deviceFormat[0];
} else {
buffer = &m_userBuffer[0][0];
channels = m_nUserChannels[0];
format = m_userFormat;
}
// Do byte swapping if necessary.
if (m_doByteSwap[0]) {
byteSwapBuffer(buffer, m_bufferSize * channels, format);
}
// Write samples to device in interleaved/non-interleaved format.
if (m_deviceInterleaved[0]) {
result = snd_pcm_writei(m_private->handle, buffer, m_bufferSize);
} else {
void *bufs[channels];
size_t offset = m_bufferSize * audio::getFormatBytes(format);
for (int32_t i=0; i<channels; i++) {
bufs[i] = (void *) (buffer + (i * offset));
}
result = snd_pcm_writen(m_private->handle, bufs, m_bufferSize);
}
if (result < (int) m_bufferSize) {
// Either an error or underrun occured.
if (result == -EPIPE) {
snd_pcm_state_t state = snd_pcm_state(m_private->handle);
if (state == SND_PCM_STATE_XRUN) {
m_private->xrun[0] = true;
result = snd_pcm_prepare(m_private->handle);
if (result < 0) {
ATA_ERROR("error preparing device after underrun, " << snd_strerror(result) << ".");
}
} else {
ATA_ERROR("error, current state is " << snd_pcm_state_name(state) << ", " << snd_strerror(result) << ".");
}
} else {
ATA_ERROR("audio write error, " << snd_strerror(result) << ".");
}
// TODO : Notuify application audio::orchestra::error_warning;
goto unlock;
}
// Check stream latency
result = snd_pcm_delay(m_private->handle, &frames);
if (result == 0 && frames > 0) {
ATA_VERBOSE("Delay in the Output " << frames << " chunk");
m_latency[0] = frames;
}
unlock:
audio::orchestra::Api::tickStreamTime();
if (doStopStream == 1) {
this->stopStream();
}
}
void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPWrite() {
if (m_state == audio::orchestra::state_closed) {
ATA_CRITICAL("the stream is closed ... this shouldn't happen!");
return; // TODO : notify appl: audio::orchestra::error_warning;
}
int32_t doStopStream = 0;
audio::Time streamTime;
std::vector<enum audio::orchestra::status> status;
if (m_private->xrun[1] == true) {
status.push_back(audio::orchestra::status_overflow);
m_private->xrun[1] = false;
}
int32_t result;
char *buffer;
int32_t channels;
snd_pcm_sframes_t frames;
audio::format format;
if (m_state == audio::orchestra::state_stopped) {
// !!! goto unlock;
}
ATA_DEBUG("UPDATE");
int32_t avail = snd_pcm_avail_update(m_private->handle);
if (avail < 0) {
ATA_ERROR("Can not get buffer data ..." << avail);
return;
}
streamTime = getStreamTime();
{
audio::Time startCall = audio::Time::now();
doStopStream = m_callback(nullptr,
audio::Time(),
&m_userBuffer[0][0],
streamTime,// + audio::Duration(m_latency[0]*1000000000LL/int64_t(m_sampleRate)),
m_bufferSize,
status);
audio::Time stopCall = audio::Time::now();
audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
audio::Duration timeProcess = stopCall - startCall;
if (timeDelay <= timeProcess) {
ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
}
}
if (doStopStream == 2) {
abortStream();
return;
}
{
std11::unique_lock<std11::mutex> lck(m_mutex);
// Setup parameters and do buffer conversion if necessary.
if (m_doConvertBuffer[0]) {
buffer = m_deviceBuffer;
convertBuffer(buffer, &m_userBuffer[0][0], m_convertInfo[0]);
channels = m_nDeviceChannels[0];
format = m_deviceFormat[0];
} else {
buffer = &m_userBuffer[0][0];
channels = m_nUserChannels[0];
format = m_userFormat;
}
// Do byte swapping if necessary.
if (m_doByteSwap[0]) {
byteSwapBuffer(buffer, m_bufferSize * channels, format);
}
#if 1
// Write samples to device in interleaved/non-interleaved format.
if (m_deviceInterleaved[0]) {
result = snd_pcm_mmap_writei(m_private->handle, buffer, m_bufferSize);
} else {
void *bufs[channels];
size_t offset = m_bufferSize * audio::getFormatBytes(format);
for (int32_t i=0; i<channels; i++) {
bufs[i] = (void *) (buffer + (i * offset));
}
result = snd_pcm_mmap_writen(m_private->handle, bufs, m_bufferSize);
}
#else
// TODO: Understand why this does not work ...
// Write samples to device in interleaved/non-interleaved format.
if (m_deviceInterleaved[0]) {
const snd_pcm_channel_area_t* myAreas = nullptr;
snd_pcm_uframes_t offset, frames;
frames = m_bufferSize;
ATA_DEBUG("START");
int err = snd_pcm_mmap_begin(m_private->handle, &myAreas, &offset, &frames);
if (err < 0) {
ATA_CRITICAL("SUPER_FAIL");
}
ATA_DEBUG("snd_pcm_mmap_begin " << offset << " frame=" << frames << " m_bufferSize=" << m_bufferSize);
ATA_DEBUG("copy " << err << " addr=" << myAreas[0].addr << " first=" << myAreas[0].first << " step=" << myAreas[0].step);
//generate_sine(myAreas, offset, frames, &phase);
// Write samples to device in interleaved/non-interleaved format.
if (m_deviceInterleaved[0]) {
memcpy(myAreas[0].addr, buffer, m_bufferSize);
result = m_bufferSize;
} else {
void *bufs[channels];
size_t offset = m_bufferSize * audio::getFormatBytes(format);
for (int32_t i=0; i<channels; i++) {
bufs[i] = (void *) (buffer + (i * offset));
}
memcpy(myAreas[0].addr, bufs, m_bufferSize);
result = m_bufferSize;
}
ATA_DEBUG("commit " << offset << " frame=" << frames);
int commitres = snd_pcm_mmap_commit(m_private->handle, offset, frames);
if ( commitres < 0
|| (snd_pcm_uframes_t)commitres != frames) {
ATA_CRITICAL("MMAP commit error: " << snd_strerror(err));
}
} else {
void *bufs[channels];
size_t offset = m_bufferSize * audio::getFormatBytes(format);
for (int32_t i=0; i<channels; i++) {
bufs[i] = (void *) (buffer + (i * offset));
}
result = snd_pcm_writen(m_private->handle, bufs, m_bufferSize);
}
#endif
// Check stream latency
result = snd_pcm_delay(m_private->handle, &frames);
if (result == 0 && frames > 0) {
ATA_VERBOSE("Delay in the Output " << frames << " chunk");
m_latency[0] = frames;
}
}
unlock:
audio::orchestra::Api::tickStreamTime();
if (doStopStream == 1) {
this->stopStream();
}
}
void audio::orchestra::api::Alsa::callbackEventOneCycleMMAPRead() {
if (m_state == audio::orchestra::state_closed) {
ATA_CRITICAL("the stream is closed ... this shouldn't happen!");
return; // TODO : notify appl: audio::orchestra::error_warning;
}
int32_t doStopStream = 0;
audio::Time streamTime;
std::vector<enum audio::orchestra::status> status;
if (m_private->xrun[0] == true) {
status.push_back(audio::orchestra::status_underflow);
m_private->xrun[0] = false;
}
int32_t result;
char *buffer;
int32_t channels;
snd_pcm_sframes_t frames;
audio::format format;
if (m_state == audio::orchestra::state_stopped) {
goto unlock;
}
{
std11::unique_lock<std11::mutex> lck(m_mutex);
// Setup parameters.
if (m_doConvertBuffer[1]) {
buffer = m_deviceBuffer;
channels = m_nDeviceChannels[1];
format = m_deviceFormat[1];
} else {
buffer = &m_userBuffer[1][0];
channels = m_nUserChannels[1];
format = m_userFormat;
}
// Read samples from device in interleaved/non-interleaved format.
if (m_deviceInterleaved[1]) {
result = snd_pcm_mmap_readi(m_private->handle, buffer, m_bufferSize);
} else {
void *bufs[channels];
size_t offset = m_bufferSize * audio::getFormatBytes(format);
for (int32_t i=0; i<channels; i++)
bufs[i] = (void *) (buffer + (i * offset));
result = snd_pcm_mmap_readn(m_private->handle, bufs, m_bufferSize);
}
{
snd_pcm_state_t state = snd_pcm_state(m_private->handle);
ATA_VERBOSE("plop: " << state);
if (state == SND_PCM_STATE_XRUN) {
ATA_ERROR("Xrun...");
}
}
// get timestamp : (to init here ...
streamTime = getStreamTime();
if (result < (int) m_bufferSize) {
// Either an error or overrun occured.
if (result == -EPIPE) {
snd_pcm_state_t state = snd_pcm_state(m_private->handle);
if (state == SND_PCM_STATE_XRUN) {
m_private->xrun[1] = true;
result = snd_pcm_prepare(m_private->handle);
if (result < 0) {
ATA_ERROR("error preparing device after overrun, " << snd_strerror(result) << ".");
}
} else {
ATA_ERROR("error, current state is " << snd_pcm_state_name(state) << ", " << snd_strerror(result) << ".");
}
} else {
ATA_ERROR("audio read error, " << snd_strerror(result) << ".");
usleep(10000);
}
// TODO : Notify application ... audio::orchestra::error_warning;
goto noInput;
}
// Do byte swapping if necessary.
if (m_doByteSwap[1]) {
byteSwapBuffer(buffer, m_bufferSize * channels, format);
}
// Do buffer conversion if necessary.
if (m_doConvertBuffer[1]) {
convertBuffer(&m_userBuffer[1][0], m_deviceBuffer, m_convertInfo[1]);
}
// Check stream latency
result = snd_pcm_delay(m_private->handle, &frames);
if (result == 0 && frames > 0) {
ATA_VERBOSE("Delay in the Input " << frames << " chunk");
m_latency[1] = frames;
}
}
noInput:
streamTime = getStreamTime();
{
audio::Time startCall = audio::Time::now();
doStopStream = m_callback(&m_userBuffer[1][0],
streamTime,// - audio::Duration(m_latency[1]*1000000000LL/int64_t(m_sampleRate)),
nullptr,
audio::Time(),
m_bufferSize,
status);
audio::Time stopCall = audio::Time::now();
audio::Duration timeDelay(0, m_bufferSize*1000000000LL/int64_t(m_sampleRate));
audio::Duration timeProcess = stopCall - startCall;
if (timeDelay <= timeProcess) {
ATA_ERROR("SOFT XRUN ... : (bufferTime) " << timeDelay.count() << " < " << timeProcess.count() << " (process time) ns");
}
}
if (doStopStream == 2) {
abortStream();
return;
}
unlock:
audio::orchestra::Api::tickStreamTime();
if (doStopStream == 1) {
this->stopStream();
}
}
bool audio::orchestra::api::Alsa::isMasterOf(audio::orchestra::Api* _api) {
audio::orchestra::api::Alsa* slave = dynamic_cast<audio::orchestra::api::Alsa*>(_api);
if (slave == nullptr) {
ATA_ERROR("NULL ptr API (not ALSA ...)");
return false;
}
if (m_state == audio::orchestra::state_running) {
ATA_ERROR("The MASTER stream is already running! ==> can not synchronize ...");
return false;
}
if (slave->m_state == audio::orchestra::state_running) {
ATA_ERROR("The SLAVE stream is already running! ==> can not synchronize ...");
return false;
}
snd_pcm_t * master = nullptr;
if (m_private->handle != nullptr) {
master = m_private->handle;
}
if (master == nullptr) {
ATA_ERROR("No ALSA handles ...");
return false;
}
ATA_INFO(" ==> plop");
if (snd_pcm_link(master, slave->m_private->handle) != 0) {
ATA_ERROR("Can not syncronize handle output");
} else {
ATA_INFO(" -------------------- LINK 0 --------------------");
}
return true;
}
#endif
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