[DEV] start rework Alsa API to support poll event and MMAP system

This commit is contained in:
Edouard DUPIN 2015-06-05 22:00:17 +02:00
parent 17d59cf370
commit 9dec54d4c7
3 changed files with 495 additions and 22 deletions

View File

@ -19,7 +19,6 @@
namespace audio {
namespace orchestra {
const std::vector<uint32_t>& genericSampleRate();
/**
* @brief airtaudio callback function prototype.
* @param _inputBuffer For input (or duplex) streams, this buffer will hold _nbChunk of input audio chunk (nullptr if no data).

View File

@ -16,7 +16,17 @@
#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"
@ -36,12 +46,15 @@ namespace audio {
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() :
synchronized(false),
runnable(false),
thread(nullptr),
threadRunning(false),
mmapInterface(false),
timeMode(timestampMode_soft) {
handles[0] = nullptr;
handles[1] = nullptr;
@ -55,7 +68,7 @@ namespace audio {
}
audio::orchestra::api::Alsa::Alsa() :
m_private(new audio::orchestra::api::AlsaPrivate()) {
m_private(std::make_shared<audio::orchestra::api::AlsaPrivate>()) {
// Nothing to do here.
}
@ -464,13 +477,13 @@ foundDevice:
}
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) {
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);
@ -522,15 +535,35 @@ bool audio::orchestra::api::Alsa::probeDeviceOpenName(const std::string& _device
ATA_ERROR("error getting pcm device (" << _deviceName << ") parameters, " << snd_strerror(result) << ".");
return false;
}
// Open stream all time in interleave mode (by default): (open in non interleave if we have no choice
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_RW_INTERLEAVED");
result = snd_pcm_hw_params_set_access(phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (result < 0) {
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_RW_NONINTERLEAVED");
result = snd_pcm_hw_params_set_access(phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
m_deviceInterleaved[modeToIdTable(_mode)] = false;
} else {
ATA_DEBUG("configure Acces: SND_PCM_ACCESS_MMAP_INTERLEAVED");
result = snd_pcm_hw_params_set_access(phandle, 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(phandle, 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(phandle, 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(phandle, 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;
}
}
}
}
if (result < 0) {
snd_pcm_close(phandle);
@ -787,6 +820,14 @@ bool audio::orchestra::api::Alsa::probeDeviceOpenName(const std::string& _device
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)]) {
bool makeBuffer = true;
bufferBytes = m_nDeviceChannels[modeToIdTable(_mode)] * audio::getFormatBytes(m_deviceFormat[modeToIdTable(_mode)]);
@ -1058,10 +1099,74 @@ void audio::orchestra::api::Alsa::alsaCallbackEvent(void *_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;
}
}
}
void audio::orchestra::api::Alsa::callbackEvent() {
etk::thread::setName("Alsa IO-" + m_name);
//Wait data with poll
/*
snd_pcm_channel_area_t *areas = nullptr;
areas = (snd_pcm_channel_area_t *)calloc(chennels, sizeof(snd_pcm_channel_area_t));
if (areas == nullptr) {
ATA_CRITICAL("No enough memory");
}
for (chn = 0; chn < channels; chn++) {
areas[chn].addr = samples;
areas[chn].first = chn * snd_pcm_format_physical_width(format);
areas[chn].step = channels * snd_pcm_format_physical_width(format);
}
*/
struct pollfd *ufds;
signed short *ptr;
int32_t err, count, cptr, init;
count = snd_pcm_poll_descriptors_count(m_private->handles[0]);
if (count <= 0) {
ATA_CRITICAL("Invalid poll descriptors count");
}
ufds = (struct pollfd*)malloc(sizeof(struct pollfd) * count);
if (ufds == nullptr) {
ATA_CRITICAL("No enough memory\n");
}
if ((err = snd_pcm_poll_descriptors(m_private->handles[0], ufds, count)) < 0) {
ATA_CRITICAL("Unable to obtain poll descriptors for playback: "<< snd_strerror(err));
}
init = 1;
while (m_private->threadRunning == true) {
callbackEventOneCycle();
err = wait_for_poll(m_private->handles[0], ufds, count);
ATA_INFO("plop " << err);
if (err < 0) {
ATA_ERROR(" POLL timeout ...");
return;
}
// 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();
}
}
}
}
@ -1148,7 +1253,371 @@ audio::Time audio::orchestra::api::Alsa::getStreamTime() {
return m_startTime + m_duration;
}
void audio::orchestra::api::Alsa::callbackEventOneCycle() {
void audio::orchestra::api::Alsa::callbackEventOneCycleRead() {
if (m_state == audio::orchestra::state_stopped) {
std11::unique_lock<std11::mutex> lck(m_mutex);
// TODO : Set this back ....
/*
while (!m_private->runnable) {
m_private->runnable_cv.wait(lck);
}
*/
usleep(1000);
if (m_state != audio::orchestra::state_running) {
return;
}
}
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->handles[1], 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->handles[1], bufs, m_bufferSize);
}
{
snd_pcm_state_t state = snd_pcm_state(m_private->handles[1]);
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->handles[1]);
if (state == SND_PCM_STATE_XRUN) {
m_private->xrun[1] = true;
result = snd_pcm_prepare(m_private->handles[1]);
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->handles[1], &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_stopped) {
std11::unique_lock<std11::mutex> lck(m_mutex);
// TODO : Set this back ....
/*
while (!m_private->runnable) {
m_private->runnable_cv.wait(lck);
}
*/
usleep(1000);
if (m_state != audio::orchestra::state_running) {
return;
}
}
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->handles[0], 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->handles[0], 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->handles[0]);
if (state == SND_PCM_STATE_XRUN) {
m_private->xrun[0] = true;
result = snd_pcm_prepare(m_private->handles[0]);
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->handles[0], &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_stopped) {
std11::unique_lock<std11::mutex> lck(m_mutex);
// TODO : Set this back ....
/*
while (!m_private->runnable) {
m_private->runnable_cv.wait(lck);
}
*/
usleep(1000);
if (m_state != audio::orchestra::state_running) {
return;
}
}
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->handles[0]);
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);
}
// 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->handles[0], &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);
memcpy(myAreas[0].addr + offset, buffer, m_bufferSize);
ATA_DEBUG("commit " << offset << " frame=" << frames);
int commitres = snd_pcm_mmap_commit(m_private->handles[0], 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->handles[0], bufs, m_bufferSize);
}
// Check stream latency
result = snd_pcm_delay(m_private->handles[0], &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_stopped) {
std11::unique_lock<std11::mutex> lck(m_mutex);
// TODO : Set this back ....
@ -1214,7 +1683,7 @@ void audio::orchestra::api::Alsa::callbackEventOneCycle() {
}
{
snd_pcm_state_t state = snd_pcm_state(m_private->handles[1]);
ATA_VERBOSE("plop : " << state);
ATA_VERBOSE("plop: " << state);
if (state == SND_PCM_STATE_XRUN) {
ATA_ERROR("Xrun...");
}
@ -1342,6 +1811,7 @@ unlock:
}
}
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) {

View File

@ -42,9 +42,13 @@ namespace audio {
// which is not a member of RtAudio. External use of this function
// will most likely produce highly undesireable results!
void callbackEvent();
void callbackEventOneCycle();
void callbackEventOneCycleRead();
void callbackEventOneCycleWrite();
void callbackEventOneCycleMMAPRead();
void callbackEventOneCycleMMAPWrite();
private:
static void alsaCallbackEvent(void* _userData);
static void alsaCallbackEventMMap(void* _userData);
private:
std11::shared_ptr<AlsaPrivate> m_private;
std::vector<audio::orchestra::DeviceInfo> m_devices;