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arts/flow/audioioalsa9.cpp

596 lines
16 KiB

/*
Copyright (C) 2001 Takashi Iwai <tiwai@suse.de>
Copyright (C) 2004 Allan Sandfeld Jensen <kde@carewolf.com>
based on audioalsa.cpp:
Copyright (C) 2000,2001 Jozef Kosoru
jozef.kosoru@pobox.sk
(C) 2000,2001 Stefan Westerfeld
stefan@space.twc.de
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; see the file COPYING.LIB. If not, write to
the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
/**
* only compile 'alsa' AudioIO class if configure thinks it is a good idea
*/
#ifdef HAVE_LIBASOUND2
#ifdef HAVE_ALSA_ASOUNDLIB_H
#include <alsa/asoundlib.h>
#elif defined(HAVE_SYS_ASOUNDLIB_H)
#include <sys/asoundlib.h>
#endif
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <iostream>
#include <algorithm>
#include "debug.h"
#include "audioio.h"
#include "audiosubsys.h"
#include "dispatcher.h"
#include "iomanager.h"
namespace Arts {
class AudioIOALSA : public AudioIO, public IONotify {
protected:
// List of file descriptors
struct poll_descriptors {
poll_descriptors() : nfds(0), pfds(0) {};
int nfds;
struct pollfd *pfds;
} audio_write_pds, audio_read_pds;
snd_pcm_t *m_pcm_playback;
snd_pcm_t *m_pcm_capture;
snd_pcm_format_t m_format;
snd_pcm_uframes_t m_period_size;
unsigned m_periods;
void startIO();
int setPcmParams(snd_pcm_t *pcm);
static int poll2iomanager(int pollTypes);
static int iomanager2poll(int ioTypes);
void getDescriptors(snd_pcm_t *pcm, poll_descriptors *pds);
void watchDescriptors(poll_descriptors *pds);
void notifyIO(int fd, int types);
int xrun(snd_pcm_t *pcm);
#ifdef HAVE_SND_PCM_RESUME
int resume(snd_pcm_t *pcm);
#endif
public:
AudioIOALSA();
void setParam(AudioParam param, int& value);
int getParam(AudioParam param);
bool open();
void close();
int read(void *buffer, int size);
int write(void *buffer, int size);
};
REGISTER_AUDIO_IO(AudioIOALSA,"alsa","Advanced Linux Sound Architecture");
}
using namespace std;
using namespace Arts;
AudioIOALSA::AudioIOALSA()
{
param(samplingRate) = 44100;
paramStr(deviceName) = "default"; // ALSA pcm device name - not file name
param(fragmentSize) = 1024;
param(fragmentCount) = 7;
param(channels) = 2;
param(direction) = directionWrite;
param(format) = 16;
/*
* default parameters
*/
m_format = SND_PCM_FORMAT_S16_LE;
m_pcm_playback = NULL;
m_pcm_capture = NULL;
}
bool AudioIOALSA::open()
{
string& _error = paramStr(lastError);
string& _deviceName = paramStr(deviceName);
int& _channels = param(channels);
int& _fragmentSize = param(fragmentSize);
int& _fragmentCount = param(fragmentCount);
int& _samplingRate = param(samplingRate);
int& _direction = param(direction);
int& _format = param(format);
m_pcm_playback = NULL;
m_pcm_capture = NULL;
/* initialize format */
switch(_format) {
case 16: // 16bit, signed little endian
m_format = SND_PCM_FORMAT_S16_LE;
break;
case 17: // 16bit, signed big endian
m_format = SND_PCM_FORMAT_S16_BE;
break;
case 8: // 8bit, unsigned
m_format = SND_PCM_FORMAT_U8;
break;
default: // test later
m_format = SND_PCM_FORMAT_UNKNOWN;
break;
}
/* open pcm device */
int err;
if (_direction & directionWrite) {
if ((err = snd_pcm_open(&m_pcm_playback, _deviceName.c_str(),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) {
_error = "device: ";
_error += _deviceName.c_str();
_error += " can't be opened for playback (";
_error += snd_strerror(err);
_error += ")";
return false;
}
snd_pcm_nonblock(m_pcm_playback, 0);
}
if (_direction & directionRead) {
if ((err = snd_pcm_open(&m_pcm_capture, _deviceName.c_str(),
SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK)) < 0) {
_error = "device: ";
_error += _deviceName.c_str();
_error += " can't be opened for capture (";
_error += snd_strerror(err);
_error += ")";
snd_pcm_close(m_pcm_playback);
return false;
}
snd_pcm_nonblock(m_pcm_capture, 0);
}
artsdebug("ALSA driver: %s", _deviceName.c_str());
/* check device capabilities */
// checkCapabilities();
/* set PCM communication parameters */
if (((_direction & directionWrite) && setPcmParams(m_pcm_playback)) ||
((_direction & directionRead) && setPcmParams(m_pcm_capture))) {
snd_pcm_close(m_pcm_playback);
snd_pcm_close(m_pcm_capture);
return false;
}
artsdebug("buffering: %d fragments with %d bytes "
"(audio latency is %1.1f ms)", _fragmentCount, _fragmentSize,
(float)(_fragmentSize*_fragmentCount) /
(float)(2.0 * _samplingRate * _channels)*1000.0);
startIO();
/* restore the format value */
switch (m_format) {
case SND_PCM_FORMAT_S16_LE:
_format = 16;
break;
case SND_PCM_FORMAT_S16_BE:
_format = 17;
break;
case SND_PCM_FORMAT_U8:
_format = 8;
break;
default:
_error = "Unknown PCM format";
return false;
}
/* start recording */
if (_direction & directionRead)
snd_pcm_start(m_pcm_capture);
return true;
}
void AudioIOALSA::close()
{
arts_debug("Closing ALSA-driver");
int& _direction = param(direction);
if ((_direction & directionRead) && m_pcm_capture) {
(void)snd_pcm_drop(m_pcm_capture);
(void)snd_pcm_close(m_pcm_capture);
m_pcm_capture = NULL;
}
if ((_direction & directionWrite) && m_pcm_playback) {
(void)snd_pcm_drop(m_pcm_playback);
(void)snd_pcm_close(m_pcm_playback);
m_pcm_playback = NULL;
}
Dispatcher::the()->ioManager()->remove(this, IOType::all);
delete[] audio_read_pds.pfds;
delete[] audio_write_pds.pfds;
audio_read_pds.pfds = NULL; audio_write_pds.pfds = NULL;
audio_read_pds.nfds = 0; audio_write_pds.nfds = 0;
}
void AudioIOALSA::setParam(AudioParam p, int& value)
{
param(p) = value;
if (m_pcm_playback != NULL) {
setPcmParams(m_pcm_playback);
}
if (m_pcm_capture != NULL) {
setPcmParams(m_pcm_capture);
}
}
int AudioIOALSA::getParam(AudioParam p)
{
snd_pcm_sframes_t avail;
switch(p) {
case canRead:
if (! m_pcm_capture) return -1;
while ((avail = snd_pcm_avail_update(m_pcm_capture)) < 0) {
if (avail == -EPIPE)
avail = xrun(m_pcm_capture);
#ifdef HAVE_SND_PCM_RESUME
else if (avail == -ESTRPIPE)
avail = resume(m_pcm_capture);
#endif
if (avail < 0) {
arts_info("Capture error: %s", snd_strerror(avail));
return -1;
}
}
return snd_pcm_frames_to_bytes(m_pcm_capture, avail);
case canWrite:
if (! m_pcm_playback) return -1;
while ((avail = snd_pcm_avail_update(m_pcm_playback)) < 0) {
if (avail == -EPIPE)
avail = xrun(m_pcm_playback);
#ifdef HAVE_SND_PCM_RESUME
else if (avail == -ESTRPIPE)
avail = resume(m_pcm_playback);
#endif
if (avail < 0) {
arts_info("Playback error: %s", snd_strerror(avail));
return -1;
}
}
return snd_pcm_frames_to_bytes(m_pcm_playback, avail);
case selectReadFD:
return -1;
case selectWriteFD:
return -1;
case autoDetect:
{
/*
* that the ALSA driver could be compiled doesn't say anything
* about whether it will work (the user might be using an OSS
* kernel driver).
* If we can open the device, it'll work - and we'll have to use
* a higher number than OSS to avoid buggy OSS emulation being used.
*/
int card = -1;
if (snd_card_next(&card) < 0 || card < 0) {
// No ALSA drivers in use...
return 0;
}
return 15;
}
default:
return param(p);
}
}
void AudioIOALSA::startIO()
{
/* get & watch PCM file descriptor(s) */
if (m_pcm_playback) {
getDescriptors(m_pcm_playback, &audio_write_pds);
watchDescriptors(&audio_write_pds);
}
if (m_pcm_capture) {
getDescriptors(m_pcm_capture, &audio_read_pds);
watchDescriptors(&audio_read_pds);
}
}
int AudioIOALSA::poll2iomanager(int pollTypes)
{
int types = 0;
if(pollTypes & POLLIN)
types |= IOType::read;
if(pollTypes & POLLOUT)
types |= IOType::write;
if(pollTypes & POLLERR)
types |= IOType::except;
return types;
}
int AudioIOALSA::iomanager2poll(int ioTypes)
{
int types = 0;
if(ioTypes & IOType::read)
types |= POLLIN;
if(ioTypes & IOType::write)
types |= POLLOUT;
if(ioTypes & IOType::except)
types |= POLLERR;
return types;
}
void AudioIOALSA::getDescriptors(snd_pcm_t *pcm, poll_descriptors *pds)
{
pds->nfds = snd_pcm_poll_descriptors_count(pcm);
pds->pfds = new struct pollfd[pds->nfds];
if (snd_pcm_poll_descriptors(pcm, pds->pfds, pds->nfds) != pds->nfds) {
arts_info("Cannot get poll descriptor(s)\n");
}
}
void AudioIOALSA::watchDescriptors(poll_descriptors *pds)
{
for(int i=0; i<pds->nfds; i++) {
// Check in which direction this handle is supposed to be watched
int types = poll2iomanager(pds->pfds[i].events);
Dispatcher::the()->ioManager()->watchFD(pds->pfds[i].fd, types, this);
}
}
int AudioIOALSA::xrun(snd_pcm_t *pcm)
{
int err;
artsdebug("xrun!!\n");
if ((err = snd_pcm_prepare(pcm)) < 0)
return err;
if (pcm == m_pcm_capture)
snd_pcm_start(pcm); // ignore error here..
return 0;
}
#ifdef HAVE_SND_PCM_RESUME
int AudioIOALSA::resume(snd_pcm_t *pcm)
{
int err;
artsdebug("resume!\n");
while ((err = snd_pcm_resume(pcm)) == -EAGAIN)
sleep(1); /* wait until suspend flag is not released */
if (err < 0) {
if ((err = snd_pcm_prepare(pcm)) < 0)
return err;
if (pcm == m_pcm_capture)
snd_pcm_start(pcm); // ignore error here..
}
return 0;
}
#endif
int AudioIOALSA::read(void *buffer, int size)
{
int frames = snd_pcm_bytes_to_frames(m_pcm_capture, size);
int length;
while ((length = snd_pcm_readi(m_pcm_capture, buffer, frames)) < 0) {
if (length == -EINTR)
continue; // Try again
else if (length == -EPIPE)
length = xrun(m_pcm_capture);
#ifdef HAVE_SND_PCM_RESUME
else if (length == -ESTRPIPE)
length = resume(m_pcm_capture);
#endif
if (length < 0) {
arts_info("Capture error: %s", snd_strerror(length));
return -1;
}
}
return snd_pcm_frames_to_bytes(m_pcm_capture, length);
}
int AudioIOALSA::write(void *buffer, int size)
{
int frames = snd_pcm_bytes_to_frames(m_pcm_playback, size);
int length;
while ((length = snd_pcm_writei(m_pcm_playback, buffer, frames)) < 0) {
if (length == -EINTR)
continue; // Try again
else if (length == -EPIPE)
length = xrun(m_pcm_playback);
#ifdef HAVE_SND_PCM_RESUME
else if (length == -ESTRPIPE)
length = resume(m_pcm_playback);
#endif
if (length < 0) {
arts_info("Playback error: %s", snd_strerror(length));
return -1;
}
}
// Start the sink if it needs it
if (snd_pcm_state( m_pcm_playback ) == SND_PCM_STATE_PREPARED)
snd_pcm_start(m_pcm_playback);
if (length == frames) // Sometimes the fragments are "odd" in alsa
return size;
else
return snd_pcm_frames_to_bytes(m_pcm_playback, length);
}
void AudioIOALSA::notifyIO(int fd, int type)
{
int todo = 0;
// Translate from iomanager-types to poll-types,
// inorder to fake a snd_pcm_poll_descriptors_revents call.
if(m_pcm_playback) {
for(int i=0; i < audio_write_pds.nfds; i++) {
if(fd == audio_write_pds.pfds[i].fd) {
audio_write_pds.pfds[i].revents = iomanager2poll(type);
todo |= AudioSubSystem::ioWrite;
}
}
if (todo & AudioSubSystem::ioWrite) {
unsigned short revents;
snd_pcm_poll_descriptors_revents(m_pcm_playback,
audio_write_pds.pfds,
audio_write_pds.nfds,
&revents);
if (! (revents & POLLOUT)) todo &= ~AudioSubSystem::ioWrite;
}
}
if(m_pcm_capture) {
for(int i=0; i < audio_read_pds.nfds; i++) {
if(fd == audio_read_pds.pfds[i].fd) {
audio_read_pds.pfds[i].revents = iomanager2poll(type);
todo |= AudioSubSystem::ioRead;
}
}
if (todo & AudioSubSystem::ioRead) {
unsigned short revents;
snd_pcm_poll_descriptors_revents(m_pcm_capture,
audio_read_pds.pfds,
audio_read_pds.nfds,
&revents);
if (! (revents & POLLIN)) todo &= ~AudioSubSystem::ioRead;
}
}
if (type & IOType::except) todo |= AudioSubSystem::ioExcept;
if (todo != 0) AudioSubSystem::the()->handleIO(todo);
}
int AudioIOALSA::setPcmParams(snd_pcm_t *pcm)
{
int &_samplingRate = param(samplingRate);
int &_channels = param(channels);
int &_fragmentSize = param(fragmentSize);
int &_fragmentCount = param(fragmentCount);
string& _error = paramStr(lastError);
snd_pcm_hw_params_t *hw;
snd_pcm_hw_params_alloca(&hw);
snd_pcm_hw_params_any(pcm, hw);
if (snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
_error = "Unable to set interleaved!";
return 1;
}
if (m_format == SND_PCM_FORMAT_UNKNOWN) {
// test the available formats
// try 16bit first, then fall back to 8bit
if (! snd_pcm_hw_params_test_format(pcm, hw, SND_PCM_FORMAT_S16_LE))
m_format = SND_PCM_FORMAT_S16_LE;
else if (! snd_pcm_hw_params_test_format(pcm, hw, SND_PCM_FORMAT_S16_BE))
m_format = SND_PCM_FORMAT_S16_BE;
else if (! snd_pcm_hw_params_test_format(pcm, hw, SND_PCM_FORMAT_U8))
m_format = SND_PCM_FORMAT_U8;
else
m_format = SND_PCM_FORMAT_UNKNOWN;
}
if (snd_pcm_hw_params_set_format(pcm, hw, m_format) < 0) {
_error = "Unable to set format!";
return 1;
}
unsigned rate = _samplingRate;
if (snd_pcm_hw_params_set_rate_near(pcm, hw, &rate, 0) < 0) {
_error = "Unable to set sampling rate!";
return 1;
}
const unsigned int tolerance = _samplingRate/10+1000;
if (abs((int)rate - (int)_samplingRate) > (int)tolerance) {
_error = "Can't set requested sampling rate!";
char details[80];
sprintf(details," (requested rate %d, got rate %d)",
_samplingRate, rate);
_error += details;
return 1;
}
_samplingRate = rate;
if (snd_pcm_hw_params_set_channels(pcm, hw, _channels) < 0) {
_error = "Unable to set channels!";
return 1;
}
m_period_size = _fragmentSize;
if (m_format != SND_PCM_FORMAT_U8)
m_period_size <<= 1;
if (_channels > 1)
m_period_size /= _channels;
if (snd_pcm_hw_params_set_period_size_near(pcm, hw, &m_period_size, 0) < 0) {
_error = "Unable to set period size!";
return 1;
}
m_periods = _fragmentCount;
if (snd_pcm_hw_params_set_periods_near(pcm, hw, &m_periods, 0) < 0) {
_error = "Unable to set periods!";
return 1;
}
if (snd_pcm_hw_params(pcm, hw) < 0) {
_error = "Unable to set hw params!";
return 1;
}
_fragmentSize = m_period_size;
_fragmentCount = m_periods;
if (m_format != SND_PCM_FORMAT_U8)
_fragmentSize >>= 1;
if (_channels > 1)
_fragmentSize *= _channels;
return 0; // ok, we're ready..
}
#endif /* HAVE_LIBASOUND2 */