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-rw-r--r--flow/audioioalsa.cpp595
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diff --git a/flow/audioioalsa.cpp b/flow/audioioalsa.cpp
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+ /*
+
+ 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 */
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