Rename file

Now that we use the 1.x ALSA APIs, naming the file `audioioalsa9.cpp`
(revering to ALSA version 0.9) is misleading.

Signed-off-by: Bobby Bingham <koorogi@koorogi.info>

1 files changed, 595 insertions, 0 deletions

diff --git a/flow/audioioalsa.cpp b/flow/audioioalsa.cpp
new file mode 100644
index 0000000..9162d41
--- /dev/null
+++ b/flow/audioioalsa.cpp
@@ -0,0 +1,595 @@
+    /*
+
+    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 */