static int
alsa_init (struct sound_device *sd)
{
const char *file;
snd_pcm_t *handle;
int err;
/* Open the sound device. Default is "default". */
if (sd->file)
file = sd->file;
else
file = DEFAULT_ALSA_SOUND_DEVICE;
snd_lib_error_set_handler ((snd_lib_error_handler_t) snd_error_quiet);
err = snd_pcm_open (&handle, file, SND_PCM_STREAM_PLAYBACK, 0);
snd_lib_error_set_handler (NULL);
if (err < 0)
return 0;
snd_pcm_close (handle);
sd->fd = -1;
sd->open = alsa_open;
sd->close = alsa_close;
sd->configure = alsa_configure;
sd->choose_format = alsa_choose_format;
sd->write = alsa_write;
sd->period_size = alsa_period_size;
return 1;
}
/* API: refresh the device list */
static pj_status_t alsa_factory_refresh(pjmedia_aud_dev_factory *f)
{
struct alsa_factory *af = (struct alsa_factory*)f;
char **hints, **n;
int err;
TRACE_((THIS_FILE, "pjmedia_snd_init: Enumerate sound devices"));
if (af->pool != NULL) {
pj_pool_release(af->pool);
af->pool = NULL;
}
af->pool = pj_pool_create(af->pf, "alsa_aud", 256, 256, NULL);
af->dev_cnt = 0;
/* Enumerate sound devices */
err = snd_device_name_hint(-1, "pcm", (void***)&hints);
if (err != 0)
return PJMEDIA_EAUD_SYSERR;
/* Set a null error handler prior to enumeration to suppress errors */
snd_lib_error_set_handler(null_alsa_error_handler);
n = hints;
while (*n != NULL) {
char *name = snd_device_name_get_hint(*n, "NAME");
if (name != NULL) {
if (0 != strcmp("null", name))
add_dev(af, name);
free(name);
}
n++;
}
/* Get the mixer name */
get_mixer_name(af);
/* Install error handler after enumeration, otherwise we'll get many
* error messages about invalid card/device ID.
*/
snd_lib_error_set_handler(alsa_error_handler);
err = snd_device_name_free_hint((void**)hints);
PJ_LOG(4,(THIS_FILE, "ALSA driver found %d devices", af->dev_cnt));
return PJ_SUCCESS;
}
static int do_alsa_open(snd_rawmidi_t **handle_p, const char *plu_name,
const char *def_dev)
{
char *dname = pcm_parse_params(config.snd_plugin_params,
plu_name, device_name_param);
const char *dev_name = dname ?: def_dev;
int ret;
snd_lib_error_set_handler(&alsa_log_handler);
ret = midoalsa_open(handle_p, dev_name);
/* reset back the error handler to not steal logs from
* other subsystems, like audio */
snd_lib_error_set_handler(NULL);
free(dname);
return ret;
}
AudioHardwareALSA::AudioHardwareALSA() :
mALSADevice(0),
mAcousticDevice(0)
{
snd_lib_error_set_handler(&ALSAErrorHandler);
mMixer = new ALSAMixer;
hw_module_t *module;
int err = hw_get_module(ALSA_HARDWARE_MODULE_ID,
(hw_module_t const**)&module);
if (err == 0) {
hw_device_t* device;
err = module->methods->open(module, ALSA_HARDWARE_NAME, &device);
if (err == 0) {
mALSADevice = (alsa_device_t *)device;
mALSADevice->init(mALSADevice, mDeviceList);
} else
LOGE("ALSA Module could not be opened!!!");
} else
LOGE("ALSA Module not found!!!");
err = hw_get_module(ACOUSTICS_HARDWARE_MODULE_ID,
(hw_module_t const**)&module);
if (err == 0) {
hw_device_t* device;
err = module->methods->open(module, ACOUSTICS_HARDWARE_NAME, &device);
if (err == 0)
mAcousticDevice = (acoustic_device_t *)device;
else
LOGE("Acoustics Module not found.");
}
}
/**************************************************************************
* ALSA_MidiInit [internal]
*
* Initializes the MIDI devices information variables
*/
LONG ALSA_MidiInit(void)
{
#ifdef HAVE_ALSA
static BOOL bInitDone = FALSE;
snd_seq_client_info_t *cinfo;
snd_seq_port_info_t *pinfo;
if (bInitDone)
return TRUE;
TRACE("Initializing the MIDI variables.\n");
bInitDone = TRUE;
/* try to open device */
if (midiOpenSeq(0) == -1) {
return TRUE;
}
#if 0 /* Debug purpose */
snd_lib_error_set_handler(error_handler);
#endif
snd_seq_client_info_alloca(&cinfo);
snd_seq_port_info_alloca(&pinfo);
/* First, search for all internal midi devices */
snd_seq_client_info_set_client(cinfo, -1);
while(snd_seq_query_next_client(midiSeq, cinfo) >= 0) {
snd_seq_port_info_set_client(pinfo, snd_seq_client_info_get_client(cinfo));
snd_seq_port_info_set_port(pinfo, -1);
while (snd_seq_query_next_port(midiSeq, pinfo) >= 0) {
int cap = snd_seq_port_info_get_capability(pinfo);
int type = snd_seq_port_info_get_type(pinfo);
if (type != SND_SEQ_PORT_TYPE_MIDI_GENERIC)
ALSA_AddMidiPort(cinfo, pinfo, cap, type);
}
}
/* Second, search for all external ports */
snd_seq_client_info_set_client(cinfo, -1);
while(snd_seq_query_next_client(midiSeq, cinfo) >= 0) {
snd_seq_port_info_set_client(pinfo, snd_seq_client_info_get_client(cinfo));
snd_seq_port_info_set_port(pinfo, -1);
while (snd_seq_query_next_port(midiSeq, pinfo) >= 0) {
int cap = snd_seq_port_info_get_capability(pinfo);
int type = snd_seq_port_info_get_type(pinfo);
if (type == SND_SEQ_PORT_TYPE_MIDI_GENERIC)
ALSA_AddMidiPort(cinfo, pinfo, cap, type);
}
}
/* close file and exit */
midiCloseSeq();
TRACE("End\n");
#endif
return TRUE;
}
AlsaBackend::~AlsaBackend()
{
Log("%s\n",__PRETTY_FUNCTION__);
snd_lib_error_set_handler(0);
snd_config_update_free_global();
std::map<AudioStream*, AlsaThread*>::iterator iter;
for (iter = p_alsathread.begin(); iter != p_alsathread.end(); iter++)
{
delete iter->second;
delete iter->first;
}
}
static int
laudio_alsa_init(laudio_status_cb cb, cfg_t *cfg_audio)
{
snd_lib_error_set_handler(logger_alsa);
status_cb = cb;
card_name = cfg_getstr(cfg_audio, "card");
mixer_name = cfg_getstr(cfg_audio, "mixer");
hdl = NULL;
mixer_hdl = NULL;
vol_elem = NULL;
return 0;
}
int
laudio_init(laudio_status_cb cb)
{
snd_lib_error_set_handler(logger_alsa);
status_cb = cb;
card_name = cfg_getstr(cfg_getsec(cfg, "audio"), "card");
mixer_name = cfg_getstr(cfg_getsec(cfg, "audio"), "mixer");
hdl = NULL;
mixer_hdl = NULL;
vol_elem = NULL;
return 0;
}
static DECLCALLBACK(int) drvHostALSAAudioInit(PPDMIHOSTAUDIO pInterface)
{
NOREF(pInterface);
LogFlowFuncEnter();
int rc = audioLoadAlsaLib();
if (RT_FAILURE(rc))
LogRel(("ALSA: Failed to load the ALSA shared library, rc=%Rrc\n", rc));
else
{
#ifdef DEBUG
snd_lib_error_set_handler(drvHostALSAAudioErrorHandler);
#endif
}
return rc;
}
/* API: destroy factory */
static pj_status_t alsa_factory_destroy(pjmedia_aud_dev_factory *f)
{
struct alsa_factory *af = (struct alsa_factory*)f;
if (af->pool)
pj_pool_release(af->pool);
if (af->base_pool) {
pj_pool_t *pool = af->base_pool;
af->base_pool = NULL;
pj_pool_release(pool);
}
/* Restore handler */
snd_lib_error_set_handler(NULL);
return PJ_SUCCESS;
}
static void
alsaspdifsink_class_init (AlsaSPDIFSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gobject_class->set_property = alsaspdifsink_set_property;
gobject_class->get_property = alsaspdifsink_get_property;
gobject_class->dispose = alsaspdifsink_dispose;
gobject_class->finalize = alsaspdifsink_finalize;
gstelement_class->change_state = alsaspdifsink_change_state;
gstelement_class->provide_clock = alsaspdifsink_provide_clock;
gstbasesink_class->event = alsaspdifsink_event;
gstbasesink_class->render = alsaspdifsink_render;
gstbasesink_class->get_times = alsaspdifsink_get_times;
gstbasesink_class->set_caps = alsaspdifsink_set_caps;
#if 0
/* We ignore the device property anyway, so don't install it
* we don't want the user supplying just any device string for us.
* At most we might want a card number and an iec958.%d device name
* to attempt */
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"ALSA device, as defined in an asound configuration file",
"default", G_PARAM_READWRITE));
#endif
g_object_class_install_property (gobject_class, PROP_CARD,
g_param_spec_int ("card", "Card",
"ALSA card number for the SPDIF device to use",
0, G_MAXINT, 0, G_PARAM_READWRITE));
snd_lib_error_set_handler (ignore_alsa_err);
}
AudioHardwareALSA::AudioHardwareALSA() :
mALSADevice(0),
mAcousticDevice(0)
{
snd_lib_error_set_handler(&ALSAErrorHandler);
mMixer = new ALSAMixer;
hw_module_t *module;
int err = hw_get_module(ALSA_HARDWARE_MODULE_ID,
(hw_module_t const**)&module);
if (err == 0) {
hw_device_t* device;
err = module->methods->open(module, ALSA_HARDWARE_NAME, &device);
if (err == 0) {
mALSADevice = (alsa_device_t *)device;
mALSADevice->init(mALSADevice, mDeviceList);
} else
LOGE("ALSA Module could not be opened!!!");
} else
LOGE("ALSA Module not found!!!");
err = hw_get_module(ACOUSTICS_HARDWARE_MODULE_ID,
(hw_module_t const**)&module);
if (err == 0) {
hw_device_t* device;
err = module->methods->open(module, ACOUSTICS_HARDWARE_NAME, &device);
if (err == 0)
mAcousticDevice = (acoustic_device_t *)device;
else
LOGE("Acoustics Module not found.");
}
mMutestate = true; //[LG_FW_P970_MERGE] - jungsoo1221.lee // 20100426 [email protected] Add the mic mute [START_LGE]
}
static int open_alsa(audio_output_t *ao)
{
const char *pcm_name;
snd_pcm_t *pcm=NULL;
printf("open_alsa with %p", ao->userptr);
#ifndef DEBUG
if(0) snd_lib_error_set_handler(error_ignorer);
#endif
pcm_name = ao->device ? ao->device : "default";
if (snd_pcm_open(&pcm, pcm_name, SND_PCM_STREAM_PLAYBACK, 0) < 0) {
if(!0) printf("cannot open device %s", pcm_name);
return -1;
}
ao->userptr = pcm;
if (ao->format != -1) {
/* we're going to play: initalize sample format */
return initialize_device(ao);
} else {
/* query mode; sample format will be set for each query */
return 0;
}
}
static gboolean
plugin_init (GstPlugin * plugin)
#endif // GSTREAMER_LITE
{
int err;
#ifndef GSTREAMER_LITE
if (!gst_element_register (plugin, "alsasrc", GST_RANK_PRIMARY,
GST_TYPE_ALSA_SRC))
return FALSE;
#endif // GSTREAMER_LITE
if (!gst_element_register (plugin, "alsasink", GST_RANK_PRIMARY,
GST_TYPE_ALSA_SINK))
return FALSE;
#ifndef GSTREAMER_LITE
if (!gst_element_register (plugin, "alsamidisrc", GST_RANK_PRIMARY,
GST_TYPE_ALSA_MIDI_SRC))
return FALSE;
#endif // GSTREAMER_LITE
GST_DEBUG_CATEGORY_INIT (alsa_debug, "alsa", 0, "alsa plugins");
#ifdef ENABLE_NLS
GST_DEBUG ("binding text domain %s to locale dir %s", GETTEXT_PACKAGE,
LOCALEDIR);
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif
err = snd_lib_error_set_handler (gst_alsa_error_wrapper);
if (err != 0)
GST_WARNING ("failed to set alsa error handler");
return TRUE;
}
AlsaBackend::AlsaBackend()
{
Log("%s\n",__PRETTY_FUNCTION__);
snd_lib_error_set_handler(alsa_error_handler);
}
/*static*/ int
alsa_init(cubeb ** context, char const * context_name)
{
cubeb * ctx;
int r;
int i;
int fd[2];
pthread_attr_t attr;
snd_pcm_t * dummy;
assert(context);
*context = NULL;
pthread_mutex_lock(&cubeb_alsa_mutex);
if (!cubeb_alsa_error_handler_set) {
snd_lib_error_set_handler(silent_error_handler);
cubeb_alsa_error_handler_set = 1;
}
pthread_mutex_unlock(&cubeb_alsa_mutex);
ctx = calloc(1, sizeof(*ctx));
assert(ctx);
ctx->ops = &alsa_ops;
r = pthread_mutex_init(&ctx->mutex, NULL);
assert(r == 0);
r = pipe(fd);
assert(r == 0);
for (i = 0; i < 2; ++i) {
fcntl(fd[i], F_SETFD, fcntl(fd[i], F_GETFD) | FD_CLOEXEC);
fcntl(fd[i], F_SETFL, fcntl(fd[i], F_GETFL) | O_NONBLOCK);
}
ctx->control_fd_read = fd[0];
ctx->control_fd_write = fd[1];
/* Force an early rebuild when alsa_run is first called to ensure fds and
nfds have been initialized. */
ctx->rebuild = 1;
r = pthread_attr_init(&attr);
assert(r == 0);
r = pthread_attr_setstacksize(&attr, 256 * 1024);
assert(r == 0);
r = pthread_create(&ctx->thread, &attr, alsa_run_thread, ctx);
assert(r == 0);
r = pthread_attr_destroy(&attr);
assert(r == 0);
/* Open a dummy PCM to force the configuration space to be evaluated so that
init_local_config_with_workaround can find and modify the default node. */
r = alsa_locked_pcm_open(&dummy, SND_PCM_STREAM_PLAYBACK, NULL);
if (r >= 0) {
alsa_locked_pcm_close(dummy);
}
ctx->is_pa = 0;
pthread_mutex_lock(&cubeb_alsa_mutex);
ctx->local_config = init_local_config_with_workaround(CUBEB_ALSA_PCM_NAME);
pthread_mutex_unlock(&cubeb_alsa_mutex);
if (ctx->local_config) {
ctx->is_pa = 1;
r = alsa_locked_pcm_open(&dummy, SND_PCM_STREAM_PLAYBACK, ctx->local_config);
/* If we got a local_config, we found a PA PCM. If opening a PCM with that
config fails with EINVAL, the PA PCM is too old for this workaround. */
if (r == -EINVAL) {
pthread_mutex_lock(&cubeb_alsa_mutex);
snd_config_delete(ctx->local_config);
pthread_mutex_unlock(&cubeb_alsa_mutex);
ctx->local_config = NULL;
} else if (r >= 0) {
alsa_locked_pcm_close(dummy);
}
}
*context = ctx;
return CUBEB_OK;
}
void CAESinkALSA::EnumerateDevicesEx(AEDeviceInfoList &list, bool force)
{
/* ensure that ALSA has been initialized */
snd_lib_error_set_handler(sndLibErrorHandler);
if(!snd_config || force)
{
if(force)
snd_config_update_free_global();
snd_config_update();
}
snd_config_t *config;
snd_config_copy(&config, snd_config);
/* Always enumerate the default device.
* Note: If "default" is a stereo device, EnumerateDevice()
* will automatically add "@" instead to enable surroundXX mangling.
* We don't want to do that if "default" can handle multichannel
* itself (e.g. in case of a pulseaudio server). */
EnumerateDevice(list, "default", "", config);
void **hints;
if (snd_device_name_hint(-1, "pcm", &hints) < 0)
{
CLog::Log(LOGINFO, "CAESinkALSA - Unable to get a list of devices");
return;
}
std::string defaultDescription;
for (void** hint = hints; *hint != NULL; ++hint)
{
char *io = snd_device_name_get_hint(*hint, "IOID");
char *name = snd_device_name_get_hint(*hint, "NAME");
char *desc = snd_device_name_get_hint(*hint, "DESC");
if ((!io || strcmp(io, "Output") == 0) && name
&& strcmp(name, "null") != 0)
{
std::string baseName = std::string(name);
baseName = baseName.substr(0, baseName.find(':'));
if (strcmp(name, "default") == 0)
{
/* added already, but lets get the description if we have one */
if (desc)
defaultDescription = desc;
}
else if (baseName == "front")
{
/* Enumerate using the surroundXX mangling */
/* do not enumerate basic "front", it is already handled
* by the default "@" entry added in the very beginning */
if (strcmp(name, "front") != 0)
EnumerateDevice(list, std::string("@") + (name+5), desc ? desc : name, config);
}
/* Do not enumerate "default", it is already enumerated above. */
/* Do not enumerate the sysdefault or surroundXX devices, those are
* always accompanied with a "front" device and it is handled above
* as "@". The below devices will be automatically used if available
* for a "@" device. */
/* Ubuntu has patched their alsa-lib so that "defaults.namehint.extended"
* defaults to "on" instead of upstream "off", causing lots of unwanted
* extra devices (many of which are not actually routed properly) to be
* found by the enumeration process. Skip them as well ("hw", "dmix",
* "plughw", "dsnoop"). */
else if (baseName != "default"
&& baseName != "sysdefault"
&& baseName != "surround40"
&& baseName != "surround41"
&& baseName != "surround50"
&& baseName != "surround51"
&& baseName != "surround71"
&& baseName != "hw"
&& baseName != "dmix"
&& baseName != "plughw"
&& baseName != "dsnoop")
{
EnumerateDevice(list, name, desc ? desc : name, config);
}
}
free(io);
free(name);
free(desc);
}
snd_device_name_free_hint(hints);
/* set the displayname for default device */
if (!list.empty() && list[0].m_deviceName == "default")
{
/* If we have one from a hint (DESC), use it */
if (!defaultDescription.empty())
list[0].m_displayName = defaultDescription;
/* Otherwise use the discovered name or (unlikely) "Default" */
else if (list[0].m_displayName.empty())
list[0].m_displayName = "Default";
}
/* lets check uniqueness, we may need to append DEV or CARD to DisplayName */
/* If even a single device of card/dev X clashes with Y, add suffixes to
* all devices of both them, for clarity. */
/* clashing card names, e.g. "NVidia", "NVidia_2" */
std::set<std::string> cardsToAppend;
/* clashing basename + cardname combinations, e.g. ("hdmi","Nvidia") */
std::set<std::pair<std::string, std::string> > devsToAppend;
for (AEDeviceInfoList::iterator it1 = list.begin(); it1 != list.end(); ++it1)
{
for (AEDeviceInfoList::iterator it2 = it1+1; it2 != list.end(); ++it2)
{
if (it1->m_displayName == it2->m_displayName
&& it1->m_displayNameExtra == it2->m_displayNameExtra)
{
/* something needs to be done */
std::string cardString1 = GetParamFromName(it1->m_deviceName, "CARD");
std::string cardString2 = GetParamFromName(it2->m_deviceName, "CARD");
if (cardString1 != cardString2)
{
/* card name differs, add identifiers to all devices */
cardsToAppend.insert(cardString1);
cardsToAppend.insert(cardString2);
continue;
}
std::string devString1 = GetParamFromName(it1->m_deviceName, "DEV");
std::string devString2 = GetParamFromName(it2->m_deviceName, "DEV");
if (devString1 != devString2)
{
/* device number differs, add identifiers to all such devices */
devsToAppend.insert(std::make_pair(it1->m_deviceName.substr(0, it1->m_deviceName.find(':')), cardString1));
devsToAppend.insert(std::make_pair(it2->m_deviceName.substr(0, it2->m_deviceName.find(':')), cardString2));
continue;
}
/* if we got here, the configuration is really weird, just append the whole device string */
it1->m_displayName += " (" + it1->m_deviceName + ")";
it2->m_displayName += " (" + it2->m_deviceName + ")";
}
}
}
for (std::set<std::string>::iterator it = cardsToAppend.begin();
it != cardsToAppend.end(); ++it)
{
for (AEDeviceInfoList::iterator itl = list.begin(); itl != list.end(); ++itl)
{
std::string cardString = GetParamFromName(itl->m_deviceName, "CARD");
if (cardString == *it)
/* "HDA NVidia (NVidia)", "HDA NVidia (NVidia_2)", ... */
itl->m_displayName += " (" + cardString + ")";
}
}
for (std::set<std::pair<std::string, std::string> >::iterator it = devsToAppend.begin();
it != devsToAppend.end(); ++it)
{
for (AEDeviceInfoList::iterator itl = list.begin(); itl != list.end(); ++itl)
{
std::string baseName = itl->m_deviceName.substr(0, itl->m_deviceName.find(':'));
std::string cardString = GetParamFromName(itl->m_deviceName, "CARD");
if (baseName == it->first && cardString == it->second)
{
std::string devString = GetParamFromName(itl->m_deviceName, "DEV");
/* "HDMI #0", "HDMI #1" ... */
itl->m_displayNameExtra += " #" + devString;
}
}
}
}
/* API: refresh the device list */
static pj_status_t alsa_factory_refresh(pjmedia_aud_dev_factory *f)
{
struct alsa_factory *af = (struct alsa_factory*)f;
char **hints, **n;
int err;
TRACE_((THIS_FILE, "pjmedia_snd_init: Enumerate sound devices"));
if (af->pool != NULL) {
pj_pool_release(af->pool);
af->pool = NULL;
}
af->pool = pj_pool_create(af->pf, "alsa_aud", 256, 256, NULL);
af->dev_cnt = 0;
/* Enumerate sound devices */
err = snd_device_name_hint(-1, "pcm", (void***)&hints);
if (err != 0)
return PJMEDIA_EAUD_SYSERR;
/* Set a null error handler prior to enumeration to suppress errors */
snd_lib_error_set_handler(null_alsa_error_handler);
n = hints;
while (*n != NULL) {
char *name = snd_device_name_get_hint(*n, "NAME");
char *desc = snd_device_name_get_hint(*n, "DESC");
if (name != NULL) {
if (strncmp("null", name, 4) == 0 ||
strncmp("front", name, 5) == 0 ||
strncmp("rear", name, 4) == 0 ||
strncmp("side", name, 4) == 0 ||
strncmp("dmix", name, 4) == 0 ||
strncmp("dsnoop", name, 6) == 0 ||
strncmp("hw", name, 2) == 0 ||
strncmp("plughw", name, 6) == 0 ||
strncmp("center_lfe", name, 10) == 0 ||
strncmp("surround40", name, 10) == 0 ||
strncmp("surround41", name, 10) == 0 ||
strncmp("surround50", name, 10) == 0 ||
strncmp("surround51", name, 10) == 0 ||
strncmp("surround71", name, 10) == 0 ||
(strncmp("default", name, 7) == 0 && strstr(name, ":CARD=") != NULL)) {
/* skip these devices, 'sysdefault' always contains the relevant information */
;
} else {
add_dev(af, name, desc);
}
free(name);
free(desc);
}
n++;
}
/* Install error handler after enumeration, otherwise we'll get many
* error messages about invalid card/device ID.
*/
snd_lib_error_set_handler(alsa_error_handler);
err = snd_device_name_free_hint((void**)hints);
PJ_LOG(4,(THIS_FILE, "ALSA driver found %d devices", af->dev_cnt));
return PJ_SUCCESS;
}
static void alsa_card_init(MSSndCard *obj){
AlsaData *ad=ms_new0(AlsaData,1);
obj->data=ad;
snd_lib_error_set_handler(alsa_error_log_handler);
}
AudioHardwareALSA::AudioHardwareALSA() :
mMixer(0),
mMixerSpdif(0),
mMixerSgtl5000(0),
mALSADevice(0),
mAcousticDevice(0)
{
snd_lib_error_set_handler(&ALSAErrorHandler);
hw_module_t *module;
char snd_sgtl5000[32], snd_spdif[32];
char **cardname = new char*[MAXCARDSNUM];
for (int i = 0; i < MAXCARDSNUM; i++) {
cardname[i] = new char[128];
memset(cardname[i],0,128);
}
int id;
int err = hw_get_module(ALSA_HARDWARE_MODULE_ID,
(hw_module_t const**)&module);
if (err == 0) {
hw_device_t* device;
err = module->methods->open(module, ALSA_HARDWARE_NAME, &device);
if (err == 0) {
mALSADevice = (alsa_device_t *)device;
mALSADevice->init(mALSADevice, mDeviceList);
} else
LOGE("ALSA Module could not be opened!!!");
} else
LOGE("ALSA Module not found!!!");
/* found out sound cards in the system and new mixer controller for them*/
err = findSoundCards(cardname);
if (err == 0) {
for (id = 0; id < MAXCARDSNUM; id++) {
if(cardname[id] && strstr(cardname[id],SPDIF)){
LOGD(" CARD NAME: %s ID %d", cardname[id],id);
sprintf(snd_spdif,"hw:0%d",id);
sprintf(snd_spdif,"hw:CARD=%d",id);
mMixerSpdif = new ALSAMixer(snd_spdif);
}else if (cardname[id] && strstr(cardname[id],SGTL5000)){
LOGD(" CARD NAME: %s ID %d", cardname[id],id);
sprintf(snd_sgtl5000,"hw:0%d",id);
sprintf(snd_sgtl5000,"hw:CARD=%d",id);
mMixerSgtl5000 = new ALSAMixer(snd_sgtl5000);
}
}
} else {
LOGE("Don't find any Sound cards, use default");
mMixerSgtl5000 = new ALSAMixer("hw:00");
mMixerSpdif = new ALSAMixer("hw:00");
}
for (int i = 0; i < MAXCARDSNUM; i++) {
delete []cardname[i];
}
delete []cardname;
mCurCard = new char[128];
if (!mCurCard)
LOGE("allocate memeory to store current sound card name fail");
memset(mCurCard,0,sizeof(mCurCard));
/* set current card as sgtl5000 default */
if(mMixerSgtl5000)
{
strcpy(mCurCard,SGTL5000);
mMixer = mMixerSgtl5000;
}else if(mMixerSpdif)
{
strcpy(mCurCard,SPDIF);
mMixer = mMixerSpdif;
}
err = hw_get_module(ACOUSTICS_HARDWARE_MODULE_ID,
(hw_module_t const**)&module);
if (err == 0) {
hw_device_t* device;
err = module->methods->open(module, ACOUSTICS_HARDWARE_NAME, &device);
if (err == 0)
mAcousticDevice = (acoustic_device_t *)device;
else
LOGE("Acoustics Module not found.");
}
}
static void
laudio_alsa_deinit(void)
{
snd_lib_error_set_handler(NULL);
}
/*
open & setup audio device
return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
unsigned int alsa_buffer_time = 500000; /* 0.5 s */
unsigned int alsa_fragcount = 16;
int err;
int block;
strarg_t device;
snd_pcm_uframes_t chunk_size;
snd_pcm_uframes_t bufsize;
snd_pcm_uframes_t boundary;
const opt_t subopts[] = {
{"block", OPT_ARG_BOOL, &block, NULL},
{"device", OPT_ARG_STR, &device, str_maxlen},
{NULL}
};
char alsa_device[ALSA_DEVICE_SIZE + 1];
// make sure alsa_device is null-terminated even when using strncpy etc.
memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
channels, format);
alsa_handler = NULL;
#if SND_LIB_VERSION >= 0x010005
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
#else
mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
#endif
snd_lib_error_set_handler(alsa_error_handler);
ao_data.samplerate = rate_hz;
ao_data.format = format;
ao_data.channels = channels;
switch (format)
{
case AF_FORMAT_S8:
alsa_format = SND_PCM_FORMAT_S8;
break;
case AF_FORMAT_U8:
alsa_format = SND_PCM_FORMAT_U8;
break;
case AF_FORMAT_U16_LE:
alsa_format = SND_PCM_FORMAT_U16_LE;
break;
case AF_FORMAT_U16_BE:
alsa_format = SND_PCM_FORMAT_U16_BE;
break;
case AF_FORMAT_AC3_LE:
case AF_FORMAT_S16_LE:
case AF_FORMAT_IEC61937_LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
case AF_FORMAT_AC3_BE:
case AF_FORMAT_S16_BE:
case AF_FORMAT_IEC61937_BE:
alsa_format = SND_PCM_FORMAT_S16_BE;
break;
case AF_FORMAT_U32_LE:
alsa_format = SND_PCM_FORMAT_U32_LE;
break;
case AF_FORMAT_U32_BE:
alsa_format = SND_PCM_FORMAT_U32_BE;
break;
case AF_FORMAT_S32_LE:
alsa_format = SND_PCM_FORMAT_S32_LE;
break;
case AF_FORMAT_S32_BE:
alsa_format = SND_PCM_FORMAT_S32_BE;
break;
case AF_FORMAT_U24_LE:
alsa_format = SND_PCM_FORMAT_U24_3LE;
break;
case AF_FORMAT_U24_BE:
alsa_format = SND_PCM_FORMAT_U24_3BE;
break;
case AF_FORMAT_S24_LE:
alsa_format = SND_PCM_FORMAT_S24_3LE;
break;
case AF_FORMAT_S24_BE:
alsa_format = SND_PCM_FORMAT_S24_3BE;
break;
case AF_FORMAT_FLOAT_LE:
alsa_format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AF_FORMAT_FLOAT_BE:
alsa_format = SND_PCM_FORMAT_FLOAT_BE;
break;
case AF_FORMAT_MU_LAW:
alsa_format = SND_PCM_FORMAT_MU_LAW;
break;
case AF_FORMAT_A_LAW:
alsa_format = SND_PCM_FORMAT_A_LAW;
break;
default:
alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
break;
}
//subdevice parsing
// set defaults
block = 1;
/* switch for spdif
* sets opening sequence for SPDIF
* sets also the playback and other switches 'on the fly'
* while opening the abstract alias for the spdif subdevice
* 'iec958'
*/
if (AF_FORMAT_IS_IEC61937(format)) {
device.str = "iec958";
mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", channels);
}
else
/* in any case for multichannel playback we should select
* appropriate device
*/
switch (channels) {
case 1:
case 2:
device.str = "default";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
break;
case 4:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
// hack - use the converter plugin
device.str = "plug:surround40";
else
device.str = "surround40";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
break;
case 6:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround51";
else
device.str = "surround51";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
break;
case 8:
if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
device.str = "plug:surround71";
else
device.str = "surround71";
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
break;
default:
device.str = "default";
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels);
}
device.len = strlen(device.str);
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
return 0;
}
parse_device(alsa_device, device.str, device.len);
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
if (!alsa_handler) {
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
int isac3 = AF_FORMAT_IS_IEC61937(format);
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
mp_msg(MSGT_AO,MSGL_V,"alsa-init: opening device in %sblocking mode\n", block ? "" : "non-");
if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
{
if (err != -EBUSY && !block) {
mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed);
if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
} else {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err));
return 0;
}
}
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err));
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: device reopened in blocking mode\n");
}
snd_pcm_hw_params_alloca(&alsa_hwparams);
snd_pcm_sw_params_alloca(&alsa_swparams);
// setting hw-parameters
if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetInitialParameters,
snd_strerror(err));
return 0;
}
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetAccessType,
snd_strerror(err));
return 0;
}
/* workaround for nonsupported formats
sets default format to S16_LE if the given formats aren't supported */
if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_INFO,
MSGTR_AO_ALSA_FormatNotSupportedByHardware, af_fmt2str_short(format));
alsa_format = SND_PCM_FORMAT_S16_LE;
if (AF_FORMAT_IS_AC3(ao_data.format))
ao_data.format = AF_FORMAT_AC3_LE;
else if (AF_FORMAT_IS_IEC61937(ao_data.format))
ao_data.format = AF_FORMAT_IEC61937_LE;
else
ao_data.format = AF_FORMAT_S16_LE;
}
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
alsa_format)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
&ao_data.channels)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels,
snd_strerror(err));
return 0;
}
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
prefer our own resampler, since that allows users to choose the resampler,
even per file if desired */
#if SND_LIB_VERSION >= 0x010009
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
0)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
&ao_data.samplerate, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2,
snd_strerror(err));
return 0;
}
bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
bytes_per_sample *= ao_data.channels;
ao_data.bps = ao_data.samplerate * bytes_per_sample;
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
&alsa_buffer_time, NULL)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear,
snd_strerror(err));
return 0;
}
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
&alsa_fragcount, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods,
snd_strerror(err));
return 0;
}
/* finally install hardware parameters */
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters,
snd_strerror(err));
return 0;
}
// end setting hw-params
// gets buffersize for control
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
{
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err));
return 0;
}
else {
ao_data.buffersize = bufsize * bytes_per_sample;
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
}
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err));
return 0;
} else {
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
}
ao_data.outburst = chunk_size * bytes_per_sample;
/* setting software parameters */
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary,
snd_strerror(err));
return 0;
}
#else
boundary = 0x7fffffff;
#endif
/* start playing when one period has been written */
if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold,
snd_strerror(err));
return 0;
}
/* disable underrun reporting */
if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold,
snd_strerror(err));
return 0;
}
#if SND_LIB_VERSION >= 0x000901
/* play silence when there is an underrun */
if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize,
snd_strerror(err));
return 0;
}
#endif
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters,
snd_strerror(err));
return 0;
}
/* end setting sw-params */
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
ao_data.samplerate, ao_data.channels, (int)bytes_per_sample, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
} // end switch alsa_handler (spdif)
alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
return 1;
} // end init
int main(int argc, char **argv)
{
int c;
snd_seq_t *seq;
int queue = 0, convert_time = 0, convert_real = 0, exclusive = 0;
int command = SUBSCRIBE;
int list_perm = 0;
int client;
int list_subs = 0;
snd_seq_port_subscribe_t *subs;
snd_seq_addr_t sender, dest;
#ifdef ENABLE_NLS
setlocale(LC_ALL, "");
textdomain(PACKAGE);
#endif
while ((c = getopt_long(argc, argv, "dior:t:elx", long_option, NULL)) != -1) {
switch (c) {
case 'd':
command = UNSUBSCRIBE;
break;
case 'i':
command = LIST;
list_perm |= LIST_INPUT;
break;
case 'o':
command = LIST;
list_perm |= LIST_OUTPUT;
break;
case 'e':
exclusive = 1;
break;
case 'r':
queue = atoi(optarg);
convert_time = 1;
convert_real = 1;
break;
case 't':
queue = atoi(optarg);
convert_time = 1;
convert_real = 0;
break;
case 'l':
list_subs = 1;
break;
case 'x':
command = REMOVE_ALL;
break;
default:
usage();
exit(1);
}
}
if (snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0) < 0) {
fprintf(stderr, _("can't open sequencer\n"));
return 1;
}
snd_lib_error_set_handler(error_handler);
switch (command) {
case LIST:
do_search_port(seq, list_perm,
list_subs ? print_port_and_subs : print_port);
snd_seq_close(seq);
return 0;
case REMOVE_ALL:
remove_all_connections(seq);
snd_seq_close(seq);
return 0;
}
/* connection or disconnection */
if (optind + 2 > argc) {
snd_seq_close(seq);
usage();
exit(1);
}
if ((client = snd_seq_client_id(seq)) < 0) {
snd_seq_close(seq);
fprintf(stderr, _("can't get client id\n"));
return 1;
}
/* set client info */
if (snd_seq_set_client_name(seq, "ALSA Connector") < 0) {
snd_seq_close(seq);
fprintf(stderr, _("can't set client info\n"));
return 1;
}
/* set subscription */
if (snd_seq_parse_address(seq, &sender, argv[optind]) < 0) {
snd_seq_close(seq);
fprintf(stderr, _("invalid sender address %s\n"), argv[optind]);
return 1;
}
if (snd_seq_parse_address(seq, &dest, argv[optind + 1]) < 0) {
snd_seq_close(seq);
fprintf(stderr, _("invalid destination address %s\n"), argv[optind + 1]);
return 1;
}
snd_seq_port_subscribe_alloca(&subs);
snd_seq_port_subscribe_set_sender(subs, &sender);
snd_seq_port_subscribe_set_dest(subs, &dest);
snd_seq_port_subscribe_set_queue(subs, queue);
snd_seq_port_subscribe_set_exclusive(subs, exclusive);
snd_seq_port_subscribe_set_time_update(subs, convert_time);
snd_seq_port_subscribe_set_time_real(subs, convert_real);
if (command == UNSUBSCRIBE) {
if (snd_seq_get_port_subscription(seq, subs) < 0) {
snd_seq_close(seq);
fprintf(stderr, _("No subscription is found\n"));
return 1;
}
if (snd_seq_unsubscribe_port(seq, subs) < 0) {
snd_seq_close(seq);
fprintf(stderr, _("Disconnection failed (%s)\n"), snd_strerror(errno));
return 1;
}
} else {
if (snd_seq_get_port_subscription(seq, subs) == 0) {
snd_seq_close(seq);
fprintf(stderr, _("Connection is already subscribed\n"));
return 1;
}
if (snd_seq_subscribe_port(seq, subs) < 0) {
snd_seq_close(seq);
fprintf(stderr, _("Connection failed (%s)\n"), snd_strerror(errno));
return 1;
}
}
snd_seq_close(seq);
return 0;
}
int
sa_stream_open(sa_stream_t *s) {
snd_output_t* out;
char* buf;
size_t bufsz;
snd_pcm_hw_params_t* hwparams;
snd_pcm_sw_params_t* swparams;
int dir;
snd_pcm_uframes_t period;
if (s == NULL) {
return SA_ERROR_NO_INIT;
}
if (s->output_unit != NULL) {
return SA_ERROR_INVALID;
}
pthread_mutex_lock(&sa_alsa_mutex);
/* Turn off debug output to stderr */
snd_lib_error_set_handler(quiet_error_handler);
if (snd_pcm_open(&s->output_unit,
"default",
SND_PCM_STREAM_PLAYBACK,
0) < 0) {
pthread_mutex_unlock(&sa_alsa_mutex);
return SA_ERROR_NO_DEVICE;
}
if (snd_pcm_set_params(s->output_unit,
#ifdef SA_LITTLE_ENDIAN
SND_PCM_FORMAT_S16_LE,
#else
SND_PCM_FORMAT_S16_BE,
#endif
SND_PCM_ACCESS_RW_INTERLEAVED,
s->n_channels,
s->rate,
1,
500000) < 0) {
snd_pcm_close(s->output_unit);
s->output_unit = NULL;
pthread_mutex_unlock(&sa_alsa_mutex);
return SA_ERROR_NOT_SUPPORTED;
}
/* ugly alsa-pulse plugin detection */
snd_output_buffer_open(&out);
snd_pcm_dump(s->output_unit, out);
bufsz = snd_output_buffer_string(out, &buf);
s->pulseaudio = bufsz >= strlen(ALSA_PA_PLUGIN) &&
strncmp(buf, ALSA_PA_PLUGIN, strlen(ALSA_PA_PLUGIN)) == 0;
snd_output_close(out);
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_hw_params_current(s->output_unit, hwparams);
snd_pcm_hw_params_get_period_size(hwparams, &period, &dir);
pthread_mutex_unlock(&sa_alsa_mutex);
return SA_SUCCESS;
}
ALSAPCMPlayer::ALSAPCMPlayer(boost::asio::io_service &_io_service) :
io_service(_io_service)
{
snd_lib_error_set_handler(alsa_error_handler_stub);
}
