main() {
long loops;
int rc;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
unsigned char *buffer;
int i=0;
/* socket setting */
int sd;
struct sockaddr_in s_addr;
int n, n_send, status;
int on = 1;
sd = socket (AF_INET, SOCK_DGRAM, 0);
bzero(&s_addr, sizeof(s_addr));
s_addr.sin_family = AF_INET;
s_addr.sin_addr.s_addr = inet_addr("192.168.42.255");
s_addr.sin_port = htons(2007);
if((status = setsockopt(sd, SOL_SOCKET, SO_BROADCAST, &on, sizeof(on))) != 0 )
{
printf("setsockopt error\n");
exit(-1);
}
/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&handle, "plughw:1,0",
SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0) {
fprintf(stderr,
"unable to open pcm device: %s\n",
snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 1);
/* 44100 bits/second sampling rate (CD quality) */
val = 44100;
snd_pcm_hw_params_set_rate_near(handle, params,
&val, &dir);
/* Set period size to 32 frames. */
frames =4800;
snd_pcm_hw_params_set_period_size_near(handle,
params, &frames, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
size = frames *1; /* 2 bytes/sample, 2 channels */
buffer = (unsigned char*) malloc(size);
/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
while (1) {
rc = snd_pcm_readi(handle, buffer, frames);
if (rc == -EPIPE) {
/* EPIPE means overrun */
fprintf(stderr, "overrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr,
"error from read: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr, "short read, read %d frames\n", rc);
}
if ((n_send = sendto(sd, buffer, strlen(buffer), 0, (struct sockaddr *)&s_addr, sizeof(s_addr))) < 0 ) {
fprintf(stderr, "sendto() error");
exit(-3);
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
close(sd);
}
bool OutputALSA::initialize(quint32 freq, ChannelMap map, Qmmp::AudioFormat format)
{
m_inited = false;
if (pcm_handle)
return false;
if (snd_pcm_open(&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) < 0)
{
qWarning ("OutputALSA: Error opening PCM device %s", pcm_name);
return false;
}
// we need to configure
uint rate = freq; /* Sample rate */
uint exact_rate = freq; /* Sample rate returned by */
/* load settings from config */
QSettings settings(Qmmp::configFile(), QSettings::IniFormat);
settings.beginGroup("ALSA");
uint buffer_time = settings.value("buffer_time",500).toUInt()*1000;
uint period_time = settings.value("period_time",100).toUInt()*1000;
bool use_pause = settings.value("use_snd_pcm_pause", false).toBool();
settings.endGroup();
snd_pcm_hw_params_t *hwparams = 0;
snd_pcm_sw_params_t *swparams = 0;
int err; //alsa error code
//hw params
snd_pcm_hw_params_alloca(&hwparams);
if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0)
{
qWarning("OutputALSA: Can not read configuration for PCM device: %s", snd_strerror(err));
return false;
}
if (m_use_mmap)
{
if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_MMAP_INTERLEAVED)) < 0)
{
qWarning("OutputALSA: Error setting mmap access: %s", snd_strerror(err));
m_use_mmap = false;
}
}
if (!m_use_mmap)
{
if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
qWarning("OutputALSA: Error setting access: %s", snd_strerror(err));
return false;
}
}
snd_pcm_format_t alsa_format = SND_PCM_FORMAT_UNKNOWN;
switch (format)
{
case Qmmp::PCM_S8:
alsa_format = SND_PCM_FORMAT_S8;
break;
case Qmmp::PCM_S16LE:
alsa_format = SND_PCM_FORMAT_S16_LE;
break;
case Qmmp::PCM_S24LE:
alsa_format = SND_PCM_FORMAT_S24_LE;
break;
case Qmmp::PCM_S32LE:
alsa_format = SND_PCM_FORMAT_S32_LE;
break;
default:
qWarning("OutputALSA: unsupported format detected");
return false;
}
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, alsa_format)) < 0)
{
qDebug("OutputALSA: Error setting format: %s", snd_strerror(err));
return false;
}
exact_rate = rate;
if ((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0)) < 0)
{
qWarning("OutputALSA: Error setting rate: %s", snd_strerror(err));
return false;
}
if (rate != exact_rate)
{
qWarning("OutputALSA: The rate %d Hz is not supported by your hardware.\n==> Using %d Hz instead.", rate, exact_rate);
rate = exact_rate;
}
uint c = map.count();
if ((err = snd_pcm_hw_params_set_channels_near(pcm_handle, hwparams, &c)) < 0)
{
qWarning("OutputALSA: Error setting channels: %s", snd_strerror(err));
return false;
}
if (c != (uint)map.count())
{
qWarning("OutputALSA: The channel number %d is not supported by your hardware", map.count());
qWarning("==> Using %d instead.", c);
}
if ((err = snd_pcm_hw_params_set_period_time_near(pcm_handle, hwparams, &period_time ,0)) < 0)
{
qWarning("OutputALSA: Error setting period time: %s", snd_strerror(err));
return false;
}
if ((err = snd_pcm_hw_params_set_buffer_time_near(pcm_handle, hwparams, &buffer_time ,0)) < 0)
{
qWarning("OutputALSA: Error setting buffer time: %s", snd_strerror(err));
return false;
}
if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0)
{
qWarning("OutputALSA: Error setting HW params: %s", snd_strerror(err));
return false;
}
//read some alsa parameters
snd_pcm_uframes_t buffer_size = 0;
snd_pcm_uframes_t period_size = 0;
if ((err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size)) < 0)
{
qWarning("OutputALSA: Error reading buffer size: %s", snd_strerror(err));
return false;
}
if ((err = snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0)) < 0)
{
qWarning("OutputALSA: Error reading period size: %s", snd_strerror(err));
return false;
}
//swparams
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(pcm_handle, swparams);
if ((err = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams,
buffer_size - period_size)) < 0)
qWarning("OutputALSA: Error setting threshold: %s", snd_strerror(err));
if ((err = snd_pcm_sw_params(pcm_handle, swparams)) < 0)
{
qWarning("OutputALSA: Error setting SW params: %s", snd_strerror(err));
return false;
}
//setup needed values
m_chunk_size = period_size;
m_can_pause = snd_pcm_hw_params_can_pause(hwparams) && use_pause;
qDebug("OutputALSA: can pause: %d", m_can_pause);
ChannelMap out_map = map;
#if (SND_LIB_VERSION >= 0x01001B)
//channel map configuration
snd_pcm_chmap_t *chmap = snd_pcm_get_chmap(pcm_handle);
if(chmap)
{
out_map.clear();
char tmp[256];
memset(tmp,0,256);
snd_pcm_chmap_print(chmap, 256, tmp);
qDebug("OutputALSA: received channel map: %s",tmp);
for(uint i = 0; i < chmap->channels; ++i)
{
if(m_alsa_channels.keys().contains(chmap->pos[i]))
out_map.append(m_alsa_channels.value(chmap->pos[i]));
else
out_map.append(Qmmp::CHAN_NULL);
}
free(chmap);
}
else
qWarning("OutputALSA: Unable to receive current channel map");
#endif
configure(exact_rate, out_map, format); //apply configuration
//create alsa prebuffer;
m_prebuf_size = 2 * snd_pcm_frames_to_bytes(pcm_handle, m_chunk_size); //buffer for two periods
m_prebuf = (uchar *)malloc(m_prebuf_size);
m_inited = true;
return true;
}
void CAESinkALSA::EnumerateDevice(AEDeviceInfoList &list, const std::string &device, const std::string &description, snd_config_t *config)
{
snd_pcm_t *pcmhandle = NULL;
if (!OpenPCMDevice(device, "", ALSA_MAX_CHANNELS, &pcmhandle, config, false))
return;
snd_pcm_info_t *pcminfo;
snd_pcm_info_alloca(&pcminfo);
memset(pcminfo, 0, snd_pcm_info_sizeof());
int err = snd_pcm_info(pcmhandle, pcminfo);
if (err < 0)
{
CLog::Log(LOGINFO, "CAESinkALSA - Unable to get pcm_info for \"%s\"", device.c_str());
snd_pcm_close(pcmhandle);
}
int cardNr = snd_pcm_info_get_card(pcminfo);
CAEDeviceInfo info;
info.m_deviceName = device;
info.m_deviceType = AEDeviceTypeFromName(device);
if (cardNr >= 0)
{
/* "HDA NVidia", "HDA Intel", "HDA ATI HDMI", "SB Live! 24-bit External", ... */
char *cardName;
if (snd_card_get_name(cardNr, &cardName) == 0)
info.m_displayName = cardName;
if (info.m_deviceType == AE_DEVTYPE_HDMI && info.m_displayName.size() > 5 &&
info.m_displayName.substr(info.m_displayName.size()-5) == " HDMI")
{
/* We already know this is HDMI, strip it */
info.m_displayName.erase(info.m_displayName.size()-5);
}
/* "CONEXANT Analog", "USB Audio", "HDMI 0", "ALC889 Digital" ... */
std::string pcminfoName = snd_pcm_info_get_name(pcminfo);
/*
* Filter "USB Audio", in those cases snd_card_get_name() is more
* meaningful already
*/
if (pcminfoName != "USB Audio")
info.m_displayNameExtra = pcminfoName;
if (info.m_deviceType == AE_DEVTYPE_HDMI)
{
/* replace, this was likely "HDMI 0" */
info.m_displayNameExtra = "HDMI";
int dev = snd_pcm_info_get_device(pcminfo);
if (dev >= 0)
{
/* lets see if we can get ELD info */
snd_ctl_t *ctlhandle;
std::stringstream sstr;
sstr << "hw:" << cardNr;
std::string strHwName = sstr.str();
if (snd_ctl_open_lconf(&ctlhandle, strHwName.c_str(), 0, config) == 0)
{
snd_hctl_t *hctl;
if (snd_hctl_open_ctl(&hctl, ctlhandle) == 0)
{
snd_hctl_load(hctl);
bool badHDMI = false;
if (!GetELD(hctl, dev, info, badHDMI))
CLog::Log(LOGDEBUG, "CAESinkALSA - Unable to obtain ELD information for device \"%s\" (not supported by device, or kernel older than 3.2)",
device.c_str());
/* snd_hctl_close also closes ctlhandle */
snd_hctl_close(hctl);
if (badHDMI)
{
/* only trust badHDMI (= unconnected or non-existent port) on Intel
* and NVIDIA where it has been confirmed to work, show the empty
* port on other systems */
if (info.m_displayName.compare(0, 9, "HDA Intel") == 0 || info.m_displayName.compare(0, 10, "HDA NVidia") == 0)
{
/* unconnected HDMI port */
CLog::Log(LOGDEBUG, "CAESinkALSA - Skipping HDMI device \"%s\" as it has no ELD data", device.c_str());
snd_pcm_close(pcmhandle);
return;
}
else
{
CLog::Log(LOGDEBUG, "CAESinkALSA - HDMI device \"%s\" may be unconnected (no ELD data)", device.c_str());
}
}
}
else
{
snd_ctl_close(ctlhandle);
}
}
}
}
else if (info.m_deviceType == AE_DEVTYPE_IEC958)
{
/* append instead of replace, pcminfoName is useful for S/PDIF */
if (!info.m_displayNameExtra.empty())
info.m_displayNameExtra += ' ';
info.m_displayNameExtra += "S/PDIF";
}
else if (info.m_displayNameExtra.empty())
{
/* for USB audio, it gets a bit confusing as there is
* - "SB Live! 24-bit External"
* - "SB Live! 24-bit External, S/PDIF"
* so add "Analog" qualifier to the first one */
info.m_displayNameExtra = "Analog";
}
/* "default" is a device that will be used for all inputs, while
* "@" will be mangled to front/default/surroundXX as necessary */
if (device == "@" || device == "default")
{
/* Make it "Default (whatever)" */
info.m_displayName = "Default (" + info.m_displayName + (info.m_displayNameExtra.empty() ? "" : " " + info.m_displayNameExtra + ")");
info.m_displayNameExtra = "";
}
}
else
{
/* virtual devices: "default", "pulse", ... */
/* description can be e.g. "PulseAudio Sound Server" - for hw devices it is
* normally uninteresting, like "HDMI Audio Output" or "Default Audio Device",
* so we only use it for virtual devices that have no better display name */
info.m_displayName = description;
}
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
memset(hwparams, 0, snd_pcm_hw_params_sizeof());
/* ensure we can get a playback configuration for the device */
if (snd_pcm_hw_params_any(pcmhandle, hwparams) < 0)
{
CLog::Log(LOGINFO, "CAESinkALSA - No playback configurations available for device \"%s\"", device.c_str());
snd_pcm_close(pcmhandle);
return;
}
/* detect the available sample rates */
for (unsigned int *rate = ALSASampleRateList; *rate != 0; ++rate)
if (snd_pcm_hw_params_test_rate(pcmhandle, hwparams, *rate, 0) >= 0)
info.m_sampleRates.push_back(*rate);
/* detect the channels available */
int channels = 0;
for (int i = ALSA_MAX_CHANNELS; i >= 1; --i)
{
/* Reopen the device if needed on the special "surroundXX" cases */
if (info.m_deviceType == AE_DEVTYPE_PCM && (i == 8 || i == 6 || i == 4))
OpenPCMDevice(device, "", i, &pcmhandle, config, false);
if (snd_pcm_hw_params_test_channels(pcmhandle, hwparams, i) >= 0)
{
channels = i;
break;
}
}
if (device == "default" && channels == 2)
{
/* This looks like the ALSA standard default stereo dmix device, we
* probably want to use "@" instead to get surroundXX. */
snd_pcm_close(pcmhandle);
EnumerateDevice(list, "@", description, config);
return;
}
CAEChannelInfo alsaChannels;
for (int i = 0; i < channels; ++i)
{
if (!info.m_channels.HasChannel(ALSAChannelMap[i]))
info.m_channels += ALSAChannelMap[i];
alsaChannels += ALSAChannelMap[i];
}
/* remove the channels from m_channels that we cant use */
info.m_channels.ResolveChannels(alsaChannels);
/* detect the PCM sample formats that are available */
for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
{
if (AE_IS_RAW(i) || i == AE_FMT_MAX)
continue;
snd_pcm_format_t fmt = AEFormatToALSAFormat(i);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
continue;
if (snd_pcm_hw_params_test_format(pcmhandle, hwparams, fmt) >= 0)
info.m_dataFormats.push_back(i);
}
snd_pcm_close(pcmhandle);
list.push_back(info);
}
static snd_pcm_t *
alsa_open (int channels, unsigned samplerate, int realtime)
{ const char * device = "default" ;
snd_pcm_t *alsa_dev = NULL ;
snd_pcm_hw_params_t *hw_params ;
snd_pcm_uframes_t buffer_size ;
snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
snd_pcm_sw_params_t *sw_params ;
int err ;
if (realtime)
{ alsa_period_size = 256 ;
alsa_buffer_frames = 3 * alsa_period_size ;
}
else
{ alsa_period_size = 1024 ;
alsa_buffer_frames = 4 * alsa_period_size ;
} ;
if ((err = snd_pcm_open (&alsa_dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{ fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
goto catch_error ;
} ;
snd_pcm_nonblock (alsa_dev, 0) ;
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{ fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_any (alsa_dev, hw_params)) < 0)
{ fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_access (alsa_dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{ fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
{ fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_rate_near (alsa_dev, hw_params, &samplerate, 0)) < 0)
{ fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_channels (alsa_dev, hw_params, channels)) < 0)
{ fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_dev, hw_params, &alsa_buffer_frames)) < 0)
{ fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params_set_period_size_near (alsa_dev, hw_params, &alsa_period_size, 0)) < 0)
{ fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_hw_params (alsa_dev, hw_params)) < 0)
{ fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
/* extra check: if we have only one period, this code won't work */
snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
if (alsa_period_size == buffer_size)
{ fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
goto catch_error ;
} ;
snd_pcm_hw_params_free (hw_params) ;
if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_sw_params_current (alsa_dev, sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
/* note: set start threshold to delay start until the ring buffer is full */
snd_pcm_sw_params_current (alsa_dev, sw_params) ;
if ((err = snd_pcm_sw_params_set_start_threshold (alsa_dev, sw_params, buffer_size)) < 0)
{ fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
goto catch_error ;
} ;
if ((err = snd_pcm_sw_params (alsa_dev, sw_params)) != 0)
{ fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
goto catch_error ;
} ;
snd_pcm_sw_params_free (sw_params) ;
snd_pcm_reset (alsa_dev) ;
catch_error :
if (err < 0 && alsa_dev != NULL)
{ snd_pcm_close (alsa_dev) ;
return NULL ;
} ;
return alsa_dev ;
} /* alsa_open */
int initAlsa(char **argv,int optind)
{
snd_pcm_hw_params_t *hw_params;
int err,n;
unsigned int Fs;
if ((err = snd_pcm_open(&capture_handle, argv[optind],
SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf(stderr, "Alsa cannot open audio device %s (%s)\n",argv[optind], snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0) {
fprintf(stderr,
"Alsa cannot allocate hardware parameter structure (%s)\n",snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0) {
fprintf(stderr,
"Alsa cannot initialize hardware parameter structure (%s)\n",snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params,SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf(stderr, "Alsa cannot set access type (%s)\n",snd_strerror(err));
return 1;
}
if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params,SND_PCM_FORMAT_S16)) < 0) {
fprintf(stderr, "Alsa cannot set sample format (%s)\n",snd_strerror(err));
return 1;
}
snd_pcm_hw_params_set_rate_resample(capture_handle, hw_params,0);
Fs=19200;
n=1;
if ((err = snd_pcm_hw_params_set_rate_near(capture_handle, hw_params, &Fs,&n)) < 0) {
fprintf(stderr, "Alsa cannot set sample rate (%s)\n",snd_strerror(err));
return 1;
}
fprintf(stderr, "Alsa sample rate %d\n",Fs);
if(snd_pcm_hw_params_get_channels (hw_params, &nbch)!=0) {
fprintf(stderr, "Alsa cannot get number of channels\n");
return 1;
}
if(nbch>4) {
fprintf(stderr, "Alsa too much channels\n");
return 1;
}
if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0) {
fprintf(stderr, "Alsa cannot set parameters (%s)\n",snd_strerror(err));
return 1;
}
snd_pcm_hw_params_free(hw_params);
if ((err = snd_pcm_prepare(capture_handle)) < 0) {
fprintf(stderr,
"Alsa cannot prepare audio interface for use (%s)\n",snd_strerror(err));
return 1;
}
for(n=0; n<nbch; n++) {
channel[n].chn=n;
channel[n].Infs=Fs;
channel[n].InBuff=malloc(MAXNBFRAMES*sizeof(sample_t));
}
for(; n<MAXNBCHANNELS; n++) channel[n].Infs=0;
return (0);
}
/**
* Create and initialize Alsa audio output device with given parameters.
*
* @param Parameters - optional parameters
* @throws AudioOutputException if output device cannot be opened
*/
AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, true, 1, 0) {
pcm_handle = NULL;
stream = SND_PCM_STREAM_PLAYBACK;
this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["CHANNELS"])->ValueAsInt();
this->uiSamplerate = ((DeviceCreationParameterInt*)Parameters["SAMPLERATE"])->ValueAsInt();
this->FragmentSize = ((DeviceCreationParameterInt*)Parameters["FRAGMENTSIZE"])->ValueAsInt();
uint Fragments = ((DeviceCreationParameterInt*)Parameters["FRAGMENTS"])->ValueAsInt();
String Card = ((DeviceCreationParameterString*)Parameters["CARD"])->ValueAsString();
dmsg(2,("Checking if hw parameters supported...\n"));
if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
pcm_name = "hw:" + Card;
}
else {
fprintf(stderr, "Warning: your soundcard doesn't support chosen hardware parameters; ");
fprintf(stderr, "trying to compensate support lack with plughw...");
fflush(stdout);
pcm_name = "plughw:" + Card;
}
dmsg(2,("HW check completed.\n"));
int err;
snd_pcm_hw_params_alloca(&hwparams); // Allocate the snd_pcm_hw_params_t structure on the stack.
/* Open PCM. The last parameter of this function is the mode. */
/* If this is set to 0, the standard mode is used. Possible */
/* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC. */
/* If SND_PCM_NONBLOCK is used, read / write access to the */
/* PCM device will return immediately. If SND_PCM_ASYNC is */
/* specified, SIGIO will be emitted whenever a period has */
/* been completely processed by the soundcard. */
if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
}
if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
}
/* Set access type. This can be either */
/* SND_PCM_ACCESS_RW_INTERLEAVED or */
/* SND_PCM_ACCESS_RW_NONINTERLEAVED. */
if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
}
/* Set sample format */
#if WORDS_BIGENDIAN
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
#else // little endian
if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
#endif
{
throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
}
int dir = 0;
/* Set sample rate. If the exact rate is not supported */
/* by the hardware, use nearest possible rate. */
#if ALSA_MAJOR > 0
if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
#else
if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
#endif
{
throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
}
if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
}
/* Set number of periods. Periods used to be called fragments. */
if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
throw AudioOutputException(String("Error setting number of ") + ToString(Fragments) + " periods: " + snd_strerror(err));
}
/* Set buffer size (in frames). The resulting latency is given by */
/* latency = periodsize * periods / (rate * bytes_per_frame) */
if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
}
/* Apply HW parameter settings to */
/* PCM device and prepare device */
if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_malloc(&swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_current(pcm_handle, swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_current: ") + snd_strerror(err));
}
if (snd_pcm_sw_params_set_stop_threshold(pcm_handle, swparams, 0xffffffff) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params_set_stop_threshold: ") + snd_strerror(err));
}
if (snd_pcm_sw_params(pcm_handle, swparams) != 0) {
throw AudioOutputException(String("Error in snd_pcm_sw_params: ") + snd_strerror(err));
}
if ((err = snd_pcm_prepare(pcm_handle)) < 0) {
throw AudioOutputException(String("Error snd_pcm_prepare: ") + snd_strerror(err));
}
// allocate Alsa output buffer
pAlsaOutputBuffer = new int16_t[uiAlsaChannels * FragmentSize];
// create audio channels for this audio device to which the sampler engines can write to
for (int i = 0; i < uiAlsaChannels; i++) this->Channels.push_back(new AudioChannel(i, FragmentSize));
if (((DeviceCreationParameterBool*)Parameters["ACTIVE"])->ValueAsBool()) {
Play();
}
}
/* Open and init the default recording device. */
void audioInit(void)
{
int rc;
int size;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir = 0;
/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&sound.handle, getenvDefault(SOUND_DEVICE_ENV,
SOUND_DEVICE),
SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0)
{
fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc));
exit(EXIT_FAILURE);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(sound.handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode. */
snd_pcm_hw_params_set_access(sound.handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format. */
snd_pcm_hw_params_set_format(sound.handle, params,
SND_PCM_FORMAT_S16_LE);
/* One channel (mono). */
snd_pcm_hw_params_set_channels(sound.handle, params, 1);
/* 44100 bits/second sampling rate (CD quality). */
val = SOUND_RATE;
snd_pcm_hw_params_set_rate_near(sound.handle, params, &val, &dir);
/* Set period size. It's best to set this to the same value as
* SOUND_SAMPLES_PER_TURN. A lower value would generate more
* hardware interrupts and thus a lower latency but that's of no use
* since we have to wait for an amount of SOUND_SAMPLES_PER_TURN
* samples anyway. */
snd_pcm_uframes_t frames = SOUND_SAMPLES_PER_TURN;
snd_pcm_hw_params_set_period_size_near(sound.handle, params,
&frames, &dir);
/* Write the parameters to the driver. */
rc = snd_pcm_hw_params(sound.handle, params);
if (rc < 0)
{
fprintf(stderr, "unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(EXIT_FAILURE);
}
/* Acquire n frames per turn. */
sound.bufferSizeFrames = SOUND_SAMPLES_PER_TURN;
size = sound.bufferSizeFrames * 2; /* 2 bytes/sample, 1 channel */
/* Initialize the buffer. */
sound.buffer = (char *)malloc(size);
sound.bufferLast = (char *)malloc(size);
sound.bufferFill = 0;
sound.bufferReady = 0;
/* Try to switch to non-blocking mode for reading. If that fails,
* print a warning and continue in blocking mode. */
rc = snd_pcm_nonblock(sound.handle, 1);
if (rc < 0)
{
fprintf(stderr, "Could not switch to non-blocking mode: %s\n",
snd_strerror(rc));
}
/* Prepare in audioRead() for the first time. */
sound.reprepare = 1;
}
static ALCboolean alsa_reset_playback(ALCdevice *device)
{
alsa_data *data = (alsa_data*)device->ExtraData;
snd_pcm_uframes_t periodSizeInFrames;
unsigned int periodLen, bufferLen;
snd_pcm_sw_params_t *sp = NULL;
snd_pcm_hw_params_t *hp = NULL;
snd_pcm_access_t access;
snd_pcm_format_t format;
unsigned int periods;
unsigned int rate;
const char *funcerr;
int allowmmap;
int err;
format = -1;
switch(device->FmtType)
{
case DevFmtByte:
format = SND_PCM_FORMAT_S8;
break;
case DevFmtUByte:
format = SND_PCM_FORMAT_U8;
break;
case DevFmtShort:
format = SND_PCM_FORMAT_S16;
break;
case DevFmtUShort:
format = SND_PCM_FORMAT_U16;
break;
case DevFmtInt:
format = SND_PCM_FORMAT_S32;
break;
case DevFmtUInt:
format = SND_PCM_FORMAT_U32;
break;
case DevFmtFloat:
format = SND_PCM_FORMAT_FLOAT;
break;
}
allowmmap = GetConfigValueBool("alsa", "mmap", 1);
periods = device->NumUpdates;
periodLen = (ALuint64)device->UpdateSize * 1000000 / device->Frequency;
bufferLen = periodLen * periods;
rate = device->Frequency;
snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
/* set interleaved access */
if(!allowmmap || snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0)
{
if(periods > 2)
{
periods--;
bufferLen = periodLen * periods;
}
CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
}
/* test and set format (implicitly sets sample bits) */
if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) < 0)
{
static const struct {
snd_pcm_format_t format;
enum DevFmtType fmttype;
} formatlist[] = {
{ SND_PCM_FORMAT_FLOAT, DevFmtFloat },
{ SND_PCM_FORMAT_S32, DevFmtInt },
{ SND_PCM_FORMAT_U32, DevFmtUInt },
{ SND_PCM_FORMAT_S16, DevFmtShort },
{ SND_PCM_FORMAT_U16, DevFmtUShort },
{ SND_PCM_FORMAT_S8, DevFmtByte },
{ SND_PCM_FORMAT_U8, DevFmtUByte },
};
size_t k;
for(k = 0;k < COUNTOF(formatlist);k++)
{
format = formatlist[k].format;
if(snd_pcm_hw_params_test_format(data->pcmHandle, hp, format) >= 0)
{
device->FmtType = formatlist[k].fmttype;
break;
}
}
}
CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
/* test and set channels (implicitly sets frame bits) */
if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)) < 0)
{
static const enum DevFmtChannels channellist[] = {
DevFmtStereo,
DevFmtQuad,
DevFmtX51,
DevFmtX71,
DevFmtMono,
};
size_t k;
for(k = 0;k < COUNTOF(channellist);k++)
{
if(snd_pcm_hw_params_test_channels(data->pcmHandle, hp, ChannelsFromDevFmt(channellist[k])) >= 0)
{
device->FmtChans = channellist[k];
break;
}
}
}
CHECK(snd_pcm_hw_params_set_channels(data->pcmHandle, hp, ChannelsFromDevFmt(device->FmtChans)));
/* set rate (implicitly constrains period/buffer parameters) */
if(snd_pcm_hw_params_set_rate_resample(data->pcmHandle, hp, 0) < 0)
ERR("Failed to disable ALSA resampler\n");
CHECK(snd_pcm_hw_params_set_rate_near(data->pcmHandle, hp, &rate, NULL));
/* set buffer time (implicitly constrains period/buffer parameters) */
CHECK(snd_pcm_hw_params_set_buffer_time_near(data->pcmHandle, hp, &bufferLen, NULL));
/* set period time (implicitly sets buffer size/bytes/time and period size/bytes) */
CHECK(snd_pcm_hw_params_set_period_time_near(data->pcmHandle, hp, &periodLen, NULL));
/* install and prepare hardware configuration */
CHECK(snd_pcm_hw_params(data->pcmHandle, hp));
/* retrieve configuration info */
CHECK(snd_pcm_hw_params_get_access(hp, &access));
CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL));
CHECK(snd_pcm_hw_params_get_periods(hp, &periods, NULL));
snd_pcm_hw_params_free(hp);
hp = NULL;
snd_pcm_sw_params_malloc(&sp);
CHECK(snd_pcm_sw_params_current(data->pcmHandle, sp));
CHECK(snd_pcm_sw_params_set_avail_min(data->pcmHandle, sp, periodSizeInFrames));
CHECK(snd_pcm_sw_params_set_stop_threshold(data->pcmHandle, sp, periodSizeInFrames*periods));
CHECK(snd_pcm_sw_params(data->pcmHandle, sp));
#undef CHECK
snd_pcm_sw_params_free(sp);
sp = NULL;
/* Increase periods by one, since the temp buffer counts as an extra
* period */
if(access == SND_PCM_ACCESS_RW_INTERLEAVED)
device->NumUpdates = periods+1;
else
device->NumUpdates = periods;
device->UpdateSize = periodSizeInFrames;
device->Frequency = rate;
SetDefaultChannelOrder(device);
return ALC_TRUE;
error:
ERR("%s failed: %s\n", funcerr, snd_strerror(err));
if(hp) snd_pcm_hw_params_free(hp);
if(sp) snd_pcm_sw_params_free(sp);
return ALC_FALSE;
}
static ALCenum alsa_open_capture(ALCdevice *Device, const ALCchar *deviceName)
{
const char *driver = NULL;
snd_pcm_hw_params_t *hp;
snd_pcm_uframes_t bufferSizeInFrames;
snd_pcm_uframes_t periodSizeInFrames;
ALboolean needring = AL_FALSE;
snd_pcm_format_t format;
const char *funcerr;
alsa_data *data;
int err;
if(deviceName)
{
size_t idx;
if(!allCaptureDevNameMap)
allCaptureDevNameMap = probe_devices(SND_PCM_STREAM_CAPTURE, &numCaptureDevNames);
for(idx = 0;idx < numCaptureDevNames;idx++)
{
if(strcmp(deviceName, allCaptureDevNameMap[idx].name) == 0)
{
driver = allCaptureDevNameMap[idx].device;
break;
}
}
if(idx == numCaptureDevNames)
return ALC_INVALID_VALUE;
}
else
{
deviceName = alsaDevice;
driver = GetConfigValue("alsa", "capture", "default");
}
data = (alsa_data*)calloc(1, sizeof(alsa_data));
err = snd_pcm_open(&data->pcmHandle, driver, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if(err < 0)
{
ERR("Could not open capture device '%s': %s\n", driver, snd_strerror(err));
free(data);
return ALC_INVALID_VALUE;
}
format = -1;
switch(Device->FmtType)
{
case DevFmtByte:
format = SND_PCM_FORMAT_S8;
break;
case DevFmtUByte:
format = SND_PCM_FORMAT_U8;
break;
case DevFmtShort:
format = SND_PCM_FORMAT_S16;
break;
case DevFmtUShort:
format = SND_PCM_FORMAT_U16;
break;
case DevFmtInt:
format = SND_PCM_FORMAT_S32;
break;
case DevFmtUInt:
format = SND_PCM_FORMAT_U32;
break;
case DevFmtFloat:
format = SND_PCM_FORMAT_FLOAT;
break;
}
funcerr = NULL;
bufferSizeInFrames = maxu(Device->UpdateSize*Device->NumUpdates,
100*Device->Frequency/1000);
periodSizeInFrames = minu(bufferSizeInFrames, 25*Device->Frequency/1000);
snd_pcm_hw_params_malloc(&hp);
#define CHECK(x) if((funcerr=#x),(err=(x)) < 0) goto error
CHECK(snd_pcm_hw_params_any(data->pcmHandle, hp));
/* set interleaved access */
CHECK(snd_pcm_hw_params_set_access(data->pcmHandle, hp, SND_PCM_ACCESS_RW_INTERLEAVED));
/* set format (implicitly sets sample bits) */
CHECK(snd_pcm_hw_params_set_format(data->pcmHandle, hp, format));
/* set channels (implicitly sets frame bits) */
CHECK(snd_pcm_hw_params_set_channels(data->pcmHandle, hp, ChannelsFromDevFmt(Device->FmtChans)));
/* set rate (implicitly constrains period/buffer parameters) */
CHECK(snd_pcm_hw_params_set_rate(data->pcmHandle, hp, Device->Frequency, 0));
/* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
if(snd_pcm_hw_params_set_buffer_size_min(data->pcmHandle, hp, &bufferSizeInFrames) < 0)
{
TRACE("Buffer too large, using intermediate ring buffer\n");
needring = AL_TRUE;
CHECK(snd_pcm_hw_params_set_buffer_size_near(data->pcmHandle, hp, &bufferSizeInFrames));
}
/* set buffer size in frame units (implicitly sets period size/bytes/time and buffer time/bytes) */
CHECK(snd_pcm_hw_params_set_period_size_near(data->pcmHandle, hp, &periodSizeInFrames, NULL));
/* install and prepare hardware configuration */
CHECK(snd_pcm_hw_params(data->pcmHandle, hp));
/* retrieve configuration info */
CHECK(snd_pcm_hw_params_get_period_size(hp, &periodSizeInFrames, NULL));
#undef CHECK
snd_pcm_hw_params_free(hp);
hp = NULL;
if(needring)
{
data->ring = CreateRingBuffer(FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType),
Device->UpdateSize*Device->NumUpdates);
if(!data->ring)
{
ERR("ring buffer create failed\n");
goto error2;
}
data->size = snd_pcm_frames_to_bytes(data->pcmHandle, periodSizeInFrames);
data->buffer = malloc(data->size);
if(!data->buffer)
{
ERR("buffer malloc failed\n");
goto error2;
}
}
Device->DeviceName = strdup(deviceName);
Device->ExtraData = data;
return ALC_NO_ERROR;
error:
ERR("%s failed: %s\n", funcerr, snd_strerror(err));
if(hp) snd_pcm_hw_params_free(hp);
error2:
free(data->buffer);
DestroyRingBuffer(data->ring);
snd_pcm_close(data->pcmHandle);
free(data);
Device->ExtraData = NULL;
return ALC_INVALID_VALUE;
}
static int
quh_alsa_config (st_quh_nfo_t *file)
{
(void) file;
#if 0
snd_pcm_hw_params_t *hw_params;
snd_pcm_format_t format;
int rate = 0;
if (snd_pcm_hw_params_malloc (&hw_params) < 0)
return -1;
if (snd_pcm_hw_params_any (handle, hw_params) < 0)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
if (snd_pcm_hw_params_set_access (handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
switch (file->size)
{
case 1:
format = SND_PCM_FORMAT_S8;
break;
case 2:
format = SND_PCM_FORMAT_S16;
break;
case 3:
format = SND_PCM_FORMAT_S24;
break;
default:
format = SND_PCM_FORMAT_S16;
break;
}
if (snd_pcm_hw_params_set_format (handle, hw_params, format) < 0)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
rate = file->rate;
if (snd_pcm_hw_params_set_rate_near (handle, hw_params, rate, 0) < 0)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
if ((float) rate * 1.05 < file->rate || (float) rate * 0.95 > file->rate)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
if (snd_pcm_hw_params_set_channels (handle, hw_params, file->channels) < 0)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
if (snd_pcm_hw_params (handle, hw_params) < 0)
{
snd_pcm_hw_params_free (hw_params);
return -1;
}
snd_pcm_hw_params_free (hw_params);
#endif
return 0;
}
bool QAudioInputPrivate::open()
{
#ifdef DEBUG_AUDIO
QTime now(QTime::currentTime());
qDebug()<<now.second()<<"s "<<now.msec()<<"ms :open()";
#endif
clockStamp.restart();
timeStamp.restart();
elapsedTimeOffset = 0;
int dir;
int err = 0;
int count=0;
unsigned int freakuency=settings.frequency();
if (!settings.isValid()) {
qWarning("QAudioOutput: open error, invalid format.");
} else if (settings.sampleRate() <= 0) {
qWarning("QAudioOutput: open error, invalid sample rate (%d).",
settings.sampleRate());
} else {
err = -1;
}
if (err == 0) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
emit errorChanged(errorState);
return false;
}
QString dev = QString(QLatin1String(m_device.constData()));
QList<QByteArray> devices = QAudioDeviceInfoInternal::availableDevices(QAudio::AudioInput);
if(dev.compare(QLatin1String("default")) == 0) {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
if (devices.size() > 0)
dev = QLatin1String(devices.first());
else
return false;
#else
dev = QLatin1String("hw:0,0");
#endif
} else {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = QLatin1String(m_device);
#else
int idx = 0;
char *name;
QString shortName = QLatin1String(m_device.mid(m_device.indexOf('=',0)+1).constData());
while(snd_card_get_name(idx,&name) == 0) {
if(qstrncmp(shortName.toLocal8Bit().constData(),name,shortName.length()) == 0)
break;
idx++;
}
dev = QString(QLatin1String("hw:%1,0")).arg(idx);
#endif
}
// Step 1: try and open the device
while((count < 5) && (err < 0)) {
err=snd_pcm_open(&handle,dev.toLocal8Bit().constData(),SND_PCM_STREAM_CAPTURE,0);
if(err < 0)
count++;
}
if (( err < 0)||(handle == 0)) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
emit stateChanged(deviceState);
return false;
}
snd_pcm_nonblock( handle, 0 );
// Step 2: Set the desired HW parameters.
snd_pcm_hw_params_alloca( &hwparams );
bool fatal = false;
QString errMessage;
unsigned int chunks = 8;
err = snd_pcm_hw_params_any( handle, hwparams );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_any: err = %1").arg(err);
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_resample( handle, hwparams, 1 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_rate_resample: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_access( handle, hwparams, access );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_access: err = %1").arg(err);
}
}
if ( !fatal ) {
err = setFormat();
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_format: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_channels( handle, hwparams, (unsigned int)settings.channels() );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_channels: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_near( handle, hwparams, &freakuency, 0 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_rate_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_buffer_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_period_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_periods_near(handle, hwparams, &chunks, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params_set_periods_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params(handle, hwparams);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioInput: snd_pcm_hw_params: err = %1").arg(err);
}
}
if( err < 0) {
qWarning()<<errMessage;
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
emit stateChanged(deviceState);
return false;
}
snd_pcm_hw_params_get_buffer_size(hwparams,&buffer_frames);
buffer_size = snd_pcm_frames_to_bytes(handle,buffer_frames);
snd_pcm_hw_params_get_period_size(hwparams,&period_frames, &dir);
period_size = snd_pcm_frames_to_bytes(handle,period_frames);
snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
// Step 3: Set the desired SW parameters.
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(handle, swparams);
snd_pcm_sw_params_set_start_threshold(handle,swparams,period_frames);
snd_pcm_sw_params_set_stop_threshold(handle,swparams,buffer_frames);
snd_pcm_sw_params_set_avail_min(handle, swparams,period_frames);
snd_pcm_sw_params(handle, swparams);
// Step 4: Prepare audio
ringBuffer.resize(buffer_size);
snd_pcm_prepare( handle );
snd_pcm_start(handle);
// Step 5: Setup timer
bytesAvailable = checkBytesReady();
if(pullMode)
connect(audioSource,SIGNAL(readyRead()),this,SLOT(userFeed()));
// Step 6: Start audio processing
chunks = buffer_size/period_size;
timer->start(period_time*chunks/2000);
errorState = QAudio::NoError;
totalTimeValue = 0;
return true;
}
bool setParameters (unsigned int sampleRate, int numChannels, int bufferSize)
{
if (handle == nullptr)
return false;
JUCE_ALSA_LOG ("ALSADevice::setParameters(" << deviceID << ", "
<< (int) sampleRate << ", " << numChannels << ", " << bufferSize << ")");
snd_pcm_hw_params_t* hwParams;
snd_pcm_hw_params_alloca (&hwParams);
if (snd_pcm_hw_params_any (handle, hwParams) < 0)
{
// this is the error message that aplay returns when an error happens here,
// it is a bit more explicit that "Invalid parameter"
error = "Broken configuration for this PCM: no configurations available";
return false;
}
if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0) // works better for plughw..
isInterleaved = true;
else if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_NONINTERLEAVED) >= 0)
isInterleaved = false;
else
{
jassertfalse;
return false;
}
enum { isFloatBit = 1 << 16, isLittleEndianBit = 1 << 17, onlyUseLower24Bits = 1 << 18 };
const int formatsToTry[] = { SND_PCM_FORMAT_FLOAT_LE, 32 | isFloatBit | isLittleEndianBit,
SND_PCM_FORMAT_FLOAT_BE, 32 | isFloatBit,
SND_PCM_FORMAT_S32_LE, 32 | isLittleEndianBit,
SND_PCM_FORMAT_S32_BE, 32,
SND_PCM_FORMAT_S24_3LE, 24 | isLittleEndianBit,
SND_PCM_FORMAT_S24_3BE, 24,
SND_PCM_FORMAT_S24_LE, 32 | isLittleEndianBit | onlyUseLower24Bits,
SND_PCM_FORMAT_S16_LE, 16 | isLittleEndianBit,
SND_PCM_FORMAT_S16_BE, 16 };
bitDepth = 0;
for (int i = 0; i < numElementsInArray (formatsToTry); i += 2)
{
if (snd_pcm_hw_params_set_format (handle, hwParams, (_snd_pcm_format) formatsToTry [i]) >= 0)
{
const int type = formatsToTry [i + 1];
bitDepth = type & 255;
converter.reset (createConverter (isInput, bitDepth,
(type & isFloatBit) != 0,
(type & isLittleEndianBit) != 0,
(type & onlyUseLower24Bits) != 0,
numChannels,
isInterleaved));
break;
}
}
if (bitDepth == 0)
{
error = "device doesn't support a compatible PCM format";
JUCE_ALSA_LOG ("Error: " + error);
return false;
}
int dir = 0;
unsigned int periods = 4;
snd_pcm_uframes_t samplesPerPeriod = (snd_pcm_uframes_t) bufferSize;
if (JUCE_ALSA_FAILED (snd_pcm_hw_params_set_rate_near (handle, hwParams, &sampleRate, 0))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_set_channels (handle, hwParams, (unsigned int ) numChannels))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_set_periods_near (handle, hwParams, &periods, &dir))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_set_period_size_near (handle, hwParams, &samplesPerPeriod, &dir))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params (handle, hwParams)))
{
return false;
}
snd_pcm_uframes_t frames = 0;
if (JUCE_ALSA_FAILED (snd_pcm_hw_params_get_period_size (hwParams, &frames, &dir))
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_get_periods (hwParams, &periods, &dir)))
latency = 0;
else
latency = (int) frames * ((int) periods - 1); // (this is the method JACK uses to guess the latency..)
JUCE_ALSA_LOG ("frames: " << (int) frames << ", periods: " << (int) periods
<< ", samplesPerPeriod: " << (int) samplesPerPeriod);
snd_pcm_sw_params_t* swParams;
snd_pcm_sw_params_alloca (&swParams);
snd_pcm_uframes_t boundary;
if (JUCE_ALSA_FAILED (snd_pcm_sw_params_current (handle, swParams))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_get_boundary (swParams, &boundary))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_silence_threshold (handle, swParams, 0))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_silence_size (handle, swParams, boundary))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_start_threshold (handle, swParams, samplesPerPeriod))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params_set_stop_threshold (handle, swParams, boundary))
|| JUCE_ALSA_FAILED (snd_pcm_sw_params (handle, swParams)))
{
return false;
}
#if JUCE_ALSA_LOGGING
// enable this to dump the config of the devices that get opened
snd_output_t* out;
snd_output_stdio_attach (&out, stderr, 0);
snd_pcm_hw_params_dump (hwParams, out);
snd_pcm_sw_params_dump (swParams, out);
#endif
numChannelsRunning = numChannels;
return true;
}
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum CodecID *codec_id)
{
AlsaData *s = ctx->priv_data;
const char *audio_device;
int res, flags = 0;
snd_pcm_format_t format;
snd_pcm_t *h;
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t buffer_size, period_size;
int64_t layout = ctx->streams[0]->codec->channel_layout;
if (ctx->filename[0] == 0) audio_device = "default";
else audio_device = ctx->filename;
if (*codec_id == CODEC_ID_NONE)
*codec_id = DEFAULT_CODEC_ID;
format = codec_id_to_pcm_format(*codec_id);
if (format == SND_PCM_FORMAT_UNKNOWN) {
av_log(ctx, AV_LOG_ERROR, "sample format 0x%04x is not supported\n", *codec_id);
return AVERROR(ENOSYS);
}
s->frame_size = av_get_bits_per_sample(*codec_id) / 8 * channels;
if (ctx->flags & AVFMT_FLAG_NONBLOCK) {
flags = SND_PCM_NONBLOCK;
}
res = snd_pcm_open(&h, audio_device, mode, flags);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot open audio device %s (%s)\n",
audio_device, snd_strerror(res));
return AVERROR(EIO);
}
res = snd_pcm_hw_params_malloc(&hw_params);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror(res));
goto fail1;
}
res = snd_pcm_hw_params_any(h, hw_params);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_access(h, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set access type (%s)\n",
snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_format(h, hw_params, format);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set sample format 0x%04x %d (%s)\n",
*codec_id, format, snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_rate_near(h, hw_params, sample_rate, 0);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set sample rate (%s)\n",
snd_strerror(res));
goto fail;
}
res = snd_pcm_hw_params_set_channels(h, hw_params, channels);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set channel count to %d (%s)\n",
channels, snd_strerror(res));
goto fail;
}
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
buffer_size = FFMIN(buffer_size, ALSA_BUFFER_SIZE_MAX);
/* TODO: maybe use ctx->max_picture_buffer somehow */
res = snd_pcm_hw_params_set_buffer_size_near(h, hw_params, &buffer_size);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set ALSA buffer size (%s)\n",
snd_strerror(res));
goto fail;
}
snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
if (!period_size)
period_size = buffer_size / 4;
res = snd_pcm_hw_params_set_period_size_near(h, hw_params, &period_size, NULL);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set ALSA period size (%s)\n",
snd_strerror(res));
goto fail;
}
s->period_size = period_size;
res = snd_pcm_hw_params(h, hw_params);
if (res < 0) {
av_log(ctx, AV_LOG_ERROR, "cannot set parameters (%s)\n",
snd_strerror(res));
goto fail;
}
snd_pcm_hw_params_free(hw_params);
if (channels > 2 && layout) {
if (find_reorder_func(s, *codec_id, layout, mode == SND_PCM_STREAM_PLAYBACK) < 0) {
char name[128];
av_get_channel_layout_string(name, sizeof(name), channels, layout);
av_log(ctx, AV_LOG_WARNING, "ALSA channel layout unknown or unimplemented for %s %s.\n",
name, mode == SND_PCM_STREAM_PLAYBACK ? "playback" : "capture");
}
if (s->reorder_func) {
s->reorder_buf_size = buffer_size;
s->reorder_buf = av_malloc(s->reorder_buf_size * s->frame_size);
if (!s->reorder_buf)
goto fail1;
}
}
s->h = h;
return 0;
fail:
snd_pcm_hw_params_free(hw_params);
fail1:
snd_pcm_close(h);
return AVERROR(EIO);
}
alsa_dev_t* alsa_open(char *name)
{
alsa_dev_t *alsa_dev = NULL;
snd_pcm_hw_params_t *params = NULL;
uint32_t val = 0;
int32_t rc = 0;
int32_t dir = 0;
alsa_dev = malloc(sizeof(alsa_dev_t));
if (!alsa_dev)
{
syslog(LOG_ERR, "Failed to create audio device: Out of memory");
exit(EXIT_FAILURE);
}
/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&alsa_dev->snd_pcm, name, SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0)
{
syslog(LOG_ERR, "Cannot open pcm device: %s", snd_strerror(rc));
exit(EXIT_FAILURE);
}
alsa_dev->name = name;
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(alsa_dev->snd_pcm, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(alsa_dev->snd_pcm, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(alsa_dev->snd_pcm, params,
SND_PCM_FORMAT_S16_LE);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(alsa_dev->snd_pcm, params, AUDIO_CHANNEL_NUM);
/* 44100 bits/second sampling rate (CD quality) */
val = AUDIO_SAMPLE_RATE;
snd_pcm_hw_params_set_rate_near(alsa_dev->snd_pcm, params, &val, &dir);
alsa_dev->frames = 1152;
snd_pcm_hw_params_set_period_size_near(alsa_dev->snd_pcm, params,
&alsa_dev->frames, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(alsa_dev->snd_pcm, params);
if (rc < 0)
{
syslog(LOG_ERR, "Cannot set hw parameters: %s", snd_strerror(rc));
exit(EXIT_FAILURE);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params, &alsa_dev->frames, &dir);
/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params, &val, &dir);
snd_pcm_prepare(alsa_dev->snd_pcm);
snd_pcm_start(alsa_dev->snd_pcm);
return alsa_dev;
}
int main()
{
int fp;
unsigned int pcm, tmp, dir;
int buff_size;
long loops;
int rc;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
snd_pcm_uframes_t frames;
char *buff;
int rate, channels, seconds;
/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0)
{
fprintf(stderr,"unable to open pcm device: %s\n", snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 1);
/* 44100 bits/second sampling rate (CD quality) */
val = 8000;
snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
/* Set period size to 32 frames. */
frames = 32;
snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0)
{
fprintf(stderr,"unable to set hw parameters: %s\n", snd_strerror(rc));
exit(1);
}
/* Use a buff large enough to hold one period */
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
size = frames * 4; /* 2 bytes/sample, 2 channels */
buff = (char *) malloc(size);
/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params, &val, &dir);
loops = 50000000 / val;
fp=open("222.wav",O_WRONLY);
while (loops > 0)
{
loops--;
bzero(buff,sizeof(buff));
rc = snd_pcm_readi(handle, buff, frames);
write(fp,buff,128);
if (rc == -EPIPE)
{
/* EPIPE means overrun */
fprintf(stderr, "overrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0)
{
fprintf(stderr,"error from read: %s\n", snd_strerror(rc));
} else if (rc != (int)frames)
{
fprintf(stderr, "short read, read %d frames\n", rc);
}
/*rc = write(1, buff, size);
if (rc != size)
fprintf(stderr,"short write: wrote %d bytes\n", rc);
*/
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buff);
return 0;
}
// We're making these functions static - there's no need to pollute the global namespace
static void alsa_init(audio_backend_pull_callback_t pull_callback)
{
snd_pcm_hw_params_t *params;
unsigned int val;
int dir=-1;
string device = cfgLoadStr("alsa", "device", "");
int rc = -1;
if (device == "" || device == "auto")
{
printf("ALSA: trying to determine audio device\n");
// trying default device
device = "default";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
// "default" didn't work, try first device
if (rc < 0)
{
device = "plughw:0,0,0";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
if (rc < 0)
{
device = "plughw:0,0";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
}
// first didn't work, try second
if (rc < 0)
{
device = "plughw:1,0";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
// try pulse audio backend
if (rc < 0)
{
device = "pulse";
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
if (rc < 0)
printf("ALSA: unable to automatically determine audio device.\n");
}
else {
rc = snd_pcm_open(&handle, device.c_str(), SND_PCM_STREAM_PLAYBACK, 0);
}
if (rc < 0)
{
fprintf(stderr, "ALSA: unable to open PCM device %s: %s\n", device.c_str(), snd_strerror(rc));
return;
}
printf("ALSA: Successfully initialized \"%s\"\n", device.c_str());
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
rc=snd_pcm_hw_params_any(handle, params);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_any %s\n", snd_strerror(rc));
return;
}
/* Set the desired hardware parameters. */
/* Interleaved mode */
rc=snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_access %s\n", snd_strerror(rc));
return;
}
/* Signed 16-bit little-endian format */
rc=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_format %s\n", snd_strerror(rc));
return;
}
/* Two channels (stereo) */
rc=snd_pcm_hw_params_set_channels(handle, params, 2);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_channels %s\n", snd_strerror(rc));
return;
}
/* 44100 bits/second sampling rate (CD quality) */
val = 44100;
rc=snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_rate_near %s\n", snd_strerror(rc));
return;
}
/* Set period size to settings.aica.BufferSize frames. */
period_size = settings.aica.BufferSize;
rc=snd_pcm_hw_params_set_period_size_near(handle, params, &period_size, &dir);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s\n", snd_strerror(rc));
return;
}
else
{
printf("ALSA: period size set to %ld\n", period_size);
}
buffer_size = (44100 * 100 /* settings.omx.Audio_Latency */ / 1000 / period_size + 1) * period_size;
rc=snd_pcm_hw_params_set_buffer_size_near(handle, params, &buffer_size);
if (rc < 0)
{
fprintf(stderr, "ALSA: Error:snd_pcm_hw_params_set_buffer_size_near %s\n", snd_strerror(rc));
return;
}
else
{
printf("ALSA: buffer size set to %ld\n", buffer_size);
}
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0)
{
fprintf(stderr, "ALSA: Unable to set hw parameters: %s\n", snd_strerror(rc));
return;
}
}
void *data_streaming(void *socket_desc)
{
int rc;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
int i;
/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&handle, "plughw:1,0",
SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0) {
fprintf(stderr,
"unable to open pcm device: %s\n",
snd_strerror(rc));
exit(1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(¶ms);
/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);
/* Set the desired hardware parameters. */
/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
/* Signed 16-bit little-endian format */
/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_U8);
/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 1);
/* 44100 bits/second sampling rate (CD quality) */
val = 32768;
snd_pcm_hw_params_set_rate_near(handle, params,
&val, &dir);
/* Set period size to 32 frames. */
frames = SIZE;
snd_pcm_hw_params_set_period_size_near(handle,
params, &frames, &dir);
/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}
/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
/* 2 bytes/sample, 2 channels */
/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
while (1) {
pthread_mutex_lock(&mutex);
rc = snd_pcm_readi(handle, buffer, frames);
pthread_kill(tid, SIGUSR1);
pthread_mutex_unlock(&mutex);
if (rc == -EPIPE) {
/* EPIPE means overrun */
fprintf(stderr, "overrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr,"error from read: %s\n",snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr, "short read, read %d frames\n", rc);
}
}
snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);
return 0;
}
bool ALSAWriter::processParams(bool *paramsCorrected)
{
const unsigned chn = getParam("chn").toUInt();
const unsigned rate = getParam("rate").toUInt();
const bool resetAudio = channels != chn || sample_rate != rate;
channels = chn;
sample_rate = rate;
if (resetAudio || err)
{
snd_pcm_hw_params_t *params;
snd_pcm_hw_params_alloca(¶ms);
close();
QString chosenDevName = devName;
if (autoFindMultichannelDevice && channels > 2)
{
bool mustAutoFind = true, forceStereo = false;
if (!snd_pcm_open(&snd, chosenDevName.toLocal8Bit(), SND_PCM_STREAM_PLAYBACK, 0))
{
if (snd_pcm_type(snd) == SND_PCM_TYPE_HW)
{
unsigned max_chn = 0;
snd_pcm_hw_params_any(snd, params);
mustAutoFind = snd_pcm_hw_params_get_channels_max(params, &max_chn) || max_chn < channels;
}
#ifdef HAVE_CHMAP
else if (paramsCorrected)
{
snd_pcm_chmap_query_t **chmaps = snd_pcm_query_chmaps(snd);
if (chmaps)
snd_pcm_free_chmaps(chmaps);
else
forceStereo = true;
}
#endif
snd_pcm_close(snd);
snd = NULL;
}
if (mustAutoFind)
{
QString newDevName;
if (channels <= 4)
newDevName = "surround40";
else if (channels <= 6)
newDevName = "surround51";
else
newDevName = "surround71";
if (!newDevName.isEmpty() && newDevName != chosenDevName)
{
if (ALSACommon::getDevices().first.contains(newDevName))
chosenDevName = newDevName;
else if (forceStereo)
{
channels = 2;
*paramsCorrected = true;
}
}
}
}
if (!chosenDevName.isEmpty())
{
bool sndOpen = !snd_pcm_open(&snd, chosenDevName.toLocal8Bit(), SND_PCM_STREAM_PLAYBACK, 0);
if (devName != chosenDevName)
{
if (sndOpen)
QMPlay2Core.logInfo("ALSA :: " + devName + "\" -> \"" + chosenDevName + "\"");
else
{
sndOpen = !snd_pcm_open(&snd, devName.toLocal8Bit(), SND_PCM_STREAM_PLAYBACK, 0);
QMPlay2Core.logInfo("ALSA :: " + tr("Cannot open") + " \"" + chosenDevName + "\", " + tr("back to") + " \"" + devName + "\"");
}
}
if (sndOpen)
{
snd_pcm_hw_params_any(snd, params);
snd_pcm_format_t fmt = SND_PCM_FORMAT_UNKNOWN;
if (!snd_pcm_hw_params_test_format(snd, params, SND_PCM_FORMAT_S32))
{
fmt = SND_PCM_FORMAT_S32;
sample_size = 4;
}
else if (!snd_pcm_hw_params_test_format(snd, params, SND_PCM_FORMAT_S16))
{
fmt = SND_PCM_FORMAT_S16;
sample_size = 2;
}
else if (!snd_pcm_hw_params_test_format(snd, params, SND_PCM_FORMAT_S8))
{
fmt = SND_PCM_FORMAT_S8;
sample_size = 1;
}
unsigned delay_us = round(delay * 1000000.0);
if (fmt != SND_PCM_FORMAT_UNKNOWN && set_snd_pcm_hw_params(snd, params, fmt, channels, sample_rate, delay_us))
{
bool err2 = false;
if (channels != chn || sample_rate != rate)
{
if (paramsCorrected)
*paramsCorrected = true;
else
err2 = true;
}
if (!err2)
{
err2 = snd_pcm_hw_params(snd, params);
if (err2 && paramsCorrected) //jakiś błąd, próba zmiany sample_rate
{
snd_pcm_hw_params_any(snd, params);
err2 = snd_pcm_hw_params_set_rate_resample(snd, params, false) || !set_snd_pcm_hw_params(snd, params, fmt, channels, sample_rate, delay_us) || snd_pcm_hw_params(snd, params);
if (!err2)
*paramsCorrected = true;
}
if (!err2)
{
modParam("delay", delay_us / 1000000.0);
if (paramsCorrected && *paramsCorrected)
{
modParam("chn", channels);
modParam("rate", sample_rate);
}
canPause = snd_pcm_hw_params_can_pause(params) && snd_pcm_hw_params_can_resume(params);
mustSwapChn = channels == 6 || channels == 8;
#ifdef HAVE_CHMAP
if (mustSwapChn)
{
snd_pcm_chmap_query_t **chmaps = snd_pcm_query_chmaps(snd);
if (chmaps)
{
for (int i = 0; chmaps[i]; ++i)
{
if (chmaps[i]->map.channels >= channels)
{
for (uint j = 0; j < channels; ++j)
{
mustSwapChn &= chmaps[i]->map.pos[j] == j + SND_CHMAP_FL;
if (!mustSwapChn)
break;
}
break;
}
}
snd_pcm_free_chmaps(chmaps);
}
}
#endif
return true;
}
}
}
}
}
err = true;
QMPlay2Core.logError("ALSA :: " + tr("Cannot open audio output stream"));
}
return readyWrite();
}
/**************************************************************************
* widOpen [internal]
*/
static DWORD widOpen(WORD wDevID, LPWAVEOPENDESC lpDesc, DWORD dwFlags)
{
WINE_WAVEDEV* wwi;
snd_pcm_hw_params_t * hw_params;
snd_pcm_sw_params_t * sw_params;
snd_pcm_access_t access;
snd_pcm_format_t format;
unsigned int rate;
unsigned int buffer_time = 500000;
unsigned int period_time = 10000;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
int flags;
snd_pcm_t * pcm;
int err;
int dir;
DWORD ret;
/* JPW TODO - review this code */
TRACE("(%u, %p, %08X);\n", wDevID, lpDesc, dwFlags);
if (lpDesc == NULL) {
WARN("Invalid Parameter !\n");
return MMSYSERR_INVALPARAM;
}
if (wDevID >= ALSA_WidNumDevs) {
TRACE("Requested device %d, but only %d are known!\n", wDevID, ALSA_WidNumDevs);
return MMSYSERR_BADDEVICEID;
}
/* only PCM format is supported so far... */
if (!ALSA_supportedFormat(lpDesc->lpFormat)) {
WARN("Bad format: tag=%04X nChannels=%d nSamplesPerSec=%d !\n",
lpDesc->lpFormat->wFormatTag, lpDesc->lpFormat->nChannels,
lpDesc->lpFormat->nSamplesPerSec);
return WAVERR_BADFORMAT;
}
if (dwFlags & WAVE_FORMAT_QUERY) {
TRACE("Query format: tag=%04X nChannels=%d nSamplesPerSec=%d !\n",
lpDesc->lpFormat->wFormatTag, lpDesc->lpFormat->nChannels,
lpDesc->lpFormat->nSamplesPerSec);
return MMSYSERR_NOERROR;
}
wwi = &WInDev[wDevID];
if (wwi->pcm != NULL) {
WARN("already allocated\n");
return MMSYSERR_ALLOCATED;
}
wwi->pcm = 0;
flags = SND_PCM_NONBLOCK;
if ( (err=snd_pcm_open(&pcm, wwi->pcmname, SND_PCM_STREAM_CAPTURE, flags)) < 0 )
{
ERR("Error open: %s\n", snd_strerror(err));
return MMSYSERR_NOTENABLED;
}
wwi->wFlags = HIWORD(dwFlags & CALLBACK_TYPEMASK);
wwi->waveDesc = *lpDesc;
ALSA_copyFormat(lpDesc->lpFormat, &wwi->format);
if (wwi->format.Format.wBitsPerSample == 0) {
WARN("Resetting zeroed wBitsPerSample\n");
wwi->format.Format.wBitsPerSample = 8 *
(wwi->format.Format.nAvgBytesPerSec /
wwi->format.Format.nSamplesPerSec) /
wwi->format.Format.nChannels;
}
hw_params = HeapAlloc( GetProcessHeap(), HEAP_ZERO_MEMORY, snd_pcm_hw_params_sizeof() );
sw_params = HeapAlloc( GetProcessHeap(), HEAP_ZERO_MEMORY, snd_pcm_sw_params_sizeof() );
snd_pcm_hw_params_any(pcm, hw_params);
#define EXIT_ON_ERROR(f,e,txt) do \
{ \
int err; \
if ( (err = (f) ) < 0) \
{ \
WARN(txt ": %s\n", snd_strerror(err)); \
ret = (e); \
goto error; \
} \
} while(0)
access = SND_PCM_ACCESS_MMAP_INTERLEAVED;
if ( ( err = snd_pcm_hw_params_set_access(pcm, hw_params, access ) ) < 0) {
WARN("mmap not available. switching to standard write.\n");
access = SND_PCM_ACCESS_RW_INTERLEAVED;
EXIT_ON_ERROR( snd_pcm_hw_params_set_access(pcm, hw_params, access ), MMSYSERR_INVALPARAM, "unable to set access for playback");
wwi->read = snd_pcm_readi;
}
else
wwi->read = snd_pcm_mmap_readi;
EXIT_ON_ERROR( snd_pcm_hw_params_set_channels(pcm, hw_params, wwi->format.Format.nChannels), WAVERR_BADFORMAT, "unable to set required channels");
if ((wwi->format.Format.wFormatTag == WAVE_FORMAT_PCM) ||
((wwi->format.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) &&
IsEqualGUID(&wwi->format.SubFormat, &KSDATAFORMAT_SUBTYPE_PCM))) {
format = (wwi->format.Format.wBitsPerSample == 8) ? SND_PCM_FORMAT_U8 :
(wwi->format.Format.wBitsPerSample == 16) ? SND_PCM_FORMAT_S16_LE :
(wwi->format.Format.wBitsPerSample == 24) ? SND_PCM_FORMAT_S24_3LE :
(wwi->format.Format.wBitsPerSample == 32) ? SND_PCM_FORMAT_S32_LE : -1;
} else if ((wwi->format.Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) &&
IsEqualGUID(&wwi->format.SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)){
format = (wwi->format.Format.wBitsPerSample == 32) ? SND_PCM_FORMAT_FLOAT_LE : -1;
} else if (wwi->format.Format.wFormatTag == WAVE_FORMAT_MULAW) {
FIXME("unimplemented format: WAVE_FORMAT_MULAW\n");
ret = WAVERR_BADFORMAT;
goto error;
} else if (wwi->format.Format.wFormatTag == WAVE_FORMAT_ALAW) {
FIXME("unimplemented format: WAVE_FORMAT_ALAW\n");
ret = WAVERR_BADFORMAT;
goto error;
} else if (wwi->format.Format.wFormatTag == WAVE_FORMAT_ADPCM) {
FIXME("unimplemented format: WAVE_FORMAT_ADPCM\n");
ret = WAVERR_BADFORMAT;
goto error;
} else {
ERR("invalid format: %0x04x\n", wwi->format.Format.wFormatTag);
ret = WAVERR_BADFORMAT;
goto error;
}
EXIT_ON_ERROR( snd_pcm_hw_params_set_format(pcm, hw_params, format), WAVERR_BADFORMAT, "unable to set required format");
rate = wwi->format.Format.nSamplesPerSec;
dir = 0;
err = snd_pcm_hw_params_set_rate_near(pcm, hw_params, &rate, &dir);
if (err < 0) {
WARN("Rate %d Hz not available for playback: %s\n", wwi->format.Format.nSamplesPerSec, snd_strerror(rate));
ret = WAVERR_BADFORMAT;
goto error;
}
if (!ALSA_NearMatch(rate, wwi->format.Format.nSamplesPerSec)) {
WARN("Rate doesn't match (requested %d Hz, got %d Hz)\n", wwi->format.Format.nSamplesPerSec, rate);
ret = WAVERR_BADFORMAT;
goto error;
}
dir=0;
EXIT_ON_ERROR( snd_pcm_hw_params_set_buffer_time_near(pcm, hw_params, &buffer_time, &dir), MMSYSERR_INVALPARAM, "unable to set buffer time");
dir=0;
EXIT_ON_ERROR( snd_pcm_hw_params_set_period_time_near(pcm, hw_params, &period_time, &dir), MMSYSERR_INVALPARAM, "unable to set period time");
EXIT_ON_ERROR( snd_pcm_hw_params(pcm, hw_params), MMSYSERR_INVALPARAM, "unable to set hw params for playback");
dir=0;
err = snd_pcm_hw_params_get_period_size(hw_params, &period_size, &dir);
err = snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size);
snd_pcm_sw_params_current(pcm, sw_params);
EXIT_ON_ERROR( snd_pcm_sw_params_set_start_threshold(pcm, sw_params, 1), MMSYSERR_ERROR, "unable to set start threshold");
EXIT_ON_ERROR( snd_pcm_sw_params_set_silence_size(pcm, sw_params, 0), MMSYSERR_ERROR, "unable to set silence size");
EXIT_ON_ERROR( snd_pcm_sw_params_set_avail_min(pcm, sw_params, period_size), MMSYSERR_ERROR, "unable to set avail min");
EXIT_ON_ERROR( snd_pcm_sw_params_set_silence_threshold(pcm, sw_params, 0), MMSYSERR_ERROR, "unable to set silence threshold");
EXIT_ON_ERROR( snd_pcm_sw_params(pcm, sw_params), MMSYSERR_ERROR, "unable to set sw params for playback");
#undef EXIT_ON_ERROR
snd_pcm_prepare(pcm);
if (TRACE_ON(wave))
ALSA_TraceParameters(hw_params, sw_params, FALSE);
/* now, we can save all required data for later use... */
if ( wwi->hw_params )
snd_pcm_hw_params_free(wwi->hw_params);
snd_pcm_hw_params_malloc(&(wwi->hw_params));
snd_pcm_hw_params_copy(wwi->hw_params, hw_params);
wwi->dwBufferSize = snd_pcm_frames_to_bytes(pcm, buffer_size);
wwi->lpQueuePtr = wwi->lpPlayPtr = wwi->lpLoopPtr = NULL;
wwi->pcm = pcm;
ALSA_InitRingMessage(&wwi->msgRing);
wwi->dwPeriodSize = snd_pcm_frames_to_bytes(pcm, period_size);
TRACE("dwPeriodSize=%u\n", wwi->dwPeriodSize);
TRACE("wBitsPerSample=%u, nAvgBytesPerSec=%u, nSamplesPerSec=%u, nChannels=%u nBlockAlign=%u!\n",
wwi->format.Format.wBitsPerSample, wwi->format.Format.nAvgBytesPerSec,
wwi->format.Format.nSamplesPerSec, wwi->format.Format.nChannels,
wwi->format.Format.nBlockAlign);
wwi->hStartUpEvent = CreateEventW(NULL, FALSE, FALSE, NULL);
wwi->hThread = CreateThread(NULL, 0, widRecorder, (LPVOID)(DWORD_PTR)wDevID, 0, &(wwi->dwThreadID));
if (wwi->hThread)
SetThreadPriority(wwi->hThread, THREAD_PRIORITY_TIME_CRITICAL);
WaitForSingleObject(wwi->hStartUpEvent, INFINITE);
CloseHandle(wwi->hStartUpEvent);
wwi->hStartUpEvent = INVALID_HANDLE_VALUE;
HeapFree( GetProcessHeap(), 0, hw_params );
HeapFree( GetProcessHeap(), 0, sw_params );
return widNotifyClient(wwi, WIM_OPEN, 0L, 0L);
error:
snd_pcm_close(pcm);
HeapFree( GetProcessHeap(), 0, hw_params );
HeapFree( GetProcessHeap(), 0, sw_params );
return ret;
}
// alsa
main (int argc, char *argv[])
{
int i;
int err;
short buf[128];
snd_pcm_t *playback_handle;
snd_pcm_hw_params_t *hw_params;
if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
fprintf (stderr, "cannot open audio device %s (%s)\n",
argv[1],
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
fprintf (stderr, "cannot set sample format (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, 44100, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set parameters (%s)\n",
snd_strerror (err));
exit (1);
}
snd_pcm_hw_params_free (hw_params);
if ((err = snd_pcm_prepare (playback_handle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
exit (1);
}
for (i = 0; i < 10; ++i) {
if ((err = snd_pcm_writei (playback_handle, buf, 128)) != 128) {
fprintf (stderr, "write to audio interface failed (%s)\n",
snd_strerror (err));
exit (1);
}
}
snd_pcm_close (playback_handle);
exit (0);
}
int main(int argc, char *argv[]) {
const char *dev;
int r, cap, count = 0;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_status_t *status;
snd_pcm_t *pcm;
unsigned rate = SAMPLE_RATE;
unsigned periods = 2;
snd_pcm_uframes_t boundary, buffer_size = SAMPLE_RATE/10; /* 100s */
int dir = 1;
int fillrate;
struct timespec start, last_timestamp = { 0, 0 };
uint64_t start_us, last_us = 0;
snd_pcm_sframes_t last_avail = 0, last_delay = 0;
struct pollfd *pollfds;
int n_pollfd;
int64_t sample_count = 0;
uint16_t *samples;
struct sched_param sp;
r = -1;
#ifdef _POSIX_PRIORITY_SCHEDULING
sp.sched_priority = 5;
r = pthread_setschedparam(pthread_self(), SCHED_RR, &sp);
#endif
if (r)
printf("Could not get RT prio. :(\n");
snd_pcm_hw_params_alloca(&hwparams);
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_status_alloca(&status);
r = clock_gettime(CLOCK_MONOTONIC, &start);
assert(r == 0);
start_us = timespec_us(&start);
dev = argc > 1 ? argv[1] : "front:0";
cap = argc > 2 ? atoi(argv[2]) : 0;
fillrate = argc > 3 ? atoi(argv[3]) : 1;
samples = calloc(fillrate, CHANNELS*sizeof(uint16_t));
assert(samples);
if (cap == 0)
r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_PLAYBACK, 0);
else
r = snd_pcm_open(&pcm, dev, SND_PCM_STREAM_CAPTURE, 0);
assert(r == 0);
r = snd_pcm_hw_params_any(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 0);
assert(r == 0);
r = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
assert(r == 0);
r = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE);
assert(r == 0);
r = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, NULL);
assert(r == 0);
r = snd_pcm_hw_params_set_channels(pcm, hwparams, CHANNELS);
assert(r == 0);
r = snd_pcm_hw_params_set_periods_integer(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_set_periods_near(pcm, hwparams, &periods, &dir);
assert(r == 0);
r = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, &buffer_size);
assert(r == 0);
r = snd_pcm_hw_params(pcm, hwparams);
assert(r == 0);
r = snd_pcm_hw_params_current(pcm, hwparams);
assert(r == 0);
r = snd_pcm_sw_params_current(pcm, swparams);
assert(r == 0);
if (cap == 0)
r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 1);
else
r = snd_pcm_sw_params_set_avail_min(pcm, swparams, 0);
assert(r == 0);
r = snd_pcm_sw_params_set_period_event(pcm, swparams, 0);
assert(r == 0);
r = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
assert(r == 0);
r = snd_pcm_sw_params_set_start_threshold(pcm, swparams, buffer_size - (buffer_size % fillrate));
assert(r == 0);
r = snd_pcm_sw_params_get_boundary(swparams, &boundary);
assert(r == 0);
r = snd_pcm_sw_params_set_stop_threshold(pcm, swparams, boundary);
assert(r == 0);
r = snd_pcm_sw_params_set_tstamp_mode(pcm, swparams, SND_PCM_TSTAMP_ENABLE);
assert(r == 0);
r = snd_pcm_sw_params(pcm, swparams);
assert(r == 0);
r = snd_pcm_prepare(pcm);
assert(r == 0);
r = snd_pcm_sw_params_current(pcm, swparams);
assert(r == 0);
/* assert(snd_pcm_hw_params_is_monotonic(hwparams) > 0); */
n_pollfd = snd_pcm_poll_descriptors_count(pcm);
assert(n_pollfd > 0);
pollfds = malloc(sizeof(struct pollfd) * n_pollfd);
assert(pollfds);
r = snd_pcm_poll_descriptors(pcm, pollfds, n_pollfd);
assert(r == n_pollfd);
printf("Starting. Buffer size is %u frames\n", (unsigned int) buffer_size);
if (cap) {
r = snd_pcm_start(pcm);
assert(r == 0);
}
for (;;) {
snd_pcm_sframes_t avail, delay;
struct timespec now, timestamp;
unsigned short revents;
int handled = 0;
uint64_t now_us, timestamp_us;
snd_pcm_state_t state;
unsigned long long pos;
r = poll(pollfds, n_pollfd, 0);
assert(r >= 0);
r = snd_pcm_poll_descriptors_revents(pcm, pollfds, n_pollfd, &revents);
assert(r == 0);
if (cap == 0)
assert((revents & ~POLLOUT) == 0);
else
assert((revents & ~POLLIN) == 0);
avail = snd_pcm_avail(pcm);
assert(avail >= 0);
r = snd_pcm_status(pcm, status);
assert(r == 0);
/* This assertion fails from time to time. ALSA seems to be broken */
/* assert(avail == (snd_pcm_sframes_t) snd_pcm_status_get_avail(status)); */
/* printf("%lu %lu\n", (unsigned long) avail, (unsigned long) snd_pcm_status_get_avail(status)); */
snd_pcm_status_get_htstamp(status, ×tamp);
delay = snd_pcm_status_get_delay(status);
state = snd_pcm_status_get_state(status);
r = clock_gettime(CLOCK_MONOTONIC, &now);
assert(r == 0);
assert(!revents || avail > 0);
if ((!cap && (avail >= fillrate)) || (cap && (unsigned)avail >= buffer_size)) {
snd_pcm_sframes_t sframes;
if (cap == 0)
sframes = snd_pcm_writei(pcm, samples, fillrate);
else
sframes = snd_pcm_readi(pcm, samples, fillrate);
assert(sframes == fillrate);
handled = fillrate;
sample_count += fillrate;
}
if (!handled &&
memcmp(×tamp, &last_timestamp, sizeof(timestamp)) == 0 &&
avail == last_avail &&
delay == last_delay) {
/* This is boring */
continue;
}
now_us = timespec_us(&now);
timestamp_us = timespec_us(×tamp);
if (cap == 0)
pos = (unsigned long long) ((sample_count - handled - delay) * 1000000LU / SAMPLE_RATE);
else
pos = (unsigned long long) ((sample_count - handled + delay) * 1000000LU / SAMPLE_RATE);
if (count++ % 50 == 0)
printf("Elapsed\tCPU\tALSA\tPos\tSamples\tavail\tdelay\trevents\thandled\tstate\n");
printf("%llu\t%llu\t%llu\t%llu\t%llu\t%li\t%li\t%i\t%i\t%i\n",
(unsigned long long) (now_us - last_us),
(unsigned long long) (now_us - start_us),
(unsigned long long) (timestamp_us ? timestamp_us - start_us : 0),
pos,
(unsigned long long) sample_count,
(signed long) avail,
(signed long) delay,
revents,
handled,
state);
if (cap == 0)
/** When this assert is hit, most likely something bad
* happened, i.e. the avail jumped suddenly. */
assert((unsigned) avail <= buffer_size);
last_avail = avail;
last_delay = delay;
last_timestamp = timestamp;
last_us = now_us;
}
return 0;
}
bool QAudioOutputPrivate::open()
{
if(opened)
return true;
#ifdef DEBUG_AUDIO
QTime now(QTime::currentTime());
qDebug()<<now.second()<<"s "<<now.msec()<<"ms :open()";
#endif
timeStamp.restart();
elapsedTimeOffset = 0;
int dir;
int err=-1;
int count=0;
unsigned int freakuency=settings.frequency();
QString dev = QString(QLatin1String(m_device.constData()));
QList<QByteArray> devices = QAudioDeviceInfoInternal::availableDevices(QAudio::AudioOutput);
if(dev.compare(QLatin1String("default")) == 0) {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = QLatin1String(devices.first());
#else
dev = QLatin1String("hw:0,0");
#endif
} else {
#if(SND_LIB_MAJOR == 1 && SND_LIB_MINOR == 0 && SND_LIB_SUBMINOR >= 14)
dev = QLatin1String(m_device);
#else
int idx = 0;
char *name;
QString shortName = QLatin1String(m_device.mid(m_device.indexOf('=',0)+1).constData());
while(snd_card_get_name(idx,&name) == 0) {
if(qstrncmp(shortName.toLocal8Bit().constData(),name,shortName.length()) == 0)
break;
idx++;
}
dev = QString(QLatin1String("hw:%1,0")).arg(idx);
#endif
}
// Step 1: try and open the device
while((count < 5) && (err < 0)) {
err=snd_pcm_open(&handle,dev.toLocal8Bit().constData(),SND_PCM_STREAM_PLAYBACK,0);
if(err < 0)
count++;
}
if (( err < 0)||(handle == 0)) {
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
return false;
}
snd_pcm_nonblock( handle, 0 );
// Step 2: Set the desired HW parameters.
snd_pcm_hw_params_alloca( &hwparams );
bool fatal = false;
QString errMessage;
unsigned int chunks = 8;
err = snd_pcm_hw_params_any( handle, hwparams );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_any: err = %1").arg(err);
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_resample( handle, hwparams, 1 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_rate_resample: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_access( handle, hwparams, access );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_access: err = %1").arg(err);
}
}
if ( !fatal ) {
err = setFormat();
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_format: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_channels( handle, hwparams, (unsigned int)settings.channels() );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_channels: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_rate_near( handle, hwparams, &freakuency, 0 );
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_rate_near: err = %1").arg(err);
}
}
if ( !fatal ) {
unsigned int maxBufferTime = 0;
unsigned int minBufferTime = 0;
unsigned int maxPeriodTime = 0;
unsigned int minPeriodTime = 0;
err = snd_pcm_hw_params_get_buffer_time_max(hwparams, &maxBufferTime, &dir);
if ( err >= 0)
err = snd_pcm_hw_params_get_buffer_time_min(hwparams, &minBufferTime, &dir);
if ( err >= 0)
err = snd_pcm_hw_params_get_period_time_max(hwparams, &maxPeriodTime, &dir);
if ( err >= 0)
err = snd_pcm_hw_params_get_period_time_min(hwparams, &minPeriodTime, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: buffer/period min and max: err = %1").arg(err);
} else {
if (maxBufferTime < buffer_time || buffer_time < minBufferTime || maxPeriodTime < period_time || minPeriodTime > period_time) {
#ifdef DEBUG_AUDIO
qDebug()<<"defaults out of range";
qDebug()<<"pmin="<<minPeriodTime<<", pmax="<<maxPeriodTime<<", bmin="<<minBufferTime<<", bmax="<<maxBufferTime;
#endif
period_time = minPeriodTime;
if (period_time*4 <= maxBufferTime) {
// Use 4 periods if possible
buffer_time = period_time*4;
chunks = 4;
} else if (period_time*2 <= maxBufferTime) {
// Use 2 periods if possible
buffer_time = period_time*2;
chunks = 2;
} else {
qWarning()<<"QAudioOutput: alsa only supports single period!";
fatal = true;
}
#ifdef DEBUG_AUDIO
qDebug()<<"used: buffer_time="<<buffer_time<<", period_time="<<period_time;
#endif
}
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_buffer_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_period_time_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params_set_periods_near(handle, hwparams, &chunks, &dir);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params_set_periods_near: err = %1").arg(err);
}
}
if ( !fatal ) {
err = snd_pcm_hw_params(handle, hwparams);
if ( err < 0 ) {
fatal = true;
errMessage = QString::fromLatin1("QAudioOutput: snd_pcm_hw_params: err = %1").arg(err);
}
}
if( err < 0) {
qWarning()<<errMessage;
errorState = QAudio::OpenError;
deviceState = QAudio::StoppedState;
return false;
}
snd_pcm_hw_params_get_buffer_size(hwparams,&buffer_frames);
buffer_size = snd_pcm_frames_to_bytes(handle,buffer_frames);
snd_pcm_hw_params_get_period_size(hwparams,&period_frames, &dir);
period_size = snd_pcm_frames_to_bytes(handle,period_frames);
snd_pcm_hw_params_get_buffer_time(hwparams,&buffer_time, &dir);
snd_pcm_hw_params_get_period_time(hwparams,&period_time, &dir);
// Step 3: Set the desired SW parameters.
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(handle, swparams);
snd_pcm_sw_params_set_start_threshold(handle,swparams,period_frames);
snd_pcm_sw_params_set_stop_threshold(handle,swparams,buffer_frames);
snd_pcm_sw_params_set_avail_min(handle, swparams,period_frames);
snd_pcm_sw_params(handle, swparams);
// Step 4: Prepare audio
if(audioBuffer == 0)
audioBuffer = new char[snd_pcm_frames_to_bytes(handle,buffer_frames)];
snd_pcm_prepare( handle );
snd_pcm_start(handle);
// Step 5: Setup callback and timer fallback
snd_async_add_pcm_handler(&ahandler, handle, async_callback, this);
bytesAvailable = bytesFree();
// Step 6: Start audio processing
timer->start(period_time/1000);
clockStamp.restart();
timeStamp.restart();
elapsedTimeOffset = 0;
errorState = QAudio::NoError;
totalTimeValue = 0;
opened = true;
return true;
}
static void
alsa_configure (struct sound_device *sd)
{
int val, err, dir;
unsigned uval;
struct alsa_params *p = (struct alsa_params *) sd->data;
snd_pcm_uframes_t buffer_size;
eassert (p->handle != 0);
err = snd_pcm_hw_params_malloc (&p->hwparams);
if (err < 0)
alsa_sound_perror ("Could not allocate hardware parameter structure", err);
err = snd_pcm_sw_params_malloc (&p->swparams);
if (err < 0)
alsa_sound_perror ("Could not allocate software parameter structure", err);
err = snd_pcm_hw_params_any (p->handle, p->hwparams);
if (err < 0)
alsa_sound_perror ("Could not initialize hardware parameter structure", err);
err = snd_pcm_hw_params_set_access (p->handle, p->hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
alsa_sound_perror ("Could not set access type", err);
val = sd->format;
err = snd_pcm_hw_params_set_format (p->handle, p->hwparams, val);
if (err < 0)
alsa_sound_perror ("Could not set sound format", err);
uval = sd->sample_rate;
err = snd_pcm_hw_params_set_rate_near (p->handle, p->hwparams, &uval, 0);
if (err < 0)
alsa_sound_perror ("Could not set sample rate", err);
val = sd->channels;
err = snd_pcm_hw_params_set_channels (p->handle, p->hwparams, val);
if (err < 0)
alsa_sound_perror ("Could not set channel count", err);
err = snd_pcm_hw_params (p->handle, p->hwparams);
if (err < 0)
alsa_sound_perror ("Could not set parameters", err);
err = snd_pcm_hw_params_get_period_size (p->hwparams, &p->period_size, &dir);
if (err < 0)
alsa_sound_perror ("Unable to get period size for playback", err);
err = snd_pcm_hw_params_get_buffer_size (p->hwparams, &buffer_size);
if (err < 0)
alsa_sound_perror ("Unable to get buffer size for playback", err);
err = snd_pcm_sw_params_current (p->handle, p->swparams);
if (err < 0)
alsa_sound_perror ("Unable to determine current swparams for playback",
err);
/* Start the transfer when the buffer is almost full */
err = snd_pcm_sw_params_set_start_threshold (p->handle, p->swparams,
(buffer_size / p->period_size)
* p->period_size);
if (err < 0)
alsa_sound_perror ("Unable to set start threshold mode for playback", err);
/* Allow the transfer when at least period_size samples can be processed */
err = snd_pcm_sw_params_set_avail_min (p->handle, p->swparams, p->period_size);
if (err < 0)
alsa_sound_perror ("Unable to set avail min for playback", err);
err = snd_pcm_sw_params (p->handle, p->swparams);
if (err < 0)
alsa_sound_perror ("Unable to set sw params for playback\n", err);
snd_pcm_hw_params_free (p->hwparams);
p->hwparams = NULL;
snd_pcm_sw_params_free (p->swparams);
p->swparams = NULL;
err = snd_pcm_prepare (p->handle);
if (err < 0)
alsa_sound_perror ("Could not prepare audio interface for use", err);
if (sd->volume > 0)
{
int chn;
snd_mixer_t *handle;
snd_mixer_elem_t *e;
if (snd_mixer_open (&handle, 0) >= 0)
{
char const *file = string_default (sd->file,
DEFAULT_ALSA_SOUND_DEVICE);
if (snd_mixer_attach (handle, file) >= 0
&& snd_mixer_load (handle) >= 0
&& snd_mixer_selem_register (handle, NULL, NULL) >= 0)
for (e = snd_mixer_first_elem (handle);
e;
e = snd_mixer_elem_next (e))
{
if (snd_mixer_selem_has_playback_volume (e))
{
long pmin, pmax, vol;
snd_mixer_selem_get_playback_volume_range (e, &pmin, &pmax);
vol = pmin + (sd->volume * (pmax - pmin)) / 100;
for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++)
snd_mixer_selem_set_playback_volume (e, chn, vol);
}
}
snd_mixer_close (handle);
}
}
}
static void* sound_thread(void* context)
{
struct SPU* c = (struct SPU*)context;
int div;
snd_pcm_t* snd;
snd_pcm_hw_params_t* hwp;
int rate = c->sampling;
int periods = 3;
snd_pcm_hw_params_alloca(&hwp);
if (snd_pcm_open(&snd, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0) {
return NULL;
}
if (snd_pcm_hw_params_any(snd, hwp) < 0) {
return NULL;
}
div = c->bit / 8;
switch (c->bit) {
case 8:
if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_U8) < 0) {
return NULL;
}
break;
case 16:
if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_S16_LE) < 0) {
return NULL;
}
break;
case 24:
if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_S24_LE) < 0) {
return NULL;
}
break;
case 32:
if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_S32_LE) < 0) {
return NULL;
}
break;
default:
return NULL;
}
if (snd_pcm_hw_params_set_rate_near(snd, hwp, &rate, 0) < 0) {
return NULL;
}
if (rate != SAMPLE_RATE) {
return NULL;
}
if (snd_pcm_hw_params_set_channels(snd, hwp, c->ch) < 0) {
return NULL;
}
if (snd_pcm_hw_params_set_periods(snd, hwp, periods, 0) < 0) {
return NULL;
}
if (snd_pcm_hw_params_set_buffer_size(snd, hwp, (periods * c->size) / 4) < 0) {
return NULL;
}
if (snd_pcm_hw_params(snd, hwp) < 0) {
return NULL;
}
while (c->alive) {
c->callback(c->buffer, c->size);
while (c->alive) {
if (snd_pcm_writei(snd, c->buffer, c->size / div) < 0) {
snd_pcm_prepare(snd);
} else {
break;
}
}
}
snd_pcm_drain(snd);
snd_pcm_close(snd);
return NULL;
}
static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate,
int nfrags, int frag_size)
{
int bufsizeforthis, err;
snd_pcm_hw_params_t* hw_params;
unsigned int tmp_uint;
snd_pcm_uframes_t tmp_snd_pcm_uframes;
if (sys_verbose)
{
if (out)
post("configuring sound output...");
else post("configuring sound input...");
}
/* set hardware parameters... */
snd_pcm_hw_params_alloca(&hw_params);
/* get the default params */
err = snd_pcm_hw_params_any(dev->a_handle, hw_params);
check_error(err, "snd_pcm_hw_params_any");
/* try to set interleaved access */
err = snd_pcm_hw_params_set_access(dev->a_handle,
hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
return (-1);
check_error(err, "snd_pcm_hw_params_set_access");
#if 0 /* enable this to print out which formats are available */
{
int i;
for (i = 0; i <= SND_PCM_FORMAT_LAST; i++)
fprintf(stderr, "%d -> %d\n",
i, snd_pcm_hw_params_test_format(dev->a_handle, hw_params, i));
}
#endif
/* Try to set 32 bit format first */
err = snd_pcm_hw_params_set_format(dev->a_handle,
hw_params, SND_PCM_FORMAT_S32);
if (err < 0)
{
/* fprintf(stderr,
"PD-ALSA: 32 bit format not available - trying 24\n"); */
err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
SND_PCM_FORMAT_S24_3LE);
if (err < 0)
{
/* fprintf(stderr,
"PD-ALSA: 32/24 bit format not available - using 16\n"); */
err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,
SND_PCM_FORMAT_S16);
check_error(err, "snd_pcm_hw_params_set_format");
dev->a_sampwidth = 2;
}
else dev->a_sampwidth = 3;
}
else dev->a_sampwidth = 4;
if (sys_verbose)
post("Sample width set to %d bytes", dev->a_sampwidth);
/* set the subformat */
err = snd_pcm_hw_params_set_subformat(dev->a_handle,
hw_params, SND_PCM_SUBFORMAT_STD);
check_error(err, "snd_pcm_hw_params_set_subformat");
/* set the number of channels */
tmp_uint = *channels;
err = snd_pcm_hw_params_set_channels_min(dev->a_handle,
hw_params, &tmp_uint);
check_error(err, "snd_pcm_hw_params_set_channels");
if (tmp_uint != (unsigned)*channels)
post("ALSA: set %s channels to %d", (out?"output":"input"), tmp_uint);
*channels = tmp_uint;
dev->a_channels = *channels;
/* set the sampling rate */
err = snd_pcm_hw_params_set_rate_min(dev->a_handle, hw_params,
(unsigned int *)rate, 0);
check_error(err, "snd_pcm_hw_params_set_rate_min (input)");
#if 0
err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir);
post("input sample rate %d", err);
#endif
/* post("frag size %d, nfrags %d", frag_size, nfrags); */
/* set "period size" */
#ifdef ALSAAPI9
err = snd_pcm_hw_params_set_period_size_near(dev->a_handle,
hw_params, (snd_pcm_uframes_t)frag_size, 0);
#else
tmp_snd_pcm_uframes = frag_size;
err = snd_pcm_hw_params_set_period_size_near(dev->a_handle,
hw_params, &tmp_snd_pcm_uframes, 0);
#endif
check_error(err, "snd_pcm_hw_params_set_period_size_near (input)");
/* set the buffer size */
#ifdef ALSAAPI9
err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle,
hw_params, nfrags * frag_size);
#else
tmp_snd_pcm_uframes = nfrags * frag_size;
err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle,
hw_params, &tmp_snd_pcm_uframes);
#endif
check_error(err, "snd_pcm_hw_params_set_buffer_size_near (input)");
err = snd_pcm_hw_params(dev->a_handle, hw_params);
check_error(err, "snd_pcm_hw_params (input)");
/* set up the buffer */
bufsizeforthis = DEFDACBLKSIZE * dev->a_sampwidth * *channels;
if (alsa_snd_buf)
{
if (alsa_snd_bufsize < bufsizeforthis)
{
if (!(alsa_snd_buf = realloc(alsa_snd_buf, bufsizeforthis)))
{
post("out of memory");
return (0);
}
memset(alsa_snd_buf, 0, bufsizeforthis);
alsa_snd_bufsize = bufsizeforthis;
}
}
else
{
if (!(alsa_snd_buf = (void *)malloc(bufsizeforthis)))
{
post("out of memory");
return (0);
}
memset(alsa_snd_buf, 0, bufsizeforthis);
alsa_snd_bufsize = bufsizeforthis;
}
return (1);
}
int sndout_alsa_start(int rate_, int stereo)
{
snd_pcm_hw_params_t *hwparams = NULL;
unsigned int rate = rate_;
int samples, shift;
int ret;
samples = rate * 40 / 1000;
for (shift = 8; (1 << shift) < samples; shift++)
;
period_size = 1 << shift;
buffer_size = 8 * period_size;
snd_pcm_hw_params_alloca(&hwparams);
ret = snd_pcm_hw_params_any(handle, hwparams);
ret |= snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
ret |= snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
ret |= snd_pcm_hw_params_set_channels(handle, hwparams, stereo ? 2 : 1);
ret |= snd_pcm_hw_params_set_rate_near(handle, hwparams, &rate, 0);
ret |= snd_pcm_hw_params_set_buffer_size_near(handle, hwparams, &buffer_size);
ret |= snd_pcm_hw_params_set_period_size_near(handle, hwparams, &period_size, NULL);
if (ret != 0) {
fprintf(stderr, PFX "failed to set hwparams\n");
goto fail;
}
ret = snd_pcm_hw_params(handle, hwparams);
if (ret != 0) {
fprintf(stderr, PFX "failed to apply hwparams: %d\n", ret);
goto fail;
}
snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
snd_pcm_hw_params_get_period_size(hwparams, &period_size, NULL);
snd_pcm_hw_params_get_channels(hwparams, &channels);
silent_period = realloc(silent_period, period_size * 2 * channels);
if (silent_period != NULL)
memset(silent_period, 0, period_size * 2 * channels);
ret = snd_pcm_prepare(handle);
if (ret != 0) {
fprintf(stderr, PFX "snd_pcm_prepare failed: %d\n", ret);
goto fail;
}
ret = snd_pcm_start(handle);
if (ret != 0) {
fprintf(stderr, PFX "snd_pcm_start failed: %d\n", ret);
goto fail;
}
failure_counter = 0;
return 0;
fail:
// to flush out redirected logs
fflush(stdout);
fflush(stderr);
return -1;
}
bool CAESinkALSA::InitializeHW(AEAudioFormat &format)
{
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_alloca(&hw_params);
memset(hw_params, 0, snd_pcm_hw_params_sizeof());
snd_pcm_hw_params_any(m_pcm, hw_params);
snd_pcm_hw_params_set_access(m_pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
unsigned int sampleRate = format.m_sampleRate;
unsigned int channelCount = format.m_channelLayout.Count();
snd_pcm_hw_params_set_rate_near (m_pcm, hw_params, &sampleRate, NULL);
snd_pcm_hw_params_set_channels_near(m_pcm, hw_params, &channelCount);
/* ensure we opened X channels or more */
if (format.m_channelLayout.Count() > channelCount)
{
CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Unable to open the required number of channels");
return false;
}
/* update the channelLayout to what we managed to open */
format.m_channelLayout.Reset();
for (unsigned int i = 0; i < channelCount; ++i)
format.m_channelLayout += ALSAChannelMap[i];
snd_pcm_format_t fmt = AEFormatToALSAFormat(format.m_dataFormat);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
{
/* if we dont support the requested format, fallback to float */
format.m_dataFormat = AE_FMT_FLOAT;
fmt = SND_PCM_FORMAT_FLOAT;
}
/* try the data format */
if (snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
{
/* if the chosen format is not supported, try each one in decending order */
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Your hardware does not support %s, trying other formats", CAEUtil::DataFormatToStr(format.m_dataFormat));
for (enum AEDataFormat i = AE_FMT_MAX; i > AE_FMT_INVALID; i = (enum AEDataFormat)((int)i - 1))
{
if (AE_IS_RAW(i) || i == AE_FMT_MAX)
continue;
if (m_passthrough && i != AE_FMT_S16BE && i != AE_FMT_S16LE)
continue;
fmt = AEFormatToALSAFormat(i);
if (fmt == SND_PCM_FORMAT_UNKNOWN || snd_pcm_hw_params_set_format(m_pcm, hw_params, fmt) < 0)
{
fmt = SND_PCM_FORMAT_UNKNOWN;
continue;
}
int fmtBits = CAEUtil::DataFormatToBits(i);
int bits = snd_pcm_hw_params_get_sbits(hw_params);
if (bits != fmtBits)
{
/* if we opened in 32bit and only have 24bits, pack into 24 */
if (fmtBits == 32 && bits == 24)
i = AE_FMT_S24NE4;
else
continue;
}
/* record that the format fell back to X */
format.m_dataFormat = i;
CLog::Log(LOGINFO, "CAESinkALSA::InitializeHW - Using data format %s", CAEUtil::DataFormatToStr(format.m_dataFormat));
break;
}
/* if we failed to find a valid output format */
if (fmt == SND_PCM_FORMAT_UNKNOWN)
{
CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Unable to find a suitable output format");
return false;
}
}
unsigned int periods;
snd_pcm_uframes_t periodSize, bufferSize;
snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufferSize);
bufferSize = std::min(bufferSize, (snd_pcm_uframes_t)8192);
periodSize = bufferSize / ALSA_PERIODS;
periods = ALSA_PERIODS;
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Request: periodSize %lu, periods %u, bufferSize %lu", periodSize, periods, bufferSize);
/* work on a copy of the hw params */
snd_pcm_hw_params_t *hw_params_copy;
snd_pcm_hw_params_alloca(&hw_params_copy);
/* try to set the buffer size then the period size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
/* try to set the period size then the buffer size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
/* try to just set the buffer size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_buffer_size_near(m_pcm, hw_params_copy, &bufferSize);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
/* try to just set the period size */
snd_pcm_hw_params_copy(hw_params_copy, hw_params);
snd_pcm_hw_params_set_period_size_near(m_pcm, hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_set_periods_near (m_pcm, hw_params_copy, &periods , NULL);
if (snd_pcm_hw_params(m_pcm, hw_params_copy) != 0)
{
CLog::Log(LOGERROR, "CAESinkALSA::InitializeHW - Failed to set the parameters");
return false;
}
}
}
}
snd_pcm_hw_params_get_period_size(hw_params_copy, &periodSize, NULL);
snd_pcm_hw_params_get_buffer_size(hw_params_copy, &bufferSize);
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Got: periodSize %lu, periods %u, bufferSize %lu", periodSize, periods, bufferSize);
/* set the format parameters */
format.m_sampleRate = sampleRate;
format.m_frames = periodSize;
format.m_frameSamples = periodSize * format.m_channelLayout.Count();
format.m_frameSize = snd_pcm_frames_to_bytes(m_pcm, 1);
m_bufferSize = (unsigned int)bufferSize;
m_timeout = std::ceil((double)(bufferSize * 1000) / (double)sampleRate);
CLog::Log(LOGDEBUG, "CAESinkALSA::InitializeHW - Setting timeout to %d ms", m_timeout);
return true;
}
static RD_BOOL
alsa_set_format(snd_pcm_t * pcm, RD_WAVEFORMATEX * pwfx)
{
snd_pcm_hw_params_t *hwparams = NULL;
int err;
unsigned int buffertime;
short samplewidth;
int audiochannels;
unsigned int rate;
samplewidth = pwfx->wBitsPerSample / 8;
if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
{
error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
return False;
}
if ((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
{
error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
return False;
}
if ((err = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
return False;
}
if (pwfx->wBitsPerSample == 16)
{
if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
{
error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
return False;
}
}
else
{
if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S8)) < 0)
{
error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
return False;
}
}
#if 0
if ((err = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 1)) < 0)
{
error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
return False;
}
#endif
rate = pwfx->nSamplesPerSec;
if ((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
{
error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
return False;
}
audiochannels = pwfx->nChannels;
if ((err = snd_pcm_hw_params_set_channels(pcm, hwparams, pwfx->nChannels)) < 0)
{
error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
return False;
}
buffertime = 500000; /* microseconds */
if ((err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams, &buffertime, 0)) < 0)
{
error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
return False;
}
if ((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
{
error("snd_pcm_hw_params: %s\n", snd_strerror(err));
return False;
}
snd_pcm_hw_params_free(hwparams);
if ((err = snd_pcm_prepare(pcm)) < 0)
{
error("snd_pcm_prepare: %s\n", snd_strerror(err));
return False;
}
reopened = True;
return True;
}
void AudioCapture::run()
{
qDebug() << "AudioCapture::run()";
snd_pcm_t * captureHandle;
snd_pcm_hw_params_t * params;
snd_pcm_hw_params_alloca(¶ms);
if (snd_pcm_open(&captureHandle, "default", SND_PCM_STREAM_CAPTURE, 0) < 0)
{
qDebug() << "error opening audio device";
return;
}
if (snd_pcm_hw_params_any(captureHandle, params) < 0)
{
qDebug() << "cannot initialize hardware paramerter structure";
return;
}
if (snd_pcm_hw_params_set_access(captureHandle, params, SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
{
qDebug() << "cannot set access type";
return;
}
if (snd_pcm_hw_params_set_format(captureHandle, params, SND_PCM_FORMAT_S16_LE) < 0)
{
qDebug() << "cannot set sample format";
return;
}
unsigned int rate = 48000;
if (snd_pcm_hw_params_set_rate_near(captureHandle, params, &rate, 0) < 0)
{
qDebug() << "cannot set sample rate";
return;
}
qDebug() << rate;
if (snd_pcm_hw_params_set_channels(captureHandle, params, 2) < 0)
{
qDebug() << "cannot set channels to 2";
return;
}
/*int periods = 2;
snd_pcm_uframes_t periodsize = 8192;
qDebug() << "settings periods";
if (snd_pcm_hw_params_set_periods(captureHandle, params, periods, 0) < 0)
{
qDebug() << "Error setting periods.";
return;
}
qDebug() << "setting buffer size to " << ((periodsize * periods)>>2);
snd_pcm_hw_params_set_buffer_size(captureHandle, params, (periodsize * periods)>>2);
qDebug() << "settings params";*/
if (snd_pcm_hw_params(captureHandle, params) < 0)
{
qDebug() << "cannot set parameters";
return;
}
//snd_pcm_hw_params_free(params);
//qDebug() <<
if (snd_pcm_prepare(captureHandle) < 0)
{
qDebug() << "cannot prepare audio interface";
}
QFile file(captureFile);
//QFile file("test.pcm");
qDebug() << file.open(QIODevice::WriteOnly);
qDebug() << "++starting loop";
int frames = 1024;
short buffer[frames * 2];
while (capturing)
{
qDebug() << "about to read samples";
int samplesRead = snd_pcm_readi(captureHandle, buffer, frames);
qDebug() << "read " << samplesRead << "samples";
if (samplesRead != frames)
{
qDebug() << "read failed";
return;
}
qDebug() << buffer;
qDebug() << file.write((const char *)buffer, frames * 2 * sizeof(short));
//file.write(QByteArray("test"));
qDebug() << "wrote samples";
}
snd_pcm_close(captureHandle);
file.close();
}
int AudioAlsa::setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access )
{
int err, dir;
// choose all parameters
if( ( err = snd_pcm_hw_params_any( m_handle, m_hwParams ) ) < 0 )
{
printf( "Broken configuration for playback: no configurations "
"available: %s\n", snd_strerror( err ) );
return err;
}
// set the interleaved read/write format
if( ( err = snd_pcm_hw_params_set_access( m_handle, m_hwParams,
_access ) ) < 0 )
{
printf( "Access type not available for playback: %s\n",
snd_strerror( err ) );
return err;
}
// set the sample format
if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
SND_PCM_FORMAT_S16_LE ) ) < 0 )
{
if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
SND_PCM_FORMAT_S16_BE ) ) < 0 )
{
printf( "Neither little- nor big-endian available for "
"playback: %s\n", snd_strerror( err ) );
return err;
}
m_convertEndian = isLittleEndian();
}
else
{
m_convertEndian = !isLittleEndian();
}
// set the count of channels
if( ( err = snd_pcm_hw_params_set_channels( m_handle, m_hwParams,
_channels ) ) < 0 )
{
printf( "Channel count (%i) not available for playbacks: %s\n"
"(Does your soundcard not support surround?)\n",
_channels, snd_strerror( err ) );
return err;
}
// set the sample rate
if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
sampleRate(), 0 ) ) < 0 )
{
if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
mixer()->baseSampleRate(), 0 ) ) < 0 )
{
printf( "Could not set sample rate: %s\n",
snd_strerror( err ) );
return err;
}
}
m_periodSize = mixer()->framesPerPeriod();
m_bufferSize = m_periodSize * 8;
dir = 0;
err = snd_pcm_hw_params_set_period_size_near( m_handle, m_hwParams,
&m_periodSize, &dir );
if( err < 0 )
{
printf( "Unable to set period size %lu for playback: %s\n",
m_periodSize, snd_strerror( err ) );
return err;
}
dir = 0;
err = snd_pcm_hw_params_get_period_size( m_hwParams, &m_periodSize,
&dir );
if( err < 0 )
{
printf( "Unable to get period size for playback: %s\n",
snd_strerror( err ) );
}
dir = 0;
err = snd_pcm_hw_params_set_buffer_size_near( m_handle, m_hwParams,
&m_bufferSize );
if( err < 0 )
{
printf( "Unable to set buffer size %lu for playback: %s\n",
m_bufferSize, snd_strerror( err ) );
return ( err );
}
err = snd_pcm_hw_params_get_buffer_size( m_hwParams, &m_bufferSize );
if( 2 * m_periodSize > m_bufferSize )
{
printf( "buffer to small, could not use\n" );
return ( err );
}
// write the parameters to device
err = snd_pcm_hw_params( m_handle, m_hwParams );
if( err < 0 )
{
printf( "Unable to set hw params for playback: %s\n",
snd_strerror( err ) );
return ( err );
}
return ( 0 ); // all ok
}