AudioOutputSettings* AudioOutputNULL::GetOutputSettings(bool /*digital*/)
{
// Pretend that we support everything
AudioFormat fmt;
int rate;
AudioOutputSettings *settings = new AudioOutputSettings();
while ((rate = settings->GetNextRate()))
{
settings->AddSupportedRate(rate);
}
while ((fmt = settings->GetNextFormat()))
{
settings->AddSupportedFormat(fmt);
}
for (uint channels = CHANNELS_MIN; channels <= CHANNELS_MAX; channels++)
{
settings->AddSupportedChannels(channels);
}
settings->setPassthrough(-1); // no passthrough
return settings;
}
AudioOutputSettings* AudioOutputWin::GetOutputSettings(bool /*digital*/)
{
AudioOutputSettings *settings = new AudioOutputSettings();
// We use WAVE_MAPPER to find a compatible device, so just claim support
// for all of the standard rates
while (DWORD rate = (DWORD)settings->GetNextRate())
settings->AddSupportedRate(rate);
// Support all standard formats
settings->AddSupportedFormat(FORMAT_U8);
settings->AddSupportedFormat(FORMAT_S16);
#if 0 // 24-bit integer is not supported
settings->AddSupportedFormat(FORMAT_S24);
#endif
#if 0 // 32-bit integer (OGG) is not supported on all platforms.
settings->AddSupportedFormat(FORMAT_S32);
#endif
#if 0 // 32-bit floating point (AC3) is not supported on all platforms.
settings->AddSupportedFormat(FORMAT_FLT);
#endif
// Guess that we can do up to 5.1
for (uint i = 2; i < 7; i++)
settings->AddSupportedChannels(i);
settings->setPassthrough(0); //Maybe passthrough
return settings;
}
AudioOutputSettings* AudioOutputJACK::GetOutputSettings(bool /*digital*/)
{
int rate = 0;
int i = 0;
const char **matching_ports = NULL;
AudioOutputSettings *settings = new AudioOutputSettings();
client = _jack_client_open();
if (!client)
{
JERROR(tr("Cannot start/connect to jack server "
"(to check supported rate/channels)"));
delete settings;
return NULL;
}
if (client)
rate = jack_get_sample_rate(client);
if (!rate)
{
JERROR(tr("Unable to retrieve jack server sample rate"));
goto err_out;
}
else
settings->AddSupportedRate(rate);
// Jack only wants float format samples (de-interleaved for preference)
settings->AddSupportedFormat(FORMAT_FLT);
// Find some Jack ports to connect to
matching_ports = _jack_get_ports();
if (!matching_ports || !matching_ports[0])
{
JERROR(tr("No ports available to connect to"));
goto err_out;
}
// Count matching ports from 2nd port upwards
i = 1;
while (matching_ports[i] && (i < JACK_CHANNELS_MAX))
{
settings->AddSupportedChannels(i+1);
VBAUDIO(QString("Adding channels: %1").arg(i+1));
i++;
}
// Currently this looks very similar to error code - duplicated for safety
free(matching_ports);
_jack_client_close(&client);
return settings;
err_out:
// Our abstracted exit point in case of error
free(matching_ports);
_jack_client_close(&client);
delete settings;
return NULL;
}
AudioOutputSettings* AudioOutputDX::GetOutputSettings(bool passthrough)
{
AudioOutputSettings *settings = new AudioOutputSettings();
DSCAPS devcaps;
devcaps.dwSize = sizeof(DSCAPS);
m_priv->InitDirectSound(passthrough);
if ((!m_priv->dsobject || !m_priv->dsound_dll) ||
FAILED(IDirectSound_GetCaps(m_priv->dsobject, &devcaps)) )
{
delete settings;
return nullptr;
}
VBAUDIO(QString("GetCaps sample rate min: %1 max: %2")
.arg(devcaps.dwMinSecondarySampleRate)
.arg(devcaps.dwMaxSecondarySampleRate));
/* We shouldn't assume we can do everything between min and max but
there's no way to test individual rates, so we'll have to */
while (DWORD rate = (DWORD)settings->GetNextRate())
if((rate >= devcaps.dwMinSecondarySampleRate) ||
(rate <= devcaps.dwMaxSecondarySampleRate))
settings->AddSupportedRate(rate);
/* We can only test for 8 and 16 bit support, DS uses float internally */
if (devcaps.dwFlags & DSCAPS_PRIMARY8BIT)
settings->AddSupportedFormat(FORMAT_U8);
if (devcaps.dwFlags & DSCAPS_PRIMARY16BIT)
settings->AddSupportedFormat(FORMAT_S16);
#if 0 // 24-bit integer is not supported
settings->AddSupportedFormat(FORMAT_S24);
#endif
#if 0 // 32-bit integer (OGG) is not supported on all platforms.
settings->AddSupportedFormat(FORMAT_S32);
#endif
#if 0 // 32-bit floating point (AC3) is not supported on all platforms.
settings->AddSupportedFormat(FORMAT_FLT);
#endif
/* No way to test anything other than mono or stereo, guess that we can do
up to 5.1 */
for (uint i = 2; i < 7; i++)
settings->AddSupportedChannels(i);
settings->setPassthrough(0); // Maybe passthrough
return settings;
}
AudioOutputSettings* AudioOutputCA::GetOutputSettings()
{
AudioOutputSettings *settings = new AudioOutputSettings();
// Seek hardware sample rate available
int rate;
int *rates = d->RatesList(d->mDeviceID);
int *p_rates = rates;
while (*p_rates > 0 && (rate = settings->GetNextRate()))
{
if (*p_rates == rate)
{
settings->AddSupportedRate(*p_rates);
p_rates++;
}
}
free(rates);
// Supported format: 16 bits audio or float
settings->AddSupportedFormat(FORMAT_S16);
settings->AddSupportedFormat(FORMAT_FLT);
bool *channels = d->ChannelsList(d->mDeviceID, passthru);
if (channels == NULL)
{
// Error retrieving list of supported channels, assume stereo only
settings->AddSupportedChannels(2);
}
else
{
for (int i = CHANNELS_MIN; i <= CHANNELS_MAX; i++)
if (channels[i])
{
Debug(QString("Support %1 channels").arg(i));
settings->AddSupportedChannels(i);
}
free(channels);
}
settings->setPassthrough(0); // Maybe passthrough
return settings;
}
AudioOutputSettings* AudioOutputWin::GetOutputSettings(void)
{
AudioOutputSettings *settings = new AudioOutputSettings();
// We use WAVE_MAPPER to find a compatible device, so just claim support
// for all of the standard rates
while (DWORD rate = (DWORD)settings->GetNextRate())
settings->AddSupportedRate(rate);
// Support all standard formats
settings->AddSupportedFormat(FORMAT_U8);
settings->AddSupportedFormat(FORMAT_S16);
settings->AddSupportedFormat(FORMAT_S24);
settings->AddSupportedFormat(FORMAT_S32);
settings->AddSupportedFormat(FORMAT_FLT);
// Guess that we can do up to 5.1
for (uint i = 2; i < 7; i++)
settings->AddSupportedChannels(i);
settings->setPassthrough(0); //Maybe passthrough
return settings;
}
AudioOutputSettings* AudioOutputOSS::GetOutputSettings(bool /*digital*/)
{
AudioOutputSettings *settings = new AudioOutputSettings();
QByteArray device = main_device.toLatin1();
audiofd = open(device.constData(), O_WRONLY | O_NONBLOCK);
AudioFormat fmt;
int rate, formats = 0;
if (audiofd < 0)
{
VBERRENO(QString("Error opening audio device (%1)").arg(main_device));
delete settings;
return NULL;
}
while ((rate = settings->GetNextRate()))
{
int rate2 = rate;
if(ioctl(audiofd, SNDCTL_DSP_SPEED, &rate2) >= 0
&& rate2 == rate)
settings->AddSupportedRate(rate);
}
if(ioctl(audiofd, SNDCTL_DSP_GETFMTS, &formats) < 0)
VBERRENO("Error retrieving formats");
else
{
int ofmt;
while ((fmt = settings->GetNextFormat()))
{
switch (fmt)
{
case FORMAT_U8: ofmt = AFMT_U8; break;
case FORMAT_S16: ofmt = AFMT_S16_NE; break;
default: continue;
}
if (formats & ofmt)
settings->AddSupportedFormat(fmt);
}
}
#if defined(AFMT_AC3)
// Check if drivers supports AC3
settings->setPassthrough(((formats & AFMT_AC3) != 0) - 1);
#endif
for (int i = 1; i <= 2; i++)
{
int channel = i;
if (ioctl(audiofd, SNDCTL_DSP_CHANNELS, &channel) >= 0 &&
channel == i)
{
settings->AddSupportedChannels(i);
}
}
close(audiofd);
audiofd = -1;
return settings;
}
AudioOutputSettings* AudioOutputALSA::GetOutputSettings(bool passthrough)
{
snd_pcm_hw_params_t *params;
snd_pcm_format_t afmt = SND_PCM_FORMAT_UNKNOWN;
AudioFormat fmt;
int rate;
int err;
AudioOutputSettings *settings = new AudioOutputSettings();
if (pcm_handle)
{
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
}
if ((err = TryOpenDevice(OPEN_FLAGS, passthrough)) < 0)
{
AERROR(QString("snd_pcm_open(\"%1\")").arg(m_lastdevice));
delete settings;
return NULL;
}
snd_pcm_hw_params_alloca(¶ms);
if ((err = snd_pcm_hw_params_any(pcm_handle, params)) < 0)
{
snd_pcm_close(pcm_handle);
if ((err = TryOpenDevice(OPEN_FLAGS&FILTER_FLAGS, passthrough)) < 0)
{
AERROR(QString("snd_pcm_open(\"%1\")").arg(m_lastdevice));
delete settings;
return NULL;
}
if ((err = snd_pcm_hw_params_any(pcm_handle, params)) < 0)
{
AERROR("No playback configurations available");
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
delete settings;
return NULL;
}
Warn("Supported audio format detection will be inacurrate "
"(using plugin?)");
}
while ((rate = settings->GetNextRate()))
if(snd_pcm_hw_params_test_rate(pcm_handle, params, rate, 0) >= 0)
settings->AddSupportedRate(rate);
while ((fmt = settings->GetNextFormat()))
{
switch (fmt)
{
case FORMAT_U8: afmt = SND_PCM_FORMAT_U8; break;
case FORMAT_S16: afmt = SND_PCM_FORMAT_S16; break;
case FORMAT_S24: afmt = SND_PCM_FORMAT_S24; break;
case FORMAT_S32: afmt = SND_PCM_FORMAT_S32; break;
case FORMAT_FLT: afmt = SND_PCM_FORMAT_FLOAT; break;
default: continue;
}
if (snd_pcm_hw_params_test_format(pcm_handle, params, afmt) >= 0)
settings->AddSupportedFormat(fmt);
}
for (uint channels = CHANNELS_MIN; channels <= CHANNELS_MAX; channels++)
if (snd_pcm_hw_params_test_channels(pcm_handle, params, channels) >= 0)
settings->AddSupportedChannels(channels);
snd_pcm_close(pcm_handle);
pcm_handle = NULL;
/* Check if name or description contains information
to know if device can accept passthrough or not */
QMap<QString, QString> *alsadevs = GetALSADevices("pcm");
while(1)
{
QString real_device = (((passthru || enc) && m_discretedigital) ?
passthru_device : main_device);
QString desc = alsadevs->value(real_device);
settings->setPassthrough(1); // yes passthrough
if (real_device.contains("digital", Qt::CaseInsensitive) ||
desc.contains("digital", Qt::CaseInsensitive))
break;
if (real_device.contains("iec958", Qt::CaseInsensitive))
break;
if (real_device.contains("spdif", Qt::CaseInsensitive))
break;
if (real_device.contains("hdmi", Qt::CaseInsensitive))
break;
settings->setPassthrough(-1); // no passthrough
// PulseAudio does not support passthrough
if (real_device.contains("pulse", Qt::CaseInsensitive) ||
desc.contains("pulse", Qt::CaseInsensitive))
break;
if (real_device.contains("analog", Qt::CaseInsensitive) ||
desc.contains("analog", Qt::CaseInsensitive))
break;
if (real_device.contains("surround", Qt::CaseInsensitive) ||
desc.contains("surround", Qt::CaseInsensitive))
break;
settings->setPassthrough(0); // maybe passthrough
break;
}
delete alsadevs;
return settings;
}
