void DAUDIO_GetFormats(INT32 mixerIndex, INT32 deviceID, int isSource, void* creator) {
snd_pcm_t* handle;
snd_pcm_format_mask_t* formatMask;
snd_pcm_format_t format;
snd_pcm_hw_params_t* hwParams;
int handledBits[MAX_BIT_INDEX+1];
int ret;
int sampleSizeInBytes, significantBits, isSigned, isBigEndian, enc;
int origSampleSizeInBytes, origSignificantBits;
int channels, minChannels, maxChannels;
int rate, bitIndex;
for (bitIndex = 0; bitIndex <= MAX_BIT_INDEX; bitIndex++) handledBits[bitIndex] = FALSE;
if (openPCMfromDeviceID(deviceID, &handle, isSource, TRUE /*query hardware*/) < 0) {
return;
}
ret = snd_pcm_format_mask_malloc(&formatMask);
if (ret != 0) {
ERROR1("snd_pcm_format_mask_malloc returned error %d\n", ret);
} else {
ret = snd_pcm_hw_params_malloc(&hwParams);
if (ret != 0) {
ERROR1("snd_pcm_hw_params_malloc returned error %d\n", ret);
} else {
ret = snd_pcm_hw_params_any(handle, hwParams);
if (ret != 0) {
ERROR1("snd_pcm_hw_params_any returned error %d\n", ret);
}
}
snd_pcm_hw_params_get_format_mask(hwParams, formatMask);
#ifdef ALSA_PCM_NEW_HW_PARAMS_API
if (ret == 0) {
ret = snd_pcm_hw_params_get_channels_min(hwParams, &minChannels);
if (ret != 0) {
ERROR1("snd_pcm_hw_params_get_channels_min returned error %d\n", ret);
}
}
if (ret == 0) {
ret = snd_pcm_hw_params_get_channels_max(hwParams, &maxChannels);
if (ret != 0) {
ERROR1("snd_pcm_hw_params_get_channels_max returned error %d\n", ret);
}
}
#else
minChannels = snd_pcm_hw_params_get_channels_min(hwParams);
maxChannels = snd_pcm_hw_params_get_channels_max(hwParams);
if (minChannels > maxChannels) {
ERROR2("MinChannels=%d, maxChannels=%d\n", minChannels, maxChannels);
}
#endif
// since we queried the hw: device, for many soundcards, it will only
// report the maximum number of channels (which is the only way to talk
// to the hw: device). Since we will, however, open the plughw: device
// when opening the Source/TargetDataLine, we can safely assume that
// also the channels 1..maxChannels are available.
#ifdef ALSA_PCM_USE_PLUGHW
minChannels = 1;
#endif
if (ret == 0) {
// plughw: supports any sample rate
rate = -1;
for (format = 0; format <= SND_PCM_FORMAT_LAST; format++) {
if (snd_pcm_format_mask_test(formatMask, format)) {
// format exists
if (getFormatFromAlsaFormat(format, &origSampleSizeInBytes,
&origSignificantBits,
&isSigned, &isBigEndian, &enc)) {
// now if we use plughw:, we can use any bit size below the
// natively supported ones. Some ALSA drivers only support the maximum
// bit size, so we add any sample rates below the reported one.
// E.g. this iteration reports support for 16-bit.
// getBitIndex will return 2, so it will add entries for
// 16-bit (bitIndex=2) and in the next do-while loop iteration,
// it will decrease bitIndex and will therefore add 8-bit support.
bitIndex = getBitIndex(origSampleSizeInBytes, origSignificantBits);
do {
if (bitIndex == 0
|| bitIndex == MAX_BIT_INDEX
|| !handledBits[bitIndex]) {
handledBits[bitIndex] = TRUE;
sampleSizeInBytes = getSampleSizeInBytes(bitIndex, origSampleSizeInBytes);
significantBits = getSignificantBits(bitIndex, origSignificantBits);
if (maxChannels - minChannels > MAXIMUM_LISTED_CHANNELS) {
// avoid too many channels explicitly listed
// just add -1, min, and max
DAUDIO_AddAudioFormat(creator, significantBits,
-1, -1, rate,
enc, isSigned, isBigEndian);
DAUDIO_AddAudioFormat(creator, significantBits,
sampleSizeInBytes * minChannels,
minChannels, rate,
enc, isSigned, isBigEndian);
DAUDIO_AddAudioFormat(creator, significantBits,
sampleSizeInBytes * maxChannels,
maxChannels, rate,
enc, isSigned, isBigEndian);
} else {
for (channels = minChannels; channels <= maxChannels; channels++) {
DAUDIO_AddAudioFormat(creator, significantBits,
(channels < 0)?-1:(sampleSizeInBytes * channels),
channels, rate,
enc, isSigned, isBigEndian);
}
}
}
#ifndef ALSA_PCM_USE_PLUGHW
// without plugin, do not add fake formats
break;
#endif
} while (--bitIndex > 0);
} else {
TRACE1("could not get format from alsa for format %d\n", format);
}
} else {
//TRACE1("Format %d not supported\n", format);
}
} // for loop
snd_pcm_hw_params_free(hwParams);
}
snd_pcm_format_mask_free(formatMask);
}
snd_pcm_close(handle);
}
void DAUDIO_GetFormats(INT32 mixerIndex, INT32 deviceID, int isSource, void* creator) {
int fd = -1;
AudioDeviceDescription desc;
am_sample_rates_t *sr;
/* hardcoded bits and channels */
int bits[] = {8, 16};
int bitsCount = 2;
int channels[] = {1, 2};
int channelsCount = 2;
/* for querying sample rates */
int err;
int ch, b, s;
TRACE2("DAUDIO_GetFormats, mixer %d, isSource=%d\n", mixerIndex, isSource);
if (getAudioDeviceDescriptionByIndex(mixerIndex, &desc, FALSE)) {
fd = open(desc.pathctl, O_RDONLY);
}
if (fd < 0) {
ERROR1("Couldn't open audio device ctl for device %d!\n", mixerIndex);
return;
}
/* get sample rates */
sr = (am_sample_rates_t*) malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(MAX_SAMPLE_RATES));
if (sr == NULL) {
ERROR1("DAUDIO_GetFormats: out of memory for mixer %d\n", (int) mixerIndex);
close(fd);
return;
}
sr->num_samp_rates = MAX_SAMPLE_RATES;
sr->type = isSource?AUDIO_PLAY:AUDIO_RECORD;
sr->samp_rates[0] = -2;
err = ioctl(fd, AUDIO_MIXER_GET_SAMPLE_RATES, sr);
if (err < 0) {
ERROR1(" DAUDIO_GetFormats: AUDIO_MIXER_GET_SAMPLE_RATES failed for mixer %d!\n",
(int)mixerIndex);
ERROR2(" -> num_sample_rates=%d sample_rates[0] = %d\n",
(int) sr->num_samp_rates,
(int) sr->samp_rates[0]);
/* Some Solaris 8 drivers fail for get sample rates!
* Do as if we support all sample rates
*/
sr->flags = MIXER_SR_LIMITS;
}
if ((sr->flags & MIXER_SR_LIMITS)
|| (sr->num_samp_rates > MAX_SAMPLE_RATES)) {
#ifdef USE_TRACE
if ((sr->flags & MIXER_SR_LIMITS)) {
TRACE1(" DAUDIO_GetFormats: floating sample rate allowed by mixer %d\n",
(int)mixerIndex);
}
if (sr->num_samp_rates > MAX_SAMPLE_RATES) {
TRACE2(" DAUDIO_GetFormats: more than %d formats. Use -1 for sample rates mixer %d\n",
MAX_SAMPLE_RATES, (int)mixerIndex);
}
#endif
/*
* Fake it to have only one sample rate: -1
*/
sr->num_samp_rates = 1;
sr->samp_rates[0] = -1;
}
close(fd);
for (ch = 0; ch < channelsCount; ch++) {
for (b = 0; b < bitsCount; b++) {
for (s = 0; s < sr->num_samp_rates; s++) {
DAUDIO_AddAudioFormat(creator,
bits[b], /* significant bits */
0, /* frameSize: let it be calculated */
channels[ch],
(float) ((int) sr->samp_rates[s]),
DAUDIO_PCM, /* encoding - let's only do PCM */
(bits[b] > 8)?TRUE:TRUE, /* isSigned */
#if (X_WORD_ORDER == TRUE)
FALSE /* little endian */
#else
(bits[b] > 8)?TRUE:FALSE /* big endian */
#endif
);
}
}
}
free(sr);
}