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quisk-kc4upr/sound_alsa.c

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/*
* This modue provides sound access for QUISK using the ALSA
* library for Linux.
*/
#include <Python.h>
#include <complex.h>
#include <math.h>
#include <alsa/asoundlib.h>
#include "quisk.h"
/*
The sample rate is in frames per second. Each frame has a number of channels,
and each channel has a sample of size sample_bytes. The channels are interleaved:
(channel0, channel1), (channel0, channel1), ...
*/
#define CORRECT_PLAY_RATE 1
extern struct sound_conf quisk_sound_state; // Current sound status
static int is_little_endian; // Test byte order; is it little-endian?
static short buffer2[SAMP_BUFFER_SIZE]; // Buffer for 2-byte samples from sound
static unsigned char buffer3[3 * SAMP_BUFFER_SIZE]; // Buffer for 3-byte samples from sound
static int buffer4[SAMP_BUFFER_SIZE]; // Buffer for 4-byte samples from sound
static int bufferz[SAMP_BUFFER_SIZE]; // Buffer for zero samples
static double mic_playbuf_util = 0.70; // Current mic play buffer utilization 0.0 to 1.0
int quisk_read_alsa(struct sound_dev * dev, complex double * cSamples)
{ // cSamples can be NULL to discard samples.
// Read sound samples from the ALSA soundcard.
// Samples are converted to 32 bits with a range of +/- CLIP32 and placed into cSamples.
int i;
snd_pcm_sframes_t frames, delay, avail;
short si, sq;
int ii, qq;
int nSamples;
if (!dev->handle)
return -1;
switch(snd_pcm_state(dev->handle)) {
case SND_PCM_STATE_RUNNING:
break;
case SND_PCM_STATE_PREPARED:
break;
case SND_PCM_STATE_XRUN:
#if DEBUG_IO
QuiskPrintTime("read_alsa: Capture overrun", 0);
#endif
snd_pcm_prepare(dev->handle);
break;
default:
#if DEBUG_IO
QuiskPrintTime("read_alsa: State UNKNOWN", 0);
#endif
break;
}
if (snd_pcm_avail_delay(dev->handle, &avail, &delay) >= 0) {
dev->dev_latency = avail + delay; // avail frames can be read plus delay frames digitized but can't be read yet
}
else {
avail = 32;
dev->dev_latency = 0;
dev->dev_error++;
#if DEBUG_IO
QuiskPrintTime("read_alsa: snd_pcm_avail_delay failed", 0);
#endif
}
if (dev->read_frames == 0) { // non-blocking: read available frames
if (avail < 32)
avail = 32; // read frames to restart from error
}
else {
avail = dev->read_frames; // size of read request
}
i = SAMP_BUFFER_SIZE * 8 / 10 / dev->num_channels; // limit read request to buffer size
if (avail > i)
avail = i;
nSamples = 0;
switch (dev->sample_bytes) {
case 2:
frames = snd_pcm_readi (dev->handle, buffer2, avail); // read samples
if ( ! cSamples)
return 0;
if (frames == -EAGAIN) { // no samples available
break;
}
else if (frames <= 0) { // error
dev->dev_error++;
#if DEBUG_IO
QuiskPrintTime("read_alsa: frames < 0", 0);
#endif
snd_pcm_prepare (dev->handle);
snd_pcm_start (dev->handle);
break;
}
for (i = 0; frames; i += dev->num_channels, frames--) {
si = buffer2[i + dev->channel_I];
sq = buffer2[i + dev->channel_Q];
if (si >= CLIP16 || si <= -CLIP16)
dev->overrange++; // assume overrange returns max int
if (sq >= CLIP16 || sq <= -CLIP16)
dev->overrange++;
ii = si << 16;
qq = sq << 16;
cSamples[nSamples] = ii + I * qq;
nSamples++;
}
break;
case 3:
frames = snd_pcm_readi (dev->handle, buffer3, avail); // read samples
if ( ! cSamples)
return 0;
if (frames == -EAGAIN) { // no samples available
break;
}
else if (frames <= 0) { // error
dev->dev_error++;
#if DEBUG_IO
QuiskPrintTime("read_alsa: frames < 0", 0);
#endif
snd_pcm_prepare (dev->handle);
snd_pcm_start (dev->handle);
break;
}
for (i = 0; frames; i += dev->num_channels, frames--) {
ii = qq = 0;
if (!is_little_endian) { // convert to big-endian
*((unsigned char *)&ii ) = buffer3[(i + dev->channel_I) * 3 + 2];
*((unsigned char *)&ii + 1) = buffer3[(i + dev->channel_I) * 3 + 1];
*((unsigned char *)&ii + 2) = buffer3[(i + dev->channel_I) * 3 ];
*((unsigned char *)&qq ) = buffer3[(i + dev->channel_Q) * 3 + 2];
*((unsigned char *)&qq + 1) = buffer3[(i + dev->channel_Q) * 3 + 1];
*((unsigned char *)&qq + 2) = buffer3[(i + dev->channel_Q) * 3 ];
}
else { // convert to little-endian
memcpy((unsigned char *)&ii + 1, buffer3 + (i + dev->channel_I) * 3, 3);
memcpy((unsigned char *)&qq + 1, buffer3 + (i + dev->channel_Q) * 3, 3);
}
if (ii >= CLIP32 || ii <= -CLIP32)
dev->overrange++; // assume overrange returns max int
if (qq >= CLIP32 || qq <= -CLIP32)
dev->overrange++;
cSamples[nSamples] = ii + I * qq;
nSamples++;
}
break;
case 4:
frames = snd_pcm_readi (dev->handle, buffer4, avail); // read samples
if ( ! cSamples)
return 0;
if (frames == -EAGAIN) { // no samples available
break;
}
else if (frames <= 0) { // error
dev->dev_error++;
#if DEBUG_IO
QuiskPrintTime("read_alsa: frames < 0", 0);
#endif
snd_pcm_prepare (dev->handle);
snd_pcm_start (dev->handle);
break;
}
for (i = 0; frames; i += dev->num_channels, frames--) {
ii = buffer4[i + dev->channel_I];
qq = buffer4[i + dev->channel_Q];
if (ii >= CLIP32 || ii <= -CLIP32)
dev->overrange++; // assume overrange returns max int
if (qq >= CLIP32 || qq <= -CLIP32)
dev->overrange++;
cSamples[nSamples] = ii + I * qq;
nSamples++;
}
break;
default:
return 0;
}
#if CORRECT_PLAY_RATE
if ( ! strcmp(dev->stream_description, "Microphone Input")) {
if (mic_playbuf_util > 0.85) { // Remove a sample
nSamples--;
#if DEBUG_IO
printf("read_alsa %s: Remove a mic sample, util %.2lf\n", dev->stream_description, mic_playbuf_util);
#endif
}
else if(cSamples && mic_playbuf_util < 0.55 && nSamples >= 2) { // Add a sample
cSamples[nSamples] = cSamples[nSamples - 1];
cSamples[nSamples - 1] = (cSamples[nSamples - 2] + cSamples[nSamples]) / 2.0;
nSamples++;
#if DEBUG_IO
printf("read_alsa %s: Add a mic sample, util %.2lf\n", dev->stream_description, mic_playbuf_util);
#endif
}
}
#endif
return nSamples;
}
void quisk_play_alsa(struct sound_dev * playdev, int nSamples,
complex double * cSamples, int report_latency, double volume)
{ // Play the samples; write them to the ALSA soundcard.
int i, n, index;
snd_pcm_sframes_t frames, delay;
int ii, qq;
#if DEBUG_IO
static int timer=0;
#endif
if (!playdev->handle || nSamples <= 0)
return;
// Note: snd_pcm_delay() is not reliable when using the "default" ALSA device and
// arbitrary sample rates. It seems to be confused by the rate conversion. So we
// are changing rates (decimate) ourselves.
//
delay = 0;
switch(snd_pcm_state(playdev->handle)) {
case SND_PCM_STATE_RUNNING:
//printf("State RUNNING\n");
snd_pcm_delay(playdev->handle, &delay); // samples left in play buffer
break;
case SND_PCM_STATE_PREPARED:
#if DEBUG_IO
//QuiskPrintTime("play_alsa: State PREPARED", 0);
#endif
snd_pcm_delay(playdev->handle, &delay);
delay = 0;
break;
case SND_PCM_STATE_XRUN:
#if DEBUG_IO
QuiskPrintTime("", 0);
printf("play_alsa: Play underrun; nSamples %d\n", nSamples);
#endif
quisk_sound_state.underrun_error++;
playdev->dev_underrun++;
snd_pcm_prepare(playdev->handle);
break;
default:
#if DEBUG_IO
QuiskPrintTime("play_alsa: State UNKNOWN", 0);
#endif
break;
}
playdev->dev_latency = delay;
if (report_latency) { // Report for main playback device
quisk_sound_state.latencyPlay = delay; // samples left in play buffer
}
if ( ! strcmp(playdev->stream_description, "I/Q Tx Sample Output")) {
mic_playbuf_util = (double)(nSamples + delay) / playdev->latency_frames;
}
// There will be additional samples available to read in the capture buffer.
index = 0;
#if CORRECT_PLAY_RATE
#if DEBUG_IO
timer += nSamples;
if (timer > playdev->sample_rate) {
timer = 0;
printf("play_alsa %s: Samples new %d old %ld total %ld latency_frames %d\n", playdev->stream_description, nSamples, delay, nSamples + delay, playdev->latency_frames);
}
#endif
if (volume == 0.0) {
n = playdev->latency_frames * 7 / 10 - delay; // samples needed to get back to the starting value of 70% full
#if DEBUG_IO
if (abs(n - nSamples) > 100)
printf("play_alsa %s: Zero volume correction nSamples %5d to %5d\n", playdev->stream_description, nSamples, n);
#endif
if (n <= 0)
nSamples = 0;
else if (n <= nSamples)
nSamples = n;
else {
n -= nSamples; // number of samples to add
if (n > 100) // arbitrary increase - should be tested
n = 100;
for (i = 0; i < n; i++)
cSamples[nSamples++] = 0.0;
}
}
else if (nSamples + delay > playdev->latency_frames * 95 / 100) {
nSamples--;
#if DEBUG_IO
printf("play_alsa %s: Remove a sample nSamples %d delay %d total %d\n", playdev->stream_description, nSamples, (int)delay, nSamples + (int)delay);
#endif
}
else if(nSamples + delay < playdev->latency_frames * 4 / 10 && nSamples >= 2) {
cSamples[nSamples] = cSamples[nSamples - 1];
cSamples[nSamples - 1] = (cSamples[nSamples - 2] + cSamples[nSamples]) / 2.0;
nSamples++;
#if DEBUG_IO
printf ("play_alsa %s: Add a sample nSamples %d delay %d total %d\n", playdev->stream_description, nSamples, (int)delay, nSamples + (int)delay);
#endif
}
#endif
if (nSamples + delay > playdev->latency_frames) {
index = nSamples + delay - playdev->latency_frames; // write only the most recent samples
quisk_sound_state.write_error++;
playdev->dev_error++;
#if DEBUG_IO
//QuiskPrintTime("", 0);
printf("play_alsa %s: Discard %d of %d samples at %ld delay\n", playdev->stream_description, index, nSamples, delay);
#endif
}
if (playdev->sample_bytes == 2) {
while (index < nSamples) {
for (i = 0, n = index; n < nSamples; i += playdev->num_channels, n++) {
ii = (int)(volume * creal(cSamples[n]) / 65536);
qq = (int)(volume * cimag(cSamples[n]) / 65536);
buffer2[i + playdev->channel_I] = (short)ii;
buffer2[i + playdev->channel_Q] = (short)qq;
}
n = n - index;
frames = snd_pcm_writei (playdev->handle, buffer2, n);
if (frames <= 0) {
#if DEBUG_IO
QuiskPrintTime("play_alsa: frames < 0", 0);
#endif
if (frames == -EPIPE) { // underrun
quisk_sound_state.underrun_error++;
playdev->dev_underrun++;
}
else {
quisk_sound_state.write_error++;
playdev->dev_error++;
}
snd_pcm_prepare(playdev->handle);
frames = snd_pcm_writei (playdev->handle, buffer2, n);
if (frames <= 0) {
index = nSamples; // give up
}
else {
index += frames;
}
}
else {
index += frames;
}
}
}
else if (playdev->sample_bytes == 3) {
while (index < nSamples) {
for (i = 0, n = index; n < nSamples; i += playdev->num_channels, n++) {
ii = (int)(volume * creal(cSamples[n]) / 256);
qq = (int)(volume * cimag(cSamples[n]) / 256);
if (!is_little_endian) { // convert to big-endian
buffer3[(i + playdev->channel_I) * 3 ] = *((unsigned char *)&ii + 2);
buffer3[(i + playdev->channel_Q) * 3 ] = *((unsigned char *)&qq + 2);
buffer3[(i + playdev->channel_I) * 3 + 1] = *((unsigned char *)&ii + 1);
buffer3[(i + playdev->channel_Q) * 3 + 1] = *((unsigned char *)&qq + 1);
buffer3[(i + playdev->channel_I) * 3 + 2] = *((unsigned char *)&ii );
buffer3[(i + playdev->channel_Q) * 3 + 2] = *((unsigned char *)&qq );
}
else { // convert to little-endian
memcpy(buffer3 + (i + playdev->channel_I) * 3, (unsigned char *)&ii, 3);
memcpy(buffer3 + (i + playdev->channel_Q) * 3, (unsigned char *)&qq, 3);
}
}
n = n - index;
frames = snd_pcm_writei (playdev->handle, buffer3, n);
if (frames <= 0) {
#if DEBUG_IO
QuiskPrintTime("play_alsa: frames < 0", 0);
#endif
if (frames == -EPIPE) { // underrun
quisk_sound_state.underrun_error++;
playdev->dev_underrun++;
}
else {
quisk_sound_state.write_error++;
playdev->dev_error++;
}
snd_pcm_prepare(playdev->handle);
frames = snd_pcm_writei (playdev->handle, buffer3, n);
if (frames <= 0) {
index = nSamples; // give up
}
else {
index += frames;
}
}
else {
index += frames;
}
}
}
else if (playdev->sample_bytes == 4) {
while (index < nSamples) {
for (i = 0, n = index; n < nSamples; i += playdev->num_channels, n++) {
ii = (int)(volume * creal(cSamples[n]));
qq = (int)(volume * cimag(cSamples[n]));
buffer4[i + playdev->channel_I] = ii;
buffer4[i + playdev->channel_Q] = qq;
}
n = n - index;
frames = snd_pcm_writei (playdev->handle, buffer4, n);
if (frames <= 0) {
#if DEBUG_IO
QuiskPrintTime("play_alsa: frames < 0", 0);
#endif
if (frames == -EPIPE) { // underrun
quisk_sound_state.underrun_error++;
playdev->dev_underrun++;
}
else {
quisk_sound_state.write_error++;
playdev->dev_error++;
}
snd_pcm_prepare(playdev->handle);
frames = snd_pcm_writei (playdev->handle, buffer4, n);
if (frames <= 0) {
index = nSamples; // give up
}
else {
index += frames;
}
}
else {
index += frames;
}
}
}
}
static int device_list(PyObject * py, snd_pcm_stream_t stream, char * name)
{ // return 1 if the card name was substituted
snd_ctl_t *handle;
int card, err, dev;
char buf100[100];
const char * card_text, * pcm_text;
snd_ctl_card_info_t *info;
snd_pcm_info_t *pcminfo;
snd_ctl_card_info_alloca(&info);
snd_pcm_info_alloca(&pcminfo);
card = -1;
if (snd_card_next(&card) < 0 || card < 0) {
printf("no soundcards found...\n");
return 0;
}
while (card >= 0) {
sprintf(buf100, "hw:%d", card);
if ((err = snd_ctl_open(&handle, buf100, 0)) < 0) {
printf("device_list: control open (%i): %s", card, snd_strerror(err));
goto next_card;
}
if ((err = snd_ctl_card_info(handle, info)) < 0) {
printf("device_list: control hardware info (%i): %s", card, snd_strerror(err));
snd_ctl_close(handle);
goto next_card;
}
dev = -1;
while (1) {
if (snd_ctl_pcm_next_device(handle, &dev)<0)
printf("device_list: snd_ctl_pcm_next_device\n");
if (dev < 0)
break;
snd_pcm_info_set_device(pcminfo, dev);
snd_pcm_info_set_subdevice(pcminfo, 0);
snd_pcm_info_set_stream(pcminfo, stream);
card_text = snd_ctl_card_info_get_name(info);
if ( ! card_text || ! card_text[0])
card_text = snd_ctl_card_info_get_id(info);
if ((err = snd_ctl_pcm_info(handle, pcminfo)) < 0) {
if (err != -ENOENT)
printf ("device_list: control digital audio info (%i): %s", card, snd_strerror(err));
continue;
}
else {
pcm_text = snd_pcm_info_get_name(pcminfo);
if ( ! pcm_text || ! pcm_text[0])
pcm_text = snd_pcm_info_get_id(pcminfo);
}
snprintf(buf100, 100, "%s %s (hw:%d,%d)", card_text, pcm_text, card, dev);
if (py) { // add to list of devices
PyList_Append(py, PyString_FromString(buf100));
}
if (name) { // return the "hw:" name
if (strstr(buf100, name)) {
snprintf(name, QUISK_SC_SIZE, "hw:%d,%d", card, dev);
snd_ctl_close(handle);
return 1;
}
}
}
snd_ctl_close(handle);
next_card:
if (snd_card_next(&card) < 0) {
printf("snd_card_next\n");
break;
}
}
return 0;
}
PyObject * quisk_sound_devices(PyObject * self, PyObject * args)
{ // Return a list of ALSA device names [pycapt, pyplay]
PyObject * pylist, * pycapt, * pyplay;
if (!PyArg_ParseTuple (args, ""))
return NULL;
// Each pycapt and pyplay is [pydev, pyname]
pylist = PyList_New(0); // list [pycapt, pyplay]
pycapt = PyList_New(0); // list of capture devices
pyplay = PyList_New(0); // list of play devices
PyList_Append(pylist, pycapt);
PyList_Append(pylist, pyplay);
device_list(pycapt, SND_PCM_STREAM_CAPTURE, NULL);
device_list(pyplay, SND_PCM_STREAM_PLAYBACK, NULL);
return pylist;
}
static snd_pcm_format_t check_formats(struct sound_dev * dev, snd_pcm_hw_params_t *hware)
{
snd_pcm_format_t format = SND_PCM_FORMAT_UNKNOWN;
dev->sample_bytes = 0;
strncpy (dev->msg1, "Available formats: ", QUISK_SC_SIZE);
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S32) == 0) {
if (!dev->sample_bytes) {
strncat(dev->msg1, "*", QUISK_SC_SIZE);
dev->sample_bytes = 4;
format = SND_PCM_FORMAT_S32;
}
strncat(dev->msg1, "S32 ", QUISK_SC_SIZE);
}
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_U32) == 0) {
strncat(dev->msg1, "U32 ", QUISK_SC_SIZE);
}
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S24) == 0) {
strncat(dev->msg1, "S24 ", QUISK_SC_SIZE);
}
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_U24) == 0) {
strncat(dev->msg1, "U24 ", QUISK_SC_SIZE);
}
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S24_3LE) == 0) {
if (!dev->sample_bytes) {
strncat(dev->msg1, "*", QUISK_SC_SIZE);
dev->sample_bytes = 3;
format = SND_PCM_FORMAT_S24_3LE;
}
strncat(dev->msg1, "S24_3LE ", QUISK_SC_SIZE);
}
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_S16) == 0) {
if (!dev->sample_bytes) {
strncat(dev->msg1, "*", QUISK_SC_SIZE);
dev->sample_bytes = 2;
format = SND_PCM_FORMAT_S16;
}
strncat(dev->msg1, "S16 ", QUISK_SC_SIZE);
}
if (snd_pcm_hw_params_test_format (dev->handle, hware, SND_PCM_FORMAT_U16) == 0) {
strncat(dev->msg1, "U16 ", QUISK_SC_SIZE);
}
if (format == SND_PCM_FORMAT_UNKNOWN)
strncat(dev->msg1, "*UNSUPPORTED", QUISK_SC_SIZE);
else
snd_pcm_hw_params_set_format (dev->handle, hware, format);
return format;
}
static int quisk_open_alsa_capture(struct sound_dev * dev)
{ // Open the ALSA soundcard for capture. Return non-zero for error.
int i, err, dir, sample_rate, mode;
int poll_size;
unsigned int ui;
char buf[QUISK_SC_SIZE];
snd_pcm_hw_params_t *hware;
snd_pcm_sw_params_t *sware;
snd_pcm_uframes_t frames;
snd_pcm_t * handle;
if ( ! dev->name[0]) // Check for null capture name
return 0;
#if DEBUG_IO
printf("*** Capture on alsa device %s\n", dev->name);
#endif
if (dev->read_frames == 0)
mode = SND_PCM_NONBLOCK;
else
mode = 0;
if ( ! strncmp (dev->name, "alsa:", 5)) { // search for the name in info strings, put device name into buf
strncpy(buf, dev->name + 5, QUISK_SC_SIZE);
device_list(NULL, SND_PCM_STREAM_CAPTURE, buf);
}
else { // just try to open the name
strncpy(buf, dev->name, QUISK_SC_SIZE);
}
for (i = 0; i < 6; i++) { // try a few times in case the device is busy
err = snd_pcm_open (&handle, buf, SND_PCM_STREAM_CAPTURE, mode);
if (err >= 0)
break;
QuiskSleepMicrosec(500000);
}
if (err < 0) {
snprintf(quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot open capture device %.40s (%.40s)",
dev->name, snd_strerror (err));
return 1;
}
dev->handle = handle;
dev->driver = DEV_DRIVER_ALSA;
if ((err = snd_pcm_sw_params_malloc (&sware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot allocate software parameter structure (%s)\n",
snd_strerror (err));
return 1;
}
if ((err = snd_pcm_hw_params_malloc (&hware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
snd_pcm_sw_params_free (sware);
return 1;
}
if ((err = snd_pcm_hw_params_any (handle, hware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot initialize capture parameters (%s)\n",
snd_strerror (err));
goto errend;
}
/* UNAVAILABLE
if ((err = snd_pcm_hw_params_set_rate_resample (handle, hware, 0)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot disable resampling (%s)\n",
snd_strerror (err));
goto errend;
}
*/
// Get some parameters to send back
if (snd_pcm_hw_params_get_rate_min(hware, &dev->rate_min, &dir) != 0)
dev->rate_min = 0; // Error
if (snd_pcm_hw_params_get_rate_max(hware, &dev->rate_max, &dir) != 0)
dev->rate_max = 0; // Error
if (snd_pcm_hw_params_get_channels_min(hware, &dev->chan_min) != 0)
dev->chan_min= 0; // Error
if (snd_pcm_hw_params_get_channels_max(hware, &dev->chan_max) != 0)
dev->chan_max= 0; // Error
#if DEBUG_IO
printf("Sample rate min %d max %d\n", dev->rate_min, dev->rate_max);
printf("Number of channels min %d max %d\n", dev->chan_min, dev->chan_max);
#endif
// Set the capture parameters
if (check_formats(dev, hware) == SND_PCM_FORMAT_UNKNOWN) {
strncpy(quisk_sound_state.msg1, dev->msg1, QUISK_SC_SIZE);
strncpy (quisk_sound_state.err_msg, "Quisk does not support your capture format.", QUISK_SC_SIZE);
goto errend;
}
strncpy(quisk_sound_state.msg1, dev->msg1, QUISK_SC_SIZE);
sample_rate = dev->sample_rate;
if (snd_pcm_hw_params_set_rate (handle, hware, sample_rate, 0) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Can not set sample rate %d",
sample_rate);
goto errend;
}
if (snd_pcm_hw_params_set_access (handle, hware, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
strncpy(quisk_sound_state.err_msg, "Interleaved access is not available", QUISK_SC_SIZE);
goto errend;
}
if (snd_pcm_hw_params_get_channels_min(hware, &ui) != 0)
ui = 0; // Error
if (dev->num_channels < ui) // increase number of channels to minimum available
dev->num_channels = ui;
if (snd_pcm_hw_params_set_channels (handle, hware, dev->num_channels) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Can not set channels to %d", dev->num_channels);
goto errend;
}
// Try to set a capture buffer larger than needed
frames = sample_rate * 200 / 1000; // buffer size in milliseconds
if (snd_pcm_hw_params_set_buffer_size_near (handle, hware, &frames) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Can not set capture buffer size");
goto errend;
}
poll_size = (int)(quisk_sound_state.data_poll_usec * 1e-6 * sample_rate + 0.5);
if (frames < poll_size * 3) { // buffer size is too small, reduce poll time
quisk_sound_state.data_poll_usec = (int)(frames * 1.e6 / sample_rate / 3 + 0.5);
#if DEBUG_IO
printf("Reduced data_poll_usec %d for small sound capture buffer\n",
quisk_sound_state.data_poll_usec);
#endif
}
#if DEBUG_IO
printf("sample rate %d\n", sample_rate);
printf("num_channels %d, %s\n", dev->num_channels, dev->msg1);
printf("Capture buffer size %d\n", (int)frames);
if (frames > SAMP_BUFFER_SIZE / dev->num_channels)
printf("Capture buffer exceeds size of sample buffers\n");
#endif
if ((err = snd_pcm_hw_params (handle, hware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set hw capture parameters (%s)\n",
snd_strerror (err));
goto errend;
}
if ((err = snd_pcm_sw_params_current (handle, sware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot get software capture parameters (%s)\n",
snd_strerror (err));
goto errend;
}
if ((err = snd_pcm_prepare (handle)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot prepare capture interface for use (%s)\n",
snd_strerror (err));
goto errend;
}
// Success
snd_pcm_hw_params_free (hware);
snd_pcm_sw_params_free (sware);
#if DEBUG_IO
printf("*** End capture on alsa device %s %s\n", dev->name, quisk_sound_state.err_msg);
#endif
return 0;
errend:
snd_pcm_hw_params_free (hware);
snd_pcm_sw_params_free (sware);
#if DEBUG_IO
printf("*** Error end for capture on alsa device %s %s\n", dev->name, quisk_sound_state.err_msg);
#endif
return 1;
}
static int quisk_open_alsa_playback(struct sound_dev * dev)
{ // Open the ALSA soundcard for playback. Return non-zero on error.
int i, err, dir, sample_rate, mode;
unsigned int ui;
char buf[QUISK_SC_SIZE];
snd_pcm_hw_params_t *hware;
snd_pcm_sw_params_t *sware;
snd_pcm_uframes_t frames;
snd_pcm_t * handle;
if ( ! dev->name[0]) // Check for null play name
return 0;
#if DEBUG_IO
printf("*** Playback on alsa device %s\n", dev->name);
printf("quisk_open_alsa_playback(): %s\n", dev->stream_description);
#endif
if (dev->read_frames == 0)
mode = SND_PCM_NONBLOCK;
else
mode = 0;
if ( ! strncmp (dev->name, "alsa:", 5)) { // search for the name in info strings, put device name into buf
strncpy(buf, dev->name + 5, QUISK_SC_SIZE);
device_list(NULL, SND_PCM_STREAM_PLAYBACK, buf);
}
else { // just try to open the name
strncpy(buf, dev->name, QUISK_SC_SIZE);
}
for (i = 0; i < 6; i++) { // try a few times in case the device is busy
err = snd_pcm_open (&handle, buf, SND_PCM_STREAM_PLAYBACK, mode);
if (err >= 0)
break;
QuiskSleepMicrosec(500000);
}
if (err < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot open playback device %.40s (%.40s)\n",
dev->name, snd_strerror (err));
return 1;
}
dev->handle = handle;
if ((err = snd_pcm_sw_params_malloc (&sware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot allocate software parameter structure (%s)\n",
snd_strerror (err));
return 1;
}
if ((err = snd_pcm_hw_params_malloc (&hware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
snd_pcm_sw_params_free (sware);
return 1;
}
if ((err = snd_pcm_hw_params_any (handle, hware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot initialize playback parameter structure (%s)\n",
snd_strerror (err));
goto errend;
}
// Get some parameters to send back
if (snd_pcm_hw_params_get_rate_min(hware, &dev->rate_min, &dir) != 0)
dev->rate_min = 0; // Error
if (snd_pcm_hw_params_get_rate_max(hware, &dev->rate_max, &dir) != 0)
dev->rate_max = 0; // Error
if (snd_pcm_hw_params_get_channels_min(hware, &dev->chan_min) != 0)
dev->chan_min= 0; // Error
if (snd_pcm_hw_params_get_channels_max(hware, &dev->chan_max) != 0)
dev->chan_max= 0; // Error
#if DEBUG_IO
printf("Sample rate min %d max %d\n", dev->rate_min, dev->rate_max);
printf("Number of channels min %d max %d\n", dev->chan_min, dev->chan_max);
#endif
// Set the playback parameters
sample_rate = dev->sample_rate;
if (snd_pcm_hw_params_set_rate (handle, hware, sample_rate, 0) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set playback rate %d",
sample_rate);
goto errend;
}
if (snd_pcm_hw_params_set_access (handle, hware, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set playback access to interleaved.");
goto errend;
}
if (snd_pcm_hw_params_get_channels_min(hware, &ui) != 0)
ui = 0; // Error
if (dev->num_channels < ui) // increase number of channels to minimum available
dev->num_channels = ui;
if (snd_pcm_hw_params_set_channels (handle, hware, dev->num_channels) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set playback channels to %d",
dev->num_channels);
goto errend;
}
if (check_formats(dev, hware) == SND_PCM_FORMAT_UNKNOWN) {
strncpy(quisk_sound_state.msg1, dev->msg1, QUISK_SC_SIZE);
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set playback format.");
goto errend;
}
// Try to set a play buffer larger than needed
frames = sample_rate * 200 / 1000; // buffer size in milliseconds
if (snd_pcm_hw_params_set_buffer_size_near (handle, hware, &frames) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Can not set playback buffer size");
goto errend;
}
dev->play_buf_size = frames;
#if DEBUG_IO
printf("num_channels %d, %s\n", dev->num_channels, dev->msg1);
printf("Playback buffer size %d\n", dev->play_buf_size);
#endif
if ((err = snd_pcm_hw_params (handle, hware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set playback hw_params (%s)\n",
snd_strerror (err));
goto errend;
}
if ((err = snd_pcm_sw_params_current (handle, sware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot get software playback parameters (%s)\n",
snd_strerror (err));
goto errend;
}
if (dev->latency_frames > dev->play_buf_size) {
dev->latency_frames = dev->play_buf_size;
#if DEBUG_IO
printf("Latency frames limited to buffer size\n");
#endif
}
#if DEBUG_IO
printf("Audio rate %d latency_frames %d\n", sample_rate, dev->latency_frames);
#endif
if (snd_pcm_sw_params_set_start_threshold (handle, sware, dev->latency_frames * 7 / 10) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set start threshold\n");
goto errend;
}
if ((err = snd_pcm_sw_params (handle, sware)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot set playback sw_params (%s)\n",
snd_strerror (err));
goto errend;
}
#if DEBUG_IO
snd_pcm_sw_params_get_silence_threshold(sware, &frames);
printf("play silence threshold %d\n", (int)frames);
snd_pcm_sw_params_get_silence_size(sware, &frames);
printf("play silence size %d\n", (int)frames);
snd_pcm_sw_params_get_start_threshold(sware, &frames);
printf("play start threshold %d\n", (int)frames);
printf ("play channels are %d %d\n", dev->channel_I, dev->channel_Q);
#endif
if ((err = snd_pcm_prepare (handle)) < 0) {
snprintf (quisk_sound_state.err_msg, QUISK_SC_SIZE, "Cannot prepare playback interface for use (%s)\n",
snd_strerror (err));
goto errend;
}
// Success
snd_pcm_hw_params_free (hware);
snd_pcm_sw_params_free (sware);
#if DEBUG_IO
printf("*** End playback on alsa device %s %s\n", dev->name, quisk_sound_state.err_msg);
#endif
return 0;
errend:
snd_pcm_hw_params_free (hware);
snd_pcm_sw_params_free (sware);
#if DEBUG_IO
printf("*** Error end for playback on alsa device %s %s\n", dev->name, quisk_sound_state.err_msg);
#endif
return 1;
}
void quisk_start_sound_alsa (struct sound_dev ** pCapture, struct sound_dev ** pPlayback)
{
struct sound_dev * pDev;
memset(bufferz, 0, sizeof(int) * SAMP_BUFFER_SIZE);
is_little_endian = 1; // Test machine byte order
if (*(char *)&is_little_endian == 1)
is_little_endian = 1;
else
is_little_endian = 0;
if (quisk_sound_state.err_msg[0])
return; // prior error
// Open the alsa playback devices
while (1) {
pDev = *pPlayback++;
if ( ! pDev)
break;
if ( ! pDev->handle && pDev->driver == DEV_DRIVER_ALSA)
if (quisk_open_alsa_playback(pDev))
return; // error
}
// Open the alsa capture devices and start them
while (1) {
pDev = *pCapture++;
if ( ! pDev)
break;
if ( ! pDev->handle && pDev->driver == DEV_DRIVER_ALSA) {
if (quisk_open_alsa_capture(pDev))
return; // error
if (pDev->handle)
snd_pcm_start((snd_pcm_t *)pDev->handle);
}
}
}
void quisk_close_sound_alsa(struct sound_dev ** pCapture, struct sound_dev ** pPlayback)
{
struct sound_dev * pDev;
while (*pCapture) {
pDev = *pCapture;
if (pDev->handle && pDev->driver == DEV_DRIVER_ALSA) {
snd_pcm_drop((snd_pcm_t *)pDev->handle);
snd_pcm_close((snd_pcm_t *)pDev->handle);
}
pDev->handle = NULL;
pDev->driver = DEV_DRIVER_NONE;
pCapture++;
}
while (*pPlayback) {
pDev = *pPlayback;
if (pDev->handle && pDev->driver == DEV_DRIVER_ALSA) {
snd_pcm_drop((snd_pcm_t *)pDev->handle);
snd_pcm_close((snd_pcm_t *)pDev->handle);
}
pDev->handle = NULL;
pDev->driver = DEV_DRIVER_NONE;
pPlayback++;
}
}
void quisk_mixer_set(char * card_name, int numid, PyObject * value, char * err_msg, int err_size)
// Set card card_name mixer control numid to value for integer, boolean, enum controls.
// If value is a float, interpret value as a decimal fraction of min/max.
{
int err;
static snd_ctl_t * handle = NULL;
snd_ctl_elem_info_t *info;
snd_ctl_elem_id_t * id;
snd_ctl_elem_value_t * control;
unsigned int idx;
long imin, imax, tmp;
snd_ctl_elem_type_t type;
unsigned int count;
snd_ctl_elem_info_alloca(&info);
snd_ctl_elem_id_alloca(&id);
snd_ctl_elem_value_alloca(&control);
err_msg[0] = 0;
snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_MIXER);
snd_ctl_elem_id_set_numid(id, numid);
//snd_ctl_elem_id_set_index(id, index);
//snd_ctl_elem_id_set_device(id, device);
//snd_ctl_elem_id_set_subdevice(id, subdevice);
if ( ! strncmp (card_name, "alsa:", 5)) { // search for the name in info strings
char buf[QUISK_SC_SIZE];
strncpy(buf, card_name + 5, QUISK_SC_SIZE);
if ( ! device_list(NULL, SND_PCM_STREAM_CAPTURE, buf)) // check capture and play names
device_list(NULL, SND_PCM_STREAM_PLAYBACK, buf);
buf[4] = 0; // Remove device nuumber
err = snd_ctl_open(&handle, buf, 0);
}
else { // just try to open the name
err = snd_ctl_open(&handle, card_name, 0);
}
if (err < 0) {
snprintf (err_msg, err_size, "Control %s open error: %s\n", card_name, snd_strerror(err));
return;
}
snd_ctl_elem_info_set_id(info, id);
if ((err = snd_ctl_elem_info(handle, info)) < 0) {
snprintf (err_msg, err_size, "Cannot find the given element from control %s\n", card_name);
return;
}
snd_ctl_elem_info_get_id(info, id);
type = snd_ctl_elem_info_get_type(info);
snd_ctl_elem_value_set_id(control, id);
count = snd_ctl_elem_info_get_count(info);
for (idx = 0; idx < count; idx++) {
switch (type) {
case SND_CTL_ELEM_TYPE_BOOLEAN:
if (PyObject_IsTrue(value))
snd_ctl_elem_value_set_boolean(control, idx, 1);
else
snd_ctl_elem_value_set_boolean(control, idx, 0);
break;
case SND_CTL_ELEM_TYPE_INTEGER:
imin = snd_ctl_elem_info_get_min(info);
imax = snd_ctl_elem_info_get_max(info);
if (PyFloat_CheckExact(value)) {
tmp = (long)(imin + (imax - imin) * PyFloat_AsDouble(value) + 0.4);
snd_ctl_elem_value_set_integer(control, idx, tmp);
}
else if(PyInt_Check(value)) {
tmp = PyInt_AsLong(value);
snd_ctl_elem_value_set_integer(control, idx, tmp);
}
else {
snprintf (err_msg, err_size, "Control %s id %d has bad value\n", card_name, numid);
}
break;
case SND_CTL_ELEM_TYPE_INTEGER64:
imin = snd_ctl_elem_info_get_min64(info);
imax = snd_ctl_elem_info_get_max64(info);
if (PyFloat_CheckExact(value)) {
tmp = (long)(imin + (imax - imin) * PyFloat_AsDouble(value) + 0.4);
snd_ctl_elem_value_set_integer64(control, idx, tmp);
}
else if(PyInt_Check(value)) {
tmp = PyInt_AsLong(value);
snd_ctl_elem_value_set_integer64(control, idx, tmp);
}
else {
snprintf (err_msg, err_size, "Control %s id %d has bad value\n", card_name, numid);
}
break;
case SND_CTL_ELEM_TYPE_ENUMERATED:
if(PyInt_Check(value)) {
tmp = PyInt_AsLong(value);
snd_ctl_elem_value_set_enumerated(control, idx, (unsigned int)tmp);
}
else {
snprintf (err_msg, err_size, "Control %s id %d has bad value\n", card_name, numid);
}
break;
default:
snprintf (err_msg, err_size, "Control %s element has unknown type\n", card_name);
break;
}
if ((err = snd_ctl_elem_write(handle, control)) < 0) {
snprintf (err_msg, err_size, "Control %s element write error: %s\n", card_name, snd_strerror(err));
return;
}
}
snd_ctl_close(handle);
return;
}
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