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;
}
void *capture_thread(void *data) {
AudioCapture *capture = (AudioCapture *)data;
// enif_keep_resource(capture);
detect_pcm(capture);
set_volume(capture, 100);
snd_pcm_open(&capture->handle, capture->pcm_name, SND_PCM_STREAM_CAPTURE, 0);
if(!capture->handle) {
fprintf(stderr, "No PCM!!\r\n");
exit(1);
}
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_malloc(&hw_params);
if(!hw_params) {
fprintf(stderr, "Damn!! No hw_params\r\n");
exit(1);
}
snd_pcm_hw_params_any(capture->handle, hw_params);
snd_pcm_hw_params_set_access(capture->handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(capture->handle, hw_params, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate_near(capture->handle, hw_params, &capture->sample_rate, 0);
snd_pcm_hw_params_set_channels(capture->handle, hw_params, capture->channels);
// int buffer_size = 16384;
// int period_time = 32000;
// int period_size = 1024;
// snd_pcm_hw_params_set_period_time_near(capture->handle, hw_params, &period_time, 0);
// snd_pcm_hw_params_set_period_size_near(capture->handle, hw_params, &period_size, 0);
// snd_pcm_hw_params_set_buffer_size_near(capture->handle, hw_params, &buffer_size);
//
capture->last_dts = 0;
fprintf(stderr, "Setting params: %p, %p\r\n", capture->handle, hw_params);
snd_pcm_hw_params(capture->handle, hw_params);
snd_pcm_hw_params_free(hw_params);
snd_pcm_prepare(capture->handle);
snd_output_t *log;
snd_output_stdio_attach(&log, stderr, 0);
snd_pcm_hw_params_dump(hw_params, log);
fprintf(stderr, "Started capture\r\n");
char *buffer = (char *)malloc(8192);
// char *ptr = buffer;
int size = 0;
while(capture->thread_started) {
int r = snd_pcm_readi(capture->handle, buffer + size, 1024);
size += r*2*capture->channels;
if(size < capture->frame_size) {
continue;
}
ErlNifEnv* env = enif_alloc_env();
ErlNifBinary frame;
enif_alloc_binary(capture->frame_size, &frame);
memmove(frame.data, buffer, frame.size);
size -= frame.size;
memmove(buffer, buffer + frame.size, size);
ErlNifUInt64 dts = (uint64_t)capture->counter*1024ULL*1000 / capture->sample_rate;
// fprintf(stderr, "A: %d -> %d\r\n", capture->counter, dts);
if(capture->last_dts > dts) {
fprintf(stderr, "Achtung! ALSA audio jump: %u, %u, %u\r\n", (unsigned)capture->counter, (unsigned)capture->last_dts, (unsigned)dts);
}
capture->last_dts = dts;
enif_send(NULL, &capture->owner_pid, env,
enif_make_tuple4(env,
enif_make_atom(env, "alsa"),
enif_make_resource(env, capture),
enif_make_uint64(env, dts),
enif_make_binary(env, &frame)
)
);
enif_release_binary(&frame);
enif_free_env(env);
capture->counter++;
}
fprintf(stderr, "Capture thread stopping\r\n");
// enif_release_resource(capture);
snd_pcm_close(capture->handle);
return 0;
}
static void *alsa_thread_init(const char *device,
unsigned rate, unsigned latency)
{
snd_pcm_uframes_t buffer_size;
snd_pcm_format_t format;
snd_pcm_hw_params_t *params = NULL;
snd_pcm_sw_params_t *sw_params = NULL;
const char *alsa_dev = device ? device : "default";
unsigned latency_usec = latency * 1000 / 2;
unsigned channels = 2;
unsigned periods = 4;
alsa_thread_t *alsa = (alsa_thread_t*)
calloc(1, sizeof(alsa_thread_t));
if (!alsa)
return NULL;
TRY_ALSA(snd_pcm_open(&alsa->pcm, alsa_dev, SND_PCM_STREAM_PLAYBACK, 0));
TRY_ALSA(snd_pcm_hw_params_malloc(¶ms));
alsa->has_float = alsathread_find_float_format(alsa->pcm, params);
format = alsa->has_float ? SND_PCM_FORMAT_FLOAT : SND_PCM_FORMAT_S16;
TRY_ALSA(snd_pcm_hw_params_any(alsa->pcm, params));
TRY_ALSA(snd_pcm_hw_params_set_access(
alsa->pcm, params, SND_PCM_ACCESS_RW_INTERLEAVED));
TRY_ALSA(snd_pcm_hw_params_set_format(alsa->pcm, params, format));
TRY_ALSA(snd_pcm_hw_params_set_channels(alsa->pcm, params, channels));
TRY_ALSA(snd_pcm_hw_params_set_rate(alsa->pcm, params, rate, 0));
TRY_ALSA(snd_pcm_hw_params_set_buffer_time_near(
alsa->pcm, params, &latency_usec, NULL));
TRY_ALSA(snd_pcm_hw_params_set_periods_near(
alsa->pcm, params, &periods, NULL));
TRY_ALSA(snd_pcm_hw_params(alsa->pcm, params));
/* Shouldn't have to bother with this,
* but some drivers are apparently broken. */
if (snd_pcm_hw_params_get_period_size(params, &alsa->period_frames, NULL))
snd_pcm_hw_params_get_period_size_min(
params, &alsa->period_frames, NULL);
RARCH_LOG("ALSA: Period size: %d frames\n", (int)alsa->period_frames);
if (snd_pcm_hw_params_get_buffer_size(params, &buffer_size))
snd_pcm_hw_params_get_buffer_size_max(params, &buffer_size);
RARCH_LOG("ALSA: Buffer size: %d frames\n", (int)buffer_size);
alsa->buffer_size = snd_pcm_frames_to_bytes(alsa->pcm, buffer_size);
alsa->period_size = snd_pcm_frames_to_bytes(alsa->pcm, alsa->period_frames);
TRY_ALSA(snd_pcm_sw_params_malloc(&sw_params));
TRY_ALSA(snd_pcm_sw_params_current(alsa->pcm, sw_params));
TRY_ALSA(snd_pcm_sw_params_set_start_threshold(
alsa->pcm, sw_params, buffer_size / 2));
TRY_ALSA(snd_pcm_sw_params(alsa->pcm, sw_params));
snd_pcm_hw_params_free(params);
snd_pcm_sw_params_free(sw_params);
alsa->fifo_lock = slock_new();
alsa->cond_lock = slock_new();
alsa->cond = scond_new();
alsa->buffer = fifo_new(alsa->buffer_size);
if (!alsa->fifo_lock || !alsa->cond_lock || !alsa->cond || !alsa->buffer)
goto error;
alsa->worker_thread = sthread_create(alsa_worker_thread, alsa);
if (!alsa->worker_thread)
{
RARCH_ERR("error initializing worker thread");
goto error;
}
return alsa;
error:
RARCH_ERR("ALSA: Failed to initialize...\n");
if (params)
snd_pcm_hw_params_free(params);
if (sw_params)
snd_pcm_sw_params_free(sw_params);
alsa_thread_free(alsa);
return NULL;
}
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;
xassert (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;
const char *file = sd->file ? sd->file : DEFAULT_ALSA_SOUND_DEVICE;
if (snd_mixer_open (&handle, 0) >= 0)
{
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 int
set_hwparams (GstAlsaSink * alsa)
{
guint rrate;
gint err;
snd_pcm_hw_params_t *params;
guint period_time, buffer_time;
snd_pcm_hw_params_malloc (¶ms);
GST_DEBUG_OBJECT (alsa, "Negotiating to %d channels @ %d Hz (format = %s) "
"SPDIF (%d)", alsa->channels, alsa->rate,
snd_pcm_format_name (alsa->format), alsa->iec958);
/* start with requested values, if we cannot configure alsa for those values,
* we set these values to -1, which will leave the default alsa values */
buffer_time = alsa->buffer_time;
period_time = alsa->period_time;
retry:
/* choose all parameters */
CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config);
/* set the interleaved read/write format */
CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access),
wrong_access);
/* set the sample format */
if (alsa->iec958) {
/* Try to use big endian first else fallback to le and swap bytes */
if (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format) < 0) {
alsa->format = SND_PCM_FORMAT_S16_LE;
alsa->need_swap = TRUE;
GST_DEBUG_OBJECT (alsa, "falling back to little endian with swapping");
} else {
alsa->need_swap = FALSE;
}
}
CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format),
no_sample_format);
/* set the count of channels */
CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels),
no_channels);
/* set the stream rate */
rrate = alsa->rate;
CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL),
no_rate);
#ifndef GST_DISABLE_GST_DEBUG
/* get and dump some limits */
{
guint min, max;
snd_pcm_hw_params_get_buffer_time_min (params, &min, NULL);
snd_pcm_hw_params_get_buffer_time_max (params, &max, NULL);
GST_DEBUG_OBJECT (alsa, "buffer time %u, min %u, max %u",
alsa->buffer_time, min, max);
snd_pcm_hw_params_get_period_time_min (params, &min, NULL);
snd_pcm_hw_params_get_period_time_max (params, &max, NULL);
GST_DEBUG_OBJECT (alsa, "period time %u, min %u, max %u",
alsa->period_time, min, max);
snd_pcm_hw_params_get_periods_min (params, &min, NULL);
snd_pcm_hw_params_get_periods_max (params, &max, NULL);
GST_DEBUG_OBJECT (alsa, "periods min %u, max %u", min, max);
}
#endif
/* now try to configure the buffer time and period time, if one
* of those fail, we fall back to the defaults and emit a warning. */
if (buffer_time != -1 && !alsa->iec958) {
/* set the buffer time */
if ((err = snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params,
&buffer_time, NULL)) < 0) {
GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set buffer time %i for playback: %s",
buffer_time, snd_strerror (err)));
/* disable buffer_time the next round */
buffer_time = -1;
goto retry;
}
GST_DEBUG_OBJECT (alsa, "buffer time %u", buffer_time);
}
if (period_time != -1 && !alsa->iec958) {
/* set the period time */
if ((err = snd_pcm_hw_params_set_period_time_near (alsa->handle, params,
&period_time, NULL)) < 0) {
GST_ELEMENT_WARNING (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set period time %i for playback: %s",
period_time, snd_strerror (err)));
/* disable period_time the next round */
period_time = -1;
goto retry;
}
GST_DEBUG_OBJECT (alsa, "period time %u", period_time);
}
/* Set buffer size and period size manually for SPDIF */
if (G_UNLIKELY (alsa->iec958)) {
snd_pcm_uframes_t buffer_size = SPDIF_BUFFER_SIZE;
snd_pcm_uframes_t period_size = SPDIF_PERIOD_SIZE;
CHECK (snd_pcm_hw_params_set_buffer_size_near (alsa->handle, params,
&buffer_size), buffer_size);
CHECK (snd_pcm_hw_params_set_period_size_near (alsa->handle, params,
&period_size, NULL), period_size);
}
/* write the parameters to device */
CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params);
/* now get the configured values */
CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size),
buffer_size);
CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, NULL),
period_size);
GST_DEBUG_OBJECT (alsa, "buffer size %lu, period size %lu", alsa->buffer_size,
alsa->period_size);
snd_pcm_hw_params_free (params);
return 0;
/* ERRORS */
no_config:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Broken configuration for playback: no configurations available: %s",
snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
wrong_access:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Access type not available for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
no_sample_format:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Sample format not available for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
no_channels:
{
gchar *msg = NULL;
if ((alsa->channels) == 1)
msg = g_strdup (_("Could not open device for playback in mono mode."));
if ((alsa->channels) == 2)
msg = g_strdup (_("Could not open device for playback in stereo mode."));
if ((alsa->channels) > 2)
msg =
g_strdup_printf (_
("Could not open device for playback in %d-channel mode."),
alsa->channels);
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, ("%s", msg),
("%s", snd_strerror (err)));
g_free (msg);
snd_pcm_hw_params_free (params);
return err;
}
no_rate:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Rate %iHz not available for playback: %s",
alsa->rate, snd_strerror (err)));
return err;
}
buffer_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get buffer size for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
period_size:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to get period size for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
set_hw_params:
{
GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL),
("Unable to set hw params for playback: %s", snd_strerror (err)));
snd_pcm_hw_params_free (params);
return err;
}
}
static GF_Err ALSA_ConfigureOutput(GF_AudioOutput*dr, u32 *SampleRate, u32 *NbChannels, u32 *nbBitsPerSample, u32 channel_cfg)
{
snd_pcm_hw_params_t *hw_params = NULL;
int err;
int nb_bufs, sr, val, period_time;
ALSAContext *ctx = (ALSAContext*)dr->opaque;
if (!ctx) return GF_BAD_PARAM;
/*close device*/
if (ctx->playback_handle) {
snd_pcm_close(ctx->playback_handle);
ctx->playback_handle = NULL;
}
if (ctx->wav_buf) free(ctx->wav_buf);
ctx->wav_buf = NULL;
err = snd_pcm_open(&ctx->playback_handle, ctx->dev_name, SND_PCM_STREAM_PLAYBACK, 0/*SND_PCM_NONBLOCK*/);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot open audio device %s: %s\n", ctx->dev_name, snd_strerror (err)) );
return GF_IO_ERR;
}
err = snd_pcm_hw_params_malloc(&hw_params);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot allocate hardware params: %s\n", snd_strerror (err)) );
goto err_exit;
}
err = snd_pcm_hw_params_any(ctx->playback_handle, hw_params);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot initialize hardware params: %s\n", snd_strerror (err)) );
goto err_exit;
}
err = snd_pcm_hw_params_set_access(ctx->playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set access type: %s\n", snd_strerror (err)) );
goto err_exit;
}
/*set output format*/
ctx->nb_ch = (int) (*NbChannels);
ctx->block_align = ctx->nb_ch;
if ((*nbBitsPerSample) == 16) {
ctx->block_align *= 2;
err = snd_pcm_hw_params_set_format(ctx->playback_handle, hw_params, SND_PCM_FORMAT_S16_LE);
} else {
err = snd_pcm_hw_params_set_format(ctx->playback_handle, hw_params, SND_PCM_FORMAT_U8);
}
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set sample format: %s\n", snd_strerror (err)) );
goto err_exit;
}
/*set output sample rate*/
if (ctx->force_sr) *SampleRate = ctx->force_sr;
sr = *SampleRate;
err = snd_pcm_hw_params_set_rate_near(ctx->playback_handle, hw_params, SampleRate, 0);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set sample rate: %s\n", snd_strerror (err)) );
goto err_exit;
}
if (sr != *SampleRate) {
GF_LOG(GF_LOG_INFO, GF_LOG_MMIO, ("[ALSA] Sample rate %d not supported, using %d instead\n", sr, *SampleRate ) );
sr = *SampleRate;
}
/*set output channels*/
err = snd_pcm_hw_params_set_channels_near(ctx->playback_handle, hw_params, NbChannels);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set channel count: %s\n", snd_strerror (err)) );
goto err_exit;
}
if (ctx->nb_ch != *NbChannels) {
GF_LOG(GF_LOG_INFO, GF_LOG_MMIO, ("[ALSA] %d channels not supported - using %d instead\n", ctx->nb_ch, *NbChannels ) );
ctx->block_align /= ctx->nb_ch;
ctx->nb_ch = *NbChannels;
ctx->block_align *= ctx->nb_ch;
}
err = snd_pcm_hw_params(ctx->playback_handle, hw_params);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set parameters: %s\n", snd_strerror (err)) );
goto err_exit;
}
/* Set number of buffers*/
snd_pcm_hw_params_get_periods_min(hw_params, &val, 0);
nb_bufs = (ctx->num_buffers>val) ? ctx->num_buffers : val;
err = snd_pcm_hw_params_set_periods_near(ctx->playback_handle, hw_params, &nb_bufs, 0);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set number of HW buffers (%d): %s\n", nb_bufs, snd_strerror(err) ));
goto err_exit;
}
/* Set total buffer size*/
if (ctx->total_duration) {
ctx->buf_size = (sr * ctx->total_duration)/1000;
} else {
ctx->buf_size = 2048;
}
ctx->buf_size /= nb_bufs;
err = snd_pcm_hw_params_set_period_size_near(ctx->playback_handle, hw_params, (snd_pcm_uframes_t *)&ctx->buf_size, 0);
if (err < 0) {
GF_LOG(GF_LOG_ERROR, GF_LOG_MMIO, ("[ALSA] Cannot set HW buffer size (%d): %s\n", ctx->buf_size, snd_strerror(err) ));
goto err_exit;
}
/*get complete buffer size*/
snd_pcm_hw_params_get_buffer_size(hw_params, (snd_pcm_uframes_t *)&ctx->buf_size);
ctx->buf_size *= ctx->block_align;
/*get period time*/
snd_pcm_hw_params_get_period_time(hw_params, &period_time, 0);
snd_pcm_hw_params_free (hw_params);
hw_params = NULL;
ctx->delay = (ctx->buf_size*1000) / (sr*ctx->block_align);
/*allocate a single buffer*/
ctx->wav_buf = malloc(ctx->buf_size*sizeof(char));
if(!ctx->wav_buf) return GF_OUT_OF_MEM;
memset(ctx->wav_buf, 0, ctx->buf_size*sizeof(char));
GF_LOG(GF_LOG_DEBUG, GF_LOG_MMIO, ("[ALSA] Setup %d ch @ %d hz - %d periods of %d us - total buffer size %d - overall delay %d ms\n", ctx->nb_ch, sr, nb_bufs, period_time, ctx->buf_size, ctx->delay));
return GF_OK;
err_exit:
if (hw_params) snd_pcm_hw_params_free(hw_params);
snd_pcm_close(ctx->playback_handle);
ctx->playback_handle = NULL;
return GF_IO_ERR;
}
AudioSourceNix::~AudioSourceNix()
{
snd_pcm_hw_params_free(params);
}
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);
}
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;
}
/**************************************************************************
* 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);
}
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 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;
}
AudioCaptureNode::~AudioCaptureNode()
{
if(buffer) delete buffer;
snd_pcm_hw_params_free(hwParams);
snd_pcm_close(captureHandle);
}
static void
init_audio(const char* devname)
{
snd_pcm_hw_params_t* hwparams = 0;
snd_pcm_sw_params_t* swparams = 0;
snd_pcm_uframes_t bufsize = 0;
unsigned int buftime = 1000000;
unsigned int pertime = 50000;
int dir = 0;
int rc;
if ((rc = snd_output_stdio_attach(&output, stderr, 0)) < 0)
exit_snd_error(rc, "log output");
if ((rc = snd_pcm_open(&audio, devname, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
exit_snd_error(rc, "opening device");
if ((rc = snd_pcm_hw_params_malloc(&hwparams)) < 0)
exit_snd_error(rc, "hardware parameters");
if ((rc = snd_pcm_hw_params_any(audio, hwparams)) < 0)
exit_snd_error(rc, "hardware parameters");
if ((rc = snd_pcm_hw_params_set_rate_resample(audio, hwparams, 0)) < 0)
exit_snd_error(rc, "hardware parameters");
if ((rc = snd_pcm_hw_params_set_access(audio, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
exit_snd_error(rc, "access type");
if ((rc = snd_pcm_hw_params_set_format(audio, hwparams, SND_PCM_FORMAT_S16)) < 0)
exit_snd_error(rc, "sample format");
if ((rc = snd_pcm_hw_params_set_channels_near(audio, hwparams, &n_channels)) < 0)
exit_snd_error(rc, "number of channels");
if ((rc = snd_pcm_hw_params_set_rate_near(audio, hwparams, &samplerate, &dir)) < 0)
exit_snd_error(rc, "sample rate");
if ((rc = snd_pcm_hw_params_set_buffer_time_near(audio, hwparams, &buftime, &dir)) < 0)
exit_snd_error(rc, "buffer time");
if ((rc = snd_pcm_hw_params_set_period_time_near(audio, hwparams, &pertime, &dir)) < 0)
exit_snd_error(rc, "period time");
if ((rc = snd_pcm_hw_params(audio, hwparams)) < 0)
exit_snd_error(rc, "applying hardware parameters");
if ((rc = snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize)) < 0)
exit_snd_error(rc, "buffer size");
if ((rc = snd_pcm_hw_params_get_period_size(hwparams, &periodsize, &dir)) < 0)
exit_snd_error(rc, "period size");
snd_pcm_hw_params_free(hwparams);
if ((rc = snd_pcm_sw_params_malloc(&swparams)) < 0)
exit_snd_error(rc, "software parameters");
if ((rc = snd_pcm_sw_params_current(audio, swparams)) < 0)
exit_snd_error(rc, "software parameters");
if ((rc = snd_pcm_sw_params_set_start_threshold(audio, swparams,
bufsize / periodsize * periodsize)) < 0)
exit_snd_error(rc, "start threshold");
if ((rc = snd_pcm_sw_params(audio, swparams)) < 0)
exit_snd_error(rc, "applying software parameters");
snd_pcm_sw_params_free(swparams);
if ((rc = snd_pcm_prepare(audio)) < 0)
exit_snd_error(rc, "preparing device");
}
av_cold int ff_alsa_open(AVFormatContext *ctx, snd_pcm_stream_t mode,
unsigned int *sample_rate,
int channels, enum AVCodecID *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;
uint64_t layout = ctx->streams[0]->codecpar->channel_layout;
if (ctx->filename[0] == 0) audio_device = "default";
else audio_device = ctx->filename;
if (*codec_id == AV_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);
}
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;
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_IO;
}
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);
/* 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);
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);
s->h = h;
return 0;
fail:
snd_pcm_hw_params_free(hw_params);
fail1:
snd_pcm_close(h);
return AVERROR_IO;
}
static void *_msynth_thread_main(void *arg)
{
int i;
int err;
int sample;
int processed;
struct sampleclock sc = {0, 0, 0.0f, 0.0f};
msynth_frame fb = NULL;
puts("synthread: started");
/* Begin initialization of ALSA */
/* snd_pcm_open(pcm_handle, device_name, stream_type, open_mode) */
err = snd_pcm_open(&pcm, config.device_name, SND_PCM_STREAM_PLAYBACK, 0);
ALSERT("opening audio device");
/* First ALSA hardware settings */
/* Allocate HW params */
err = snd_pcm_hw_params_malloc(&hw_p);
ALSERT("allocating hw params");
/* Get default HW parameters */
err = snd_pcm_hw_params_any(pcm, hw_p);
ALSERT("requesting hw default params");
/* Disable software resampling */
err = snd_pcm_hw_params_set_rate_resample(pcm, hw_p, config.resample);
ALSERT("disabling software resampling");
/* Configure sample rate */
if (config.srate != -1)
srate = config.srate;
else {
/* Get maximum hardware samplerate */
err = snd_pcm_hw_params_get_rate_max(hw_p, &srate, &dir);
ALSERT("requesting maximum hardware samplerate");
}
/* Set sample rate
* snd_pcm_hw_params_set_rate_near(pcn, hw_p, *rate,
* 0 (== set exact rate))
*/
err = snd_pcm_hw_params_set_rate_near(pcm, hw_p, &srate, 0);
ALSERT("setting samplerate");
printf("synthread: Detected samplerate of %u\n", srate);
/* RW interleaved access (means we will use the snd_pcm_writei function) */
err = snd_pcm_hw_params_set_access(pcm, hw_p,
SND_PCM_ACCESS_RW_INTERLEAVED);
ALSERT("setting access mode");
/* native-endian 16-bit signed sample format */
err = snd_pcm_hw_params_set_format(pcm, hw_p, SND_PCM_FORMAT_S16);
ALSERT("setting sample format");
/* Switch to 2.0ch audio */
err = snd_pcm_hw_params_set_channels(pcm, hw_p, 2);
ALSERT("switching to 2.0ch audio");
if (config.buffer_time != -1) {
/* Set configured buffer time */
err = snd_pcm_hw_params_set_buffer_time_near(pcm, hw_p,
&config.buffer_time, &dir);
ALSERT("setting buffer time");
/* Fetch resulting size */
err = snd_pcm_hw_params_get_buffer_size(hw_p, &buffer_size);
ALSERT("getting buffer size");
} else {
/* Since we want the interactive synthesizer to be very responsive
* select a minimum buffer size.
*/
err = snd_pcm_hw_params_set_buffer_size_first(pcm, hw_p, &buffer_size);
ALSERT("setting buffer size");
}
printf("synthread: Selected buffersize of %lu\n", buffer_size);
printf("synthread: Response delay is approximately %.2f ms\n",
(double)buffer_size / (double)srate * 1000.0);
if (config.period_time != -1) {
err = snd_pcm_hw_params_set_period_time_near(pcm, hw_p,
&config.period_time, &dir);
ALSERT("setting period time");
err = snd_pcm_hw_params_get_period_size(hw_p, &period_size, &dir);
ALSERT("getting period size");
} else {
/* Select a period time of half the buffer time
* since this improves processing performance
*/
err = snd_pcm_hw_params_set_period_size_last(pcm, hw_p, &period_size, &dir);
ALSERT("setting period size");
}
if (dir)
printf("synthread: Selected period size near %lu\n", period_size);
else
printf("synthread: Selected period size of %lu\n", period_size);
/* write hw parameters to device */
err = snd_pcm_hw_params(pcm, hw_p);
ALSERT("writing hw params");
/* Begin ALSA software side setup */
/* Allocate SW params */
err = snd_pcm_sw_params_malloc(&sw_p);
ALSERT("allocating sw params");
/* Request software current settings */
err = snd_pcm_sw_params_current(pcm, sw_p);
ALSERT("requesting current sw params");
/* Automatically start playing after the first period is written */
err = snd_pcm_sw_params_set_start_threshold(pcm, sw_p, period_size);
ALSERT("settings start threshold");
/* XRUN when buffer is empty */
err = snd_pcm_sw_params_set_stop_threshold(pcm, sw_p, buffer_size * 2);
ALSERT("setting stop threshold");
/* Block synthesizer when there is not at least period frames available */
err = snd_pcm_sw_params_set_avail_min(pcm, sw_p, period_size);
ALSERT("setting minimum free frames");
/* Write software parameters */
err = snd_pcm_sw_params(pcm, sw_p);
ALSERT("writing sw params");
printf("synthread: Audio system configured.\n");
/* Prepare device for playback */
err = snd_pcm_prepare(pcm);
ALSERT("preparing device");
/* Allocate a period sized framebuffer
* (yup an audio buffer is in ALSA speak indeed called a framebuffer)
*/
fb = malloc(sizeof(struct _msynth_frame) * period_size);
if (!fb) {
perror("malloc framebuffer failed");
exit(1);
}
/* Set sampleclock to 0 */
sc.samples = 0;
sc.samplerate = srate;
sc.cycle = 0.0f;
sc.seconds = 0.0f;
/* Initially the <root> variable describing the flow of sound within
* the synthesizer is configured to be a NULL signal. (silence)
*/
/* Signal successful completion of initialization */
pthread_mutex_lock(&mutex);
pthread_cond_signal(&cond);
pthread_mutex_unlock(&mutex);
/* -------------- Main loop --------------- */
while (!shutdown) {
/* Only during generation we need the synth tree to be static */
synth_lock_graphs();
for (i = 0; i < period_size; i++) {
/* Evaluate variables */
ssv_eval(sc);
sample = (int)(32767.5 * ssv_get_var_eval("left") * volume);
/* Clip samples */
if (sample > 32767) sample = 32767;
if (sample < -32768) sample = -32768;
fb[i].left = (short)sample;
sample = (int)(32767.5 * ssv_get_var_eval("right") * volume);
/* Clip samples */
if (sample > 32767) sample = 32767;
if (sample < -32768) sample = -32768;
fb[i].right = (short)sample;
sc = sc_from_samples(sc.samplerate, sc.samples + 1);
}
synth_unlock_graphs();
/* Send audio to sound card */
processed = 0;
while (processed != period_size) {
err = snd_pcm_writei(pcm, fb + processed, period_size - processed);
/* Retry on interruption by signal */
if (err == -EAGAIN)
continue;
/* Recover from XRUN/suspend */
if (err < 0) {
err = synth_recover(err);
ALSERT("sending audio");
continue;
}
/* Update processed samples */
processed += err;
}
}
puts("synthread: shutting down");
/* Dump any statistics recorded */
if (recover_resumes)
printf("synthread: Device was resumed %i times\n", recover_resumes);
if (recover_xruns)
printf("synthread: %i xrun recoveries were needed\n", recover_xruns);
printf("synthread: processed %i samples\n", sc.samples);
/* Wait for playback to complete */
err = snd_pcm_drain(pcm);
ALSERT("draining device");
/* Shutdown PCM device */
err = snd_pcm_close(pcm);
ALSERT("closing audio device");
/* Clean up parameters */
snd_pcm_hw_params_free(hw_p);
snd_pcm_sw_params_free(sw_p);
return NULL;
}
snd_pcm_t *
setup_alsa_capture()
{
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
int err;
char *device = DEFAULT_DEVICE;
uint32_t rate = RATE;
uint32_t channels = CHANNELS;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
if ((err = snd_pcm_open (&capture_handle, device, SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fatal_tragedy(1, "cannot open audio device %s (%s)", device, snd_strerror (err));
}
log_info("audio interface opened");
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fatal_tragedy(1 , "cannot allocate hardware parameter structure (%s)", snd_strerror (err));
}
log_info("hw_params allocated");
if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0) {
fatal_tragedy(1, "cannot initialize hardware parameter structure (%s)", snd_strerror (err));
}
log_info("hw_params initialized");
if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fatal_tragedy(1, "cannot set access type (%s)", snd_strerror (err));
}
log_info("hw_params access setted");
if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, format)) < 0) {
fatal_tragedy(1, "cannot set sample format (%s)", snd_strerror (err));
}
fprintf(stdout, "hw_params format setted");
if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &rate, 0)) < 0) {
fatal_tragedy(1, "cannot set sample rate (%s)", snd_strerror (err));
}
log_info("hw_params rate setted");
if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, channels)) < 0) {
fatal_tragedy(1, "cannot set channel count (%s)", snd_strerror (err));
}
log_info("hw_params channels setted");
if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0) {
fatal_tragedy(1, "cannot set parameters (%s)", snd_strerror (err));
}
fprintf(stdout, "hw_params setted");
snd_pcm_hw_params_free (hw_params);
if ((err = snd_pcm_prepare (capture_handle)) < 0) {
fatal_tragedy(1, "cannot prepare audio interface for use (%s)", snd_strerror (err));
}
log_info("audio interface prepared");
return capture_handle;
}
static snd_pcm_t *alsa_open(snd_pcm_t **pcm_handle, int stream_type,
snd_pcm_uframes_t *buffer_size) {
char *pcm_name = "chumix";
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
unsigned int buffer_time;
int status;
snd_pcm_uframes_t xfer_align;
// Open the device. Experimenting with duplex mode.
fprintf(stderr, "Opening PCM device\n");
status = snd_pcm_open(pcm_handle, pcm_name,
stream_type, 0);
if(status < 0) {
fprintf(stderr, "Unable to open audio device: %s\n",
snd_strerror(status));
goto error;
}
// Determine what the hardware can do.
fprintf(stderr, "Allocating hw params\n");
snd_pcm_hw_params_malloc(&hwparams);
status = snd_pcm_hw_params_any(*pcm_handle, hwparams);
if(status < 0) {
fprintf(stderr, "Unable to get hardware parameters: %s\n",
snd_strerror(status));
goto error;
}
// Set up interleaved audio.
fprintf(stderr, "Setting up interleaved audio\n");
status = snd_pcm_hw_params_set_access(*pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if(status < 0) {
fprintf(stderr, "Unable to set interleaved mode: %s\n",
snd_strerror(status));
goto error;
}
// Set the audio format.
fprintf(stderr, "Setting format\n");
status = snd_pcm_hw_params_set_format(*pcm_handle, hwparams,
SND_PCM_FORMAT_S16_LE);
if(status<0) {
fprintf(stderr, "Unable to set audio format: %s\n",
snd_strerror(status));
goto error;
}
// Set stereo audio.
fprintf(stderr, "Setting channels\n");
status = snd_pcm_hw_params_set_channels(*pcm_handle, hwparams, 2);
if(status<0) {
fprintf(stderr, "Unable to set stereo mode: %s\n",
snd_strerror(status));
goto error;
}
// Set sample rate.
fprintf(stderr, "Setting audio rate\n");
int rate = 44100;
status = snd_pcm_hw_params_set_rate_near(*pcm_handle, hwparams,
&rate, NULL);
if(status<0) {
fprintf(stderr, "Unable to set sample rate to 44100: %s\n",
snd_strerror(status));
goto error;
}
// Set buffer time to a reasonable value.
// Probably won't work for playback, as it's fixed in /etc/asound.conf
buffer_time = BUFFER_TIME;
status = snd_pcm_hw_params_set_buffer_time_near(*pcm_handle, hwparams,
&buffer_time, 0);
snd_pcm_hw_params_get_buffer_size(hwparams, buffer_size);
// Write the settings out to the audio card.
fprintf(stderr, "Setting hw params\n");
status = snd_pcm_hw_params(*pcm_handle, hwparams);
if(status<0) {
fprintf(stderr, "Unable to set audio parameters: %s\n",
snd_strerror(status));
goto error;
}
snd_pcm_hw_params_free(hwparams);
status = snd_pcm_prepare(*pcm_handle);
if(status<0) {
fprintf(stderr, "Unable to prepare audio device: %s\n",
snd_strerror(status));
goto error;
}
return *pcm_handle;
error:
if(*pcm_handle) {
snd_pcm_drain(*pcm_handle);
snd_pcm_close(*pcm_handle);
}
return NULL;
}
/*------------------------------------------------------------------------------
* Open the audio source
*----------------------------------------------------------------------------*/
bool
AlsaDspSource :: open ( void ) throw ( Exception )
{
unsigned int u;
snd_pcm_format_t format;
snd_pcm_hw_params_t *hwParams;
if ( isOpen() ) {
return false;
}
switch ( getBitsPerSample() ) {
case 8:
format = SND_PCM_FORMAT_S8;
break;
case 16:
format = SND_PCM_FORMAT_S16;
break;
default:
return false;
}
if (snd_pcm_open(&captureHandle, pcmName, SND_PCM_STREAM_CAPTURE, 0) < 0) {
captureHandle = 0;
return false;
}
if (snd_pcm_hw_params_malloc(&hwParams) < 0) {
close();
throw Exception( __FILE__, __LINE__, "can't alloc hardware "\
"parameter structure");
}
if (snd_pcm_hw_params_any(captureHandle, hwParams) < 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't initialize hardware "\
"parameter structure");
}
if (snd_pcm_hw_params_set_access(captureHandle, hwParams,
SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set access type");
}
if (snd_pcm_hw_params_set_format(captureHandle, hwParams, format) < 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set sample format");
}
u = getSampleRate();
if (snd_pcm_hw_params_set_rate_near(captureHandle, hwParams, &u, 0) < 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set sample rate", u);
}
u = getChannel();
if (snd_pcm_hw_params_set_channels(captureHandle, hwParams, u) < 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set channels", u);
}
u = 4;
if (snd_pcm_hw_params_set_periods_near(captureHandle, hwParams, &u, 0)
< 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set interrupt frequency");
}
u = getBufferTime();
if (snd_pcm_hw_params_set_buffer_time_near(captureHandle, hwParams, &u, 0)
< 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set buffer size");
}
if (snd_pcm_hw_params(captureHandle, hwParams) < 0) {
snd_pcm_hw_params_free(hwParams);
close();
throw Exception( __FILE__, __LINE__, "can't set hardware parameters");
}
snd_pcm_hw_params_free(hwParams);
if (snd_pcm_prepare(captureHandle) < 0) {
close();
throw Exception( __FILE__, __LINE__, "can't prepare audio interface "\
"for use");
}
bytesPerFrame = getChannel() * getBitsPerSample() / 8;
return true;
}
short *Audio_ALSA::open (AudioConfig &cfg, const char *)
{
AudioConfig tmpCfg;
if (_audioHandle != NULL)
{
_errorString = "ERROR: Device already in use";
return NULL;
}
snd_pcm_uframes_t buffer_frames;
snd_pcm_hw_params_t *hw_params = 0;
if (snd_pcm_open (&_audioHandle, "default", SND_PCM_STREAM_PLAYBACK, 0))
{
_errorString = "ERROR: Could not open audio device.";
goto open_error;
}
// May later be replaced with driver defaults.
tmpCfg = cfg;
if (snd_pcm_hw_params_malloc (&hw_params))
{
_errorString = "ERROR: could not malloc hwparams.";
goto open_error;
}
if (snd_pcm_hw_params_any (_audioHandle, hw_params))
{
_errorString = "ERROR: could not initialize hw params";
goto open_error;
}
if (snd_pcm_hw_params_set_access (_audioHandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED))
{
_errorString = "ERROR: could not set access type";
goto open_error;
}
if (snd_pcm_hw_params_set_format (_audioHandle, hw_params, SND_PCM_FORMAT_S16_LE))
{
_errorString = "ERROR: could not set sample format";
goto open_error;
}
if (snd_pcm_hw_params_set_channels (_audioHandle, hw_params, tmpCfg.channels))
{
_errorString = "ERROR: could not set channel count";
goto open_error;
}
{ // Gentoo bug #98769, comment 4
unsigned int rate = tmpCfg.frequency;
if (snd_pcm_hw_params_set_rate_near (_audioHandle, hw_params, &rate, 0))
{
_errorString = "ERROR: could not set sample rate";
goto open_error;
}
}
_alsa_to_frames_divisor = tmpCfg.channels;
buffer_frames = 4096;
snd_pcm_hw_params_set_period_size_near(_audioHandle, hw_params, &buffer_frames, 0);
if (snd_pcm_hw_params (_audioHandle, hw_params))
{
_errorString = "ERROR: could not set hw parameters";
goto open_error;
}
snd_pcm_hw_params_free (hw_params);
hw_params = 0;
if (snd_pcm_prepare (_audioHandle))
{
_errorString = "ERROR: could not prepare audio interface for use";
goto open_error;
}
tmpCfg.bufSize = buffer_frames * _alsa_to_frames_divisor;
#ifdef HAVE_EXCEPTIONS
_sampleBuffer = new(std::nothrow) short[tmpCfg.bufSize];
#else
_sampleBuffer = new short[tmpCfg.bufSize];
#endif
if (!_sampleBuffer)
{
_errorString = "AUDIO: Unable to allocate memory for sample buffers.";
goto open_error;
}
// Setup internal Config
_settings = tmpCfg;
// Update the users settings
getConfig (cfg);
return _sampleBuffer;
open_error:
if (hw_params)
snd_pcm_hw_params_free (hw_params);
if (_audioHandle != NULL)
close ();
perror ("ALSA");
return NULL;
}
int main(int argc, char** argv)
{
snd_pcm_t *handle = NULL;
snd_pcm_hw_params_t *params = NULL;
snd_pcm_uframes_t frames = 64;
int dir = 0, size, loops;
unsigned int val;
int ret = 0, n = 0;
unsigned char *buf = NULL;
FILE *fpcm = NULL;
unsigned int rate;
int format;
int channel;
int time;
/*
int i;
for(i = 0; i < argc; i++)
AUDIO_ALSA_DBG("param%d: %s\n", i, *(argv + i));
*/
if(7 != argc)
{
AUDIO_ALSA_DBG("\n");
AUDIO_ALSA_DBG("param1 snd dev: alsa device name: default, plughw:0,0 or plughw:1,0\n");
AUDIO_ALSA_DBG("param2 filename: file used to save audio pcm data\n");
AUDIO_ALSA_DBG("param3 sampleRate: 8000 -- 48000 \n");
AUDIO_ALSA_DBG("param4 channel: 1:mono, 2:stereo\n");
AUDIO_ALSA_DBG("param5 format: 1:SND_PCM_FORMAT_U8, 2:SND_PCM_FORMAT_S16_LE\n");
AUDIO_ALSA_DBG("param6 record time: record time(unit: second)\n\n");
return -1;
}
rate = strtoul(*(argv + 3), NULL, 10);
channel = strtoul(*(argv + 4), NULL, 10);
format = strtoul(*(argv + 5), NULL, 10);
time = strtoul(*(argv + 6), NULL, 10);
if(rate < 8000 || rate > 48000)
{
AUDIO_ALSA_DBG("param2 sampleRate error: 8000 -- 48000 \n");
ret = -1;
goto error;
}
if(channel < 1 || channel > 2)
{
AUDIO_ALSA_DBG("param3 channel error: 1:mono, 2:stereo\n");
ret = -1;
goto error;
}
if(format < SND_PCM_FORMAT_U8 || format > SND_PCM_FORMAT_S16_LE)
{
AUDIO_ALSA_DBG("param4 format error: 1:SND_PCM_FORMAT_U8, 2:SND_PCM_FORMAT_S16_LE\n");
ret = -1;
goto error;
}
AUDIO_ALSA_DBG("rate = %u channel = %s format = %s time = %d\n", rate, (1 == channel) ? "mono" : "stereo",
(1 == format) ? "SND_PCM_FORMAT_U8" : "SND_PCM_FORMAT_S16_LE", time);
if((ret = snd_pcm_open(&handle, *(argv + 1), SND_PCM_STREAM_CAPTURE, 0)) < 0)
{
AUDIO_ALSA_DBG("snd_pcm_open error: %s\n", snd_strerror(ret));
handle = NULL;
goto error;
}
snd_pcm_hw_params_malloc(¶ms);
if(!params)
{
AUDIO_ALSA_DBG("snd_pcm_hw_params_alloca error ... \n");
ret = -1;
goto error;
}
snd_pcm_hw_params_any(handle, params);
snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
if((ret = snd_pcm_hw_params_set_format(handle ,params, format)) < 0)
{
AUDIO_ALSA_DBG("format set error: %s\n", snd_strerror(ret));
goto error;
}
if((ret = snd_pcm_hw_params_set_channels(handle, params, channel)) < 0)
{
AUDIO_ALSA_DBG("channel set error: %s\n", snd_strerror(ret));
goto error;
}
if((ret = snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir)) < 0)
{
AUDIO_ALSA_DBG("sampleRate set error: %s\n", snd_strerror(ret));
goto error;
}
snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir);
if((ret = snd_pcm_hw_params(handle, params)) < 0)
{
AUDIO_ALSA_DBG("snd_pcm_hw_params error: %s\n", snd_strerror(ret));
goto error;
}
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
size = frames * channel * format;
buf = (unsigned char*)malloc(size);
if(!buf)
{
AUDIO_ALSA_DBG("malloc buf error: buf size = %d\n", size);
ret = -1;
goto error;
}
snd_pcm_hw_params_get_period_time(params, &val, &dir);
loops = time * 1000000 / val;
AUDIO_ALSA_DBG("loops = %d\n", loops);
fpcm = fopen(*(argv + 2), "wb");
if(!fpcm)
{
AUDIO_ALSA_DBG("Can not open file %s ... \n", *(argv + 2));
goto error;
}
while(loops-- > 0)
{
n = snd_pcm_readi(handle, buf, frames);
if(-EPIPE == n)
{
AUDIO_ALSA_DBG("snd_pcm_readi over run ... \n");
snd_pcm_prepare(handle);
continue;
}
else if(n < 0)
{
AUDIO_ALSA_DBG("snd_pcm_readi error: %s\n", snd_strerror(n));
ret = -1;
break;
}
else if(n != frames)
{
AUDIO_ALSA_DBG("snd_pcm_readi read frames %d, less than %u\n", n, (unsigned int)frames);
}
fwrite(buf, 1, size, fpcm);
}
error:
if(params)
snd_pcm_hw_params_free(params);
if(handle)
snd_pcm_close(handle);
if(buf)
free(buf);
if(fpcm)
fclose(fpcm);
return ret;
}
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);
}
static void OpenAlsa()
{
int i;
int err;
short buf[128];
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
int card=0;
int dev=0;
if ((err = snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 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 (capture_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 (capture_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 (capture_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 (capture_handle, hw_params, FMT_dwSamplesPerSec, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params (capture_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 (capture_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_readi (capture_handle, buf, 128)) != 128) {
fprintf (stderr, "read from audio interface failed (%s)\n",
snd_strerror (err));
exit (1);
}
}
snd_pcm_close (capture_handle);
exit (0);
}
eBool tfAlsa::openDevice(const eChar *device, eU32 freq, eU32 bits, eU32 channels)
{
eInt err = 0;
if (device == eNULL)
return eFALSE;
printf ("opening device '%s' for playback\n", device);
if ((err = snd_pcm_open (&m_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
fprintf (stderr, "cannot open audio device '%s' for playback(%s)\n", device, snd_strerror (err));
return eFALSE;
}
snd_pcm_hw_params_t *hw_params;
snd_pcm_format_t format;
snd_pcm_uframes_t buffer_size = BUFFERSIZE;
snd_pcm_uframes_t period_size = PERIODSIZE;
printf("setting device to %i hz %i bits %i channels\n", freq, bits, channels);
if (bits == 16)
format = SND_PCM_FORMAT_S16_LE;
else
format = SND_PCM_FORMAT_S8;
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_any (m_handle, hw_params)) < 0)
{
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_set_access (m_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_set_format (m_handle, hw_params, format)) < 0)
{
fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_set_rate_near (m_handle, hw_params, (unsigned int *)&freq, 0)) < 0)
{
fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_set_channels (m_handle, hw_params, channels)) < 0)
{
fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_set_buffer_size_near (m_handle, hw_params, &buffer_size)) < 0)
{
fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params_set_period_size_near (m_handle, hw_params, &period_size, NULL)) < 0)
{
fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err));
return false;
}
if ((err = snd_pcm_hw_params (m_handle, hw_params)) < 0)
{
fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err));
return false;
}
snd_pcm_hw_params_free (hw_params);
return eTRUE;
}
static int
palsa_set_hw_params (ddb_waveformat_t *fmt) {
snd_pcm_hw_params_t *hw_params = NULL;
int err = 0;
memcpy (&plugin.fmt, fmt, sizeof (ddb_waveformat_t));
if (!plugin.fmt.channels) {
// generic format
plugin.fmt.bps = 16;
plugin.fmt.is_float = 0;
plugin.fmt.channels = 2;
plugin.fmt.samplerate = 44100;
plugin.fmt.channelmask = 3;
}
snd_pcm_nonblock(audio, 0);
snd_pcm_drain (audio);
snd_pcm_nonblock(audio, 1);
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
goto error;
}
if ((err = snd_pcm_hw_params_any (audio, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
goto error;
}
if ((err = snd_pcm_hw_params_set_access (audio, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
goto error;
}
snd_pcm_format_t sample_fmt;
switch (plugin.fmt.bps) {
case 8:
sample_fmt = SND_PCM_FORMAT_S8;
break;
case 16:
#if WORDS_BIGENDIAN
sample_fmt = SND_PCM_FORMAT_S16_BE;
#else
sample_fmt = SND_PCM_FORMAT_S16_LE;
#endif
break;
case 24:
#if WORDS_BIGENDIAN
sample_fmt = SND_PCM_FORMAT_S24_3BE;
#else
sample_fmt = SND_PCM_FORMAT_S24_3LE;
#endif
break;
case 32:
if (plugin.fmt.is_float) {
#if WORDS_BIGENDIAN
sample_fmt = SND_PCM_FORMAT_FLOAT_BE;
#else
sample_fmt = SND_PCM_FORMAT_FLOAT_LE;
#endif
}
else {
#if WORDS_BIGENDIAN
sample_fmt = SND_PCM_FORMAT_S32_BE;
#else
sample_fmt = SND_PCM_FORMAT_S32_LE;
#endif
}
break;
}
if ((err = snd_pcm_hw_params_set_format (audio, hw_params, sample_fmt)) < 0) {
fprintf (stderr, "cannot set sample format to %d bps (error: %s), trying all supported formats\n", plugin.fmt.bps, snd_strerror (err));
int fmt_cnt[] = { 16, 24, 32, 32, 8 };
#if WORDS_BIGENDIAN
int fmt[] = { SND_PCM_FORMAT_S16_BE, SND_PCM_FORMAT_S24_3BE, SND_PCM_FORMAT_S32_BE, SND_PCM_FORMAT_FLOAT_BE, SND_PCM_FORMAT_S8, -1 };
#else
int fmt[] = { SND_PCM_FORMAT_S16_LE, SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S32_LE, SND_PCM_FORMAT_FLOAT_LE, SND_PCM_FORMAT_S8, -1 };
#endif
// 1st try formats with higher bps
int i = 0;
for (i = 0; fmt[i] != -1; i++) {
if (fmt[i] != sample_fmt && fmt_cnt[i] > plugin.fmt.bps) {
if (snd_pcm_hw_params_set_format (audio, hw_params, fmt[i]) >= 0) {
fprintf (stderr, "Found compatible format %d bps\n", fmt_cnt[i]);
sample_fmt = fmt[i];
break;
}
}
}
if (fmt[i] == -1) {
// next try formats with lower bps
i = 0;
for (i = 0; fmt[i] != -1; i++) {
if (fmt[i] != sample_fmt && fmt_cnt[i] < plugin.fmt.bps) {
if (snd_pcm_hw_params_set_format (audio, hw_params, fmt[i]) >= 0) {
fprintf (stderr, "Found compatible format %d bps\n", fmt_cnt[i]);
sample_fmt = fmt[i];
break;
}
}
}
}
if (fmt[i] == -1) {
fprintf (stderr, "Fallback format could not be found\n");
goto error;
}
}
snd_pcm_hw_params_get_format (hw_params, &sample_fmt);
trace ("chosen sample format: %04Xh\n", (int)sample_fmt);
int val = plugin.fmt.samplerate;
int ret = 0;
if ((err = snd_pcm_hw_params_set_rate_resample (audio, hw_params, conf_alsa_resample)) < 0) {
fprintf (stderr, "cannot setup resampling (%s)\n",
snd_strerror (err));
goto error;
}
if ((err = snd_pcm_hw_params_set_rate_near (audio, hw_params, &val, &ret)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
goto error;
}
plugin.fmt.samplerate = val;
trace ("chosen samplerate: %d Hz\n", val);
int chanmin, chanmax;
snd_pcm_hw_params_get_channels_min (hw_params, &chanmin);
snd_pcm_hw_params_get_channels_max (hw_params, &chanmax);
trace ("minchan: %d, maxchan: %d\n", chanmin, chanmax);
int nchan = plugin.fmt.channels;
if (nchan > chanmax) {
nchan = chanmax;
}
else if (nchan < chanmin) {
nchan = chanmin;
}
trace ("setting chan=%d\n", nchan);
if ((err = snd_pcm_hw_params_set_channels (audio, hw_params, nchan)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
}
snd_pcm_hw_params_get_channels (hw_params, &nchan);
trace ("alsa channels: %d\n", nchan);
req_buffer_size = deadbeef->conf_get_int ("alsa.buffer", DEFAULT_BUFFER_SIZE);
req_period_size = deadbeef->conf_get_int ("alsa.period", DEFAULT_PERIOD_SIZE);
buffer_size = req_buffer_size;
period_size = req_period_size;
trace ("trying buffer size: %d frames\n", (int)buffer_size);
trace ("trying period size: %d frames\n", (int)period_size);
snd_pcm_hw_params_set_buffer_size_near (audio, hw_params, &buffer_size);
snd_pcm_hw_params_set_period_size_near (audio, hw_params, &period_size, NULL);
trace ("alsa buffer size: %d frames\n", (int)buffer_size);
trace ("alsa period size: %d frames\n", (int)period_size);
if ((err = snd_pcm_hw_params (audio, hw_params)) < 0) {
fprintf (stderr, "cannot set parameters (%s)\n",
snd_strerror (err));
goto error;
}
plugin.fmt.is_float = 0;
switch (sample_fmt) {
case SND_PCM_FORMAT_S8:
plugin.fmt.bps = 8;
break;
case SND_PCM_FORMAT_S16_BE:
case SND_PCM_FORMAT_S16_LE:
plugin.fmt.bps = 16;
break;
case SND_PCM_FORMAT_S24_3BE:
case SND_PCM_FORMAT_S24_3LE:
plugin.fmt.bps = 24;
break;
case SND_PCM_FORMAT_S32_BE:
case SND_PCM_FORMAT_S32_LE:
plugin.fmt.bps = 32;
break;
case SND_PCM_FORMAT_FLOAT_LE:
case SND_PCM_FORMAT_FLOAT_BE:
plugin.fmt.bps = 32;
plugin.fmt.is_float = 1;
break;
}
trace ("chosen bps: %d (%s)\n", plugin.fmt.bps, plugin.fmt.is_float ? "float" : "int");
plugin.fmt.channels = nchan;
plugin.fmt.channelmask = 0;
if (nchan == 1) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT;
}
if (nchan == 2) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT;
}
if (nchan == 3) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT | DDB_SPEAKER_LOW_FREQUENCY;
}
if (nchan == 4) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT | DDB_SPEAKER_BACK_LEFT | DDB_SPEAKER_BACK_RIGHT;
}
if (nchan == 5) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT | DDB_SPEAKER_BACK_LEFT | DDB_SPEAKER_BACK_RIGHT | DDB_SPEAKER_FRONT_CENTER;
}
if (nchan == 6) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT | DDB_SPEAKER_BACK_LEFT | DDB_SPEAKER_BACK_RIGHT | DDB_SPEAKER_FRONT_CENTER | DDB_SPEAKER_LOW_FREQUENCY;
}
if (nchan == 7) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT | DDB_SPEAKER_BACK_LEFT | DDB_SPEAKER_BACK_RIGHT | DDB_SPEAKER_FRONT_CENTER | DDB_SPEAKER_SIDE_LEFT | DDB_SPEAKER_SIDE_RIGHT;
}
if (nchan == 8) {
plugin.fmt.channelmask = DDB_SPEAKER_FRONT_LEFT | DDB_SPEAKER_FRONT_RIGHT | DDB_SPEAKER_BACK_LEFT | DDB_SPEAKER_BACK_RIGHT | DDB_SPEAKER_FRONT_CENTER | DDB_SPEAKER_SIDE_LEFT | DDB_SPEAKER_SIDE_RIGHT | DDB_SPEAKER_LOW_FREQUENCY;
}
error:
if (err < 0) {
memset (&plugin.fmt, 0, sizeof (ddb_waveformat_t));
}
if (hw_params) {
snd_pcm_hw_params_free (hw_params);
}
return err;
}
static snd_pcm_t * bg_alsa_open(const char * card,
gavl_audio_format_t * format,
snd_pcm_stream_t stream,
gavl_time_t buffer_time,
int * convert_4_3)
{
unsigned int i_tmp;
int dir, err;
snd_pcm_hw_params_t *hw_params = NULL;
// snd_pcm_sw_params_t *sw_params = NULL;
snd_pcm_t *ret = NULL;
snd_pcm_uframes_t buffer_size;
snd_pcm_uframes_t period_size;
snd_pcm_uframes_t period_size_min = 0;
snd_pcm_uframes_t period_size_max = 0;
snd_pcm_uframes_t buffer_size_min = 0;
snd_pcm_uframes_t buffer_size_max = 0;
/* We open in non blocking mode so our process won't hang if the card is
busy */
if((err = snd_pcm_open(&ret,
card,
stream, // SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_CAPTURE
SND_PCM_NONBLOCK // SND_PCM_ASYNC
) < 0))
{
ret = NULL;
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_open failed for device %s (%s)",
card, snd_strerror(err));
goto fail;
}
/* Now, set blocking mode */
snd_pcm_nonblock(ret, 0);
if(snd_pcm_hw_params_malloc(&hw_params) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_malloc failed");
goto fail;
}
if(snd_pcm_hw_params_any(ret, hw_params) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_any failed");
goto fail;
}
/* Interleave mode */
if(snd_pcm_hw_params_set_access(ret, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_access failed");
goto fail;
}
else
format->interleave_mode = GAVL_INTERLEAVE_ALL;
/* Sample format */
switch(format->sample_format)
{
case GAVL_SAMPLE_S8:
case GAVL_SAMPLE_U8:
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S8) < 0)
// if(1)
{
/* Soundcard support no 8-bit, try 16 */
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S16) < 0)
{
/* Hopeless */
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_format failed");
goto fail;
}
else
format->sample_format = GAVL_SAMPLE_S16;
}
else
format->sample_format = GAVL_SAMPLE_S8;
break;
case GAVL_SAMPLE_S16:
case GAVL_SAMPLE_U16:
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S16) < 0)
{
/* Hopeless */
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_format failed");
goto fail;
}
else
format->sample_format = GAVL_SAMPLE_S16;
break;
case GAVL_SAMPLE_S32:
#ifndef WORDS_BIGENDIAN
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S32_LE) < 0)
#else
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S32_BE) < 0)
#endif
{
/* Soundcard supports no 32-bit, try 24
(more probably supported but needs conversion) */
#ifndef WORDS_BIGENDIAN
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S24_3LE) < 0)
#else
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S24_3BE) < 0)
#endif
{
/* No 24 bit, try 16 */
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S16) < 0)
{
/* Hopeless */
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_format failed");
goto fail;
}
else
format->sample_format = GAVL_SAMPLE_S16;
}
else
{
format->sample_format = GAVL_SAMPLE_S32;
if(convert_4_3)
*convert_4_3 = 1;
}
}
else
format->sample_format = GAVL_SAMPLE_S32;
break;
case GAVL_SAMPLE_FLOAT:
case GAVL_SAMPLE_DOUBLE:
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_FLOAT) < 0)
{
#ifndef WORDS_BIGENDIAN
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S32_LE) < 0)
#else
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S32_BE) < 0)
#endif
{
/* Soundcard supports no 32-bit, try 24
(more probably supported but needs conversion) */
#ifndef WORDS_BIGENDIAN
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S24_3LE) < 0)
#else
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S24_3BE) < 0)
#endif
{
/* No 24 bit, try 16 */
if(snd_pcm_hw_params_set_format(ret, hw_params, SND_PCM_FORMAT_S16) < 0)
{
/* Hopeless */
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_format failed");
goto fail;
}
else
format->sample_format = GAVL_SAMPLE_S16;
}
else
{
format->sample_format = GAVL_SAMPLE_S32;
if(convert_4_3)
*convert_4_3 = 1;
}
}
else
format->sample_format = GAVL_SAMPLE_S32;
}
else
format->sample_format = GAVL_SAMPLE_FLOAT;
break;
case GAVL_SAMPLE_NONE:
goto fail;
break;
}
/* Channels */
if(snd_pcm_hw_params_set_channels(ret, hw_params,
format->num_channels) < 0)
{
if(format->num_channels == 1) /* Mono doesn't work, try stereo */
{
if(snd_pcm_hw_params_set_channels(ret, hw_params,
2) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN,
"snd_pcm_hw_params_set_channels failed (Format has %d channels)",
format->num_channels);
goto fail;
}
else
{
format->num_channels = 2;
format->channel_locations[0] = GAVL_CHID_FRONT_LEFT;
format->channel_locations[1] = GAVL_CHID_FRONT_RIGHT;
}
}
else
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN,
"snd_pcm_hw_params_set_channels failed (Format has %d channels)",
format->num_channels);
goto fail;
}
}
/* Switch off driver side resampling */
#if SND_LIB_VERSION >= 0x010009
snd_pcm_hw_params_set_rate_resample(ret, hw_params, 0);
#endif
/* Samplerate */
i_tmp = format->samplerate;
if(snd_pcm_hw_params_set_rate_near(ret, hw_params,
&i_tmp,
0) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_rate_near failed");
goto fail;
}
if(format->samplerate != i_tmp)
{
bg_log(BG_LOG_INFO, LOG_DOMAIN,
"Samplerate %d not supported by device %s, using %d",
format->samplerate, card, i_tmp);
}
format->samplerate = i_tmp;
dir = 0;
/* Buffer size */
snd_pcm_hw_params_get_buffer_size_min(hw_params, &buffer_size_min);
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size_max);
dir=0;
snd_pcm_hw_params_get_period_size_min(hw_params, &period_size_min,&dir);
dir=0;
snd_pcm_hw_params_get_period_size_max(hw_params, &period_size_max,&dir);
buffer_size = gavl_time_to_samples(format->samplerate, buffer_time);
if(buffer_size > buffer_size_max)
buffer_size = buffer_size_max;
if(buffer_size < buffer_size_min)
buffer_size = buffer_size_min;
period_size = buffer_size / 8;
buffer_size = period_size * 8;
dir = 0;
if(snd_pcm_hw_params_set_period_size_near(ret, hw_params, &period_size, &dir) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_period_size failed");
goto fail;
}
dir = 0;
snd_pcm_hw_params_get_period_size(hw_params, &period_size, &dir);
dir = 0;
if(snd_pcm_hw_params_set_buffer_size_near(ret, hw_params, &buffer_size) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params_set_buffer_size failed");
goto fail;
}
snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size);
format->samples_per_frame = period_size;
/* Write params to card */
if(snd_pcm_hw_params (ret, hw_params) < 0)
{
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "snd_pcm_hw_params failed");
goto fail;
}
snd_pcm_hw_params_free(hw_params);
gavl_set_channel_setup(format);
// gavl_audio_format_dump(format);
return ret;
fail:
bg_log(BG_LOG_ERROR, LOG_DOMAIN, "Alsa initialization failed");
if(ret)
snd_pcm_close(ret);
if(hw_params)
snd_pcm_hw_params_free(hw_params);
return NULL;
}
static int
laudio_alsa_open(void)
{
snd_pcm_hw_params_t *hw_params;
snd_pcm_uframes_t bufsize;
int ret;
hw_params = NULL;
ret = snd_pcm_open(&hdl, card_name, SND_PCM_STREAM_PLAYBACK, 0);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not open playback device: %s\n", snd_strerror(ret));
return -1;
}
/* HW params */
ret = snd_pcm_hw_params_malloc(&hw_params);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not allocate hw params: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_hw_params_any(hdl, hw_params);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not retrieve hw params: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_hw_params_set_access(hdl, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set access method: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_hw_params_set_format(hdl, hw_params, SND_PCM_FORMAT_S16_LE);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set S16LE format: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_hw_params_set_channels(hdl, hw_params, 2);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set stereo output: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_hw_params_set_rate(hdl, hw_params, 44100, 0);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Hardware doesn't support 44.1 kHz: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufsize);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not get max buffer size: %s\n", snd_strerror(ret));
goto out_fail;
}
DPRINTF(E_DBG, L_LAUDIO, "Max buffer size is %lu samples\n", bufsize);
ret = snd_pcm_hw_params_set_buffer_size_max(hdl, hw_params, &bufsize);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set buffer size to max: %s\n", snd_strerror(ret));
goto out_fail;
}
DPRINTF(E_DBG, L_LAUDIO, "Buffer size is %lu samples\n", bufsize);
ret = snd_pcm_hw_params(hdl, hw_params);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set hw params: %s\n", snd_strerror(ret));
goto out_fail;
}
snd_pcm_hw_params_free(hw_params);
hw_params = NULL;
pcm_pos = 0;
pcm_last_error = 0;
pcm_recovery = 0;
pcm_buf_threshold = (bufsize / AIRTUNES_V2_PACKET_SAMPLES) * AIRTUNES_V2_PACKET_SAMPLES;
ret = mixer_open();
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not open mixer\n");
goto out_fail;
}
update_status(LAUDIO_OPEN);
return 0;
out_fail:
if (hw_params)
snd_pcm_hw_params_free(hw_params);
snd_pcm_close(hdl);
hdl = NULL;
return -1;
}
